From jayesh1017 at gmail.com Thu Feb 1 00:53:48 2018 From: jayesh1017 at gmail.com (Jayesh Nambiar) Date: Thu, 1 Feb 2018 11:23:48 +0530 Subject: [OpenSIPS-Users] user location in mongodb Message-ID: Hello, I am running opensips 2.3.2 stable. I wanted to know if mongodb can be used for storing the user location information with opensips cachedb_mongodb module? Quick googling didnt give me any clarity if it is possible or not. Before doing anything, I wanted to know if this is possible with opensips. Thanks, - Jayesh -------------- next part -------------- An HTML attachment was scrubbed... URL: From liviu at opensips.org Thu Feb 1 05:04:25 2018 From: liviu at opensips.org (Liviu Chircu) Date: Thu, 1 Feb 2018 12:04:25 +0200 Subject: [OpenSIPS-Users] user location in mongodb In-Reply-To: References: Message-ID: <05aa43ec-9ccf-8ba4-742d-2a36e0f45ded@opensips.org> Hi Jayesh, Absolutely yes! In fact, MongoDB is currently the only supported NoSQL backend to work with usrloc! The idea is that its document-oriented representation of data makes it easy to equivalate to an SQL backend, hence Vlad wrote the "db_cachedb" NoSQL -> SQL translation interface and also equipped the "cachedb_mongodb" module with an initial implementation of this interface. So, to sum up: make sure to load both "db_cachedb" and "cachedb_mongodb", while configuring then in a similar fashion to the module docs [1]. You should get the usrloc working in no time! Cheers, [1]: http://www.opensips.org/html/docs/modules/2.4.x/db_cachedb.html#idp243888 Liviu Chircu OpenSIPS Developer http://www.opensips-solutions.com On 01.02.2018 07:53, Jayesh Nambiar wrote: > Hello, > I am running opensips 2.3.2 stable. I wanted to know if mongodb can be > used for storing the user location information with opensips > cachedb_mongodb module? > Quick googling didnt give me any clarity if it is possible or not. > Before doing anything, I wanted to know if this is possible with opensips. > > Thanks, > > - Jayesh > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From netaskd at gmail.com Thu Feb 1 10:34:26 2018 From: netaskd at gmail.com (Dmitry) Date: Thu, 1 Feb 2018 18:34:26 +0300 Subject: [OpenSIPS-Users] Handling 488 on a leg B In-Reply-To: <1f74b65d-5ac7-7bcd-653c-c419882785f0@opensips.org> References: <000a01d399c9$c4ccaae0$4e6600a0$@gmail.com> <1f74b65d-5ac7-7bcd-653c-c419882785f0@opensips.org> Message-ID: <424a01d39b72$252783c0$6f768b40$@gmail.com> Hello Bogdan, Thank you for advice. I catched 488 and can send INVITE from failure_route, but I can’t modify body inside it. The main goal is change SAVP profile to AVP when 488 received. I tried to do it via rtpengine_offer function. It doesn’t work. I guess, it try to read the body from response (488 in my case) but not from request. I tried to rewrite body via remove_body_part and add_body_part functions, it also doesn’t work. So, I don’t understand how exactly rewrite body in failure_route before do t_relay(). In addition, I tried making a new branch (via append_branch function and via branch_route). It just duplicate my re-INVITE without changing the body. Could you please drop a hint (example on few lines of code) how to do it? Thanks. From: Bogdan-Andrei Iancu [mailto:bogdan at opensips.org] Sent: Wednesday, January 31, 2018 6:55 PM To: OpenSIPS users mailling list ; Dmitry Subject: Re: [OpenSIPS-Users] Handling 488 on a leg B Hello Dmitry, What you want to do is called (in SIP) serial forking. Use failure_route{} to catch the 488 and to add a create a new branch (with modified body) . See: http://www.opensips.org/Documentation/Script-Routes-2-3#toc3 Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Summit 2018 http://www.opensips.org/events/Summit-2018Amsterdam On 01/30/2018 02:56 PM, Dmitry wrote: Hello all, Could you say how can I re-send INVITE with changed SDP to B-leg after receive 488 from it? Call flow: -> A-leg INVITE <- A-leg 100 -> B-leg INVITE <- B-leg 100 <- B-leg 488 – I catch it, and want to send INVITE again with properly SDP -> B-leg ACK <-A-leg 488 -> B-leg ACK Thanks for any advice. Cheers! _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Thu Feb 1 11:51:11 2018 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Thu, 1 Feb 2018 18:51:11 +0200 Subject: [OpenSIPS-Users] Handling 488 on a leg B In-Reply-To: <424a01d39b72$252783c0$6f768b40$@gmail.com> References: <000a01d399c9$c4ccaae0$4e6600a0$@gmail.com> <1f74b65d-5ac7-7bcd-653c-c419882785f0@opensips.org> <424a01d39b72$252783c0$6f768b40$@gmail.com> Message-ID: Hi Dmitry, In failure route you re-process the original request (INVITE) and not the 488 reply - the reply is just the trigger. Do you change the INVITE SDP before the first attempt (which ends with 488) ? Or you change it for the first time in the failure route ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Summit 2018 http://www.opensips.org/events/Summit-2018Amsterdam On 02/01/2018 05:34 PM, Dmitry wrote: > > Hello Bogdan, > > Thank you for advice. > > I catched 488 and can send INVITE from failure_route, but I can’t > modify body inside it. > > The main goal is change SAVP profile to AVP when 488 received. > > I tried to do it via rtpengine_offer function. It doesn’t work. I > guess, it try to read the body from response (488 in my case) but not > from request. > > I tried to rewrite body via remove_body_part and add_body_part > functions, it also doesn’t work. > > So, I don’t understand how exactly rewrite body in failure_route > before do t_relay(). > > In addition, I tried making a new branch (via append_branch function > and via branch_route). It just duplicate my re-INVITE without changing > the body. > > Could you please drop a hint (example on few lines of code) how to do it? > > Thanks. > > *From:*Bogdan-Andrei Iancu [mailto:bogdan at opensips.org] > *Sent:* Wednesday, January 31, 2018 6:55 PM > *To:* OpenSIPS users mailling list ; Dmitry > > *Subject:* Re: [OpenSIPS-Users] Handling 488 on a leg B > > Hello Dmitry, > > What you want to do is called (in SIP) serial forking. Use > failure_route{} to catch the 488 and to add a create a new branch > (with modified body) . > > See: http://www.opensips.org/Documentation/Script-Routes-2-3#toc3 > > Regards, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > OpenSIPS Summit 2018 > http://www.opensips.org/events/Summit-2018Amsterdam > > On 01/30/2018 02:56 PM, Dmitry wrote: > > Hello all, > > Could you say how can I re-send INVITE with changed SDP to B-leg > after receive 488 from it? > > Call flow: > > -> A-leg INVITE > > <- A-leg 100 > > -> B-leg INVITE > > <- B-leg 100 > > <- B-leg 488 – I catch it, and want to send INVITE again with > properly SDP > > -> B-leg ACK > > <-A-leg 488 > > -> B-leg ACK > > Thanks for any advice. > > Cheers! > > > > > _______________________________________________ > > Users mailing list > > Users at lists.opensips.org > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Thu Feb 1 11:53:44 2018 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Thu, 1 Feb 2018 18:53:44 +0200 Subject: [OpenSIPS-Users] Potential memory leak in siptrace module In-Reply-To: <1D7B55AB4979E6498B3AC325C40E84360102C18BB6@ORD2MBX02F.mex05.mlsrvr.com> References: <1D7B55AB4979E6498B3AC325C40E84360102C18BB6@ORD2MBX02F.mex05.mlsrvr.com> Message-ID: <67b64a89-f845-03dd-6f67-b95226b815c2@opensips.org> Hi Steve, Thank you for your report. Could you provide the exact revision of your OpenSIPS (opensips -V) ? Also what requests are you calling the sip_trace() for (INVITES ? PUBLISH? all?) and what are the input paramters ? Thanks, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Summit 2018 http://www.opensips.org/events/Summit-2018Amsterdam On 01/31/2018 07:48 PM, Steve Brisson wrote: > > I've found that when I call the sip_trace method, in the main request > routing block, the shmem used_size and fragments continually increases > until the process crashes from an out of memory condition. > > By enabling memory alloc debugging, I confirmed that the leaked > allocation is the shm_malloc of a trace_info_t structure in the > sip_trace_w method [siptrace.c:1539]. This piece of memory is supposed > to be freed in the dialog terminated callback but because of the logic > in the sip_trace_w method, trace_flags gets modified to > TRACE_TRANSACTION and the dialog callbacks are never registered. > > I haven't worked out the details of a proper fix yet. My workaround > has been to remove the siptrace module from my config. > > OpenSIPS config details: > > opensips 2.3.2 (x86_64/linux) > > #siptrace module config > > modparam("siptrace", "trace_on", 0) > > modparam("siptrace", "trace_id", > "[tid]uri=mysql://db-user:db-pass at db-host/db-name;table=sip_trace;") > > #siptrace method call in main request branch > > sip_trace("tid", "d", "sip"); > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From steve.brisson at librestream.com Thu Feb 1 12:30:03 2018 From: steve.brisson at librestream.com (Steve Brisson) Date: Thu, 1 Feb 2018 17:30:03 +0000 Subject: [OpenSIPS-Users] Potential memory leak in siptrace module In-Reply-To: <67b64a89-f845-03dd-6f67-b95226b815c2@opensips.org> References: <1D7B55AB4979E6498B3AC325C40E84360102C18BB6@ORD2MBX02F.mex05.mlsrvr.com> <67b64a89-f845-03dd-6f67-b95226b815c2@opensips.org> Message-ID: <1D7B55AB4979E6498B3AC325C40E84360102C18F04@ORD2MBX02F.mex05.mlsrvr.com> Hi, In answer to your questions... 1. $ opensips -V version: opensips 2.3.2 (x86_64/linux) flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535 poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. main.c compiled on 15:38:40 Nov 24 2017 with gcc 5.4.0 I built from commit 9e497ea4ab03c67cb073bdf0dccef2b16bff80e2. I've since diff'd with the HEAD and I haven't seen any changes in the siptrace code that would have helped my particular issue. 2. The call to sip_trace is in the main request route branch and it gets called for all requests. input parameters: sip_trace("tid", "d", "sip"); module config: modparam("siptrace", "trace_on", 0) modparam("siptrace", "trace_id", "[tid]uri=mysql://db-user:db-pass at db-host/db-name;table=sip_trace;") From: Bogdan-Andrei Iancu [mailto:bogdan at opensips.org] Sent: Thursday, February 1, 2018 10:54 AM To: OpenSIPS users mailling list ; Steve Brisson Subject: Re: [OpenSIPS-Users] Potential memory leak in siptrace module Hi Steve, Thank you for your report. Could you provide the exact revision of your OpenSIPS (opensips -V) ? Also what requests are you calling the sip_trace() for (INVITES ? PUBLISH? all?) and what are the input paramters ? Thanks, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Summit 2018 http://www.opensips.org/events/Summit-2018Amsterdam On 01/31/2018 07:48 PM, Steve Brisson wrote: I've found that when I call the sip_trace method, in the main request routing block, the shmem used_size and fragments continually increases until the process crashes from an out of memory condition. By enabling memory alloc debugging, I confirmed that the leaked allocation is the shm_malloc of a trace_info_t structure in the sip_trace_w method [siptrace.c:1539]. This piece of memory is supposed to be freed in the dialog terminated callback but because of the logic in the sip_trace_w method, trace_flags gets modified to TRACE_TRANSACTION and the dialog callbacks are never registered. I haven't worked out the details of a proper fix yet. My workaround has been to remove the siptrace module from my config. OpenSIPS config details: opensips 2.3.2 (x86_64/linux) #siptrace module config modparam("siptrace", "trace_on", 0) modparam("siptrace", "trace_id", "[tid]uri=mysql://db-user:db-pass at db-host/db-name;table=sip_trace;") #siptrace method call in main request branch sip_trace("tid", "d", "sip"); _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From govoiper at gmail.com Thu Feb 1 12:56:49 2018 From: govoiper at gmail.com (SamyGo) Date: Thu, 1 Feb 2018 12:56:49 -0500 Subject: [OpenSIPS-Users] I have some doubts on contact fix in opensips-1.11 . In-Reply-To: <59e36d28-a483-314c-3997-d37b40fe8e8f@opensips.org> References: <59e36d28-a483-314c-3997-d37b40fe8e8f@opensips.org> Message-ID: :D @ Bogdan. Good one. On Wed, Jan 31, 2018 at 10:51 AM, Bogdan-Andrei Iancu wrote: > Hi Sasmita, > > Of course it fixes only the first contact - afterall the function is > called fix_contact() and not fix_contactS() > > But please ignore me, this was a silly joke :D.. > > Yes, it seems that fix_contact() is limited at handling the first only. > But you can use the fix_nated_contact() function from nathelper module, > which handles all contacts. > > Regards > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > OpenSIPS Summit 2018 > http://www.opensips.org/events/Summit-2018Amsterdam > > On 01/31/2018 02:23 PM, Sasmita Panda wrote: > > Hi All , > > I am using opensips-1.11 and trying to fix contact in the proxy > before saving in DB . This is the bellow code for contact fix . > > if (client_nat_test("1")) { > fix_contact(); > }; > > As for my observation , when I am sending multiple contact in a > single REGISTER message , Its only updating the first contact and wont > update the rest . > > Just wanted to know why this is happening . Every contact of mine in > the REGISTER message have private IP but one get updated and rest remain > same . > > Please do help me . Thanks in advance . > > > *Thanks & Regards* > *Sasmita Panda* > *Network Testing and Software Engineer* > *3CLogic , ph:07827611765* > > > _______________________________________________ > Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From brian.southworth at clocom.uk Fri Feb 2 04:14:33 2018 From: brian.southworth at clocom.uk (=?UTF-8?Q?Brian_Southworth?=) Date: Fri, 2 Feb 2018 09:14:33 +0000 Subject: [OpenSIPS-Users] [15066] WARNING:rr:after_strict: no socket found to match RR [1][XXX.XXX.XXX.XXX:5060] References: Message-ID: <0102016155cbc5e4-c3c66a50-ab65-4987-803e-cd61fbfd5647-000000@eu-west-1.amazonses.com> I get this when trying to transfer calls using the B2BUA: [15066] WARNING:rr:after_strict: no socket found to match RR [1][xxx.xxx.xxx.xxx:5060]   When I try looking on the mailing list there are no other similar posts, could you please shed some light on what maybe causing this please.   Regards,   Brian Southworth -------------- next part -------------- An HTML attachment was scrubbed... URL: From netaskd at gmail.com Fri Feb 2 05:53:16 2018 From: netaskd at gmail.com (Dmitry) Date: Fri, 2 Feb 2018 13:53:16 +0300 Subject: [OpenSIPS-Users] Handling 488 on a leg B In-Reply-To: References: <000a01d399c9$c4ccaae0$4e6600a0$@gmail.com> <1f74b65d-5ac7-7bcd-653c-c419882785f0@opensips.org> <424a01d39b72$252783c0$6f768b40$@gmail.com> Message-ID: <429c01d39c14$087d3a60$1977af20$@gmail.com> Hi Bogdan, I change both of them. The first INVITE and the second (after 488). The first apply successfully, but the second (that in the failure_route) does not. My script looks like this: route { … t_on_reply("ONREPLY”); if (isbflagset(NAT_BFLAG) && af!=INET6) { force_rport(); # add rport to first line Via: route(RTPENGINE_OFFER); # make offer } … t_relay(); … } route[RTPENGINE_OFFER] { … $avp(rtp_profile) = "trust-address replace-origin replace-session-connection ICE=remove DTLS=off RTP/SAVP"; if (rtpengine_offer("$avp(rtp_profile)","$avp(rtp_socket)","$avp(rtp_body)")) { remove_body_part(); add_body_part("$avp(rtp_body)", "application/sdp"); # first changing leg-B - successfully } … } onreply_route[ONREPLY] { … t_on_reply("RTPENGINE_ANSWER"); # positive t_on_failure("RTPENGINE_488"); # negative … } failure_route[RTPENGINE_488] { … if (t_check_status("488") ) { # try to change SDP in leg-B after 488 received - unsuccessfully # it returns: ERROR:rtpengine:rtpe_function_call: can't extract body from the message. # from which message? if (rtpengine_offer("trust-address replace-origin replace-session-connection ICE=remove DTLS=off RTP/AVP","$avp(rtp_socket)","$avp(rtp_body)")) { remove_body_part(); add_body_part("$avp(rtp_body)", "application/sdp"); } # # just replace with re.subst $avp(rtp_body) = $(avp(rtp_body){re.subst,/SAVP/AVP/g}); $avp(rtp_body) = $(avp(rtp_body){re.subst,/a=crypto.*//g}); # if I use remove_body_part(), then add_body_part() is not adds content. INVITE sends without body. remove_body_part(); # if I don’t use remove_body_part(), then INVITE contains TWO sdp body with RTP/SAVP and RTP/AVP add_body_part("$avp(rtp_body)", "application/sdp"); # xlog("rtp_body=$avp(rtp_body)\n"); # xlog for $avp(rtp_body) shows unchanged content (with RTP/SAVP) … t_relay(); } … } From: Bogdan-Andrei Iancu [mailto:bogdan at opensips.org] Sent: Thursday, February 1, 2018 7:51 PM To: Dmitry ; 'OpenSIPS users mailling list' Subject: Re: [OpenSIPS-Users] Handling 488 on a leg B Hi Dmitry, In failure route you re-process the original request (INVITE) and not the 488 reply - the reply is just the trigger. Do you change the INVITE SDP before the first attempt (which ends with 488) ? Or you change it for the first time in the failure route ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Summit 2018 http://www.opensips.org/events/Summit-2018Amsterdam On 02/01/2018 05:34 PM, Dmitry wrote: Hello Bogdan, Thank you for advice. I catched 488 and can send INVITE from failure_route, but I can’t modify body inside it. The main goal is change SAVP profile to AVP when 488 received. I tried to do it via rtpengine_offer function. It doesn’t work. I guess, it try to read the body from response (488 in my case) but not from request. I tried to rewrite body via remove_body_part and add_body_part functions, it also doesn’t work. So, I don’t understand how exactly rewrite body in failure_route before do t_relay(). In addition, I tried making a new branch (via append_branch function and via branch_route). It just duplicate my re-INVITE without changing the body. Could you please drop a hint (example on few lines of code) how to do it? Thanks. From: Bogdan-Andrei Iancu [mailto:bogdan at opensips.org] Sent: Wednesday, January 31, 2018 6:55 PM To: OpenSIPS users mailling list ; Dmitry Subject: Re: [OpenSIPS-Users] Handling 488 on a leg B Hello Dmitry, What you want to do is called (in SIP) serial forking. Use failure_route{} to catch the 488 and to add a create a new branch (with modified body) . See: http://www.opensips.org/Documentation/Script-Routes-2-3#toc3 Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Summit 2018 http://www.opensips.org/events/Summit-2018Amsterdam On 01/30/2018 02:56 PM, Dmitry wrote: Hello all, Could you say how can I re-send INVITE with changed SDP to B-leg after receive 488 from it? Call flow: -> A-leg INVITE <- A-leg 100 -> B-leg INVITE <- B-leg 100 <- B-leg 488 – I catch it, and want to send INVITE again with properly SDP -> B-leg ACK <-A-leg 488 -> B-leg ACK Thanks for any advice. Cheers! _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From razvan at opensips.org Fri Feb 2 06:08:09 2018 From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=) Date: Fri, 2 Feb 2018 13:08:09 +0200 Subject: [OpenSIPS-Users] OpenSIPS at FOSDEM 2018 In-Reply-To: <02edf02e-6ede-6317-2707-17d24270a465@opensips.org> References: <02edf02e-6ede-6317-2707-17d24270a465@opensips.org> Message-ID: <5cdae5ec-b6dc-6720-55d7-d625b632217b@opensips.org> On 01/30/2018 07:27 PM, Răzvan Crainea wrote: > Hi, everyone! > > This year the OpenSIPS team is going to be present again at FOSDEM, the > largest open-source conference in Europe. > > Our talk for FOSDEM 2018 is going to present one of the main attraction > of the upcoming OpenSIPS release: *Real Time Clustering* using OpenSIPS > 2.4. > > Join us for this talk on Saturday, 15:00, in Room H.1309 (Van Rijn)[1] > to find out what are the new and powerful tools we've been working on > for the OpenSIPS Clustering! Mind my typo, our talk is on Sunday, 15:00, in the RTC devroom, H.1309 (Van Rijn)[1]. > > [1] https://fosdem.org/2018/schedule/event/opensips/ > See you there! Cheers, -- Răzvan Crainea OpenSIPS Core Developer http://www.opensips-solutions.com OpenSIPS Summit 2018 http://www.opensips.org/events/Summit-2018Amsterdam From bogdan at opensips.org Fri Feb 2 06:18:04 2018 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Fri, 2 Feb 2018 13:18:04 +0200 Subject: [OpenSIPS-Users] OpenSIPS Summit 50% off during Fosdem weekend Message-ID: <7448f4d5-7b3d-f78b-c577-2b479090131a@opensips.org> *Fosdem, here we come!* Are you a Fosdem , an Open Source or an OpenSIPS fan? Don't mis us Sunday, 15:00, in Room H.1309 , talking about */R/*/*eal Time Clustering*/ using OpenSIPS 2.4 !! And we gave a gift for you all - save 50% for OpenSIPS Summit and OpenSIPS Training by registering during *Fosdem* weekend 02-04 of February with token *b3e737ac-9019-493f-8888-9f6c16af3bc9* *Register now * and secure your seat for the OpenSIPS Summit ! -- Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Fri Feb 2 06:29:29 2018 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Fri, 2 Feb 2018 13:29:29 +0200 Subject: [OpenSIPS-Users] [15066] WARNING:rr:after_strict: no socket found to match RR [1][XXX.XXX.XXX.XXX:5060] In-Reply-To: <0102016155cbc5e4-c3c66a50-ab65-4987-803e-cd61fbfd5647-000000@eu-west-1.amazonses.com> References: <0102016155cbc5e4-c3c66a50-ab65-4987-803e-cd61fbfd5647-000000@eu-west-1.amazonses.com> Message-ID: Hi Brian, That warning means OpenSIPS found a Route header (while doing loose_route) that is suppose to be of its own, but the network information from the header does not match any of the OpenSIPS SIP listeners. Best regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Summit 2018 http://www.opensips.org/events/Summit-2018Amsterdam On 02/02/2018 11:14 AM, Brian Southworth wrote: > > I get this when trying to transfer calls using the B2BUA: > > [15066] WARNING:rr:after_strict: no socket found to match RR > [1][xxx.xxx.xxx.xxx:5060] > > When I try looking on the mailing list there are no other similar > posts, could you please shed some light on what maybe causing this please. > > Regards, > > Brian Southworth > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From brian.southworth at clocom.uk Fri Feb 2 06:38:28 2018 From: brian.southworth at clocom.uk (=?UTF-8?Q?Brian_Southworth?=) Date: Fri, 2 Feb 2018 11:38:28 +0000 Subject: [OpenSIPS-Users] [15066] WARNING:rr:after_strict: no socket found to match RR [1][XXX.XXX.XXX.XXX:5060] In-Reply-To: References: <0102016155cbc5e4-c3c66a50-ab65-4987-803e-cd61fbfd5647-000000@eu-west-1.amazonses.com> Message-ID: <01020161564f88ab-94ee6ed7-b89e-4d6e-866d-99d682719034-000000@eu-west-1.amazonses.com> Hi Bogdan,   Thank you very much, so this doesn’t have any impact on why the call being transferred are dropped ?   I am trying to transfer a call using the refer method as that is what the cisco phones use.   The network is setup like so opensips proxy à asterisk gateway à carrier   Scenario:   Inbound call comes into the phone like so: carrier à ast à proxy à phone A Phone A needs to transfer call to phone B: Phone A dials phone B à phone B picks up à phone A presses xfer button and call is dropped.   Any help would be appreciated.   Regards,   Brian Southworth   From: Bogdan-Andrei Iancu [mailto:bogdan at opensips.org] Sent: 02 February 2018 11:29 To: OpenSIPS users mailling list ; Brian Southworth Subject: Re: [OpenSIPS-Users] [15066] WARNING:rr:after_strict: no socket found to match RR [1][XXX.XXX.XXX.XXX:5060]   Hi Brian, That warning means OpenSIPS found a Route header (while doing loose_route) that is suppose to be of its own, but the network information from the header does not match any of the OpenSIPS SIP listeners. Best regards, Bogdan-Andrei Iancu   OpenSIPS Founder and Developer   http://www.opensips-solutions.com OpenSIPS Summit 2018   http://www.opensips.org/events/Summit-2018Amsterdam On 02/02/2018 11:14 AM, Brian Southworth wrote: I get this when trying to transfer calls using the B2BUA: [15066] WARNING:rr:after_strict: no socket found to match RR [1][xxx.xxx.xxx.xxx:5060]   When I try looking on the mailing list there are no other similar posts, could you please shed some light on what maybe causing this please.   Regards,   Brian Southworth _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users   -------------- next part -------------- An HTML attachment was scrubbed... URL: From razvan at opensips.org Fri Feb 2 06:50:39 2018 From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=) Date: Fri, 2 Feb 2018 13:50:39 +0200 Subject: [OpenSIPS-Users] Potential memory leak in siptrace module In-Reply-To: <1D7B55AB4979E6498B3AC325C40E84360102C18F04@ORD2MBX02F.mex05.mlsrvr.com> References: <1D7B55AB4979E6498B3AC325C40E84360102C18BB6@ORD2MBX02F.mex05.mlsrvr.com> <67b64a89-f845-03dd-6f67-b95226b815c2@opensips.org> <1D7B55AB4979E6498B3AC325C40E84360102C18F04@ORD2MBX02F.mex05.mlsrvr.com> Message-ID: <9acd06b5-5864-1f60-2354-f8d4ce8d886e@opensips.org> Hi, Steve! We actually found out a memory leak related to transactions tracing. Unfortunately the fix is not that easy, so it will take a while until we figure out the right way to solve this. I will keep you updated. Best regards, Răzvan On 02/01/2018 07:30 PM, Steve Brisson wrote: > Hi, > > In answer to your questions… > > 1. > > $ opensips -V > > version: opensips 2.3.2 (x86_64/linux) > > flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, > F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT > > ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, > MAX_URI_SIZE 1024, BUF_SIZE 65535 > > poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. > > main.c compiled on 15:38:40 Nov 24 2017 with gcc 5.4.0 > > I built from commit 9e497ea4ab03c67cb073bdf0dccef2b16bff80e2. I've since > diff'd with the HEAD and I haven't seen any changes in the siptrace code > that would have helped my particular issue. > > 2. > > The call to sip_trace is in the main request route branch and it gets > called for all requests. > > input parameters: > > sip_trace("tid", "d", "sip"); > > module config: > > modparam("siptrace", "trace_on", 0) > > modparam("siptrace", "trace_id", > "[tid]uri=mysql://db-user:db-pass at db-host/db-name;table=sip_trace;") > > *From:*Bogdan-Andrei Iancu [mailto:bogdan at opensips.org] > *Sent:* Thursday, February 1, 2018 10:54 AM > *To:* OpenSIPS users mailling list ; Steve > Brisson > *Subject:* Re: [OpenSIPS-Users] Potential memory leak in siptrace module > > Hi Steve, > > Thank you for your report. Could you provide the exact revision of your > OpenSIPS (opensips -V) ? > > Also what requests are you calling the sip_trace() for (INVITES ? > PUBLISH? all?) and what are the input paramters ? > > Thanks, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > > http://www.opensips-solutions.com > > OpenSIPS Summit 2018 > > http://www.opensips.org/events/Summit-2018Amsterdam > > On 01/31/2018 07:48 PM, Steve Brisson wrote: > > I've found that when I call the sip_trace method, in the main > request routing block, the shmem used_size and fragments continually > increases until the process crashes from an out of memory condition. > > By enabling memory alloc debugging, I confirmed that the leaked > allocation is the shm_malloc of a trace_info_t structure in the > sip_trace_w method [siptrace.c:1539]. This piece of memory is > supposed to be freed in the dialog terminated callback but because > of the logic in the sip_trace_w method, trace_flags gets modified to > TRACE_TRANSACTION and the dialog callbacks are never registered. > > I haven't worked out the details of a proper fix yet. My workaround > has been to remove the siptrace module from my config. > > OpenSIPS config details: > > opensips 2.3.2 (x86_64/linux) > > #siptrace module config > > modparam("siptrace", "trace_on", 0) > > modparam("siptrace", "trace_id", > "[tid]uri=mysql://db-user:db-pass at db-host/db-name;table=sip_trace;") > > #siptrace method call in main request branch > > sip_trace("tid", "d", "sip"); > > > > > _______________________________________________ > > Users mailing list > > Users at lists.opensips.org > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -- Răzvan Crainea OpenSIPS Core Developer http://www.opensips-solutions.com OpenSIPS Summit 2018 http://www.opensips.org/events/Summit-2018Amsterdam From bogdan at opensips.org Fri Feb 2 09:17:38 2018 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Fri, 2 Feb 2018 16:17:38 +0200 Subject: [OpenSIPS-Users] Handling 488 on a leg B In-Reply-To: <429c01d39c14$087d3a60$1977af20$@gmail.com> References: <000a01d399c9$c4ccaae0$4e6600a0$@gmail.com> <1f74b65d-5ac7-7bcd-653c-c419882785f0@opensips.org> <424a01d39b72$252783c0$6f768b40$@gmail.com> <429c01d39c14$087d3a60$1977af20$@gmail.com> Message-ID: Hi Dmitry, In order to push per-branch changes (changes which are specific only to a certainbranch and not to all of them), you need to use the branch route- see http://www.opensips.org/Documentation/Script-Routes-2-3#toc2 So, put the code for changing the body via rtpengine into a branch route. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Summit 2018 http://www.opensips.org/events/Summit-2018Amsterdam On 02/02/2018 12:53 PM, Dmitry wrote: > > Hi Bogdan, > > I change both of them. The first INVITE and the second (after 488). > > The first apply successfully, but the second (that in the > failure_route) does not. > > My script looks like this: > > route { > > … > > t_on_reply("ONREPLY”); > > if (isbflagset(NAT_BFLAG) && af!=INET6) { > > force_rport(); # add rport to first line Via: > > route(RTPENGINE_OFFER); # make offer > > } > > … > > t_relay(); > > … > > } > > route[RTPENGINE_OFFER] { > > … > > $avp(rtp_profile) = "trust-address replace-origin > replace-session-connection ICE=remove DTLS=off RTP/SAVP"; > > if > (rtpengine_offer("$avp(rtp_profile)","$avp(rtp_socket)","$avp(rtp_body)")) > { > > remove_body_part(); > > add_body_part("$avp(rtp_body)", "application/sdp"); # first changing > leg-B - successfully > > } > > … > > } > > onreply_route[ONREPLY] { > > … > > t_on_reply("RTPENGINE_ANSWER"); # positive > > t_on_failure("RTPENGINE_488"); # negative > > … > > } > > failure_route[RTPENGINE_488] { > > … > > if (t_check_status("488") ) { > > # try to change SDP in leg-B after 488 received - unsuccessfully > > # it returns: ERROR:rtpengine:rtpe_function_call: can't extract body > from the message. > > # from which message? > > if (rtpengine_offer("trust-address replace-origin > replace-session-connection ICE=remove DTLS=off > RTP/AVP","$avp(rtp_socket)","$avp(rtp_body)")) { > > remove_body_part(); > > add_body_part("$avp(rtp_body)", "application/sdp"); > > } > > # > > # just replace with re.subst > > $avp(rtp_body) = $(avp(rtp_body){re.subst,/SAVP/AVP/g}); > > $avp(rtp_body) = $(avp(rtp_body){re.subst,/a=crypto.*//g}); > > # if I use remove_body_part(), then add_body_part() is not adds > content. INVITE sends without body. > > remove_body_part(); > > # if I don’t use remove_body_part(), then INVITE contains TWO sdp body > with RTP/SAVP and RTP/AVP > > add_body_part("$avp(rtp_body)", "application/sdp"); > > # > > xlog("rtp_body=$avp(rtp_body)\n"); > > # xlog for $avp(rtp_body) shows unchanged content (with RTP/SAVP) > > … > > t_relay(); > > } > > … > > } > > *From:*Bogdan-Andrei Iancu [mailto:bogdan at opensips.org] > *Sent:* Thursday, February 1, 2018 7:51 PM > *To:* Dmitry ; 'OpenSIPS users mailling list' > > *Subject:* Re: [OpenSIPS-Users] Handling 488 on a leg B > > Hi Dmitry, > > In failure route you re-process the original request (INVITE) and not > the 488 reply - the reply is just the trigger. > Do you change the INVITE SDP before the first attempt (which ends with > 488) ? Or you change it for the first time in the failure route ? > > Regards, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > OpenSIPS Summit 2018 > http://www.opensips.org/events/Summit-2018Amsterdam > > On 02/01/2018 05:34 PM, Dmitry wrote: > > Hello Bogdan, > > Thank you for advice. > > I catched 488 and can send INVITE from failure_route, but I can’t > modify body inside it. > > The main goal is change SAVP profile to AVP when 488 received. > > I tried to do it via rtpengine_offer function. It doesn’t work. I > guess, it try to read the body from response (488 in my case) but > not from request. > > I tried to rewrite body via remove_body_part and add_body_part > functions, it also doesn’t work. > > So, I don’t understand how exactly rewrite body in failure_route > before do t_relay(). > > In addition, I tried making a new branch (via append_branch > function and via branch_route). It just duplicate my re-INVITE > without changing the body. > > Could you please drop a hint (example on few lines of code) how to > do it? > > Thanks. > > *From:*Bogdan-Andrei Iancu [mailto:bogdan at opensips.org] > *Sent:* Wednesday, January 31, 2018 6:55 PM > *To:* OpenSIPS users mailling list > ; Dmitry > > *Subject:* Re: [OpenSIPS-Users] Handling 488 on a leg B > > Hello Dmitry, > > What you want to do is called (in SIP) serial forking. Use > failure_route{} to catch the 488 and to add a create a new branch > (with modified body) . > > See: http://www.opensips.org/Documentation/Script-Routes-2-3#toc3 > > Regards, > > > Bogdan-Andrei Iancu > > > > OpenSIPS Founder and Developer > > http://www.opensips-solutions.com > > OpenSIPS Summit 2018 > > http://www.opensips.org/events/Summit-2018Amsterdam > > On 01/30/2018 02:56 PM, Dmitry wrote: > > Hello all, > > Could you say how can I re-send INVITE with changed SDP to > B-leg after receive 488 from it? > > Call flow: > > -> A-leg INVITE > > <- A-leg 100 > > -> B-leg INVITE > > <- B-leg 100 > > <- B-leg 488 – I catch it, and want to send INVITE again with > properly SDP > > -> B-leg ACK > > <-A-leg 488 > > -> B-leg ACK > > Thanks for any advice. > > Cheers! > > > > > > _______________________________________________ > > Users mailing list > > Users at lists.opensips.org > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Fri Feb 2 09:19:53 2018 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Fri, 2 Feb 2018 16:19:53 +0200 Subject: [OpenSIPS-Users] [15066] WARNING:rr:after_strict: no socket found to match RR [1][XXX.XXX.XXX.XXX:5060] In-Reply-To: <01020161564f88ab-94ee6ed7-b89e-4d6e-866d-99d682719034-000000@eu-west-1.amazonses.com> References: <0102016155cbc5e4-c3c66a50-ab65-4987-803e-cd61fbfd5647-000000@eu-west-1.amazonses.com> <01020161564f88ab-94ee6ed7-b89e-4d6e-866d-99d682719034-000000@eu-west-1.amazonses.com> Message-ID: Hi Brian, Maybe thatwarning points toa routing error that prevents the REFER to be route to carrier - make a sip capture to be sure the REFER from A is properly routed and accepted by the carrier. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Summit 2018 http://www.opensips.org/events/Summit-2018Amsterdam On 02/02/2018 01:38 PM, Brian Southworth wrote: > > Hi Bogdan, > > Thank you very much, so this doesn’t have any impact on why the call > being transferred are dropped ? > > I am trying to transfer a call using the refer method as that is what > the cisco phones use. > > The network is setup like so opensips proxy àasterisk gateway àcarrier > > Scenario: > > Inbound call comes into the phone like so: carrier àast àproxy àphone A > > Phone A needs to transfer call to phone B: Phone A dials phone B > àphone B picks up àphone A presses xfer button and call is dropped. > > Any help would be appreciated. > > Regards, > > Brian Southworth > > *From:*Bogdan-Andrei Iancu [mailto:bogdan at opensips.org] > *Sent:* 02 February 2018 11:29 > *To:* OpenSIPS users mailling list ; Brian > Southworth > *Subject:* Re: [OpenSIPS-Users] [15066] WARNING:rr:after_strict: no > socket found to match RR [1][XXX.XXX.XXX.XXX:5060] > > Hi Brian, > > That warning means OpenSIPS found a Route header (while doing > loose_route) that is suppose to be of its own, but the network > information from the header does not match any of the OpenSIPS SIP > listeners. > > Best regards, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > OpenSIPS Summit 2018 > http://www.opensips.org/events/Summit-2018Amsterdam > > On 02/02/2018 11:14 AM, Brian Southworth wrote: > > I get this when trying to transfer calls using the B2BUA: > > [15066] WARNING:rr:after_strict: no socket found to match RR > [1][xxx.xxx.xxx.xxx:5060] > > When I try looking on the mailing list there are no other similar > posts, could you please shed some light on what maybe causing this > please. > > Regards, > > Brian Southworth > > > > > _______________________________________________ > > Users mailing list > > Users at lists.opensips.org > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -------------- next part -------------- An HTML attachment was scrubbed... URL: From brian.southworth at clocom.uk Fri Feb 2 09:27:08 2018 From: brian.southworth at clocom.uk (=?UTF-8?Q?Brian_Southworth?=) Date: Fri, 2 Feb 2018 14:27:08 +0000 Subject: [OpenSIPS-Users] [15066] WARNING:rr:after_strict: no socket found to match RR [1][XXX.XXX.XXX.XXX:5060] In-Reply-To: References: <0102016155cbc5e4-c3c66a50-ab65-4987-803e-cd61fbfd5647-000000@eu-west-1.amazonses.com> <01020161564f88ab-94ee6ed7-b89e-4d6e-866d-99d682719034-000000@eu-west-1.amazonses.com> Message-ID: <0102016156e9f5f9-6a281c81-ae20-4a1d-9362-2e98a1fff3e0-000000@eu-west-1.amazonses.com> Maybe I am doing this wrong but I wanted the B2BUA module to handle the refer and bridge the calls. I have the B2bUA working now. However my issue is that its not able to send the replies.   incoming reply b2b_reply (B2B.222.7591351.1517580641) Feb  2 14:10:47 [22664] ERROR:tm:_reply_light: failed to generate 408 reply when a final 200 was sent out Feb  2 14:10:47 [22664] ERROR:b2b_entities:b2b_send_reply: failed to send reply with tm Feb  2 14:10:47 [22664] ERROR:b2b_logic:b2b_logic_notify_reply: Sending reply failed - 408, [B2B.452.342.1517580641]   Do you need anything else to help me debug this ? I am not sure why its failing to pass the reply with tm, I have enabled the param: modparam("tm", "pass_provisional_replies", 1)   I should also note that I am using the load balancer module also. This normally deals with all call distribution. In and out.   Regards,   Brian Southworth   From: Bogdan-Andrei Iancu [mailto:bogdan at opensips.org] Sent: 02 February 2018 14:20 To: Brian Southworth ; OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] [15066] WARNING:rr:after_strict: no socket found to match RR [1][XXX.XXX.XXX.XXX:5060]   Hi Brian, Maybe that warning points to a routing error that prevents the REFER to be route to carrier - make a sip capture to be sure the REFER from A is properly routed and accepted by the carrier. Regards, Bogdan-Andrei Iancu   OpenSIPS Founder and Developer   http://www.opensips-solutions.com OpenSIPS Summit 2018   http://www.opensips.org/events/Summit-2018Amsterdam On 02/02/2018 01:38 PM, Brian Southworth wrote: Hi Bogdan,   Thank you very much, so this doesn’t have any impact on why the call being transferred are dropped ?   I am trying to transfer a call using the refer method as that is what the cisco phones use.   The network is setup like so opensips proxy à asterisk gateway à carrier   Scenario:   Inbound call comes into the phone like so: carrier à ast à proxy à phone A Phone A needs to transfer call to phone B: Phone A dials phone B à phone B picks up à phone A presses xfer button and call is dropped.   Any help would be appreciated.   Regards,   Brian Southworth   From: Bogdan-Andrei Iancu [mailto:bogdan at opensips.org ] Sent: 02 February 2018 11:29 To: OpenSIPS users mailling list ; Brian Southworth Subject: Re: [OpenSIPS-Users] [15066] WARNING:rr:after_strict: no socket found to match RR [1][XXX.XXX.XXX.XXX:5060]   Hi Brian, That warning means OpenSIPS found a Route header (while doing loose_route) that is suppose to be of its own, but the network information from the header does not match any of the OpenSIPS SIP listeners. Best regards, Bogdan-Andrei Iancu   OpenSIPS Founder and Developer   http://www.opensips-solutions.com OpenSIPS Summit 2018   http://www.opensips.org/events/Summit-2018Amsterdam On 02/02/2018 11:14 AM, Brian Southworth wrote: I get this when trying to transfer calls using the B2BUA: [15066] WARNING:rr:after_strict: no socket found to match RR [1][xxx.xxx.xxx.xxx:5060]   When I try looking on the mailing list there are no other similar posts, could you please shed some light on what maybe causing this please.   Regards,   Brian Southworth _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users     -------------- next part -------------- An HTML attachment was scrubbed... URL: From vishalmpai at gmail.com Fri Feb 2 10:17:40 2018 From: vishalmpai at gmail.com (Vishal Pai) Date: Fri, 02 Feb 2018 15:17:40 +0000 Subject: [OpenSIPS-Users] Upgrade opensips Message-ID: Hello all, I want to upgrade my existing system developed on opensips version 1.10 to opensips version 2.2. What is best possible to do this without any issue. -------------- next part -------------- An HTML attachment was scrubbed... URL: From netaskd at gmail.com Fri Feb 2 11:00:42 2018 From: netaskd at gmail.com (Dmitry) Date: Fri, 2 Feb 2018 19:00:42 +0300 Subject: [OpenSIPS-Users] Handling 488 on a leg B In-Reply-To: References: <000a01d399c9$c4ccaae0$4e6600a0$@gmail.com> <1f74b65d-5ac7-7bcd-653c-c419882785f0@opensips.org> <424a01d39b72$252783c0$6f768b40$@gmail.com> <429c01d39c14$087d3a60$1977af20$@gmail.com> Message-ID: <42e101d39c3e$fb3646a0$f1a2d3e0$@gmail.com> Hi Bogdan, Thank you indeed for the help! After moving the first changing body to branch (and second too), it works perfect. From: Bogdan-Andrei Iancu [mailto:bogdan at opensips.org] Sent: Friday, February 2, 2018 5:18 PM To: Dmitry ; 'OpenSIPS users mailling list' Subject: Re: [OpenSIPS-Users] Handling 488 on a leg B Hi Dmitry, In order to push per-branch changes (changes which are specific only to a certain branch and not to all of them), you need to use the branch route - see http://www.opensips.org/Documentation/Script-Routes-2-3#toc2 So, put the code for changing the body via rtpengine into a branch route. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Summit 2018 http://www.opensips.org/events/Summit-2018Amsterdam On 02/02/2018 12:53 PM, Dmitry wrote: Hi Bogdan, I change both of them. The first INVITE and the second (after 488). The first apply successfully, but the second (that in the failure_route) does not. My script looks like this: route { … t_on_reply("ONREPLY”); if (isbflagset(NAT_BFLAG) && af!=INET6) { force_rport(); # add rport to first line Via: route(RTPENGINE_OFFER); # make offer } … t_relay(); … } route[RTPENGINE_OFFER] { … $avp(rtp_profile) = "trust-address replace-origin replace-session-connection ICE=remove DTLS=off RTP/SAVP"; if (rtpengine_offer("$avp(rtp_profile)","$avp(rtp_socket)","$avp(rtp_body)")) { remove_body_part(); add_body_part("$avp(rtp_body)", "application/sdp"); # first changing leg-B - successfully } … } onreply_route[ONREPLY] { … t_on_reply("RTPENGINE_ANSWER"); # positive t_on_failure("RTPENGINE_488"); # negative … } failure_route[RTPENGINE_488] { … if (t_check_status("488") ) { # try to change SDP in leg-B after 488 received - unsuccessfully # it returns: ERROR:rtpengine:rtpe_function_call: can't extract body from the message. # from which message? if (rtpengine_offer("trust-address replace-origin replace-session-connection ICE=remove DTLS=off RTP/AVP","$avp(rtp_socket)","$avp(rtp_body)")) { remove_body_part(); add_body_part("$avp(rtp_body)", "application/sdp"); } # # just replace with re.subst $avp(rtp_body) = $(avp(rtp_body){re.subst,/SAVP/AVP/g}); $avp(rtp_body) = $(avp(rtp_body){re.subst,/a=crypto.*//g}); # if I use remove_body_part(), then add_body_part() is not adds content. INVITE sends without body. remove_body_part(); # if I don’t use remove_body_part(), then INVITE contains TWO sdp body with RTP/SAVP and RTP/AVP add_body_part("$avp(rtp_body)", "application/sdp"); # xlog("rtp_body=$avp(rtp_body)\n"); # xlog for $avp(rtp_body) shows unchanged content (with RTP/SAVP) … t_relay(); } … } From: Bogdan-Andrei Iancu [mailto:bogdan at opensips.org] Sent: Thursday, February 1, 2018 7:51 PM To: Dmitry ; 'OpenSIPS users mailling list' Subject: Re: [OpenSIPS-Users] Handling 488 on a leg B Hi Dmitry, In failure route you re-process the original request (INVITE) and not the 488 reply - the reply is just the trigger. Do you change the INVITE SDP before the first attempt (which ends with 488) ? Or you change it for the first time in the failure route ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Summit 2018 http://www.opensips.org/events/Summit-2018Amsterdam On 02/01/2018 05:34 PM, Dmitry wrote: Hello Bogdan, Thank you for advice. I catched 488 and can send INVITE from failure_route, but I can’t modify body inside it. The main goal is change SAVP profile to AVP when 488 received. I tried to do it via rtpengine_offer function. It doesn’t work. I guess, it try to read the body from response (488 in my case) but not from request. I tried to rewrite body via remove_body_part and add_body_part functions, it also doesn’t work. So, I don’t understand how exactly rewrite body in failure_route before do t_relay(). In addition, I tried making a new branch (via append_branch function and via branch_route). It just duplicate my re-INVITE without changing the body. Could you please drop a hint (example on few lines of code) how to do it? Thanks. From: Bogdan-Andrei Iancu [mailto:bogdan at opensips.org] Sent: Wednesday, January 31, 2018 6:55 PM To: OpenSIPS users mailling list ; Dmitry Subject: Re: [OpenSIPS-Users] Handling 488 on a leg B Hello Dmitry, What you want to do is called (in SIP) serial forking. Use failure_route{} to catch the 488 and to add a create a new branch (with modified body) . See: http://www.opensips.org/Documentation/Script-Routes-2-3#toc3 Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Summit 2018 http://www.opensips.org/events/Summit-2018Amsterdam On 01/30/2018 02:56 PM, Dmitry wrote: Hello all, Could you say how can I re-send INVITE with changed SDP to B-leg after receive 488 from it? Call flow: -> A-leg INVITE <- A-leg 100 -> B-leg INVITE <- B-leg 100 <- B-leg 488 – I catch it, and want to send INVITE again with properly SDP -> B-leg ACK <-A-leg 488 -> B-leg ACK Thanks for any advice. Cheers! _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Fri Feb 2 11:49:30 2018 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Fri, 2 Feb 2018 18:49:30 +0200 Subject: [OpenSIPS-Users] [15066] WARNING:rr:after_strict: no socket found to match RR [1][XXX.XXX.XXX.XXX:5060] In-Reply-To: <0102016156e9f5f9-6a281c81-ae20-4a1d-9362-2e98a1fff3e0-000000@eu-west-1.amazonses.com> References: <0102016155cbc5e4-c3c66a50-ab65-4987-803e-cd61fbfd5647-000000@eu-west-1.amazonses.com> <01020161564f88ab-94ee6ed7-b89e-4d6e-866d-99d682719034-000000@eu-west-1.amazonses.com> <0102016156e9f5f9-6a281c81-ae20-4a1d-9362-2e98a1fff3e0-000000@eu-west-1.amazonses.com> Message-ID: <3615c9bb-1619-dd03-4354-c511a9e68dd0@opensips.org> I'm a bit confused. The original report was on a record_route() / loose_route() matter. But you say you have opensips as B2B, so the RR mechanism must not be used in such a case- you act either as a end-point, either as a proxy - you cannot be both for the same call. Now you have this b2b error, during a call transfer scenario. and you mentioned LB also :)...so I'm a bit confused - could please try to put all these pieces together, so I can understand what you are doing? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Summit 2018 http://www.opensips.org/events/Summit-2018Amsterdam On 02/02/2018 04:27 PM, Brian Southworth wrote: > > Maybe I am doing this wrong but I wanted the B2BUA module to handle > the refer and bridge the calls. > > I have the B2bUA working now. However my issue is that its not able to > send the replies. > > incoming reply > > b2b_reply (B2B.222.7591351.1517580641) > > Feb 2 14:10:47 [22664] ERROR:tm:_reply_light: failed to generate 408 > reply when a final 200 was sent out > > Feb 2 14:10:47 [22664] ERROR:b2b_entities:b2b_send_reply: failed to > send reply with tm > > Feb 2 14:10:47 [22664] ERROR:b2b_logic:b2b_logic_notify_reply: > Sending reply failed - 408, [B2B.452.342.1517580641] > > Do you need anything else to help me debug this ? I am not sure why > its failing to pass the reply with tm, I have enabled the param: > > modparam("tm","pass_provisional_replies",1) > > I should also note that I am using the load balancer module also. This > normally deals with all call distribution. In and out. > > Regards, > > Brian Southworth > > *From:*Bogdan-Andrei Iancu [mailto:bogdan at opensips.org] > *Sent:* 02 February 2018 14:20 > *To:* Brian Southworth ; OpenSIPS users > mailling list > *Subject:* Re: [OpenSIPS-Users] [15066] WARNING:rr:after_strict: no > socket found to match RR [1][XXX.XXX.XXX.XXX:5060] > > Hi Brian, > > Maybe that warning points to a routing error that prevents the REFER > to be route to carrier - make a sip capture to be sure the REFER from > A is properly routed and accepted by the carrier. > > Regards, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > OpenSIPS Summit 2018 > http://www.opensips.org/events/Summit-2018Amsterdam > > On 02/02/2018 01:38 PM, Brian Southworth wrote: > > Hi Bogdan, > > Thank you very much, so this doesn’t have any impact on why the > call being transferred are dropped ? > > I am trying to transfer a call using the refer method as that is > what the cisco phones use. > > The network is setup like so opensips proxy àasterisk gateway àcarrier > > Scenario: > > Inbound call comes into the phone like so: carrier àast àproxy > àphone A > > Phone A needs to transfer call to phone B: Phone A dials phone B > àphone B picks up àphone A presses xfer button and call is dropped. > > Any help would be appreciated. > > Regards, > > Brian Southworth > > *From:*Bogdan-Andrei Iancu [mailto:bogdan at opensips.org] > *Sent:* 02 February 2018 11:29 > *To:* OpenSIPS users mailling list > ; Brian Southworth > > *Subject:* Re: [OpenSIPS-Users] [15066] WARNING:rr:after_strict: > no socket found to match RR [1][XXX.XXX.XXX.XXX:5060] > > Hi Brian, > > That warning means OpenSIPS found a Route header (while doing > loose_route) that is suppose to be of its own, but the network > information from the header does not match any of the OpenSIPS SIP > listeners. > > Best regards, > > > Bogdan-Andrei Iancu > > > > OpenSIPS Founder and Developer > > http://www.opensips-solutions.com > > OpenSIPS Summit 2018 > > http://www.opensips.org/events/Summit-2018Amsterdam > > On 02/02/2018 11:14 AM, Brian Southworth wrote: > > I get this when trying to transfer calls using the B2BUA: > > [15066] WARNING:rr:after_strict: no socket found to match RR > [1][xxx.xxx.xxx.xxx:5060] > > When I try looking on the mailing list there are no other > similar posts, could you please shed some light on what maybe > causing this please. > > Regards, > > Brian Southworth > > > > > > _______________________________________________ > > Users mailing list > > Users at lists.opensips.org > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -------------- next part -------------- An HTML attachment was scrubbed... URL: From brian.southworth at clocom.uk Fri Feb 2 12:03:28 2018 From: brian.southworth at clocom.uk (=?UTF-8?Q?Brian_Southworth?=) Date: Fri, 2 Feb 2018 17:03:28 +0000 Subject: [OpenSIPS-Users] [15066] WARNING:rr:after_strict: no socket found to match RR [1][XXX.XXX.XXX.XXX:5060] In-Reply-To: <3615c9bb-1619-dd03-4354-c511a9e68dd0@opensips.org> References: <0102016155cbc5e4-c3c66a50-ab65-4987-803e-cd61fbfd5647-000000@eu-west-1.amazonses.com> <01020161564f88ab-94ee6ed7-b89e-4d6e-866d-99d682719034-000000@eu-west-1.amazonses.com> <0102016156e9f5f9-6a281c81-ae20-4a1d-9362-2e98a1fff3e0-000000@eu-west-1.amazonses.com> <3615c9bb-1619-dd03-4354-c511a9e68dd0@opensips.org> Message-ID: <01020161577912d4-87a753fa-ece7-491a-98cb-4d8b9f479e22-000000@eu-west-1.amazonses.com> Sorry my apologies.   So from the beginning opensips acts as an authorization proxy which passes the call on to an asterisk box based on load (using load balancer). I am trying to get the opensips proxy to handle call transfers and I thought the b2bua would be the best way. Initially the refer was sent to the asterisk box.   On inbound calls The call comes in from the carrier goes to asterisk, asterisk then passes the sip invite to the proxy which then rings the sip phone.   What I wish to achieve is a way to transfer an inbound call to an internal extension or external number.   Example: Caller A receives call à caller A places call on hold and dials caller B à caller B picks up à caller A presses cisco xfer and call is passed to caller B   I was hoping to achieve this using the proxy or asterisk box if possible.   I hope this helps.   Regards,   Brian Southworth   From: Bogdan-Andrei Iancu [mailto:bogdan at opensips.org] Sent: 02 February 2018 16:50 To: Brian Southworth ; OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] [15066] WARNING:rr:after_strict: no socket found to match RR [1][XXX.XXX.XXX.XXX:5060]   I'm a bit confused. The original report was on a record_route() / loose_route() matter. But you say you have opensips as B2B, so the RR mechanism must not be used in such a case - you act either as a end-point, either as a proxy - you cannot be both for the same call. Now you have this b2b error, during a call transfer scenario. and you mentioned LB also :)...so I'm a bit confused - could please try to put all these pieces together, so I can understand what you are doing ? Regards, Bogdan-Andrei Iancu   OpenSIPS Founder and Developer   http://www.opensips-solutions.com OpenSIPS Summit 2018   http://www.opensips.org/events/Summit-2018Amsterdam On 02/02/2018 04:27 PM, Brian Southworth wrote: Maybe I am doing this wrong but I wanted the B2BUA module to handle the refer and bridge the calls. I have the B2bUA working now. However my issue is that its not able to send the replies.   incoming reply b2b_reply (B2B.222.7591351.1517580641) Feb  2 14:10:47 [22664] ERROR:tm:_reply_light: failed to generate 408 reply when a final 200 was sent out Feb  2 14:10:47 [22664] ERROR:b2b_entities:b2b_send_reply: failed to send reply with tm Feb  2 14:10:47 [22664] ERROR:b2b_logic:b2b_logic_notify_reply: Sending reply failed - 408, [B2B.452.342.1517580641]   Do you need anything else to help me debug this ? I am not sure why its failing to pass the reply with tm, I have enabled the param: modparam("tm", "pass_provisional_replies", 1)   I should also note that I am using the load balancer module also. This normally deals with all call distribution. In and out.   Regards,   Brian Southworth   From: Bogdan-Andrei Iancu [mailto:bogdan at opensips.org ] Sent: 02 February 2018 14:20 To: Brian Southworth ; OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] [15066] WARNING:rr:after_strict: no socket found to match RR [1][XXX.XXX.XXX.XXX:5060]   Hi Brian, Maybe that warning points to a routing error that prevents the REFER to be route to carrier - make a sip capture to be sure the REFER from A is properly routed and accepted by the carrier. Regards, Bogdan-Andrei Iancu   OpenSIPS Founder and Developer   http://www.opensips-solutions.com OpenSIPS Summit 2018   http://www.opensips.org/events/Summit-2018Amsterdam On 02/02/2018 01:38 PM, Brian Southworth wrote: Hi Bogdan,   Thank you very much, so this doesn’t have any impact on why the call being transferred are dropped ?   I am trying to transfer a call using the refer method as that is what the cisco phones use.   The network is setup like so opensips proxy à asterisk gateway à carrier   Scenario:   Inbound call comes into the phone like so: carrier à ast à proxy à phone A Phone A needs to transfer call to phone B: Phone A dials phone B à phone B picks up à phone A presses xfer button and call is dropped.   Any help would be appreciated.   Regards,   Brian Southworth   From: Bogdan-Andrei Iancu [mailto:bogdan at opensips.org ] Sent: 02 February 2018 11:29 To: OpenSIPS users mailling list ; Brian Southworth Subject: Re: [OpenSIPS-Users] [15066] WARNING:rr:after_strict: no socket found to match RR [1][XXX.XXX.XXX.XXX:5060]   Hi Brian, That warning means OpenSIPS found a Route header (while doing loose_route) that is suppose to be of its own, but the network information from the header does not match any of the OpenSIPS SIP listeners. Best regards, Bogdan-Andrei Iancu   OpenSIPS Founder and Developer   http://www.opensips-solutions.com OpenSIPS Summit 2018   http://www.opensips.org/events/Summit-2018Amsterdam On 02/02/2018 11:14 AM, Brian Southworth wrote: I get this when trying to transfer calls using the B2BUA: [15066] WARNING:rr:after_strict: no socket found to match RR [1][xxx.xxx.xxx.xxx:5060]   When I try looking on the mailing list there are no other similar posts, could you please shed some light on what maybe causing this please.   Regards,   Brian Southworth _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users       -------------- next part -------------- An HTML attachment was scrubbed... URL: From royee.tichauer at vonage.com Mon Feb 5 07:09:00 2018 From: royee.tichauer at vonage.com (Royee Tichauer) Date: Mon, 05 Feb 2018 12:09:00 +0000 Subject: [OpenSIPS-Users] Registration Loss on opensips SBC Message-ID: Hi, We are using opensips 2.1 as an SBC component and we see that we have loss of registrations happening once in a while. When we captured SIP we saw that some messages are being delayed by a couple of seconds. We enabled 'exec_execute_message=50000' and saw in the logs that opensips core methods like 't_realy' and 'save' . I looked at SAR during this time and there are no indications of CPU running high during this time. Can you help me figure out this issue? Thanks, Royee Logs: Feb 4 17:07:23 sbc.company.com /usr/sbin/opensips[29827]: WARNING:core:log_expiry: threshold exceeded : msg processing took too long - 8912776 us.Source : SIP/2.0 200 OK#015#012Via: SIP/2.0/UDP 5060;received=;rport=5060;branch=z9hG4bK7eec.23d21053.0#015#012Via: SIP/2.0/UDP :5060;rport=5060;received=;branch=z9hG4bK2715962793#015#012From: "" @.company.com:5060 >;tag=646749010#015#012To: "###" >;tag=8b4547f60a700f827fe775c80b65be1b.efed#015#012Call-ID: 0_2258842442@#015#012CSeq : 3575 REGISTER#015#012Contact: :5060 >;expires=45;received="sip: :5060 "#015#012Server: Vonage#015#012Content-Length: 0#015#012#015#012 Feb 4 17:07:23 sbc.amz1.company.com /usr/sbin/opensips[29827]: WARNING:core:log_expiry: #1 is a module action : save - 8912725us - line 914 Feb 4 17:07:23 sbc.amz1.company.com /usr/sbin/opensips[29827]: WARNING:core:log_expiry: #2 is a core action : 80 - 17us - line 1103 Feb 4 17:07:23 sbc.amz1.company.com /usr/sbin/opensips[29827]: WARNING:core:log_expiry: #3 is a core action : 72 - 5us - line 887 Feb 4 17:07:23 sbc.amz1.company.com /usr/sbin/opensips[29820]: WARNING:core:log_expiry: threshold exceeded : msg processing took too long - 5355695 us.Source : REGISTER sip:####.accounts.company.com:5060 SIP/2.0#015#012Via: SIP/2.0/UDP :5060;branch=z9hG4bK2715962793#015#012From: "###" < sip:#####@sip-####.accounts.company.com:5060 >;tag=646749010#015#012To: "####" >#015#012Call-ID: 0_2258842442@######015#012CSeq : 3575 REGISTER#015#012Contact: <####@:5060 >#015#012Authorization: Digest username="####", realm="####.accounts.company.com ", nonce="5a773de000000b19dc7d21205f1edfda504fac90296b04fb", uri="sip: sip-#######.accounts.company.com:5060 ", response="df8d6cbfd178a36913a1464920da3f06", algorithm=MD5#015#012Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE#015#012Max-Forwards: 69#015#012User-Agent: #########015#012Expires: 3600#015#012Allow-Events: talk,hold,conference,refer,check-sync#015#012Content-Length: 0#015#012#015#012 Feb 4 17:07:23 sbc.amz1.company.com /usr/sbin/opensips[29827]: WARNING:core:log_expiry: #4 is a core action : 72 - 5us - line 1086 Feb 4 17:07:23 sbc.amz1.company.com /usr/sbin/opensips[29820]: WARNING:core:log_expiry: #1 is a module action : t_relay - 5355333us - line 465 Feb 4 17:07:23 sbc.amz1.company.com /usr/sbin/opensips[29827]: WARNING:core:log_expiry: #5 is a module action : ds_is_in_list - 4us - line 943 Feb 4 17:07:23 sbc.amz1.company.com /usr/sbin/opensips[29817]: WARNING:core:log_expiry: #1 is a module action : t_relay - 7375370us - line 465 Feb 4 17:07:23 sbc.amz1.company.com /usr/sbin/opensips[29820]: WARNING:core:log_expiry: #2 is a core action : 78 - 105us - line 604 Feb 4 17:07:23 sbc.amz1.company.com /usr/sbin/opensips[29817]: WARNING:core:log_expiry: #2 is a core action : 78 - 100us - line 604 Feb 4 17:07:23 sbc.amz1.company.com /usr/sbin/opensips[29820]: WARNING:core:log_expiry: #3 is a core action : 78 - 99us - line 635 Feb 4 17:07:23 sbc.amz1.company.com /usr/sbin/opensips[29817]: WARNING:core:log_expiry: #3 is a core action : 78 - 98us - line 635 -------------- next part -------------- An HTML attachment was scrubbed... URL: From abdoul.osseni at gmail.com Mon Feb 5 09:42:04 2018 From: abdoul.osseni at gmail.com (=?UTF-8?Q?Abdoul_Oss=C3=A9ni?=) Date: Mon, 5 Feb 2018 15:42:04 +0100 Subject: [OpenSIPS-Users] False Answer Supervision Message-ID: Hello guys, Is there a method to detect FAS (False Answer Supervision) in OpenSips ? Regards Abdoul OSSENI Ingénieur DevOps chez Néo-Soft Co-Fondateur de ON SERVICES Tél : +33 601 135 167 -------------- next part -------------- An HTML attachment was scrubbed... URL: From abdoul.osseni at gmail.com Mon Feb 5 10:02:20 2018 From: abdoul.osseni at gmail.com (=?UTF-8?Q?Abdoul_Oss=C3=A9ni?=) Date: Mon, 5 Feb 2018 16:02:20 +0100 Subject: [OpenSIPS-Users] Dectect FAS: False Answer Supervision Message-ID: Hello guys, Is there a method to detect FAS (False Answer Supervision) in OpenSips ? Regards Abdoul OSSENI Ingénieur DevOps chez Néo-Soft Co-Fondateur de ON SERVICES Tél : +33 601 135 167 -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Mon Feb 5 10:47:24 2018 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Mon, 5 Feb 2018 17:47:24 +0200 Subject: [OpenSIPS-Users] [15066] WARNING:rr:after_strict: no socket found to match RR [1][XXX.XXX.XXX.XXX:5060] In-Reply-To: <01020161577912d4-87a753fa-ece7-491a-98cb-4d8b9f479e22-000000@eu-west-1.amazonses.com> References: <0102016155cbc5e4-c3c66a50-ab65-4987-803e-cd61fbfd5647-000000@eu-west-1.amazonses.com> <01020161564f88ab-94ee6ed7-b89e-4d6e-866d-99d682719034-000000@eu-west-1.amazonses.com> <0102016156e9f5f9-6a281c81-ae20-4a1d-9362-2e98a1fff3e0-000000@eu-west-1.amazonses.com> <3615c9bb-1619-dd03-4354-c511a9e68dd0@opensips.org> <01020161577912d4-87a753fa-ece7-491a-98cb-4d8b9f479e22-000000@eu-west-1.amazonses.com> Message-ID: Hi Brian, Keep in mind that you cannot make opensips act in the same time as proxy (as required by the load balancer) and as a end-point (as required by the B2BUA). Ideally is to run the two services (LB and B2B) on two opensips instancesin a chain. Best regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Summit 2018 http://www.opensips.org/events/Summit-2018Amsterdam On 02/02/2018 07:03 PM, Brian Southworth wrote: > > Sorry my apologies. > > So from the beginning opensips acts as an authorization proxy which > passes the call on to an asterisk box based on load (using load balancer). > > I am trying to get the opensips proxy to handle call transfers and I > thought the b2bua would be the best way. Initially the refer was sent > to the asterisk box. > > On inbound calls > > The call comes in from the carrier goes to asterisk, asterisk then > passes the sip invite to the proxy which then rings the sip phone. > > What I wish to achieve is a way to transfer an inbound call to an > internal extension or external number. > > Example: > > Caller A receives call àcaller A places call on hold and dials caller > B àcaller B picks up àcaller A presses cisco xfer and call is passed > to caller B > > I was hoping to achieve this using the proxy or asterisk box if possible. > > I hope this helps. > > Regards, > > Brian Southworth > > *From:*Bogdan-Andrei Iancu [mailto:bogdan at opensips.org] > *Sent:* 02 February 2018 16:50 > *To:* Brian Southworth ; OpenSIPS users > mailling list > *Subject:* Re: [OpenSIPS-Users] [15066] WARNING:rr:after_strict: no > socket found to match RR [1][XXX.XXX.XXX.XXX:5060] > > I'm a bit confused. The original report was on a record_route() / > loose_route() matter. But you say you have opensips as B2B, so the RR > mechanism must not be used in such a case - you act either as a > end-point, either as a proxy - you cannot be both for the same call. > > Now you have this b2b error, during a call transfer scenario. and you > mentioned LB also :)...so I'm a bit confused - could please try to put > all these pieces together, so I can understand what you are doing ? > > Regards, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > OpenSIPS Summit 2018 > http://www.opensips.org/events/Summit-2018Amsterdam > > On 02/02/2018 04:27 PM, Brian Southworth wrote: > > Maybe I am doing this wrong but I wanted the B2BUA module to > handle the refer and bridge the calls. > > I have the B2bUA working now. However my issue is that its not > able to send the replies. > > incoming reply > > b2b_reply (B2B.222.7591351.1517580641) > > Feb 2 14:10:47 [22664] ERROR:tm:_reply_light: failed to generate > 408 reply when a final 200 was sent out > > Feb 2 14:10:47 [22664] ERROR:b2b_entities:b2b_send_reply: failed > to send reply with tm > > Feb 2 14:10:47 [22664] ERROR:b2b_logic:b2b_logic_notify_reply: > Sending reply failed - 408, [B2B.452.342.1517580641] > > Do you need anything else to help me debug this ? I am not sure > why its failing to pass the reply with tm, I have enabled the param: > > modparam("tm", "pass_provisional_replies", 1) > > I should also note that I am using the load balancer module also. > This normally deals with all call distribution. In and out. > > Regards, > > Brian Southworth > > *From:*Bogdan-Andrei Iancu [mailto:bogdan at opensips.org] > *Sent:* 02 February 2018 14:20 > *To:* Brian Southworth > ; OpenSIPS users mailling list > > *Subject:* Re: [OpenSIPS-Users] [15066] WARNING:rr:after_strict: > no socket found to match RR [1][XXX.XXX.XXX.XXX:5060] > > Hi Brian, > > Maybe that warning points to a routing error that prevents the > REFER to be route to carrier - make a sip capture to be sure the > REFER from A is properly routed and accepted by the carrier. > > Regards, > > > Bogdan-Andrei Iancu > > > > OpenSIPS Founder and Developer > > http://www.opensips-solutions.com > > OpenSIPS Summit 2018 > > http://www.opensips.org/events/Summit-2018Amsterdam > > On 02/02/2018 01:38 PM, Brian Southworth wrote: > > Hi Bogdan, > > Thank you very much, so this doesn’t have any impact on why > the call being transferred are dropped ? > > I am trying to transfer a call using the refer method as that > is what the cisco phones use. > > The network is setup like so opensips proxy àasterisk gateway > àcarrier > > Scenario: > > Inbound call comes into the phone like so: carrier àast àproxy > àphone A > > Phone A needs to transfer call to phone B: Phone A dials phone > B àphone B picks up àphone A presses xfer button and call is > dropped. > > Any help would be appreciated. > > Regards, > > Brian Southworth > > *From:*Bogdan-Andrei Iancu [mailto:bogdan at opensips.org] > *Sent:* 02 February 2018 11:29 > *To:* OpenSIPS users mailling list > ; Brian Southworth > > *Subject:* Re: [OpenSIPS-Users] [15066] > WARNING:rr:after_strict: no socket found to match RR > [1][XXX.XXX.XXX.XXX:5060] > > Hi Brian, > > That warning means OpenSIPS found a Route header (while doing > loose_route) that is suppose to be of its own, but the network > information from the header does not match any of the OpenSIPS > SIP listeners. > > Best regards, > > > > Bogdan-Andrei Iancu > > > > OpenSIPS Founder and Developer > > http://www.opensips-solutions.com > > OpenSIPS Summit 2018 > > http://www.opensips.org/events/Summit-2018Amsterdam > > On 02/02/2018 11:14 AM, Brian Southworth wrote: > > I get this when trying to transfer calls using the B2BUA: > > [15066] WARNING:rr:after_strict: no socket found to match > RR [1][xxx.xxx.xxx.xxx:5060] > > When I try looking on the mailing list there are no other > similar posts, could you please shed some light on what > maybe causing this please. > > Regards, > > Brian Southworth > > > > > > > _______________________________________________ > > Users mailing list > > Users at lists.opensips.org > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Mon Feb 5 10:50:15 2018 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Mon, 5 Feb 2018 17:50:15 +0200 Subject: [OpenSIPS-Users] Registration Loss on opensips SBC In-Reply-To: References: Message-ID: Hi Royee, what are the time values reported for 't_relay' and 'save' ? the 50ms threashold may not be so relevant for explaining delays of seconds. Best regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Summit 2018 http://www.opensips.org/events/Summit-2018Amsterdam On 02/05/2018 02:09 PM, Royee Tichauer via Users wrote: > Hi, > > We are using opensips 2.1 as an SBC component and we see that we have > loss of registrations happening once in a while. When we captured SIP > we saw that some messages are being delayed by a couple of seconds. We > enabled 'exec_execute_message=50000' and saw in the logs that opensips > core methods like 't_realy' and 'save' . I looked at SAR during this > time and there are no indications of CPU running high during this time. > > Can you help me figure out this issue? > > Thanks, > Royee > > Logs: > > Feb4 17:07:23 sbc.company.com > /usr/sbin/opensips[29827]: > WARNING:core:log_expiry: threshold exceeded : msg processing took too > long - 8912776 us.Source : SIP/2.0 200 OK#015#012Via: SIP/2.0/UDP > 5060;received=;rport=5060;branch=z9hG4bK7eec.23d21053.0#015#012Via: > SIP/2.0/UDP > :5060;rport=5060;received=;branch=z9hG4bK2715962793#015#012From: > "" @.company.com:5060 > >;tag=646749010#015#012To: > "###" >;tag=8b4547f60a700f827fe775c80b65be1b.efed#015#012Call-ID: > 0_2258842442@#015#012CSeq > : 3575 > REGISTER#015#012Contact: :5060 > >;expires=45;received="sip::5060 > "#015#012Server: > Vonage#015#012Content-Length: 0#015#012#015#012 > > Feb4 17:07:23 sbc.amz1.company.com > /usr/sbin/opensips[29827]: > WARNING:core:log_expiry: #1 is a module action : save - 8912725us - > line 914 > > Feb4 17:07:23 sbc.amz1.company.com > /usr/sbin/opensips[29827]: > WARNING:core:log_expiry: #2 is a core action : 80 - 17us - line 1103 > > Feb4 17:07:23 sbc.amz1.company.com > /usr/sbin/opensips[29827]: > WARNING:core:log_expiry: #3 is a core action : 72 - 5us - line 887 > > Feb4 17:07:23 sbc.amz1.company.com > /usr/sbin/opensips[29820]: > WARNING:core:log_expiry: threshold exceeded : msg processing took too > long - 5355695 us.Source : REGISTER sip:####.accounts.company.com:5060 > SIP/2.0#015#012Via: > SIP/2.0/UDP :5060;branch=z9hG4bK2715962793#015#012From: "###" > >;tag=646749010#015#012To: > "####" >#015#012Call-ID: > 0_2258842442@######015#012CSeq > : 3575 > REGISTER#015#012Contact: <####@:5060 > >#015#012Authorization: > Digest username="####", realm="####.accounts.company.com > ", > nonce="5a773de000000b19dc7d21205f1edfda504fac90296b04fb", > uri="sip:sip-#######.accounts.company.com:5060 > ", > response="df8d6cbfd178a36913a1464920da3f06", > algorithm=MD5#015#012Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, > OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, > MESSAGE#015#012Max-Forwards: 69#015#012User-Agent: > #########015#012Expires: 3600#015#012Allow-Events: > talk,hold,conference,refer,check-sync#015#012Content-Length: > 0#015#012#015#012 > > > Feb4 17:07:23 sbc.amz1.company.com > /usr/sbin/opensips[29827]: > WARNING:core:log_expiry: #4 is a core action : 72 - 5us - line 1086 > > Feb4 17:07:23 sbc.amz1.company.com > /usr/sbin/opensips[29820]: > WARNING:core:log_expiry: #1 is a module action : t_relay - 5355333us - > line 465 > > Feb4 17:07:23 sbc.amz1.company.com > /usr/sbin/opensips[29827]: > WARNING:core:log_expiry: #5 is a module action : ds_is_in_list - 4us - > line 943 > > Feb4 17:07:23 sbc.amz1.company.com > /usr/sbin/opensips[29817]: > WARNING:core:log_expiry: #1 is a module action : t_relay - 7375370us - > line 465 > > Feb4 17:07:23 sbc.amz1.company.com > /usr/sbin/opensips[29820]: > WARNING:core:log_expiry: #2 is a core action : 78 - 105us - line 604 > > Feb4 17:07:23 sbc.amz1.company.com > /usr/sbin/opensips[29817]: > WARNING:core:log_expiry: #2 is a core action : 78 - 100us - line 604 > > Feb4 17:07:23 sbc.amz1.company.com > /usr/sbin/opensips[29820]: > WARNING:core:log_expiry: #3 is a core action : 78 - 99us - line 635 > > Feb4 17:07:23 sbc.amz1.company.com > /usr/sbin/opensips[29817]: > WARNING:core:log_expiry: #3 is a core action : 78 - 98us - line 635 > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From brian.southworth at clocom.uk Mon Feb 5 10:52:08 2018 From: brian.southworth at clocom.uk (=?UTF-8?Q?Brian_Southworth?=) Date: Mon, 5 Feb 2018 15:52:08 +0000 Subject: [OpenSIPS-Users] [15066] WARNING:rr:after_strict: no socket found to match RR [1][XXX.XXX.XXX.XXX:5060] In-Reply-To: References: <0102016155cbc5e4-c3c66a50-ab65-4987-803e-cd61fbfd5647-000000@eu-west-1.amazonses.com> <01020161564f88ab-94ee6ed7-b89e-4d6e-866d-99d682719034-000000@eu-west-1.amazonses.com> <0102016156e9f5f9-6a281c81-ae20-4a1d-9362-2e98a1fff3e0-000000@eu-west-1.amazonses.com> <3615c9bb-1619-dd03-4354-c511a9e68dd0@opensips.org> <01020161577912d4-87a753fa-ece7-491a-98cb-4d8b9f479e22-000000@eu-west-1.amazonses.com> Message-ID: <0102016166aad91b-15d647cd-2313-46ff-a6e7-ca6e77c13741-000000@eu-west-1.amazonses.com> I think I get it now thank you Bogdan. So I would forward the traffic using the opensips proxy, using the if (is_method(“refer”)) to the opensips box that would be the B2BUA? To bridge the call ?.   Also look forward to Opensips summit in may 😊 ill see you all there got it booked Saturday 😊   Regards,   Brian Southworth   From: Bogdan-Andrei Iancu [mailto:bogdan at opensips.org] Sent: 05 February 2018 15:47 To: Brian Southworth ; OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] [15066] WARNING:rr:after_strict: no socket found to match RR [1][XXX.XXX.XXX.XXX:5060]   Hi Brian, Keep in mind that you cannot make opensips act in the same time as proxy (as required by the load balancer) and as a end-point (as required by the B2BUA). Ideally is to run the two services (LB and B2B) on two opensips instances in a chain. Best regards, Bogdan-Andrei Iancu   OpenSIPS Founder and Developer   http://www.opensips-solutions.com OpenSIPS Summit 2018   http://www.opensips.org/events/Summit-2018Amsterdam On 02/02/2018 07:03 PM, Brian Southworth wrote: Sorry my apologies.   So from the beginning opensips acts as an authorization proxy which passes the call on to an asterisk box based on load (using load balancer). I am trying to get the opensips proxy to handle call transfers and I thought the b2bua would be the best way. Initially the refer was sent to the asterisk box.   On inbound calls The call comes in from the carrier goes to asterisk, asterisk then passes the sip invite to the proxy which then rings the sip phone.   What I wish to achieve is a way to transfer an inbound call to an internal extension or external number.   Example: Caller A receives call à caller A places call on hold and dials caller B à caller B picks up à caller A presses cisco xfer and call is passed to caller B   I was hoping to achieve this using the proxy or asterisk box if possible.   I hope this helps.   Regards,   Brian Southworth   From: Bogdan-Andrei Iancu [mailto:bogdan at opensips.org ] Sent: 02 February 2018 16:50 To: Brian Southworth ; OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] [15066] WARNING:rr:after_strict: no socket found to match RR [1][XXX.XXX.XXX.XXX:5060]   I'm a bit confused. The original report was on a record_route() / loose_route() matter. But you say you have opensips as B2B, so the RR mechanism must not be used in such a case - you act either as a end-point, either as a proxy - you cannot be both for the same call. Now you have this b2b error, during a call transfer scenario. and you mentioned LB also :)...so I'm a bit confused - could please try to put all these pieces together, so I can understand what you are doing ? Regards, Bogdan-Andrei Iancu   OpenSIPS Founder and Developer   http://www.opensips-solutions.com OpenSIPS Summit 2018   http://www.opensips.org/events/Summit-2018Amsterdam On 02/02/2018 04:27 PM, Brian Southworth wrote: Maybe I am doing this wrong but I wanted the B2BUA module to handle the refer and bridge the calls. I have the B2bUA working now. However my issue is that its not able to send the replies.   incoming reply b2b_reply (B2B.222.7591351.1517580641) Feb  2 14:10:47 [22664] ERROR:tm:_reply_light: failed to generate 408 reply when a final 200 was sent out Feb  2 14:10:47 [22664] ERROR:b2b_entities:b2b_send_reply: failed to send reply with tm Feb  2 14:10:47 [22664] ERROR:b2b_logic:b2b_logic_notify_reply: Sending reply failed - 408, [B2B.452.342.1517580641]   Do you need anything else to help me debug this ? I am not sure why its failing to pass the reply with tm, I have enabled the param: modparam("tm", "pass_provisional_replies", 1)   I should also note that I am using the load balancer module also. This normally deals with all call distribution. In and out.   Regards,   Brian Southworth   From: Bogdan-Andrei Iancu [mailto:bogdan at opensips.org ] Sent: 02 February 2018 14:20 To: Brian Southworth ; OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] [15066] WARNING:rr:after_strict: no socket found to match RR [1][XXX.XXX.XXX.XXX:5060]   Hi Brian, Maybe that warning points to a routing error that prevents the REFER to be route to carrier - make a sip capture to be sure the REFER from A is properly routed and accepted by the carrier. Regards, Bogdan-Andrei Iancu   OpenSIPS Founder and Developer   http://www.opensips-solutions.com OpenSIPS Summit 2018   http://www.opensips.org/events/Summit-2018Amsterdam On 02/02/2018 01:38 PM, Brian Southworth wrote: Hi Bogdan,   Thank you very much, so this doesn’t have any impact on why the call being transferred are dropped ?   I am trying to transfer a call using the refer method as that is what the cisco phones use.   The network is setup like so opensips proxy à asterisk gateway à carrier   Scenario:   Inbound call comes into the phone like so: carrier à ast à proxy à phone A Phone A needs to transfer call to phone B: Phone A dials phone B à phone B picks up à phone A presses xfer button and call is dropped.   Any help would be appreciated.   Regards,   Brian Southworth   From: Bogdan-Andrei Iancu [mailto:bogdan at opensips.org ] Sent: 02 February 2018 11:29 To: OpenSIPS users mailling list ; Brian Southworth Subject: Re: [OpenSIPS-Users] [15066] WARNING:rr:after_strict: no socket found to match RR [1][XXX.XXX.XXX.XXX:5060]   Hi Brian, That warning means OpenSIPS found a Route header (while doing loose_route) that is suppose to be of its own, but the network information from the header does not match any of the OpenSIPS SIP listeners. Best regards, Bogdan-Andrei Iancu   OpenSIPS Founder and Developer   http://www.opensips-solutions.com OpenSIPS Summit 2018   http://www.opensips.org/events/Summit-2018Amsterdam On 02/02/2018 11:14 AM, Brian Southworth wrote: I get this when trying to transfer calls using the B2BUA: [15066] WARNING:rr:after_strict: no socket found to match RR [1][xxx.xxx.xxx.xxx:5060]   When I try looking on the mailing list there are no other similar posts, could you please shed some light on what maybe causing this please.   Regards,   Brian Southworth _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users         -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Mon Feb 5 10:57:54 2018 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Mon, 5 Feb 2018 17:57:54 +0200 Subject: [OpenSIPS-Users] Dectect FAS: False Answer Supervision In-Reply-To: References: Message-ID: Hi Abdoul, AFAIK, you cannot detect FAS from signaling level. Even by looking at the RTP level it will be difficult. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Summit 2018 http://www.opensips.org/events/Summit-2018Amsterdam On 02/05/2018 05:02 PM, Abdoul Osséni wrote: > Hello guys, > > Is there a method to detect FAS (False Answer Supervision) in OpenSips ? > > Regards > > Abdoul OSSENI > Ingénieur DevOps chez Néo-Soft > Co-Fondateur de ON SERVICES > Tél : +33 601 135 167 > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From royee.tichauer at vonage.com Mon Feb 5 11:20:15 2018 From: royee.tichauer at vonage.com (Royee Tichauer) Date: Mon, 05 Feb 2018 16:20:15 +0000 Subject: [OpenSIPS-Users] Registration Loss on opensips SBC In-Reply-To: References: Message-ID: Hi Bogdan, as the logs show: save - 8912725us t_relay - 5355333us Let me know if you have any ideas, Thanks, Royee On Mon, Feb 5, 2018 at 5:50 PM Bogdan-Andrei Iancu wrote: > Hi Royee, > > what are the time values reported for 't_relay' and 'save' ? the 50ms > threashold may not be so relevant for explaining delays of seconds. > > Best regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > OpenSIPS Summit 2018 > http://www.opensips.org/events/Summit-2018Amsterdam > > On 02/05/2018 02:09 PM, Royee Tichauer via Users wrote: > > Hi, > > We are using opensips 2.1 as an SBC component and we see that we have loss > of registrations happening once in a while. When we captured SIP we saw > that some messages are being delayed by a couple of seconds. We enabled > 'exec_execute_message=50000' and saw in the logs that opensips core methods > like 't_realy' and 'save' . I looked at SAR during this time and there are > no indications of CPU running high during this time. > > Can you help me figure out this issue? > > Thanks, > Royee > > Logs: > > Feb 4 17:07:23 sbc.company.com > /usr/sbin/opensips[29827]: WARNING:core:log_expiry: threshold exceeded : > msg processing took too long - 8912776 us.Source : SIP/2.0 200 > OK#015#012Via: SIP/2.0/UDP > 5060;received=;rport=5060;branch=z9hG4bK7eec.23d21053.0#015#012Via: > SIP/2.0/UDP > :5060;rport=5060;received=;branch=z9hG4bK2715962793#015#012From: "" > @.company.com:5060 > >;tag=646749010#015#012To: > "###" > >;tag=8b4547f60a700f827fe775c80b65be1b.efed#015#012Call-ID: > 0_2258842442@#015#012CSeq > : 3575 > REGISTER#015#012Contact: :5060 > >;expires=45;received="sip: > :5060 "#015#012Server: > Vonage#015#012Content-Length: 0#015#012#015#012 > > Feb 4 17:07:23 sbc.amz1.company.com > /usr/sbin/opensips[29827]: > WARNING:core:log_expiry: #1 is a module action : save - 8912725us - line > 914 > > Feb 4 17:07:23 sbc.amz1.company.com > /usr/sbin/opensips[29827]: > WARNING:core:log_expiry: #2 is a core action : 80 - 17us - line 1103 > > Feb 4 17:07:23 sbc.amz1.company.com > /usr/sbin/opensips[29827]: > WARNING:core:log_expiry: #3 is a core action : 72 - 5us - line 887 > > Feb 4 17:07:23 sbc.amz1.company.com > /usr/sbin/opensips[29820]: > WARNING:core:log_expiry: threshold exceeded : msg processing took too > long - 5355695 us.Source : REGISTER sip:####.accounts.company.com:5060 > SIP/2.0#015#012Via: > SIP/2.0/UDP :5060;branch=z9hG4bK2715962793#015#012From: "###" < > sip:#####@sip-####.accounts.company.com:5060 > >;tag=646749010#015#012To: > "####" > >#015#012Call-ID: 0_2258842442@######015#012CSeq > : 3575 > REGISTER#015#012Contact: <####@:5060 > >#015#012Authorization: Digest > username="####", realm="####.accounts.company.com > ", > nonce="5a773de000000b19dc7d21205f1edfda504fac90296b04fb", uri="sip: > sip-#######.accounts.company.com:5060 > ", > response="df8d6cbfd178a36913a1464920da3f06", algorithm=MD5#015#012Allow: > INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, > SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE#015#012Max-Forwards: > 69#015#012User-Agent: #########015#012Expires: 3600#015#012Allow-Events: > talk,hold,conference,refer,check-sync#015#012Content-Length: > 0#015#012#015#012 > > > Feb 4 17:07:23 sbc.amz1.company.com > /usr/sbin/opensips[29827]: > WARNING:core:log_expiry: #4 is a core action : 72 - 5us - line 1086 > > Feb 4 17:07:23 sbc.amz1.company.com > /usr/sbin/opensips[29820]: > WARNING:core:log_expiry: #1 is a module action : t_relay - 5355333us - > line 465 > > Feb 4 17:07:23 sbc.amz1.company.com > /usr/sbin/opensips[29827]: > WARNING:core:log_expiry: #5 is a module action : ds_is_in_list - 4us - > line 943 > > Feb 4 17:07:23 sbc.amz1.company.com > /usr/sbin/opensips[29817]: > WARNING:core:log_expiry: #1 is a module action : t_relay - 7375370us - > line 465 > > Feb 4 17:07:23 sbc.amz1.company.com > /usr/sbin/opensips[29820]: > WARNING:core:log_expiry: #2 is a core action : 78 - 105us - line 604 > > Feb 4 17:07:23 sbc.amz1.company.com > /usr/sbin/opensips[29817]: > WARNING:core:log_expiry: #2 is a core action : 78 - 100us - line 604 > > Feb 4 17:07:23 sbc.amz1.company.com > /usr/sbin/opensips[29820]: > WARNING:core:log_expiry: #3 is a core action : 78 - 99us - line 635 > > Feb 4 17:07:23 sbc.amz1.company.com > /usr/sbin/opensips[29817]: > WARNING:core:log_expiry: #3 is a core action : 78 - 98us - line 635 > > > _______________________________________________ > Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Mon Feb 5 11:34:32 2018 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Mon, 5 Feb 2018 18:34:32 +0200 Subject: [OpenSIPS-Users] Registration Loss on opensips SBC In-Reply-To: References: Message-ID: Hi Royee, Well, you need to break for each function the possible types of actions. Like for save(): - actual internal save() - DB insert if realtime mode used - potential events triggering - do you use any usrloc related events ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Summit 2018 http://www.opensips.org/events/Summit-2018Amsterdam On 02/05/2018 06:20 PM, Royee Tichauer wrote: > Hi Bogdan, as the logs show: > > save - 8912725us > t_relay - 5355333us > > Let me know if you have any ideas, > Thanks, > Royee > > > > > On Mon, Feb 5, 2018 at 5:50 PM Bogdan-Andrei Iancu > > wrote: > > Hi Royee, > > what are the time values reported for 't_relay' and 'save' ? the > 50ms threashold may not be so relevant for explaining delays of > seconds. > > Best regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > OpenSIPS Summit 2018 > http://www.opensips.org/events/Summit-2018Amsterdam > > On 02/05/2018 02:09 PM, Royee Tichauer via Users wrote: >> Hi, >> >> We are using opensips 2.1 as an SBC component and we see that we >> have loss of registrations happening once in a while. When we >> captured SIP we saw that some messages are being delayed by a >> couple of seconds. We enabled 'exec_execute_message=50000' and >> saw in the logs that opensips core methods like 't_realy' and >> 'save' . I looked at SAR during this time and there are no >> indications of CPU running high during this time. >> >> Can you help me figure out this issue? >> >> Thanks, >> Royee >> >> Logs: >> >> Feb4 17:07:23 sbc.company.com >> /usr/sbin/opensips[29827]: >> WARNING:core:log_expiry: threshold exceeded : msg processing took >> too long - 8912776 us.Source : SIP/2.0 200 OK#015#012Via: >> SIP/2.0/UDP >> 5060;received=;rport=5060;branch=z9hG4bK7eec.23d21053.0#015#012Via: >> SIP/2.0/UDP >> :5060;rport=5060;received=;branch=z9hG4bK2715962793#015#012From: >> "" @.company.com:5060 >> >;tag=646749010#015#012To: >> "###" > >;tag=8b4547f60a700f827fe775c80b65be1b.efed#015#012Call-ID: >> 0_2258842442@#015#012CSeq >> : 3575 >> REGISTER#015#012Contact: :5060 >> >;expires=45;received="sip::5060 >> "#015#012Server: >> Vonage#015#012Content-Length: 0#015#012#015#012 >> >> Feb4 17:07:23 sbc.amz1.company.com >> /usr/sbin/opensips[29827]: >> WARNING:core:log_expiry: #1 is a module action : save - >> 8912725us - line 914 >> >> Feb4 17:07:23 sbc.amz1.company.com >> /usr/sbin/opensips[29827]: >> WARNING:core:log_expiry: #2 is a core action : 80 - 17us - line 1103 >> >> Feb4 17:07:23 sbc.amz1.company.com >> /usr/sbin/opensips[29827]: >> WARNING:core:log_expiry: #3 is a core action : 72 - 5us - line 887 >> >> Feb4 17:07:23 sbc.amz1.company.com >> /usr/sbin/opensips[29820]: >> WARNING:core:log_expiry: threshold exceeded : msg processing took >> too long - 5355695 us.Source : REGISTER >> sip:####.accounts.company.com:5060 >> SIP/2.0#015#012Via: >> SIP/2.0/UDP :5060;branch=z9hG4bK2715962793#015#012From: "###" >> > >;tag=646749010#015#012To: >> "####" > >#015#012Call-ID: >> 0_2258842442@######015#012CSeq >> : 3575 >> REGISTER#015#012Contact: <####@:5060 >> >#015#012Authorization: >> Digest username="####", realm="####.accounts.company.com >> ", >> nonce="5a773de000000b19dc7d21205f1edfda504fac90296b04fb", >> uri="sip:sip-#######.accounts.company.com:5060 >> ", >> response="df8d6cbfd178a36913a1464920da3f06", >> algorithm=MD5#015#012Allow: INVITE, INFO, PRACK, ACK, BYE, >> CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, >> UPDATE, MESSAGE#015#012Max-Forwards: 69#015#012User-Agent: >> #########015#012Expires: 3600#015#012Allow-Events: >> talk,hold,conference,refer,check-sync#015#012Content-Length: >> 0#015#012#015#012 >> >> >> Feb4 17:07:23 sbc.amz1.company.com >> >> /usr/sbin/opensips[29827]: WARNING:core:log_expiry: #4 is a core >> action : 72 - 5us - line 1086 >> >> Feb4 17:07:23 sbc.amz1.company.com >> /usr/sbin/opensips[29820]: >> WARNING:core:log_expiry: #1 is a module action : t_relay - >> 5355333us - line 465 >> >> Feb4 17:07:23 sbc.amz1.company.com >> /usr/sbin/opensips[29827]: >> WARNING:core:log_expiry: #5 is a module action : ds_is_in_list - >> 4us - line 943 >> >> Feb4 17:07:23 sbc.amz1.company.com >> /usr/sbin/opensips[29817]: >> WARNING:core:log_expiry: #1 is a module action : t_relay - >> 7375370us - line 465 >> >> Feb4 17:07:23 sbc.amz1.company.com >> /usr/sbin/opensips[29820]: >> WARNING:core:log_expiry: #2 is a core action : 78 - 105us - line 604 >> >> Feb4 17:07:23 sbc.amz1.company.com >> /usr/sbin/opensips[29817]: >> WARNING:core:log_expiry: #2 is a core action : 78 - 100us - line 604 >> >> Feb4 17:07:23 sbc.amz1.company.com >> /usr/sbin/opensips[29820]: >> WARNING:core:log_expiry: #3 is a core action : 78 - 99us - line 635 >> >> Feb4 17:07:23 sbc.amz1.company.com >> /usr/sbin/opensips[29817]: >> WARNING:core:log_expiry: #3 is a core action : 78 - 98us - line 635 >> >> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -------------- next part -------------- An HTML attachment was scrubbed... URL: From royee.tichauer at vonage.com Mon Feb 5 11:44:46 2018 From: royee.tichauer at vonage.com (Royee Tichauer) Date: Mon, 05 Feb 2018 16:44:46 +0000 Subject: [OpenSIPS-Users] Registration Loss on opensips SBC In-Reply-To: References: Message-ID: Bogdan, Not sure I follow about breaking each function. What do you mean by 'actual internal save()'? DB insert is not used. We use: event_route[E_UL_CONTACT_UPDATE] event_route[E_UL_CONTACT_DELETE] event_route[E_UL_CONTACT_INSERT] btw, we also use the mi_xmlrpc_ng constantly for monitoring if that might have something to do with it (I saw a comment on opensips 1.6 that there was an opensips blocking issue in some cases). Royee On Mon, Feb 5, 2018 at 6:34 PM Bogdan-Andrei Iancu wrote: > Hi Royee, > > Well, you need to break for each function the possible types of actions. > Like for save(): > - actual internal save() > - DB insert if realtime mode used > - potential events triggering - do you use any usrloc related events ? > > Regards, > > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > OpenSIPS Summit 2018 > http://www.opensips.org/events/Summit-2018Amsterdam > > On 02/05/2018 06:20 PM, Royee Tichauer wrote: > > Hi Bogdan, as the logs show: > > save - 8912725us > t_relay - 5355333us > > Let me know if you have any ideas, > Thanks, > Royee > > > > > On Mon, Feb 5, 2018 at 5:50 PM Bogdan-Andrei Iancu > wrote: > >> Hi Royee, >> >> what are the time values reported for 't_relay' and 'save' ? the 50ms >> threashold may not be so relevant for explaining delays of seconds. >> >> Best regards, >> >> Bogdan-Andrei Iancu >> >> OpenSIPS Founder and Developer >> http://www.opensips-solutions.com >> OpenSIPS Summit 2018 >> http://www.opensips.org/events/Summit-2018Amsterdam >> >> On 02/05/2018 02:09 PM, Royee Tichauer via Users wrote: >> >> Hi, >> >> We are using opensips 2.1 as an SBC component and we see that we have >> loss of registrations happening once in a while. When we captured SIP we >> saw that some messages are being delayed by a couple of seconds. We enabled >> 'exec_execute_message=50000' and saw in the logs that opensips core methods >> like 't_realy' and 'save' . I looked at SAR during this time and there are >> no indications of CPU running high during this time. >> >> Can you help me figure out this issue? >> >> Thanks, >> Royee >> >> Logs: >> >> Feb 4 17:07:23 sbc.company.com >> /usr/sbin/opensips[29827]: WARNING:core:log_expiry: threshold exceeded >> : msg processing took too long - 8912776 us.Source : SIP/2.0 200 >> OK#015#012Via: SIP/2.0/UDP >> 5060;received=;rport=5060;branch=z9hG4bK7eec.23d21053.0#015#012Via: >> SIP/2.0/UDP >> :5060;rport=5060;received=;branch=z9hG4bK2715962793#015#012From: "" >> @.company.com:5060 >> >;tag=646749010#015#012To: >> "###" > >> >;tag=8b4547f60a700f827fe775c80b65be1b.efed#015#012Call-ID: >> 0_2258842442@#015#012CSeq >> : 3575 >> REGISTER#015#012Contact: :5060 >> >;expires=45;received="sip: >> :5060 "#015#012Server: >> Vonage#015#012Content-Length: 0#015#012#015#012 >> >> Feb 4 17:07:23 sbc.amz1.company.com >> /usr/sbin/opensips[29827]: >> WARNING:core:log_expiry: #1 is a module action : save - 8912725us - line >> 914 >> >> Feb 4 17:07:23 sbc.amz1.company.com >> /usr/sbin/opensips[29827]: >> WARNING:core:log_expiry: #2 is a core action : 80 - 17us - line 1103 >> >> Feb 4 17:07:23 sbc.amz1.company.com >> /usr/sbin/opensips[29827]: >> WARNING:core:log_expiry: #3 is a core action : 72 - 5us - line 887 >> >> Feb 4 17:07:23 sbc.amz1.company.com >> /usr/sbin/opensips[29820]: >> WARNING:core:log_expiry: threshold exceeded : msg processing took too >> long - 5355695 us.Source : REGISTER sip:####.accounts.company.com:5060 >> SIP/2.0#015#012Via: >> SIP/2.0/UDP :5060;branch=z9hG4bK2715962793#015#012From: "###" < >> sip:#####@sip-####.accounts.company.com:5060 >> >;tag=646749010#015#012To: >> "####" > >> >#015#012Call-ID: 0_2258842442@######015#012CSeq >> : 3575 >> REGISTER#015#012Contact: <####@:5060 >> >#015#012Authorization: Digest >> username="####", realm="####.accounts.company.com >> ", >> nonce="5a773de000000b19dc7d21205f1edfda504fac90296b04fb", uri="sip: >> sip-#######.accounts.company.com:5060 >> ", >> response="df8d6cbfd178a36913a1464920da3f06", algorithm=MD5#015#012Allow: >> INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, >> SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE#015#012Max-Forwards: >> 69#015#012User-Agent: #########015#012Expires: 3600#015#012Allow-Events: >> talk,hold,conference,refer,check-sync#015#012Content-Length: >> 0#015#012#015#012 >> >> >> Feb 4 17:07:23 sbc.amz1.company.com >> /usr/sbin/opensips[29827]: >> WARNING:core:log_expiry: #4 is a core action : 72 - 5us - line 1086 >> >> Feb 4 17:07:23 sbc.amz1.company.com >> /usr/sbin/opensips[29820]: >> WARNING:core:log_expiry: #1 is a module action : t_relay - 5355333us - >> line 465 >> >> Feb 4 17:07:23 sbc.amz1.company.com >> /usr/sbin/opensips[29827]: >> WARNING:core:log_expiry: #5 is a module action : ds_is_in_list - 4us - >> line 943 >> >> Feb 4 17:07:23 sbc.amz1.company.com >> /usr/sbin/opensips[29817]: >> WARNING:core:log_expiry: #1 is a module action : t_relay - 7375370us - >> line 465 >> >> Feb 4 17:07:23 sbc.amz1.company.com >> /usr/sbin/opensips[29820]: >> WARNING:core:log_expiry: #2 is a core action : 78 - 105us - line 604 >> >> Feb 4 17:07:23 sbc.amz1.company.com >> /usr/sbin/opensips[29817]: >> WARNING:core:log_expiry: #2 is a core action : 78 - 100us - line 604 >> >> Feb 4 17:07:23 sbc.amz1.company.com >> /usr/sbin/opensips[29820]: >> WARNING:core:log_expiry: #3 is a core action : 78 - 99us - line 635 >> >> Feb 4 17:07:23 sbc.amz1.company.com >> /usr/sbin/opensips[29817]: >> WARNING:core:log_expiry: #3 is a core action : 78 - 98us - line 635 >> >> >> _______________________________________________ >> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: From pasandev at ymail.com Mon Feb 5 05:29:16 2018 From: pasandev at ymail.com (Pasan Meemaduma) Date: Mon, 5 Feb 2018 10:29:16 +0000 (UTC) Subject: [OpenSIPS-Users] fix documentation link References: <25734421.3205880.1517826556288.ref@mail.yahoo.com> Message-ID: <25734421.3205880.1517826556288@mail.yahoo.com> Hi could you please fix the hyperlink for PROTO_BIN ? its refer to same link as PROTO_HEP http://www.opensips.org/Documentation/Modules-2-3 | | | | | | | | | | | openSIPS | Documentation / Modules - 2.3 | | | | -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Mon Feb 5 12:22:58 2018 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Mon, 5 Feb 2018 19:22:58 +0200 Subject: [OpenSIPS-Users] Registration Loss on opensips SBC In-Reply-To: References: Message-ID: Hi Royee, I don't think it is related to xmlrpc - there are separate connection, separate processes for this. I would rather suspect the event routes - the event is in-line handled in usrloc, so the save() will cover the delivery/handling of the contact related events. And if you do something time consuming in event_route[E_UL_CONTACT_INSERT] lets say, this will reflect as processing time for the save() function. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Summit 2018 http://www.opensips.org/events/Summit-2018Amsterdam On 02/05/2018 06:44 PM, Royee Tichauer wrote: > > Bogdan, > > Not sure I follow about breaking each function. What do you mean by > 'actual internal save()'? > > DB insert is not used. > > We use: > > event_route[E_UL_CONTACT_UPDATE] > > event_route[E_UL_CONTACT_DELETE] > > event_route[E_UL_CONTACT_INSERT] > > btw, we also use the mi_xmlrpc_ng constantly for monitoring if that > might have something to do with it (I saw a comment on opensips 1.6 > that there was an opensips blocking issue in some cases). > > Royee > > > > On Mon, Feb 5, 2018 at 6:34 PM Bogdan-Andrei Iancu > > wrote: > > Hi Royee, > > Well, you need to break for each function the possible types of > actions. Like for save(): > - actual internal save() > - DB insert if realtime mode used > - potential events triggering - do you use any usrloc related > events ? > > Regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > OpenSIPS Summit 2018 > http://www.opensips.org/events/Summit-2018Amsterdam > > On 02/05/2018 06:20 PM, Royee Tichauer wrote: >> Hi Bogdan, as the logs show: >> >> save - 8912725us >> t_relay - 5355333us >> >> Let me know if you have any ideas, >> Thanks, >> Royee >> >> >> >> >> On Mon, Feb 5, 2018 at 5:50 PM Bogdan-Andrei Iancu >> > wrote: >> >> Hi Royee, >> >> what are the time values reported for 't_relay' and 'save' ? >> the 50ms threashold may not be so relevant for explaining >> delays of seconds. >> >> Best regards, >> >> Bogdan-Andrei Iancu >> >> OpenSIPS Founder and Developer >> http://www.opensips-solutions.com >> OpenSIPS Summit 2018 >> http://www.opensips.org/events/Summit-2018Amsterdam >> >> On 02/05/2018 02:09 PM, Royee Tichauer via Users wrote: >>> Hi, >>> >>> We are using opensips 2.1 as an SBC component and we see >>> that we have loss of registrations happening once in a >>> while. When we captured SIP we saw that some messages are >>> being delayed by a couple of seconds. We enabled >>> 'exec_execute_message=50000' and saw in the logs that >>> opensips core methods like 't_realy' and 'save' . I looked >>> at SAR during this time and there are no indications of CPU >>> running high during this time. >>> >>> Can you help me figure out this issue? >>> >>> Thanks, >>> Royee >>> >>> Logs: >>> >>> Feb4 17:07:23 sbc.company.com >>> /usr/sbin/opensips[29827]: >>> WARNING:core:log_expiry: threshold exceeded : msg processing >>> took too long - 8912776 us.Source : SIP/2.0 200 >>> OK#015#012Via: SIP/2.0/UDP >>> 5060;received=;rport=5060;branch=z9hG4bK7eec.23d21053.0#015#012Via: >>> SIP/2.0/UDP >>> :5060;rport=5060;received=;branch=z9hG4bK2715962793#015#012From: >>> "" @.company.com:5060 >>> >;tag=646749010#015#012To: >>> "###" >> >;tag=8b4547f60a700f827fe775c80b65be1b.efed#015#012Call-ID: >>> 0_2258842442@#015#012CSeq >>> : 3575 >>> REGISTER#015#012Contact: :5060 >>> >;expires=45;received="sip::5060 >>> "#015#012Server: >>> Vonage#015#012Content-Length: 0#015#012#015#012 >>> >>> Feb4 17:07:23 sbc.amz1.company.com >>> /usr/sbin/opensips[29827]: >>> WARNING:core:log_expiry: #1 is a module action : save - >>> 8912725us - line 914 >>> >>> Feb4 17:07:23 sbc.amz1.company.com >>> /usr/sbin/opensips[29827]: >>> WARNING:core:log_expiry: #2 is a core action : 80 - 17us - >>> line 1103 >>> >>> Feb4 17:07:23 sbc.amz1.company.com >>> /usr/sbin/opensips[29827]: >>> WARNING:core:log_expiry: #3 is a core action : 72 - 5us - >>> line 887 >>> >>> Feb4 17:07:23 sbc.amz1.company.com >>> /usr/sbin/opensips[29820]: >>> WARNING:core:log_expiry: threshold exceeded : msg processing >>> took too long - 5355695 us.Source : REGISTER >>> sip:####.accounts.company.com:5060 >>> SIP/2.0#015#012Via: >>> SIP/2.0/UDP :5060;branch=z9hG4bK2715962793#015#012From: >>> "###" >> >;tag=646749010#015#012To: >>> "####" >> >#015#012Call-ID: >>> 0_2258842442@######015#012CSeq >>> : 3575 >>> REGISTER#015#012Contact: <####@:5060 >>> >#015#012Authorization: >>> Digest username="####", realm="####.accounts.company.com >>> ", >>> nonce="5a773de000000b19dc7d21205f1edfda504fac90296b04fb", >>> uri="sip:sip-#######.accounts.company.com:5060 >>> ", >>> response="df8d6cbfd178a36913a1464920da3f06", >>> algorithm=MD5#015#012Allow: INVITE, INFO, PRACK, ACK, BYE, >>> CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, >>> PUBLISH, UPDATE, MESSAGE#015#012Max-Forwards: >>> 69#015#012User-Agent: #########015#012Expires: >>> 3600#015#012Allow-Events: >>> talk,hold,conference,refer,check-sync#015#012Content-Length: >>> 0#015#012#015#012 >>> >>> >>> Feb4 17:07:23 sbc.amz1.company.com >>> >>> /usr/sbin/opensips[29827]: WARNING:core:log_expiry: #4 is a >>> core action : 72 - 5us - line 1086 >>> >>> Feb4 17:07:23 sbc.amz1.company.com >>> /usr/sbin/opensips[29820]: >>> WARNING:core:log_expiry: #1 is a module action : t_relay - >>> 5355333us - line 465 >>> >>> Feb4 17:07:23 sbc.amz1.company.com >>> /usr/sbin/opensips[29827]: >>> WARNING:core:log_expiry: #5 is a module action : >>> ds_is_in_list - 4us - line 943 >>> >>> Feb4 17:07:23 sbc.amz1.company.com >>> /usr/sbin/opensips[29817]: >>> WARNING:core:log_expiry: #1 is a module action : t_relay - >>> 7375370us - line 465 >>> >>> Feb4 17:07:23 sbc.amz1.company.com >>> /usr/sbin/opensips[29820]: >>> WARNING:core:log_expiry: #2 is a core action : 78 - 105us - >>> line 604 >>> >>> Feb4 17:07:23 sbc.amz1.company.com >>> /usr/sbin/opensips[29817]: >>> WARNING:core:log_expiry: #2 is a core action : 78 - 100us - >>> line 604 >>> >>> Feb4 17:07:23 sbc.amz1.company.com >>> /usr/sbin/opensips[29820]: >>> WARNING:core:log_expiry: #3 is a core action : 78 - 99us - >>> line 635 >>> >>> Feb4 17:07:23 sbc.amz1.company.com >>> /usr/sbin/opensips[29817]: >>> WARNING:core:log_expiry: #3 is a core action : 78 - 98us - >>> line 635 >>> >>> >>> >>> _______________________________________________ >>> Users mailing list >>> Users at lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: From olle at zaark.com Mon Feb 5 13:08:58 2018 From: olle at zaark.com (Olle Frimanson) Date: Mon, 5 Feb 2018 19:08:58 +0100 Subject: [OpenSIPS-Users] Username in DB URL Message-ID: <013d01d39eac$651ded60$2f59c820$@zaark.com> Hi, we are experiencing an issue with a roleout in Microsoft Azure, where the DB username contains a "@" sign. This seems to be the default in MySQL in Azure. Is there anyway to escape this the URL for the clusterer module? If you use mysql client it's typical something like Mysql -h host.mysql.azure.com -u opensips at host -popensips opensips, which works fine In clustererar where we use this this translates to: modparam("clusterer", "db_url","mysql://opensips at host:opensips at host.mysql.azure.com/opensips") which fails Any ideas on a workaround on this? BR / Olle -------------- next part -------------- An HTML attachment was scrubbed... URL: From abalashov at evaristesys.com Mon Feb 5 13:11:27 2018 From: abalashov at evaristesys.com (Alex Balashov) Date: Mon, 5 Feb 2018 13:11:27 -0500 Subject: [OpenSIPS-Users] Username in DB URL In-Reply-To: <013d01d39eac$651ded60$2f59c820$@zaark.com> References: <013d01d39eac$651ded60$2f59c820$@zaark.com> Message-ID: <20180205181126.GA17654@tlaquepaque.localdomain> Hah! Yeah, it's a well-known proble, same if '@' appears in the password. I'd love to know if there's a solution too, as there appears to be no means of escaping these. On Mon, Feb 05, 2018 at 07:08:58PM +0100, Olle Frimanson wrote: > Hi, we are experiencing an issue with a roleout in Microsoft Azure, where > the DB username contains a "@" sign. This seems to be the default in MySQL > in Azure. Is there anyway to escape this the URL for the clusterer module? > > > > If you use mysql client it's typical something like > > > > Mysql -h host.mysql.azure.com -u opensips at host -popensips opensips, which > works fine > > > > In clustererar where we use this this translates to: > > > > modparam("clusterer", > "db_url","mysql://opensips at host:opensips at host.mysql.azure.com/opensips") > > > > which fails > > > > Any ideas on a workaround on this? > > > > BR / Olle > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Alex Balashov | Principal | Evariste Systems LLC Tel: +1-706-510-6800 / +1-800-250-5920 (toll-free) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/ From h323 at ramdyne.nl Mon Feb 5 13:14:52 2018 From: h323 at ramdyne.nl (Andreas Sikkema) Date: Mon, 5 Feb 2018 19:14:52 +0100 Subject: [OpenSIPS-Users] Problem with topology hiding when hairpinning a call Message-ID: Hi, I'm playing with the topology hiding module and I am running into a problem when a call is sent from the inside of OpenSIPS to the outside, which is then received back on the outside of OpenSIPS to sent to the inside of OpenSIP. Calls being answered on the inside or outside work fine, it's just with hairpinning I have this problem. The call setup works fine, but as soon as the call is answered, I am seeing a lot of lines like this: Feb 5 18:52:38 [31118] WARNING:dialog:dlg_onroute: tight matching failed for BYE with callid='ee75e7e72504b88f57959fe48d3faacc at 192.168.xxx.xxx'/46, ftag='1640688292'/10, ttag='1456089971'/10 and direction=0 Feb 5 18:52:38 [31118] WARNING:dialog:dlg_onroute: dialog identification elements are callid='BB_KhVSWzZ+NWR9RVVaMXFoNXpHXFtqLzVndxRWCDIoMzAPQVxcfXhma2FBV1p9fA--'/71, caller tag='1640688292'/10, callee tag='1456089971'/10 Is there something obvious I am doing wrong? The config is basically the generic topology hiding module example. I was hoping to not have a seperate instance for both directions ;-) Thanks! -- Andreas -------------- next part -------------- An HTML attachment was scrubbed... URL: From abdoul.osseni at gmail.com Mon Feb 5 13:19:04 2018 From: abdoul.osseni at gmail.com (=?UTF-8?Q?Abdoul_Oss=C3=A9ni?=) Date: Mon, 5 Feb 2018 19:19:04 +0100 Subject: [OpenSIPS-Users] Dectect FAS: False Answer Supervision In-Reply-To: References: Message-ID: Thank you for your feedback. Abdoul OSSENI Ingénieur DevOps chez Néo-Soft Co-Fondateur de ON SERVICES Tél : +33 601 135 167 2018-02-05 16:57 GMT+01:00 Bogdan-Andrei Iancu : > Hi Abdoul, > > AFAIK, you cannot detect FAS from signaling level. Even by looking at the > RTP level it will be difficult. > > Regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > OpenSIPS Summit 2018 > http://www.opensips.org/events/Summit-2018Amsterdam > > On 02/05/2018 05:02 PM, Abdoul Osséni wrote: > > Hello guys, > > Is there a method to detect FAS (False Answer Supervision) in OpenSips ? > > Regards > > Abdoul OSSENI > Ingénieur DevOps chez Néo-Soft > Co-Fondateur de ON SERVICES > Tél : +33 601 135 167 > > > _______________________________________________ > Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From royee.tichauer at vonage.com Tue Feb 6 03:30:18 2018 From: royee.tichauer at vonage.com (Royee Tichauer) Date: Tue, 06 Feb 2018 08:30:18 +0000 Subject: [OpenSIPS-Users] Registration Loss on opensips SBC In-Reply-To: References: Message-ID: Thanks for lead Bogdan we will look into it. Do you also have any idea why the t_relay was blocking for so long? Royee On Mon, Feb 5, 2018 at 7:23 PM Bogdan-Andrei Iancu wrote: > Hi Royee, > > I don't think it is related to xmlrpc - there are separate connection, > separate processes for this. > > I would rather suspect the event routes - the event is in-line handled in > usrloc, so the save() will cover the delivery/handling of the contact > related events. And if you do something time consuming in > event_route[E_UL_CONTACT_INSERT] lets say, this will reflect as processing > time for the save() function. > > Regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > OpenSIPS Summit 2018 > http://www.opensips.org/events/Summit-2018Amsterdam > > On 02/05/2018 06:44 PM, Royee Tichauer wrote: > > Bogdan, > > Not sure I follow about breaking each function. What do you mean by > 'actual internal save()'? > > DB insert is not used. > > We use: > > event_route[E_UL_CONTACT_UPDATE] > > event_route[E_UL_CONTACT_DELETE] > > event_route[E_UL_CONTACT_INSERT] > > btw, we also use the mi_xmlrpc_ng constantly for monitoring if that might > have something to do with it (I saw a comment on opensips 1.6 that there > was an opensips blocking issue in some cases). > > Royee > > > > On Mon, Feb 5, 2018 at 6:34 PM Bogdan-Andrei Iancu > wrote: > >> Hi Royee, >> >> Well, you need to break for each function the possible types of actions. >> Like for save(): >> - actual internal save() >> - DB insert if realtime mode used >> - potential events triggering - do you use any usrloc related events ? >> >> Regards, >> >> >> Bogdan-Andrei Iancu >> >> OpenSIPS Founder and Developer >> http://www.opensips-solutions.com >> OpenSIPS Summit 2018 >> http://www.opensips.org/events/Summit-2018Amsterdam >> >> On 02/05/2018 06:20 PM, Royee Tichauer wrote: >> >> Hi Bogdan, as the logs show: >> >> save - 8912725us >> t_relay - 5355333us >> >> Let me know if you have any ideas, >> Thanks, >> Royee >> >> >> >> >> On Mon, Feb 5, 2018 at 5:50 PM Bogdan-Andrei Iancu >> wrote: >> >>> Hi Royee, >>> >>> what are the time values reported for 't_relay' and 'save' ? the 50ms >>> threashold may not be so relevant for explaining delays of seconds. >>> >>> Best regards, >>> >>> Bogdan-Andrei Iancu >>> >>> OpenSIPS Founder and Developer >>> http://www.opensips-solutions.com >>> OpenSIPS Summit 2018 >>> http://www.opensips.org/events/Summit-2018Amsterdam >>> >>> On 02/05/2018 02:09 PM, Royee Tichauer via Users wrote: >>> >>> Hi, >>> >>> We are using opensips 2.1 as an SBC component and we see that we have >>> loss of registrations happening once in a while. When we captured SIP we >>> saw that some messages are being delayed by a couple of seconds. We enabled >>> 'exec_execute_message=50000' and saw in the logs that opensips core methods >>> like 't_realy' and 'save' . I looked at SAR during this time and there are >>> no indications of CPU running high during this time. >>> >>> Can you help me figure out this issue? >>> >>> Thanks, >>> Royee >>> >>> Logs: >>> >>> Feb 4 17:07:23 sbc.company.com >>> /usr/sbin/opensips[29827]: WARNING:core:log_expiry: threshold exceeded >>> : msg processing took too long - 8912776 us.Source : SIP/2.0 200 >>> OK#015#012Via: SIP/2.0/UDP >>> 5060;received=;rport=5060;branch=z9hG4bK7eec.23d21053.0#015#012Via: >>> SIP/2.0/UDP >>> :5060;rport=5060;received=;branch=z9hG4bK2715962793#015#012From: "" >>> @.company.com:5060 >>> >;tag=646749010#015#012To: >>> "###" >> >>> >;tag=8b4547f60a700f827fe775c80b65be1b.efed#015#012Call-ID: >>> 0_2258842442@#015#012CSeq >>> : 3575 >>> REGISTER#015#012Contact: :5060 >>> >;expires=45;received="sip: >>> :5060 "#015#012Server: >>> Vonage#015#012Content-Length: 0#015#012#015#012 >>> >>> Feb 4 17:07:23 sbc.amz1.company.com >>> /usr/sbin/opensips[29827]: >>> WARNING:core:log_expiry: #1 is a module action : save - 8912725us - >>> line 914 >>> >>> Feb 4 17:07:23 sbc.amz1.company.com >>> /usr/sbin/opensips[29827]: >>> WARNING:core:log_expiry: #2 is a core action : 80 - 17us - line 1103 >>> >>> Feb 4 17:07:23 sbc.amz1.company.com >>> /usr/sbin/opensips[29827]: >>> WARNING:core:log_expiry: #3 is a core action : 72 - 5us - line 887 >>> >>> Feb 4 17:07:23 sbc.amz1.company.com >>> /usr/sbin/opensips[29820]: >>> WARNING:core:log_expiry: threshold exceeded : msg processing took too >>> long - 5355695 us.Source : REGISTER sip:####.accounts.company.com:5060 >>> SIP/2.0#015#012Via: >>> SIP/2.0/UDP :5060;branch=z9hG4bK2715962793#015#012From: "###" < >>> sip:#####@sip-####.accounts.company.com:5060 >>> >;tag=646749010#015#012To: >>> "####" >> >>> >#015#012Call-ID: 0_2258842442@######015#012CSeq >>> : 3575 >>> REGISTER#015#012Contact: <####@:5060 >>> >#015#012Authorization: Digest >>> username="####", realm="####.accounts.company.com >>> ", >>> nonce="5a773de000000b19dc7d21205f1edfda504fac90296b04fb", uri="sip: >>> sip-#######.accounts.company.com:5060 >>> ", >>> response="df8d6cbfd178a36913a1464920da3f06", algorithm=MD5#015#012Allow: >>> INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, >>> SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE#015#012Max-Forwards: >>> 69#015#012User-Agent: #########015#012Expires: 3600#015#012Allow-Events: >>> talk,hold,conference,refer,check-sync#015#012Content-Length: >>> 0#015#012#015#012 >>> >>> >>> Feb 4 17:07:23 sbc.amz1.company.com >>> /usr/sbin/opensips[29827]: >>> WARNING:core:log_expiry: #4 is a core action : 72 - 5us - line 1086 >>> >>> Feb 4 17:07:23 sbc.amz1.company.com >>> /usr/sbin/opensips[29820]: >>> WARNING:core:log_expiry: #1 is a module action : t_relay - 5355333us - >>> line 465 >>> >>> Feb 4 17:07:23 sbc.amz1.company.com >>> /usr/sbin/opensips[29827]: >>> WARNING:core:log_expiry: #5 is a module action : ds_is_in_list - 4us - >>> line 943 >>> >>> Feb 4 17:07:23 sbc.amz1.company.com >>> /usr/sbin/opensips[29817]: >>> WARNING:core:log_expiry: #1 is a module action : t_relay - 7375370us - >>> line 465 >>> >>> Feb 4 17:07:23 sbc.amz1.company.com >>> /usr/sbin/opensips[29820]: >>> WARNING:core:log_expiry: #2 is a core action : 78 - 105us - line 604 >>> >>> Feb 4 17:07:23 sbc.amz1.company.com >>> /usr/sbin/opensips[29817]: >>> WARNING:core:log_expiry: #2 is a core action : 78 - 100us - line 604 >>> >>> Feb 4 17:07:23 sbc.amz1.company.com >>> /usr/sbin/opensips[29820]: >>> WARNING:core:log_expiry: #3 is a core action : 78 - 99us - line 635 >>> >>> Feb 4 17:07:23 sbc.amz1.company.com >>> /usr/sbin/opensips[29817]: >>> WARNING:core:log_expiry: #3 is a core action : 78 - 98us - line 635 >>> >>> >>> _______________________________________________ >>> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >>> >>> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Tue Feb 6 07:12:31 2018 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Tue, 6 Feb 2018 14:12:31 +0200 Subject: [OpenSIPS-Users] Registration Loss on opensips SBC In-Reply-To: References: Message-ID: <59b08ffb-b8f6-521d-3bc6-2735104a51b7@opensips.org> Hi Royee, So, my primary suspicion on the save() points to the event routes - to confirm this, you can use the benchmark module to log the required time to execute those routes. And whenever you have an alert on save(), you can correlate it with the execution time for the event routes. On the t_relay() part, it may be more complicated, as the TM module has a complex system of callbacks that are triggered (inline) . And there are many modules using the TM callbacks , any of them potentially delaying the t_relay() action. Another reason may be DNS lookups, if your relaying is DNS based (and not IP based). Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Summit 2018 http://www.opensips.org/events/Summit-2018Amsterdam On 02/06/2018 10:30 AM, Royee Tichauer wrote: > Thanks for lead Bogdan we will look into it. > > Do you also have any idea why the t_relay was blocking for so long? > > Royee > > On Mon, Feb 5, 2018 at 7:23 PM Bogdan-Andrei Iancu > > wrote: > > Hi Royee, > > I don't think it is related to xmlrpc - there are separate > connection, separate processes for this. > > I would rather suspect the event routes - the event is in-line > handled in usrloc, so the save() will cover the delivery/handling > of the contact related events. And if you do something time > consuming in event_route[E_UL_CONTACT_INSERT] lets say, this will > reflect as processing time for the save() function. > > Regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > OpenSIPS Summit 2018 > http://www.opensips.org/events/Summit-2018Amsterdam > > On 02/05/2018 06:44 PM, Royee Tichauer wrote: >> >> Bogdan, >> >> Not sure I follow about breaking each function. What do you mean >> by 'actual internal save()'? >> >> DB insert is not used. >> >> We use: >> >> event_route[E_UL_CONTACT_UPDATE] >> >> event_route[E_UL_CONTACT_DELETE] >> >> event_route[E_UL_CONTACT_INSERT] >> >> btw, we also use the mi_xmlrpc_ng constantly for monitoring if >> that might have something to do with it (I saw a comment on >> opensips 1.6 that there was an opensips blocking issue in some >> cases). >> >> Royee >> >> >> >> On Mon, Feb 5, 2018 at 6:34 PM Bogdan-Andrei Iancu >> > wrote: >> >> Hi Royee, >> >> Well, you need to break for each function the possible types >> of actions. Like for save(): >> - actual internal save() >> - DB insert if realtime mode used >> - potential events triggering - do you use any usrloc >> related events ? >> >> Regards, >> >> Bogdan-Andrei Iancu >> >> OpenSIPS Founder and Developer >> http://www.opensips-solutions.com >> OpenSIPS Summit 2018 >> http://www.opensips.org/events/Summit-2018Amsterdam >> >> On 02/05/2018 06:20 PM, Royee Tichauer wrote: >>> Hi Bogdan, as the logs show: >>> >>> save - 8912725us >>> t_relay - 5355333us >>> >>> Let me know if you have any ideas, >>> Thanks, >>> Royee >>> >>> >>> >>> >>> On Mon, Feb 5, 2018 at 5:50 PM Bogdan-Andrei Iancu >>> > wrote: >>> >>> Hi Royee, >>> >>> what are the time values reported for 't_relay' and >>> 'save' ? the 50ms threashold may not be so relevant for >>> explaining delays of seconds. >>> >>> Best regards, >>> >>> Bogdan-Andrei Iancu >>> >>> OpenSIPS Founder and Developer >>> http://www.opensips-solutions.com >>> OpenSIPS Summit 2018 >>> http://www.opensips.org/events/Summit-2018Amsterdam >>> >>> On 02/05/2018 02:09 PM, Royee Tichauer via Users wrote: >>>> Hi, >>>> >>>> We are using opensips 2.1 as an SBC component and we >>>> see that we have loss of registrations happening once >>>> in a while. When we captured SIP we saw that some >>>> messages are being delayed by a couple of seconds. We >>>> enabled 'exec_execute_message=50000' and saw in the >>>> logs that opensips core methods like 't_realy' and >>>> 'save' . I looked at SAR during this time and there are >>>> no indications of CPU running high during this time. >>>> >>>> Can you help me figure out this issue? >>>> >>>> Thanks, >>>> Royee >>>> >>>> Logs: >>>> -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Tue Feb 6 12:20:29 2018 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Tue, 6 Feb 2018 19:20:29 +0200 Subject: [OpenSIPS-Users] Username in DB URL In-Reply-To: <013d01d39eac$651ded60$2f59c820$@zaark.com> References: <013d01d39eac$651ded60$2f59c820$@zaark.com> Message-ID: <682eceac-7b46-0a50-1f34-9ba5734c6814@opensips.org> Hi Olle, As Alex said there is no work around for this for now. Still the problem can be addressed by coding and it is not so hard to do. The question is what is the best/proper type of escaping to be supported into the URL - I guess the %xy is the proper one for an URL, as per RFC 1738 ). Have you installed opensips form sources ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Summit 2018 http://www.opensips.org/events/Summit-2018Amsterdam On 02/05/2018 08:08 PM, Olle Frimanson wrote: > > Hi, we are experiencing an issue with a roleout in Microsoft Azure, > where the DB username contains a ”@” sign. This seems to be the > default in MySQL in Azure. Is there anyway to escape this the URL for > the clusterer module? > > If you use mysql client it’s typical something like > > Mysql -h host.mysql.azure.com -u opensips at host -popensips opensips, > which works fine > > In clustererar where we use this this translates to: > > modparam("clusterer", > "db_url","mysql://opensips at host:opensips at host.mysql.azure.com/opensips") > > which fails > > Any ideas on a workaround on this? > > BR / Olle > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Tue Feb 6 12:24:05 2018 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Tue, 6 Feb 2018 19:24:05 +0200 Subject: [OpenSIPS-Users] Problem with topology hiding when hairpinning a call In-Reply-To: References: Message-ID: <93c8cdd5-d78f-f43e-b2bd-c921c79108af@opensips.org> Hi Andreas, Do you do callid changing with TH ? also the TH is done on both passes through OpenSIPS ? Best regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Summit 2018 http://www.opensips.org/events/Summit-2018Amsterdam On 02/05/2018 08:14 PM, Andreas Sikkema wrote: > Hi, > > I'm playing with the topology hiding module and I am running into a > problem when a call is sent from the inside of OpenSIPS to the > outside, which is then received back on the outside of OpenSIPS to > sent to the inside of OpenSIP. Calls being answered on the inside or > outside work fine, it's just with hairpinning I have this problem. > > The call setup works fine, but as soon as the call is answered, I am > seeing a lot of lines like this: > > Feb 5 18:52:38 [31118] WARNING:dialog:dlg_onroute: tight matching > failed for BYE with > callid='ee75e7e72504b88f57959fe48d3faacc at 192.168.xxx.xxx'/46, > ftag='1640688292'/10, ttag='1456089971'/10 and direction=0 > Feb 5 18:52:38 [31118] WARNING:dialog:dlg_onroute: dialog > identification elements are > callid='BB_KhVSWzZ+NWR9RVVaMXFoNXpHXFtqLzVndxRWCDIoMzAPQVxcfXhma2FBV1p9fA--'/71, > caller tag='1640688292'/10, callee tag='1456089971'/10 > > Is there something obvious I am doing wrong? The config is basically > the generic topology hiding module example. I was hoping to not have a > seperate instance for both directions ;-) > > Thanks! > > -- > Andreas > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Tue Feb 6 12:31:03 2018 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Tue, 6 Feb 2018 19:31:03 +0200 Subject: [OpenSIPS-Users] fix documentation link In-Reply-To: <25734421.3205880.1517826556288@mail.yahoo.com> References: <25734421.3205880.1517826556288.ref@mail.yahoo.com> <25734421.3205880.1517826556288@mail.yahoo.com> Message-ID: Hi Pasan, Thanks for reporting, we fixed the page. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Summit 2018 http://www.opensips.org/events/Summit-2018Amsterdam On 02/05/2018 12:29 PM, Pasan Meemaduma via Users wrote: > Hi > > could you please fix the hyperlink for *PROTO_BIN* > ? its > refer to same link as *PROTO_HEP* > > > > > http://www.opensips.org/Documentation/Modules-2-3 > > > > > > > > openSIPS | Documentation / Modules - 2.3 > > > > > > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Tue Feb 6 13:26:11 2018 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Tue, 6 Feb 2018 20:26:11 +0200 Subject: [OpenSIPS-Users] =?utf-8?q?Two_training_sessions=2C_two_projects?= =?utf-8?q?=2C_one_event_=E2=80=93_OpenSIPS_Summit_2018?= Message-ID: <780c3d9a-41b3-3263-1a1e-37488cb91e02@opensips.org> TheOpenSIPS Summit 2018 is hosting in Amsterdam two different training sessions, dedicated to two different VoIP projects – an officialOpenSIPS training and an official FreeSWITCH training. */Yes, that is correct and it is apremiere– an official and certified FreeSWITCH training taking place for the first time in Europe!/* So, this year, the OpenSIPS Summit will not only gather more Open Source projects from SIP and RTC ecosystem, but will expand with its training opportunities – more reasons not to mis the OpenSIPS Summit in Amsterdam, 1st of May – justregister now and here . The forth day (4th of May) of the event is entirely dedicated to trainings. Official trainer will run (in parallel) dedicated trainings, one for OpenSIPS, another for FreeSWITCH. See here the full description of the trainings, as content and setup: https://blog.opensips.org/2018/02/06/two-training-sessions-two-projects-one-event-opensips-summit-2018/ See you in Amsterdam !! -- Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Summit 2018 http://www.opensips.org/events/Summit-2018Amsterdam -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Wed Feb 7 04:38:06 2018 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Wed, 7 Feb 2018 11:38:06 +0200 Subject: [OpenSIPS-Users] [15066] WARNING:rr:after_strict: no socket found to match RR [1][XXX.XXX.XXX.XXX:5060] In-Reply-To: <0102016166aad91b-15d647cd-2313-46ff-a6e7-ca6e77c13741-000000@eu-west-1.amazonses.com> References: <01020161564f88ab-94ee6ed7-b89e-4d6e-866d-99d682719034-000000@eu-west-1.amazonses.com> <0102016156e9f5f9-6a281c81-ae20-4a1d-9362-2e98a1fff3e0-000000@eu-west-1.amazonses.com> <3615c9bb-1619-dd03-4354-c511a9e68dd0@opensips.org> <01020161577912d4-87a753fa-ece7-491a-98cb-4d8b9f479e22-000000@eu-west-1.amazonses.com> <0102016166aad91b-15d647cd-2313-46ff-a6e7-ca6e77c13741-000000@eu-west-1.amazonses.com> Message-ID: Hi Brian, Which partyis generating the REFER ? the asterisk boxes from behind the LB ? or the carrier side ? and yes, see you in Amsterdam !! Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Summit 2018 http://www.opensips.org/events/Summit-2018Amsterdam On 02/05/2018 05:52 PM, Brian Southworth wrote: > > I think I get it now thank you Bogdan. > > So I would forward the traffic using the opensips proxy, using the if > (is_method(“refer”)) to the opensips box that would be the B2BUA? To > bridge the call ?. > > Also look forward to Opensips summit in may 😊ill see you all there > got it booked Saturday 😊 > > Regards, > > Brian Southworth > > *From:*Bogdan-Andrei Iancu [mailto:bogdan at opensips.org] > *Sent:* 05 February 2018 15:47 > *To:* Brian Southworth ; OpenSIPS users > mailling list > *Subject:* Re: [OpenSIPS-Users] [15066] WARNING:rr:after_strict: no > socket found to match RR [1][XXX.XXX.XXX.XXX:5060] > > Hi Brian, > > Keep in mind that you cannot make opensips act in the same time as > proxy (as required by the load balancer) and as a end-point (as > required by the B2BUA). Ideally is to run the two services (LB and > B2B) on two opensips instances in a chain. > > Best regards, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > OpenSIPS Summit 2018 > http://www.opensips.org/events/Summit-2018Amsterdam > > On 02/02/2018 07:03 PM, Brian Southworth wrote: > > Sorry my apologies. > > So from the beginning opensips acts as an authorization proxy > which passes the call on to an asterisk box based on load (using > load balancer). > > I am trying to get the opensips proxy to handle call transfers and > I thought the b2bua would be the best way. Initially the refer was > sent to the asterisk box. > > On inbound calls > > The call comes in from the carrier goes to asterisk, asterisk then > passes the sip invite to the proxy which then rings the sip phone. > > What I wish to achieve is a way to transfer an inbound call to an > internal extension or external number. > > Example: > > Caller A receives call àcaller A places call on hold and dials > caller B àcaller B picks up àcaller A presses cisco xfer and call > is passed to caller B > > I was hoping to achieve this using the proxy or asterisk box if > possible. > > I hope this helps. > > Regards, > > Brian Southworth > > *From:*Bogdan-Andrei Iancu [mailto:bogdan at opensips.org] > *Sent:* 02 February 2018 16:50 > *To:* Brian Southworth > ; OpenSIPS users mailling list > > *Subject:* Re: [OpenSIPS-Users] [15066] WARNING:rr:after_strict: > no socket found to match RR [1][XXX.XXX.XXX.XXX:5060] > > I'm a bit confused. The original report was on a record_route() / > loose_route() matter. But you say you have opensips as B2B, so the > RR mechanism must not be used in such a case - you act either as a > end-point, either as a proxy - you cannot be both for the same call. > > Now you have this b2b error, during a call transfer scenario. and > you mentioned LB also :)...so I'm a bit confused - could please > try to put all these pieces together, so I can understand what you > are doing ? > > Regards, > > > Bogdan-Andrei Iancu > > > > OpenSIPS Founder and Developer > > http://www.opensips-solutions.com > > OpenSIPS Summit 2018 > > http://www.opensips.org/events/Summit-2018Amsterdam > > On 02/02/2018 04:27 PM, Brian Southworth wrote: > > Maybe I am doing this wrong but I wanted the B2BUA module to > handle the refer and bridge the calls. > > I have the B2bUA working now. However my issue is that its not > able to send the replies. > > incoming reply > > b2b_reply (B2B.222.7591351.1517580641) > > Feb 2 14:10:47 [22664] ERROR:tm:_reply_light: failed to > generate 408 reply when a final 200 was sent out > > Feb 2 14:10:47 [22664] ERROR:b2b_entities:b2b_send_reply: > failed to send reply with tm > > Feb 2 14:10:47 [22664] > ERROR:b2b_logic:b2b_logic_notify_reply: Sending reply failed - > 408, [B2B.452.342.1517580641] > > Do you need anything else to help me debug this ? I am not > sure why its failing to pass the reply with tm, I have enabled > the param: > > modparam("tm", "pass_provisional_replies", 1) > > I should also note that I am using the load balancer module > also. This normally deals with all call distribution. In and out. > > Regards, > > Brian Southworth > > *From:*Bogdan-Andrei Iancu [mailto:bogdan at opensips.org] > *Sent:* 02 February 2018 14:20 > *To:* Brian Southworth > ; OpenSIPS users mailling > list > *Subject:* Re: [OpenSIPS-Users] [15066] > WARNING:rr:after_strict: no socket found to match RR > [1][XXX.XXX.XXX.XXX:5060] > > Hi Brian, > > Maybe that warning points to a routing error that prevents the > REFER to be route to carrier - make a sip capture to be sure > the REFER from A is properly routed and accepted by the carrier. > > Regards, > > > > Bogdan-Andrei Iancu > > > > OpenSIPS Founder and Developer > > http://www.opensips-solutions.com > > OpenSIPS Summit 2018 > > http://www.opensips.org/events/Summit-2018Amsterdam > > On 02/02/2018 01:38 PM, Brian Southworth wrote: > > Hi Bogdan, > > Thank you very much, so this doesn’t have any impact on > why the call being transferred are dropped ? > > I am trying to transfer a call using the refer method as > that is what the cisco phones use. > > The network is setup like so opensips proxy àasterisk > gateway àcarrier > > Scenario: > > Inbound call comes into the phone like so: carrier àast > àproxy àphone A > > Phone A needs to transfer call to phone B: Phone A dials > phone B àphone B picks up àphone A presses xfer button and > call is dropped. > > Any help would be appreciated. > > Regards, > > Brian Southworth > > *From:*Bogdan-Andrei Iancu [mailto:bogdan at opensips.org] > *Sent:* 02 February 2018 11:29 > *To:* OpenSIPS users mailling list > > ; Brian Southworth > > > *Subject:* Re: [OpenSIPS-Users] [15066] > WARNING:rr:after_strict: no socket found to match RR > [1][XXX.XXX.XXX.XXX:5060] > > Hi Brian, > > That warning means OpenSIPS found a Route header (while > doing loose_route) that is suppose to be of its own, but > the network information from the header does not match any > of the OpenSIPS SIP listeners. > > Best regards, > > > > > Bogdan-Andrei Iancu > > > > OpenSIPS Founder and Developer > > http://www.opensips-solutions.com > > OpenSIPS Summit 2018 > > http://www.opensips.org/events/Summit-2018Amsterdam > > On 02/02/2018 11:14 AM, Brian Southworth wrote: > > I get this when trying to transfer calls using the B2BUA: > > [15066] WARNING:rr:after_strict: no socket found to > match RR [1][xxx.xxx.xxx.xxx:5060] > > When I try looking on the mailing list there are no > other similar posts, could you please shed some light > on what maybe causing this please. > > Regards, > > Brian Southworth > > > > > > > > _______________________________________________ > > Users mailing list > > Users at lists.opensips.org > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -------------- next part -------------- An HTML attachment was scrubbed... URL: From brian.southworth at clocom.uk Wed Feb 7 06:32:14 2018 From: brian.southworth at clocom.uk (=?UTF-8?Q?Brian_Southworth?=) Date: Wed, 7 Feb 2018 11:32:14 +0000 Subject: [OpenSIPS-Users] [15066] WARNING:rr:after_strict: no socket found to match RR [1][XXX.XXX.XXX.XXX:5060] In-Reply-To: References: <01020161564f88ab-94ee6ed7-b89e-4d6e-866d-99d682719034-000000@eu-west-1.amazonses.com> <0102016156e9f5f9-6a281c81-ae20-4a1d-9362-2e98a1fff3e0-000000@eu-west-1.amazonses.com> <3615c9bb-1619-dd03-4354-c511a9e68dd0@opensips.org> <01020161577912d4-87a753fa-ece7-491a-98cb-4d8b9f479e22-000000@eu-west-1.amazonses.com> <0102016166aad91b-15d647cd-2313-46ff-a6e7-ca6e77c13741-000000@eu-west-1.amazonses.com> Message-ID: <0102016170099e99-226313be-8ec4-44e6-bdb4-525eef01def5-000000@eu-west-1.amazonses.com> Hi Bogdan,   The Cisco phone, generates the refer once you press the xfer button when inside a call. Caller opensips asteriskCarrier (cisco) Regards,   Brian Southworth Communications Developer 111 Wilmslow Road Handforth Wilmslow SK9 3ER   T: 03333 446677 W: www.clocom.uk           Like us on Facebook Follow us on Twitter           Clocom is a green company. Think, do you need to print this email?   This message contains confidential information and is intended only for the individual named. If you are not the named addressee you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately by e-mail if you have received this e-mail by mistake and delete this e-mail from your system. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version. Clocom UK Ltd, 111 Wilmslow Road, Handforth, Cheshire, SK9 3ER www.clocom.uk   From: Bogdan-Andrei Iancu [mailto:bogdan at opensips.org] Sent: 07 February 2018 09:38 To: Brian Southworth ; OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] [15066] WARNING:rr:after_strict: no socket found to match RR [1][XXX.XXX.XXX.XXX:5060]   Hi Brian, Which party is generating the REFER ? the asterisk boxes from behind the LB ? or the carrier side ? and yes, see you in Amsterdam !! Regards, Bogdan-Andrei Iancu   OpenSIPS Founder and Developer http://www.openutions.com OpenSIPS Summit 2018 http://www.opensips.org/events/Summit-2018Amsterdam On 02/05/2018 05:52 PM, Brian Southworth wrote: I think I get it now thank you Bogdan. So I would forward the traffic using the opensips proxy, using the if (is_method(refer)) to the opensips box that would be the B2BUA? To bridge the call ?.   Also look forward to Opensips summit in may 😊 ill see you all there got it booked Saturday 😊   Regards,   Brian Southworth   From: Bogdan-Andrei Iancu [mailto:bogdan at opensips.org ] Sent: 05 February 2018 15:47 To: Brian Southworth ; OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] [15066] WARNING:rr:after_strict: no socket found to match RR [1][XXX.XXX.XXX.XXX:5060]   Hi Brian, Keep in mind that you cannot make opensips act in the same time as proxy (as required by the load balancer) and as a end-point (as required by the B2BUA). Ideally is to run the two services (LB and B2B) on two opensips instances in a chain. Best regards, Bogdan-Andrei Iancu   OpenSIPS Founder and Developer   http://www.opensips-solutions.com OpenSIPS Summit 2018   http://www.opensips.org/events/Summit-2018Amsterdam On 02/02/2018 07:03 PM, Brian Southworth wrote: Sorry my apologies.   So from the beginning opensips acts as an authorization proxy which passes the call on to an asterisk box based on load (using load balancer). I am trying to get the opensips proxy to handle call transfers and I thought the b2bua would be the best way. Initially the refer was sent to the asterisk box.   On inbound calls The call comes in from the carrier goes to asterisk, asterisk then passes the sip invite to the proxy which then rings the sip phone.   What I wish to achieve is a way to transfer an inbound call to an internal extension or external number.   Example: Caller A receives call caller A places call on hold and dials caller B caller B picks up caller A presses cisco xfer and call is passed to caller B   I was hoping to achieve this using the proxy or asterisk box if possible.   I hope this helps.   Regards,   Brian Southworth   From: Bogdan-Andrei Iancu [mailto:bogdan at opensips.org ] Sent: 02 February 2018 16:50 To: Brian Southworth ; OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] [15066] WARNING:rr:after_strict: no socket found to match RR [1][XXX.XXX.XXX.XXX:5060]   I'm a bit confused. The original report was on a record_route() / loose_route() matter. But you say you have opensips as B2B, so the RR mechanism must not be used in such a case - you act either as a end-point, either as a proxy - you cannot be both for the same call. Now you have this b2b error, during a call transfer scenario. and you mentioned LB also :)...so I'm a bit confused - could please try to put all these pieces together, so I can understand what you are doing ? Regards, Bogdan-Andrei Iancu   OpenSIPS Founder and Developer   http://www.opensips-solutions.com OpenSIPS Summit 2018   http://www.opensips.org/events/Summit-2018Amsterdam On 02/02/2018 04:27 PM, Brian Southworth wrote: Maybe I am doing this wrong but I wanted the B2BUA module to handle the refer and bridge the calls. I have the B2bUA working now. However my issue is that its not able to send the replies.   incoming reply b2b_reply (B2B.222.7591351.1517580641) Feb  2 14:10:47 [22664] ERROR:tm:_reply_light: failed to generate 408 reply when a final 200 was sent out Feb  2 14:10:47 [22664] ERROR:b2b_entities:b2b_send_reply: failed to send reply with tm Feb  2 14:10:47 [22664] ERROR:b2b_logic:b2b_logic_notify_reply: Sending reply failed - 408, [B2B.452.342.1517580641]   Do you need anything else to help me debug this ? I am not sure why its failing to pass the reply with tm, I have enabled the param: modparam("tm", "pass_provisional_replies", 1)   I should also note that I am using the load balancer module also. This normally deals with all call distribution. In and out.   Regards,   Brian Southworth   From: Bogdan-Andrei Iancu [mailto:bogdan at opensips.org ] Sent: 02 February 2018 14:20 To: Brian Southworth ; OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] [15066] WARNING:rr:after_strict: no socket found to match RR [1][XXX.XXX.XXX.XXX:5060]   Hi Brian, Maybe that warning points to a routing error that prevents the REFER to be route to carrier - make a sip capture to be sure the REFER from A is properly routed and accepted by the carrier. Regards, Bogdan-Andrei Iancu   OpenSIPS Founder and Developer   http://www.opensips-solutions.com OpenSIPS Summit 2018   http://www.opensips.org/events/Summit-2018Amsterdam On 02/02/2018 01:38 PM, Brian Southworth wrote: Hi Bogdan,   Thank you very much, so this doesnt have any impact on why the call being transferred are dropped ?   I am trying to transfer a call using the refer method as that is what the cisco phones use.   The network is setup like so opensips proxy asterisk gateway carrier   Scenario:   Inbound call comes into the phone like so: carrier ast proxy phone A Phone A needs to transfer call to phone B: Phone A dials phone B phone B picks up phone A presses xfer button and call is dropped.   Any help would be appreciated.   Regards,   Brian Southworth   From: Bogdan-Andrei Iancu [mailto:bogdan at opensips.org ] Sent: 02 February 2018 11:29 To: OpenSIPS users mailling list ; Brian Southworth Subject: Re: [OpenSIPS-Users] [15066] WARNING:rr:after_strict: no socket found to match RR [1][XXX.XXX.XXX.XXX:5060]   Hi Brian, That warning means OpenSIPS found a Route header (while doing loose_route) that is suppose to be of its own, but the network information from the header does not match any of the OpenSIPS SIP listeners. Best regards, Bogdan-Andrei Iancu   OpenSIPS Founder and Developer   http://www.opensips-solutions.com OpenSIPS Summit 2018   http://www.opensips.org/events/Summit-2018Amsterdam On 02/02/2018 11:14 AM, Brian Southworth wrote: I get this when trying to transfer calls using the B2BUA: [15066] WARNING:rr:after_strict: no socket found to match RR [1][xxx.xxx.xxx.xxx:5060]   When I try looking on the mailing list there are no other similar posts, could you please shed some light on what maybe causing this please.   Regards,   Brian Southworth _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users           -------------- next part -------------- An HTML attachment was scrubbed... URL: -------------- next part -------------- A non-text attachment was scrubbed... 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Name: not available Type: image/png Size: 2999 bytes Desc: not available URL: From bogdan at opensips.org Wed Feb 7 06:35:57 2018 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Wed, 7 Feb 2018 13:35:57 +0200 Subject: [OpenSIPS-Users] [15066] WARNING:rr:after_strict: no socket found to match RR [1][XXX.XXX.XXX.XXX:5060] In-Reply-To: <0102016170099e99-226313be-8ec4-44e6-bdb4-525eef01def5-000000@eu-west-1.amazonses.com> References: <0102016156e9f5f9-6a281c81-ae20-4a1d-9362-2e98a1fff3e0-000000@eu-west-1.amazonses.com> <3615c9bb-1619-dd03-4354-c511a9e68dd0@opensips.org> <01020161577912d4-87a753fa-ece7-491a-98cb-4d8b9f479e22-000000@eu-west-1.amazonses.com> <0102016166aad91b-15d647cd-2313-46ff-a6e7-ca6e77c13741-000000@eu-west-1.amazonses.com> <0102016170099e99-226313be-8ec4-44e6-bdb4-525eef01def5-000000@eu-west-1.amazonses.com> Message-ID: <431edb91-6ccd-6d7b-7be9-b8730b4a98ee@opensips.org> So the target of the refer is to another Asterisk or may be also back to the carrier ? Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Summit 2018 http://www.opensips.org/events/Summit-2018Amsterdam On 02/07/2018 01:32 PM, Brian Southworth wrote: > > Hi Bogdan, > > The Cisco phone, generates the refer once you press the xfer button > when inside a call. > > Caller àopensipsàasteriskàCarrier > > (cisco) > > Regards, > > Brian Southworth > > Communications Developer > > cid:image001.png at 01D22CAC.1DCB8580 > > 111 Wilmslow Road > > Handforth > > Wilmslow > > SK9 3ER > > T: 03333 446677 > > W: www.clocom.uk __ > > > > > > > > > > cid:image002.png at 01CDDC62.D8483910 > > > > Like us on Facebook > > > > cid:image003.png at 01CDDC62.D8483910 > > > > Follow us on Twitter > > > > cid:image004.png at 01CDDC62.D8483910 > > > > > > > > > > Clocom is a *green* company. Think, do you need to print this email? > > This message contains confidential information and is intended only > for the individual named. If you are not the named addressee you > should not disseminate, distribute or copy this e-mail. Please notify > the sender immediately by e-mail if you have received this e-mail by > mistake and delete this e-mail from your system. E-mail transmission > cannot be guaranteed to be secure or error-free as information could > be intercepted, corrupted, lost, destroyed, arrive late or incomplete, > or contain viruses. The sender therefore does not accept liability for > any errors or omissions in the contents of this message, which arise > as a result of e-mail transmission. If verification is required please > request a hard-copy version. Clocom UK Ltd, 111 Wilmslow Road, > Handforth, Cheshire, SK9 3ER www.clocom.uk __ > > *From:*Bogdan-Andrei Iancu [mailto:bogdan at opensips.org] > *Sent:* 07 February 2018 09:38 > *To:* Brian Southworth ; OpenSIPS users > mailling list > *Subject:* Re: [OpenSIPS-Users] [15066] WARNING:rr:after_strict: no > socket found to match RR [1][XXX.XXX.XXX.XXX:5060] > > Hi Brian, > > Which party is generating the REFER ? the asterisk boxes from behind > the LB ? or the carrier side ? > > and yes, see you in Amsterdam !! > > Regards, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > OpenSIPS Summit 2018 > http://www.opensips.org/events/Summit-2018Amsterdam > > On 02/05/2018 05:52 PM, Brian Southworth wrote: > > I think I get it now thank you Bogdan. > > So I would forward the traffic using the opensips proxy, using the > if (is_method(“refer”)) to the opensips box that would be the > B2BUA? To bridge the call ?. > > Also look forward to Opensips summit in may 😊ill see you all > there got it booked Saturday 😊 > > Regards, > > Brian Southworth > > *From:*Bogdan-Andrei Iancu [mailto:bogdan at opensips.org] > *Sent:* 05 February 2018 15:47 > *To:* Brian Southworth > ; OpenSIPS users mailling list > > *Subject:* Re: [OpenSIPS-Users] [15066] WARNING:rr:after_strict: > no socket found to match RR [1][XXX.XXX.XXX.XXX:5060] > > Hi Brian, > > Keep in mind that you cannot make opensips act in the same time as > proxy (as required by the load balancer) and as a end-point (as > required by the B2BUA). Ideally is to run the two services (LB and > B2B) on two opensips instances in a chain. > > Best regards, > > > Bogdan-Andrei Iancu > > > > OpenSIPS Founder and Developer > > http://www.opensips-solutions.com > > OpenSIPS Summit 2018 > > http://www.opensips.org/events/Summit-2018Amsterdam > > On 02/02/2018 07:03 PM, Brian Southworth wrote: > > Sorry my apologies. > > So from the beginning opensips acts as an authorization proxy > which passes the call on to an asterisk box based on load > (using load balancer). > > I am trying to get the opensips proxy to handle call transfers > and I thought the b2bua would be the best way. Initially the > refer was sent to the asterisk box. > > On inbound calls > > The call comes in from the carrier goes to asterisk, asterisk > then passes the sip invite to the proxy which then rings the > sip phone. > > What I wish to achieve is a way to transfer an inbound call to > an internal extension or external number. > > Example: > > Caller A receives call àcaller A places call on hold and dials > caller B àcaller B picks up àcaller A presses cisco xfer and > call is passed to caller B > > I was hoping to achieve this using the proxy or asterisk box > if possible. > > I hope this helps. > > Regards, > > Brian Southworth > > *From:*Bogdan-Andrei Iancu [mailto:bogdan at opensips.org] > *Sent:* 02 February 2018 16:50 > *To:* Brian Southworth > ; OpenSIPS users mailling > list > *Subject:* Re: [OpenSIPS-Users] [15066] > WARNING:rr:after_strict: no socket found to match RR > [1][XXX.XXX.XXX.XXX:5060] > > I'm a bit confused. The original report was on a > record_route() / loose_route() matter. But you say you have > opensips as B2B, so the RR mechanism must not be used in such > a case - you act either as a end-point, either as a proxy - > you cannot be both for the same call. > > Now you have this b2b error, during a call transfer scenario. > and you mentioned LB also :)...so I'm a bit confused - could > please try to put all these pieces together, so I can > understand what you are doing ? > > Regards, > > > > Bogdan-Andrei Iancu > > > > OpenSIPS Founder and Developer > > http://www.opensips-solutions.com > > OpenSIPS Summit 2018 > > http://www.opensips.org/events/Summit-2018Amsterdam > > On 02/02/2018 04:27 PM, Brian Southworth wrote: > > Maybe I am doing this wrong but I wanted the B2BUA module > to handle the refer and bridge the calls. > > I have the B2bUA working now. However my issue is that its > not able to send the replies. > > incoming reply > > b2b_reply (B2B.222.7591351.1517580641) > > Feb 2 14:10:47 [22664] ERROR:tm:_reply_light: failed to > generate 408 reply when a final 200 was sent out > > Feb 2 14:10:47 [22664] ERROR:b2b_entities:b2b_send_reply: > failed to send reply with tm > > Feb 2 14:10:47 [22664] > ERROR:b2b_logic:b2b_logic_notify_reply: Sending reply > failed - 408, [B2B.452.342.1517580641] > > Do you need anything else to help me debug this ? I am not > sure why its failing to pass the reply with tm, I have > enabled the param: > > modparam("tm", "pass_provisional_replies", 1) > > I should also note that I am using the load balancer > module also. This normally deals with all call > distribution. In and out. > > Regards, > > Brian Southworth > > *From:*Bogdan-Andrei Iancu [mailto:bogdan at opensips.org] > *Sent:* 02 February 2018 14:20 > *To:* Brian Southworth > ; OpenSIPS users > mailling list > > *Subject:* Re: [OpenSIPS-Users] [15066] > WARNING:rr:after_strict: no socket found to match RR > [1][XXX.XXX.XXX.XXX:5060] > > Hi Brian, > > Maybe that warning points to a routing error that prevents > the REFER to be route to carrier - make a sip capture to > be sure the REFER from A is properly routed and accepted > by the carrier. > > Regards, > > > > > Bogdan-Andrei Iancu > > > > OpenSIPS Founder and Developer > > http://www.opensips-solutions.com > > OpenSIPS Summit 2018 > > http://www.opensips.org/events/Summit-2018Amsterdam > > On 02/02/2018 01:38 PM, Brian Southworth wrote: > > Hi Bogdan, > > Thank you very much, so this doesn’t have any impact > on why the call being transferred are dropped ? > > I am trying to transfer a call using the refer method > as that is what the cisco phones use. > > The network is setup like so opensips proxy àasterisk > gateway àcarrier > > Scenario: > > Inbound call comes into the phone like so: carrier > àast àproxy àphone A > > Phone A needs to transfer call to phone B: Phone A > dials phone B àphone B picks up àphone A presses xfer > button and call is dropped. > > Any help would be appreciated. > > Regards, > > Brian Southworth > > *From:*Bogdan-Andrei Iancu [mailto:bogdan at opensips.org] > *Sent:* 02 February 2018 11:29 > *To:* OpenSIPS users mailling list > > ; Brian Southworth > > > *Subject:* Re: [OpenSIPS-Users] [15066] > WARNING:rr:after_strict: no socket found to match RR > [1][XXX.XXX.XXX.XXX:5060] > > Hi Brian, > > That warning means OpenSIPS found a Route header > (while doing loose_route) that is suppose to be of its > own, but the network information from the header does > not match any of the OpenSIPS SIP listeners. > > Best regards, > > > > > > Bogdan-Andrei Iancu > > > > OpenSIPS Founder and Developer > > http://www.opensips-solutions.com > > OpenSIPS Summit 2018 > > http://www.opensips.org/events/Summit-2018Amsterdam > > On 02/02/2018 11:14 AM, Brian Southworth wrote: > > I get this when trying to transfer calls using the > B2BUA: > > [15066] WARNING:rr:after_strict: no socket found > to match RR [1][xxx.xxx.xxx.xxx:5060] > > When I try looking on the mailing list there are > no other similar posts, could you please shed some > light on what maybe causing this please. > > Regards, > > Brian Southworth > > > > > > > > > _______________________________________________ > > Users mailing list > > Users at lists.opensips.org > > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -------------- next part -------------- An HTML attachment was scrubbed... URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: ckggjhhhanaaaked.png Type: image/png Size: 9352 bytes Desc: not available URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: mghoipepgkigfljp.png Type: image/png Size: 2708 bytes Desc: not available URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: gmcbkbpblaojglbj.png Type: image/png Size: 234 bytes Desc: not available URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: lpiemiegflaniagj.png Type: image/png Size: 2999 bytes Desc: not available URL: From goup2010 at gmail.com Wed Feb 7 16:20:48 2018 From: goup2010 at gmail.com (Dragomir Haralambiev) Date: Wed, 7 Feb 2018 23:20:48 +0200 Subject: [OpenSIPS-Users] LoadBalancer destination managment Message-ID: Hello, I'm using Opensips 2.2 LTS latest version from git. I am trying to understand how exactly the load balancer module senses when a destination is out of order. As per documentation if the destination 'probe_mode' is set to 2 to always probe the destination on the defined interval and re-enable the destination if it responds as expected to the probe. However, the opposite doesn't seem to be happening. If the destination is offline (does not respond to probes at all), load balancer module keeps the destination online, regardless of the fact destination does not respond at all. How can I sense this situation (destination offline) and disable the destination if that is the case (no sip response at all)? Best regards, Dragomir -------------- next part -------------- An HTML attachment was scrubbed... URL: From goup2010 at gmail.com Thu Feb 8 03:08:07 2018 From: goup2010 at gmail.com (Dragomir Haralambiev) Date: Thu, 8 Feb 2018 10:08:07 +0200 Subject: [OpenSIPS-Users] =?utf-8?q?=28no_subject=29?= Message-ID: Hello, In my log I see : ERROR: ....non-ASCII escaped character in '%09004915213874522' @ 0 What I can do to remove all not digit simvols from dialing number? Thanks, Dragomir -------------- next part -------------- An HTML attachment was scrubbed... URL: From brian.southworth at clocom.uk Thu Feb 8 10:10:55 2018 From: brian.southworth at clocom.uk (=?UTF-8?Q?Brian_Southworth?=) Date: Thu, 8 Feb 2018 15:10:55 +0000 Subject: [OpenSIPS-Users] Opensips to recognise and send DTMF or relay References: Message-ID: <0102016175f83312-de46e7dc-f7be-41cd-9876-e3491bde40ce-000000@eu-west-1.amazonses.com> Hi All,   How would I go about getting a opensips as a proxy to recognise the DTMF tones or forward them onto asterisk ?   Regards,   Brian Southworth Communications Developer 111 Wilmslow Road Handforth Wilmslow SK9 3ER   T: 03333 446677 W: www.clocom.uk           Like us on Facebook Follow us on Twitter           Clocom is a green company. Think, do you need to print this email?   This message contains confidential information and is intended only for the individual named. If you are not the named addressee you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately by e-mail if you have received this e-mail by mistake and delete this e-mail from your system. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version. Clocom UK Ltd, 111 Wilmslow Road, Handforth, Cheshire, SK9 3ER www.clocom.uk   -------------- next part -------------- An HTML attachment was scrubbed... URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/png Size: 9352 bytes Desc: not available URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/png Size: 2708 bytes Desc: not available URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/png Size: 234 bytes Desc: not available URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/png Size: 2999 bytes Desc: not available URL: From goup2010 at gmail.com Thu Feb 8 10:20:32 2018 From: goup2010 at gmail.com (Dragomir Haralambiev) Date: Thu, 8 Feb 2018 17:20:32 +0200 Subject: [OpenSIPS-Users] Opensips to recognise and send DTMF or relay In-Reply-To: <0102016175f83312-de46e7dc-f7be-41cd-9876-e3491bde40ce-000000@eu-west-1.amazonses.com> References: <0102016175f83312-de46e7dc-f7be-41cd-9876-e3491bde40ce-000000@eu-west-1.amazonses.com> Message-ID: Hi, Opensips is signaling proxy. You need media proxy to make this. Regards, Dragomir 2018-02-08 17:10 GMT+02:00 Brian Southworth : > Hi All, > > > > How would I go about getting a opensips as a proxy to recognise the DTMF > tones or forward them onto asterisk ? > > > > Regards, > > > > Brian Southworth > > Communications Developer > > [image: cid:image001.png at 01D22CAC.1DCB8580] > > 111 Wilmslow Road > > Handforth > > Wilmslow > > SK9 3ER > > > > T: 03333 446677 > > W: www.clocom.uk > > > > > > > > > > > > [image: cid:image002.png at 01CDDC62.D8483910] > > > Like us on Facebook > > [image: cid:image003.png at 01CDDC62.D8483910] > > Follow us on Twitter > > [image: cid:image004.png at 01CDDC62.D8483910] > > > > > > > > > > > > > Clocom is a *green* company. Think, do you need to print this email? > > > > This message contains confidential information and is intended only for > the individual named. If you are not the named addressee you should not > disseminate, distribute or copy this e-mail. Please notify the sender > immediately by e-mail if you have received this e-mail by mistake and > delete this e-mail from your system. E-mail transmission cannot be > guaranteed to be secure or error-free as information could be intercepted, > corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. > The sender therefore does not accept liability for any errors or omissions > in the contents of this message, which arise as a result of e-mail > transmission. If verification is required please request a hard-copy > version. Clocom UK Ltd, 111 Wilmslow Road, Handforth, Cheshire, SK9 3ER > www.clocom.uk > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > -------------- next part -------------- An HTML attachment was scrubbed... URL: -------------- next part -------------- A non-text attachment was scrubbed... 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Name: not available Type: image/png Size: 234 bytes Desc: not available URL: From brian.southworth at clocom.uk Thu Feb 8 10:26:58 2018 From: brian.southworth at clocom.uk (=?UTF-8?Q?Brian_Southworth?=) Date: Thu, 8 Feb 2018 15:26:58 +0000 Subject: [OpenSIPS-Users] Opensips to recognise and send DTMF or relay In-Reply-To: References: <0102016175f83312-de46e7dc-f7be-41cd-9876-e3491bde40ce-000000@eu-west-1.amazonses.com> Message-ID: <010201617606e53e-4bcff799-8d73-4c37-a23f-da793b085635-000000@eu-west-1.amazonses.com> Hi Dragomir,   Sorry for my lack of knowledge I am still learning as I go (which is why ill be attending summit this year 😊 ) So If i install media proxy this should relay the DTMF tones ?  would I need to install the media proxy on asterisk box and the opensips proxy ?   Again sorry for all the questions some of this is still very new to me. Many thanks.   Regards,   Brian Southworth From: Users [mailto:users-bounces at lists.opensips.org] On Behalf Of Dragomir Haralambiev Sent: 08 February 2018 15:21 To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] Opensips to recognise and send DTMF or relay   Hi,   Opensips is signaling proxy. You need media proxy to make this.   Regards, Dragomir   2018-02-08 17:10 GMT+02:00 Brian Southworth >: Hi All,   How would I go about getting a opensips as a proxy to recognise the DTMF tones or forward them onto asterisk ?   Regards,   Brian Southworth Communications Developer 111 Wilmslow Road Handforth Wilmslow SK9 3ER   T: 03333 446677 W: www.clocom.uk           Like us on Facebook Follow us on Twitter           Clocom is a green company. Think, do you need to print this email?   This message contains confidential information and is intended only for the individual named. If you are not the named addressee you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately by e-mail if you have received this e-mail by mistake and delete this e-mail from your system. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version. Clocom UK Ltd, 111 Wilmslow Road, Handforth, Cheshire, SK9 3ER www.clocom.uk   _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users   _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... 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Name: not available Type: image/png Size: 2999 bytes Desc: not available URL: From liviu at opensips.org Thu Feb 8 12:05:50 2018 From: liviu at opensips.org (Liviu Chircu) Date: Thu, 8 Feb 2018 19:05:50 +0200 Subject: [OpenSIPS-Users] Opensips to recognise and send DTMF or relay In-Reply-To: <010201617606e53e-4bcff799-8d73-4c37-a23f-da793b085635-000000@eu-west-1.amazonses.com> References: <0102016175f83312-de46e7dc-f7be-41cd-9876-e3491bde40ce-000000@eu-west-1.amazonses.com> <010201617606e53e-4bcff799-8d73-4c37-a23f-da793b085635-000000@eu-west-1.amazonses.com> Message-ID: Hi Brian, Since Asterisk forces the media to pass through it, this will automatically include the DTMF tones. Now, how to detect them in Asterisk and/or decide further is beyond me :) there are definitely people around here (or #asterisk IRC) who can give hints if that is what you're looking for. Cheers, Liviu Chircu OpenSIPS Developer http://www.opensips-solutions.com On 08.02.2018 17:26, Brian Southworth wrote: > > Hi Dragomir, > > Sorry for my lack of knowledge I am still learning as I go (which is > why ill be attending summit this year 😊) > > So If i install media proxy this should relay the DTMF tones ? would I > need to install the media proxy on asterisk box and the opensips proxy ? > > Again sorry for all the questions some of this is still very new to me. > > Many thanks. > > Regards, > -------------- next part -------------- An HTML attachment was scrubbed... URL: From morozov_a at at-home.ru Fri Feb 9 02:52:19 2018 From: morozov_a at at-home.ru (=?UTF-8?B?0JzQvtGA0L7Qt9C+0LIg0JDQu9C10LrRgdCw0L3QtNGA?=) Date: Fri, 9 Feb 2018 10:52:19 +0300 Subject: [OpenSIPS-Users] Fraud Detection counters. Message-ID: Hello, community. Can you help me? I`ve installed opensips version 2.1.15 with the fraud detection module and I`ve got a question about counter of sequence calls. I configured a rule for international calls, and the counter is summing values after pass to next period. Is there another way to reset the counter, except to make another rule for matching the calls? Regards, Alex. From liviu at opensips.org Fri Feb 9 03:44:27 2018 From: liviu at opensips.org (Liviu Chircu) Date: Fri, 9 Feb 2018 10:44:27 +0200 Subject: [OpenSIPS-Users] Fraud Detection counters. In-Reply-To: References: Message-ID: <3e17f4df-063a-5250-f8bc-4fda96d6f52a@opensips.org> Hi, This is a known bug, and it should be fixed starting with OpenSIPS 2.2, so I recommend you test with a 2.2 or 2.3 OpenSIPS. Regards, Liviu Chircu OpenSIPS Developer http://www.opensips-solutions.com On 09.02.2018 09:52, Морозов Александр wrote: > Hello, community. > > Can you help me? > > I`ve installed opensips version 2.1.15 with the fraud detection module > and I`ve got a question about counter of sequence calls. > > I configured a rule for international calls, and the counter is > summing values after pass to next period. > > Is there another way to reset the counter, except to make another rule > for matching the calls? > > Regards, Alex. > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users From morozov_a at at-home.ru Fri Feb 9 05:58:25 2018 From: morozov_a at at-home.ru (=?UTF-8?B?0JzQvtGA0L7Qt9C+0LIg0JDQu9C10LrRgdCw0L3QtNGA?=) Date: Fri, 9 Feb 2018 13:58:25 +0300 Subject: [OpenSIPS-Users] Fraud Detection counters. In-Reply-To: <3e17f4df-063a-5250-f8bc-4fda96d6f52a@opensips.org> References: <3e17f4df-063a-5250-f8bc-4fda96d6f52a@opensips.org> Message-ID: <4db03e01-842b-c449-3554-f4f27a7b56d5@at-home.ru> I just now tried it on OpenSIPS 2.3.2. Now counter of sequence calls dont reset after matching another rule. Could you explain, how it must work? Regards. 09.02.2018 11:44, Liviu Chircu пишет: > Hi, > > This is a known bug, and it should be fixed starting with OpenSIPS > 2.2, so I recommend you test with a 2.2 or 2.3 OpenSIPS. > > Regards, > > Liviu Chircu > OpenSIPS Developer > http://www.opensips-solutions.com > > On 09.02.2018 09:52, Морозов Александр wrote: >> Hello, community. >> >> Can you help me? >> >> I`ve installed opensips version 2.1.15 with the fraud detection >> module and I`ve got a question about counter of sequence calls. >> >> I configured a rule for international calls, and the counter is >> summing values after pass to next period. >> >> Is there another way to reset the counter, except to make another >> rule for matching the calls? >> >> Regards, Alex. >> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users From art666 at hotmail.com Fri Feb 9 06:43:22 2018 From: art666 at hotmail.com (Arto Kuiri) Date: Fri, 9 Feb 2018 11:43:22 +0000 Subject: [OpenSIPS-Users] gflags module script and MI command difference Message-ID: Hi, I noticed that in opensips script gflags and MI commands gflags is off by 1. if i have this kind of check: if(is_gflag("1")) { xlog("** MAINTANCE MODE \n"); } and I try to set it on with MI command: opensipsctl fifo set_gflag 1 it wont work, but this works: opensipsctl fifo set_gflag 2 is this intended? If it is, I think documentation should mention it? Tested this with 2.4.x Regards, Arto Kuiri -------------- next part -------------- An HTML attachment was scrubbed... URL: From liviu at opensips.org Fri Feb 9 06:52:37 2018 From: liviu at opensips.org (Liviu Chircu) Date: Fri, 9 Feb 2018 13:52:37 +0200 Subject: [OpenSIPS-Users] Fraud Detection counters. In-Reply-To: <4db03e01-842b-c449-3554-f4f27a7b56d5@at-home.ru> References: <3e17f4df-063a-5250-f8bc-4fda96d6f52a@opensips.org> <4db03e01-842b-c449-3554-f4f27a7b56d5@at-home.ru> Message-ID: The idea is that it counts the current consecutive attempt count for the last dialed number, for each user. IIRC, this stat does not change across interval changes. Liviu Chircu OpenSIPS Developer http://www.opensips-solutions.com On 09.02.2018 12:58, Морозов Александр wrote: > I just now tried it on OpenSIPS 2.3.2. > > Now counter of sequence calls dont reset after matching another rule. > Could you explain, how it must work? > > Regards. > > 09.02.2018 11:44, Liviu Chircu пишет: >> Hi, >> >> This is a known bug, and it should be fixed starting with OpenSIPS >> 2.2, so I recommend you test with a 2.2 or 2.3 OpenSIPS. >> >> Regards, >> >> Liviu Chircu >> OpenSIPS Developer >> http://www.opensips-solutions.com >> >> On 09.02.2018 09:52, Морозов Александр wrote: >>> Hello, community. >>> >>> Can you help me? >>> >>> I`ve installed opensips version 2.1.15 with the fraud detection >>> module and I`ve got a question about counter of sequence calls. >>> >>> I configured a rule for international calls, and the counter is >>> summing values after pass to next period. >>> >>> Is there another way to reset the counter, except to make another >>> rule for matching the calls? >>> >>> Regards, Alex. >>> >>> >>> _______________________________________________ >>> Users mailing list >>> Users at lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users From brian.southworth at clocom.uk Fri Feb 9 11:45:28 2018 From: brian.southworth at clocom.uk (=?UTF-8?Q?Brian_Southworth?=) Date: Fri, 9 Feb 2018 16:45:28 +0000 Subject: [OpenSIPS-Users] Error References: Message-ID: <010201617b751df7-79d9b432-b9c4-4f93-8ff9-a6a7b9e087a4-000000@eu-west-1.amazonses.com> dlg_validate_dialog: Script error - validate function before having a dialog   I am trying to fix contact header on the called, ive fixed the register users used to register with a private ip and not local. Now when inbound calls happen they don’t get them due to the call being send to a local ip that isn’t local.   So ive used the validate dialog code chunk and I get back the above   Regards,   Brian Southworth Communications Developer 111 Wilmslow Road Handforth Wilmslow SK9 3ER   T: 03333 446677 W: www.clocom.uk           Like us on Facebook Follow us on Twitter           Clocom is a green company. Think, do you need to print this email?   This message contains confidential information and is intended only for the individual named. If you are not the named addressee you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately by e-mail if you have received this e-mail by mistake and delete this e-mail from your system. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version. Clocom UK Ltd, 111 Wilmslow Road, Handforth, Cheshire, SK9 3ER www.clocom.uk   -------------- next part -------------- An HTML attachment was scrubbed... URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/png Size: 9352 bytes Desc: not available URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/png Size: 2708 bytes Desc: not available URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/png Size: 234 bytes Desc: not available URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/png Size: 2999 bytes Desc: not available URL: From bogdan at opensips.org Mon Feb 12 09:23:11 2018 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Mon, 12 Feb 2018 16:23:11 +0200 Subject: [OpenSIPS-Users] [15066] WARNING:rr:after_strict: no socket found to match RR [1][XXX.XXX.XXX.XXX:5060] In-Reply-To: <01020161707cfe19-1fd9ab6c-de81-4654-a78e-ed2bcef21d4a-000000@eu-west-1.amazonses.com> References: <01020161577912d4-87a753fa-ece7-491a-98cb-4d8b9f479e22-000000@eu-west-1.amazonses.com> <0102016166aad91b-15d647cd-2313-46ff-a6e7-ca6e77c13741-000000@eu-west-1.amazonses.com> <0102016170099e99-226313be-8ec4-44e6-bdb4-525eef01def5-000000@eu-west-1.amazonses.com> <431edb91-6ccd-6d7b-7be9-b8730b4a98ee@opensips.org> <01020161707cfe19-1fd9ab6c-de81-4654-a78e-ed2bcef21d4a-000000@eu-west-1.amazonses.com> Message-ID: <3767b068-006a-74e5-ba2d-91c8df58f73d@opensips.org> Hi Brian, In this case, I guess that the OPenSIPS B2B (handling the REFER) should sit between the OpenSIPS LB and Asterisk - again , this is the case only if the result of the transfer is a call to an Asterisk box too. If the call may be redirected back to a carrier, the OpenSIPS B2B should sit in front of the LB. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Summit 2018 http://www.opensips.org/events/Summit-2018Amsterdam On 02/07/2018 03:38 PM, Brian Southworth wrote: > > Opensips handles the refer sending it to the asterisk box > > Regards, > > Brian Southworth > > Communications Developer > > > 111 Wilmslow Road > > Handforth > > Wilmslow > > SK9 3ER > > T: 03333 446677 > > W: www.clocom.uk __ > > > > > > > > > > __ > > *From:*Bogdan-Andrei Iancu [mailto:bogdan at opensips.org] > *Sent:* 07 February 2018 11:36 > *To:* Brian Southworth ; OpenSIPS users > mailling list > *Subject:* Re: [OpenSIPS-Users] [15066] WARNING:rr:after_strict: no > socket found to match RR [1][XXX.XXX.XXX.XXX:5060] > > So the target of the refer is to another Asterisk or may be also back > to the carrier ? > > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > OpenSIPS Summit 2018 > http://www.opensips.org/events/Summit-2018Amsterdam > > On 02/07/2018 01:32 PM, Brian Southworth wrote: > > Hi Bogdan, > > The Cisco phone, generates the refer once you press the xfer > button when inside a call. > > Caller àopensipsàasteriskàCarrier > > (cisco) > > Regards, > > Brian Southworth > > T: 03333 446677 > > W: www.clocom.uk __ > > > > > > > > > > > > > > > > > > > __ > > *From:*Bogdan-Andrei Iancu [mailto:bogdan at opensips.org] > *Sent:* 07 February 2018 09:38 > *To:* Brian Southworth > ; OpenSIPS users mailling list > > *Subject:* Re: [OpenSIPS-Users] [15066] WARNING:rr:after_strict: > no socket found to match RR [1][XXX.XXX.XXX.XXX:5060] > > Hi Brian, > > Which party is generating the REFER ? the asterisk boxes from > behind the LB ? or the carrier side ? > > and yes, see you in Amsterdam !! > > Regards, > > > Bogdan-Andrei Iancu > > > > OpenSIPS Founder and Developer > > http://www.opensips-solutions.com > > OpenSIPS Summit 2018 > > http://www.opensips.org/events/Summit-2018Amsterdam > > On 02/05/2018 05:52 PM, Brian Southworth wrote: > > I think I get it now thank you Bogdan. > > So I would forward the traffic using the opensips proxy, using > the if (is_method(“refer”)) to the opensips box that would be > the B2BUA? To bridge the call ?. > > Also look forward to Opensips summit in may 😊ill see you all > there got it booked Saturday 😊 > > Regards, > > Brian Southworth > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From brian.southworth at clocom.uk Mon Feb 12 09:41:09 2018 From: brian.southworth at clocom.uk (=?UTF-8?Q?Brian_Southworth?=) Date: Mon, 12 Feb 2018 14:41:09 +0000 Subject: [OpenSIPS-Users] [15066] WARNING:rr:after_strict: no socket found to match RR [1][XXX.XXX.XXX.XXX:5060] In-Reply-To: <3767b068-006a-74e5-ba2d-91c8df58f73d@opensips.org> References: <01020161577912d4-87a753fa-ece7-491a-98cb-4d8b9f479e22-000000@eu-west-1.amazonses.com> <0102016166aad91b-15d647cd-2313-46ff-a6e7-ca6e77c13741-000000@eu-west-1.amazonses.com> <0102016170099e99-226313be-8ec4-44e6-bdb4-525eef01def5-000000@eu-west-1.amazonses.com> <431edb91-6ccd-6d7b-7be9-b8730b4a98ee@opensips.org> <01020161707cfe19-1fd9ab6c-de81-4654-a78e-ed2bcef21d4a-000000@eu-west-1.amazonses.com> <3767b068-006a-74e5-ba2d-91c8df58f73d@opensips.org> Message-ID: <010201618a766184-e83756da-fe3e-4360-a418-a284690bd605-000000@eu-west-1.amazonses.com> Hi Bogan,   Thanks for the reply, so are you saying the load balancer will send the call over to the B2B and then to asterisk ? Again sorry for my lack of knowledge there is still a lot I don’t understand or know.   Regards,   Brian Southworth Communications Developer   From: Bogdan-Andrei Iancu [mailto:bogdan at opensips.org] Sent: 12 February 2018 14:23 To: Brian Southworth ; OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] [15066] WARNING:rr:after_strict: no socket found to match RR [1][XXX.XXX.XXX.XXX:5060]   Hi Brian, In this case, I guess that the OPenSIPS B2B (handling the REFER) should sit between the OpenSIPS LB and Asterisk - again , this is the case only if the result of the transfer is a call to an Asterisk box too. If the call may be redirected back to a carrier, the OpenSIPS B2B should sit in front of the LB. Regards, Bogdan-Andrei Iancu   OpenSIPS Founder and Developer   http://www.opensips-solutions.com OpenSIPS Summit 2018   http://www.opensips.org/events/Summit-2018Amsterdam On 02/07/2018 03:38 PM, Brian Southworth wrote: Opensips handles the refer sending it to the asterisk box   Regards,   Brian Southworth Communications Developer 111 Wilmslow Road Handforth Wilmslow SK9 3ER   T: 03333 446677 W: www.clocom.uk           From: Bogdan-Andrei Iancu [mailto:bogdan at opensips.org ] Sent: 07 February 2018 11:36 To: Brian Southworth ; OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] [15066] WARNING:rr:after_strict: no socket found to match RR [1][XXX.XXX.XXX.XXX:5060]   So the target of the refer is to another Asterisk or may be also back to the carrier ? Bogdan-Andrei Iancu   OpenSIPS Founder and Developer   http://www.opensips-solutions.com OpenSIPS Summit 2018   http://www.opensips.org/events/Summit-2018Amsterdam On 02/07/2018 01:32 PM, Brian Southworth wrote: Hi Bogdan,   The Cisco phone, generates the refer once you press the xfer button when inside a call. Caller àopensipsà asteriskàCarrier (cisco) Regards,   Brian Southworth   T: 03333 446677 W: www.clocom.uk       From: Bogdan-Andrei Iancu [mailto:bogdan at opensips.org ] Sent: 07 February 2018 09:38 To: Brian Southworth ; OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] [15066] WARNING:rr:after_strict: no socket found to match RR [1][XXX.XXX.XXX.XXX:5060]   Hi Brian, Which party is generating the REFER ? the asterisk boxes from behind the LB ? or the carrier side ? and yes, see you in Amsterdam !! Regards, Bogdan-Andrei Iancu   OpenSIPS Founder and Developer   http://www.opensips-solutions.com OpenSIPS Summit 2018   http://www.opensips.org/events/Summit-2018Amsterdam On 02/05/2018 05:52 PM, Brian Southworth wrote: I think I get it now thank you Bogdan. So I would forward the traffic using the opensips proxy, using the if (is_method(“refer”)) to the opensips box that would be the B2BUA? To bridge the call ?.   Also look forward to Opensips summit in may 😊 ill see you all there got it booked Saturday 😊   Regards,   Brian Southworth     -------------- next part -------------- An HTML attachment was scrubbed... URL: From brian.southworth at clocom.uk Mon Feb 12 09:48:54 2018 From: brian.southworth at clocom.uk (=?UTF-8?Q?Brian_Southworth?=) Date: Mon, 12 Feb 2018 14:48:54 +0000 Subject: [OpenSIPS-Users] Database search inside opensips References: Message-ID: <010201618a7d7ba6-785d11ea-57be-48e8-9c50-770df62d0a0d-000000@eu-west-1.amazonses.com> Hi Opensips,   Sorry for all my emails lately.   Is there any way I can run a database search in opensips config while a route is being processed.   Example scenario: Company has over 15 clients each with their own user id (lets call this a callgroup ID), so the call group id needs to be changed based on the extension calling to match the correct call group. $rU->accountcode (this is just an example).   I just thought being able to change this on the fly would be easier than writing loads of new call groups into the Even based routing.   Regards,   Brian Southworth Communications Developer -------------- next part -------------- An HTML attachment was scrubbed... URL: From rosenberg11219 at gmail.com Mon Feb 12 10:05:44 2018 From: rosenberg11219 at gmail.com (Schneur Rosenberg) Date: Mon, 12 Feb 2018 17:05:44 +0200 Subject: [OpenSIPS-Users] Port changes Message-ID: I have this interesting scenario, caller sends call to our OpenSIPS who actsd as a loadbalancer which sends the call to a gateway for termination and the gateway sets Session-Expires: 1800;refresher=uas, the caller sends call the call through port 1090 and the rport in the Via shows rport=1090 and so far everything is ok, but after 15 minutes the gateway sends a reinvite, OpenSIPS sends it to the client and the client sends a OK and OpenSIPS properly sends the OK to the gateway, the gateway sends a ACK to OpenSIPS, but here the problem starts OpenSIPS sends the ACK to the port in the Contact header which is not the same port as the rport, the clients router does not recognize the packet and blocks it, the client sends multiple OK's to OpenSIPS who ignores it because he has already moved on, as far as OpenSIPS is concerned the ACK has already been sent to the next hop, and after approx 30 seconds the client does not receive the ACK and sends a BYE and the call terminates. Why is OpenSIPS changing the port to the Contact port? and what can I do to fix it?

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From rosenberg11219 at gmail.com Mon Feb 12 10:20:53 2018 From: rosenberg11219 at gmail.com (Schneur Rosenberg) Date: Mon, 12 Feb 2018 17:20:53 +0200 Subject: [OpenSIPS-Users] Database search inside opensips In-Reply-To: <010201618a7d7ba6-785d11ea-57be-48e8-9c50-770df62d0a0d-000000@eu-west-1.amazonses.com> References: <010201618a7d7ba6-785d11ea-57be-48e8-9c50-770df62d0a0d-000000@eu-west-1.amazonses.com> Message-ID: You can use avp_db_query

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On Mon, Feb 12, 2018 at 4:48 PM, Brian Southworth wrote: > Hi Opensips, > > > > Sorry for all my emails lately. > > > > Is there any way I can run a database search in opensips config while a > route is being processed. > > > > Example scenario: Company has over 15 clients each with their own user id > (lets call this a callgroup ID), so the call group id needs to be changed > based on the extension calling to match the correct call group. > > $rU->accountcode (this is just an example). > > > > I just thought being able to change this on the fly would be easier than > writing loads of new call groups into the Even based routing. > > > > Regards, > > > > Brian Southworth > > Communications Developer > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > From bogdan at opensips.org Mon Feb 12 16:35:32 2018 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Mon, 12 Feb 2018 23:35:32 +0200 Subject: [OpenSIPS-Users] [15066] WARNING:rr:after_strict: no socket found to match RR [1][XXX.XXX.XXX.XXX:5060] In-Reply-To: <010201618a766184-e83756da-fe3e-4360-a418-a284690bd605-000000@eu-west-1.amazonses.com> References: <0102016166aad91b-15d647cd-2313-46ff-a6e7-ca6e77c13741-000000@eu-west-1.amazonses.com> <0102016170099e99-226313be-8ec4-44e6-bdb4-525eef01def5-000000@eu-west-1.amazonses.com> <431edb91-6ccd-6d7b-7be9-b8730b4a98ee@opensips.org> <01020161707cfe19-1fd9ab6c-de81-4654-a78e-ed2bcef21d4a-000000@eu-west-1.amazonses.com> <3767b068-006a-74e5-ba2d-91c8df58f73d@opensips.org> <010201618a766184-e83756da-fe3e-4360-a418-a284690bd605-000000@eu-west-1.amazonses.com> Message-ID: Yes, the chain is: carrier -> LB -> B2B -> Asterisk. And when Asterisk generates an REFER (for call transfer) , the REFER will be received by B2B which will handle it (it will generate a new call leg) without changing anything on the leg connecting the b2b to the carrier. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Summit 2018 http://www.opensips.org/events/Summit-2018Amsterdam On 02/12/2018 04:41 PM, Brian Southworth wrote: > > Hi Bogan, > > Thanks for the reply, so are you saying the load balancer will send > the call over to the B2B and then to asterisk ? > > Again sorry for my lack of knowledge there is still a lot I don’t > understand or know. > > Regards, > > Brian Southworth > > Communications Developer > > *From:*Bogdan-Andrei Iancu [mailto:bogdan at opensips.org] > *Sent:* 12 February 2018 14:23 > *To:* Brian Southworth ; OpenSIPS users > mailling list > *Subject:* Re: [OpenSIPS-Users] [15066] WARNING:rr:after_strict: no > socket found to match RR [1][XXX.XXX.XXX.XXX:5060] > > Hi Brian, > > In this case, I guess that the OPenSIPS B2B (handling the REFER) > should sit between the OpenSIPS LB and Asterisk - again , this is the > case only if the result of the transfer is a call to an Asterisk box > too. If the call may be redirected back to a carrier, the OpenSIPS B2B > should sit in front of the LB. > > Regards, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > OpenSIPS Summit 2018 > http://www.opensips.org/events/Summit-2018Amsterdam > > On 02/07/2018 03:38 PM, Brian Southworth wrote: > > Opensips handles the refer sending it to the asterisk box > > Regards, > > Brian Southworth > > Communications Developer > > > > 111 Wilmslow Road > > Handforth > > Wilmslow > > SK9 3ER > > T: 03333 446677 > > W: www.clocom.uk __ > > > > > > > > > > __ > > *From:*Bogdan-Andrei Iancu [mailto:bogdan at opensips.org] > *Sent:* 07 February 2018 11:36 > *To:* Brian Southworth > ; OpenSIPS users mailling list > > *Subject:* Re: [OpenSIPS-Users] [15066] WARNING:rr:after_strict: > no socket found to match RR [1][XXX.XXX.XXX.XXX:5060] > > So the target of the refer is to another Asterisk or may be also > back to the carrier ? > > > > Bogdan-Andrei Iancu > > > > OpenSIPS Founder and Developer > > http://www.opensips-solutions.com > > OpenSIPS Summit 2018 > > http://www.opensips.org/events/Summit-2018Amsterdam > > On 02/07/2018 01:32 PM, Brian Southworth wrote: > > Hi Bogdan, > > The Cisco phone, generates the refer once you press the xfer > button when inside a call. > > Caller àopensipsàasteriskàCarrier > > (cisco) > > Regards, > > Brian Southworth > > T: 03333 446677 > > W: www.clocom.uk __ > > > > > > > > > > > > > > > > > > > *From:*Bogdan-Andrei Iancu [mailto:bogdan at opensips.org] > *Sent:* 07 February 2018 09:38 > *To:* Brian Southworth > ; OpenSIPS users mailling > list > *Subject:* Re: [OpenSIPS-Users] [15066] > WARNING:rr:after_strict: no socket found to match RR > [1][XXX.XXX.XXX.XXX:5060] > > Hi Brian, > > Which party is generating the REFER ? the asterisk boxes from > behind the LB ? or the carrier side ? > > and yes, see you in Amsterdam !! > > Regards, > > > > Bogdan-Andrei Iancu > > > > OpenSIPS Founder and Developer > > http://www.opensips-solutions.com > > OpenSIPS Summit 2018 > > http://www.opensips.org/events/Summit-2018Amsterdam > > On 02/05/2018 05:52 PM, Brian Southworth wrote: > > I think I get it now thank you Bogdan. > > So I would forward the traffic using the opensips proxy, > using the if (is_method(“refer”)) to the opensips box that > would be the B2BUA? To bridge the call ?. > > Also look forward to Opensips summit in may 😊ill see you > all there got it booked Saturday 😊 > > Regards, > > Brian Southworth > -------------- next part -------------- An HTML attachment was scrubbed... URL: From pasandev at ymail.com Mon Feb 12 20:39:58 2018 From: pasandev at ymail.com (Pasan Meemaduma) Date: Tue, 13 Feb 2018 01:39:58 +0000 (UTC) Subject: [OpenSIPS-Users] acc extra_fields param References: <978310918.155131.1518485998683.ref@mail.yahoo.com> Message-ID: <978310918.155131.1518485998683@mail.yahoo.com> Hi List, I'm working on getting my opensips 1.10 -> 2.3.3 :) As db_extra param is not there anymore I tried to use extra_fields param in acc module and it appears it can hold string values only. Is it the case ? I tried to use it to hold an integer value and it comes up as empty string ''. Is it a bug or purpose set only to accept string data only ? cause my db column has integer type for it. following is my config modparam("acc", "extra_fields", "db: fu->from_uri; ru->to_uri; islocal->islocal; callid->\"exe-callid\"; direction->direction") $acc_extra(islocal) = 1; When call hangups it failed to insert the cdr record to acc table as islocal comes up as ''. Ex:-ERROR:db_postgres:db_postgres_submit_query: 0x7fd07fb45d18 PQsendQuery Error: ERROR:  invalid input syntax for integer: ""#012LINE 1: ...sip:xx at xx;transport=UDP','','376ab2...#012                                                             ^#012 Query: insert into acc (method,from_tag,to_tag,callid,sip_code,sip_reason,time,from_uri,to_uri,islocal,"exe-callid",direction,setuptime,created,duration,ms_duration ) values ('INVITE','101c267c','464b920d','6m1aTeUcif5ctoXI7ZSq9A..','200','OK','2018-02-09 17:55:41','sip:xx at xx;transport=UDP','sip:xx at xxx;transport=UDP','','376ab2da-0d66-11e8-9410-bb1806756cbd','in',11,'2018-02-09 17:55:30',4,4298) -------------- next part -------------- An HTML attachment was scrubbed... URL: From pasandev at ymail.com Mon Feb 12 20:49:49 2018 From: pasandev at ymail.com (Pasan Meemaduma) Date: Tue, 13 Feb 2018 01:49:49 +0000 (UTC) Subject: [OpenSIPS-Users] Port changes In-Reply-To: References: Message-ID: <630967390.118289.1518486589820@mail.yahoo.com> Hi Schneur, do you have force_rport() in your opensips.cfg ? hard to make any other comments without looking at your config. On Monday, 12 February 2018, 20:36, Schneur Rosenberg wrote: I have this interesting scenario, caller sends call to our OpenSIPS who actsd as a loadbalancer which sends the call to a gateway for termination and the gateway sets Session-Expires: 1800;refresher=uas, the caller sends call the call through port 1090 and the rport in the Via shows rport=1090 and so far everything is ok, but after 15 minutes the gateway sends a reinvite,  OpenSIPS sends it to the client and the client sends a OK and OpenSIPS properly sends the OK to the gateway, the gateway sends a ACK to OpenSIPS, but here the problem starts OpenSIPS sends the ACK to the port in the Contact header which is not the same port as the rport, the clients router does not recognize the packet and blocks it, the client sends multiple OK's to OpenSIPS who ignores it because he has already moved on, as far as OpenSIPS is concerned the ACK has already been sent to the next hop, and after approx 30 seconds the client does not receive the ACK and sends a BYE and the call terminates. Why is OpenSIPS changing the port to the Contact port? and what can I do to fix it?

                       
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_______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From matangiraa at afri-com.net Tue Feb 13 01:51:53 2018 From: matangiraa at afri-com.net (abisai matangira) Date: Tue, 13 Feb 2018 08:51:53 +0200 Subject: [OpenSIPS-Users] Prefered Media Proxy or RTP Proxy Message-ID: Hi Which is the preferred or stable Media proxy for Opensips from below RTP Proxy Media Proxy From bogdan at opensips.org Tue Feb 13 04:29:34 2018 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Tue, 13 Feb 2018 11:29:34 +0200 Subject: [OpenSIPS-Users] gflags module script and MI command difference In-Reply-To: References: Message-ID: <8ccd86c8-3172-c1d0-9a34-a05db7ae52b9@opensips.org> Hi Arto, No, there should be no offset at all. The is_gflag function takes as parameter the index of the bit to be checked, while the MI functions (like set_gflag) take as parameter a bitmask with all the bits to be operated with. So, if you setting the bitmask 2 -> you set bit index 1 (2<<1) . With bitmask 1 you set the bit index 0 (2<<0). Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Summit 2018 http://www.opensips.org/events/Summit-2018Amsterdam On 02/09/2018 01:43 PM, Arto Kuiri wrote: > > Hi, > > > I noticed that in opensips script gflags and MI commands gflags is off > by 1. > > > if i have this kind of check: > > > if(is_gflag("1")) { > xlog("** MAINTANCE MODE \n"); > } > > > and I try to set it on with MI command: > > opensipsctl fifo set_gflag 1 > > it wont work, but this works: > > opensipsctl fifo set_gflag 2 > > is this intended? If it is, I think documentation should mention it? > > Tested this with 2.4.x > > Regards, > Arto Kuiri > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Tue Feb 13 04:34:43 2018 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Tue, 13 Feb 2018 11:34:43 +0200 Subject: [OpenSIPS-Users] Error In-Reply-To: <010201617b751df7-79d9b432-b9c4-4f93-8ff9-a6a7b9e087a4-000000@eu-west-1.amazonses.com> References: <010201617b751df7-79d9b432-b9c4-4f93-8ff9-a6a7b9e087a4-000000@eu-west-1.amazonses.com> Message-ID: Hi Brian, The idea is the function needs a dialog context in order to work - as this function is to be used for sequential requests, you have to be sure your request matched its dialog before using this function. The dialog matching (for sequential) is done via loose_route() (if you do not use topo hidding) , via dialog_match() or via topology_hiding_match() if using topo hiding. Best regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Summit 2018 http://www.opensips.org/events/Summit-2018Amsterdam On 02/09/2018 06:45 PM, Brian Southworth wrote: > > dlg_validate_dialog: Script error - validate function before having a > dialog > > I am trying to fix contact header on the called, ive fixed the > register users used to register with a private ip and not local. > > Now when inbound calls happen they don’t get them due to the call > being send to a local ip that isn’t local. > > So ive used the validate dialog code chunk and I get back the above > > Regards, > > Brian Southworth > > Communications Developer > > cid:image001.png at 01D22CAC.1DCB8580 > > 111 Wilmslow Road > > Handforth > > Wilmslow > > SK9 3ER > > T: 03333 446677 > > W: www.clocom.uk __ > > > > > > > > > > cid:image002.png at 01CDDC62.D8483910 > > > > Like us on Facebook > > > > cid:image003.png at 01CDDC62.D8483910 > > > > Follow us on Twitter > > > > cid:image004.png at 01CDDC62.D8483910 > > > > > > > > > > Clocom is a *green* company. Think, do you need to print this email? > > This message contains confidential information and is intended only > for the individual named. If you are not the named addressee you > should not disseminate, distribute or copy this e-mail. Please notify > the sender immediately by e-mail if you have received this e-mail by > mistake and delete this e-mail from your system. E-mail transmission > cannot be guaranteed to be secure or error-free as information could > be intercepted, corrupted, lost, destroyed, arrive late or incomplete, > or contain viruses. The sender therefore does not accept liability for > any errors or omissions in the contents of this message, which arise > as a result of e-mail transmission. If verification is required please > request a hard-copy version. Clocom UK Ltd, 111 Wilmslow Road, > Handforth, Cheshire, SK9 3ER www.clocom.uk __ > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: ahaajploaadpaeje.png Type: image/png Size: 9352 bytes Desc: not available URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: hklmcoodbhmdpgkn.png Type: image/png Size: 2708 bytes Desc: not available URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: njnhpbmhfedjngop.png Type: image/png Size: 234 bytes Desc: not available URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: idkcjnfokheeeckj.png Type: image/png Size: 2999 bytes Desc: not available URL: From bogdan at opensips.org Tue Feb 13 04:56:00 2018 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Tue, 13 Feb 2018 11:56:00 +0200 Subject: [OpenSIPS-Users] Prefered Media Proxy or RTP Proxy In-Reply-To: References: Message-ID: <3800e743-bad4-c9c1-f1e1-d5b31acc0c9f@opensips.org> Hi, All three engines are stable and properly work: rtpproxy mediaproxy rtpengine it is up to you to pick the one that does the job for you (in terms of fulfilling your needs). Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Summit 2018 http://www.opensips.org/events/Summit-2018Amsterdam On 02/13/2018 08:51 AM, abisai matangira wrote: > Hi > > Which is the preferred or stable Media proxy for Opensips from below > > RTP Proxy > > Media Proxy > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users From bogdan at opensips.org Tue Feb 13 05:04:36 2018 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Tue, 13 Feb 2018 12:04:36 +0200 Subject: [OpenSIPS-Users] Port changes In-Reply-To: References: Message-ID: <648da842-8997-2c0a-6775-a57eb7622b7e@opensips.org> Hi Schneur, The ACK is routed back via the Route set (the Record Route headers from the initial Invite) and the Contact received in the final reply (in the 200 OK in your case). This is called in-dialog routing and is exclusively based on Route set (RR headers + Contact). The rport as part of via is used only for routing back the replies for a requests (in transaction routing) and has no impact on the in-dialog routing. So, if the RURI in ACK (which should be the Contact in the received 200 OK) is broken, maybe you should do a fix_nated_contact() for the Contact when handling the 200 OK reply. Best regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Summit 2018 http://www.opensips.org/events/Summit-2018Amsterdam On 02/12/2018 05:05 PM, Schneur Rosenberg wrote: > I have this interesting scenario, caller sends call to our OpenSIPS > who actsd as a loadbalancer which sends the call to a gateway for > termination and the gateway sets Session-Expires: 1800;refresher=uas, > the caller sends call the call through port 1090 and the rport in the > Via shows rport=1090 and so far everything is ok, but after 15 minutes > the gateway sends a reinvite, OpenSIPS sends it to the client and the > client sends a OK and OpenSIPS properly sends the OK to the gateway, > the gateway sends a ACK to OpenSIPS, but here the problem starts > OpenSIPS sends the ACK to the port in the Contact header which is not > the same port as the rport, the clients router does not recognize the > packet and blocks it, the client sends multiple OK's to OpenSIPS who > ignores it because he has already moved on, as far as OpenSIPS is > concerned the ACK has already been sent to the next hop, and after > approx 30 seconds the client does not receive the ACK and sends a BYE > and the call terminates. > > Why is OpenSIPS changing the port to the Contact port? and what can I > do to fix it? > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users From razvan at opensips.org Tue Feb 13 05:35:19 2018 From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=) Date: Tue, 13 Feb 2018 12:35:19 +0200 Subject: [OpenSIPS-Users] OpenSIPS Organization @ GSoC 2018 Message-ID: <3e4b3501-c1bf-87df-91e0-6319cf12fbf3@opensips.org> Hello, everyone! We are very excited to announce you that we've just been accepted as an organization for this year's Google Summer of Code program[1]! We are looking forward to getting students involved in our community and work together with them on our projects. You can find the ideas we have proposed for them here[2]. If you're a student and you want to get involved into the GSoC project, don't hesitate to contact us over email, or on IRC and make a proposal for one of the projects! [1] https://summerofcode.withgoogle.com/organizations/5634263836262400/ [2] http://www.opensips.org/Development/IdeasList Cheers, -- Răzvan Crainea OpenSIPS Core Developer http://www.opensips-solutions.com OpenSIPS Summit 2018 http://www.opensips.org/events/Summit-2018Amsterdam From rosenberg11219 at gmail.com Tue Feb 13 05:37:27 2018 From: rosenberg11219 at gmail.com (Schneur Rosenberg) Date: Tue, 13 Feb 2018 12:37:27 +0200 Subject: [OpenSIPS-Users] Port changes In-Reply-To: <648da842-8997-2c0a-6775-a57eb7622b7e@opensips.org> References: <648da842-8997-2c0a-6775-a57eb7622b7e@opensips.org> Message-ID: Bogdan its only happening when the clients SIP ALG fixes the contact, when the contact is broken I'm already using fix_nated_contact, do you think I should use fix_nated_contact even when the contact was already fixed by the SIP ALG? The clients router does not support disabling SIP ALG, I can try using a non standard port, I've sen some routers ALG ignoring the packets when sent to alternative port. I feel like its a bug in OpenSIPS, because the reinvite is routed to the proper port, the issue starts with the ACK for the reinvite, I will email the trace directly to you. On Tue, Feb 13, 2018 at 12:04 PM, Bogdan-Andrei Iancu wrote: > Hi Schneur, > > The ACK is routed back via the Route set (the Record Route headers from the > initial Invite) and the Contact received in the final reply (in the 200 OK > in your case). This is called in-dialog routing and is exclusively based on > Route set (RR headers + Contact). > > The rport as part of via is used only for routing back the replies for a > requests (in transaction routing) and has no impact on the in-dialog > routing. > > So, if the RURI in ACK (which should be the Contact in the received 200 OK) > is broken, maybe you should do a fix_nated_contact() for the Contact when > handling the 200 OK reply. > > Best regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > OpenSIPS Summit 2018 > http://www.opensips.org/events/Summit-2018Amsterdam > > > On 02/12/2018 05:05 PM, Schneur Rosenberg wrote: >> >> I have this interesting scenario, caller sends call to our OpenSIPS >> who actsd as a loadbalancer which sends the call to a gateway for >> termination and the gateway sets Session-Expires: 1800;refresher=uas, >> the caller sends call the call through port 1090 and the rport in the >> Via shows rport=1090 and so far everything is ok, but after 15 minutes >> the gateway sends a reinvite, OpenSIPS sends it to the client and the >> client sends a OK and OpenSIPS properly sends the OK to the gateway, >> the gateway sends a ACK to OpenSIPS, but here the problem starts >> OpenSIPS sends the ACK to the port in the Contact header which is not >> the same port as the rport, the clients router does not recognize the >> packet and blocks it, the client sends multiple OK's to OpenSIPS who >> ignores it because he has already moved on, as far as OpenSIPS is >> concerned the ACK has already been sent to the next hop, and after >> approx 30 seconds the client does not receive the ACK and sends a BYE >> and the call terminates. >> >> Why is OpenSIPS changing the port to the Contact port? and what can I >> do to fix it? >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > From bogdan at opensips.org Tue Feb 13 07:08:11 2018 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Tue, 13 Feb 2018 14:08:11 +0200 Subject: [OpenSIPS-Users] acc extra_fields param In-Reply-To: <978310918.155131.1518485998683@mail.yahoo.com> References: <978310918.155131.1518485998683.ref@mail.yahoo.com> <978310918.155131.1518485998683@mail.yahoo.com> Message-ID: Hi Pasan, The "extra" support in ACC is always producing output as strings (when performing the DB query). Nevertheless this should not prevent you pushing integer or string values into the $acc_extra() variable. But, it seems you discovered a BUG here :D...if the pushed value is a static int (from the script), it is not properly converted :(. I just pushed a fix on GIT repo (master and 2.3) - so please update you checkout or if using packages, use the nightly build (of course, wait for the night :) first ) Best regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Summit 2018 http://www.opensips.org/events/Summit-2018Amsterdam On 02/13/2018 03:39 AM, Pasan Meemaduma via Users wrote: > Hi List, > > I'm working on getting my opensips 1.10 -> 2.3.3 :) > > As db_extra param is not there anymore I tried to use extra_fields > param in acc module and it appears it can hold string values only. Is > it the case ? > > I tried to use it to hold an integer value and it comes up as empty > string ''. > > Is it a bug or purpose set only to accept string data only ? cause my > db column has integer type for it. > > following is my config > > modparam("acc", "extra_fields", "db: fu->from_uri; ru->to_uri; > islocal->islocal; callid->\"exe-callid\"; direction->direction") > > > > $acc_extra(islocal) = 1; > > > When call hangups it failed to insert the cdr record to acc table as > islocal comes up as ''. > > Ex:- > ERROR:db_postgres:db_postgres_submit_query: 0x7fd07fb45d18 PQsendQuery > Error: ERROR: invalid input syntax for integer: ""#012LINE 1: > ...sip:xx at xx;transport=UDP','','376ab2...#012 ^#012 Query: insert into > acc > (method,from_tag,to_tag,callid,sip_code,sip_reason,time,from_uri,to_uri,islocal,"exe-callid",direction,setuptime,created,duration,ms_duration > ) values > ('INVITE','101c267c','464b920d','6m1aTeUcif5ctoXI7ZSq9A..','200','OK','2018-02-09 > 17:55:41','sip:xx at xx;transport=UDP','sip:xx at xxx;transport=UDP','','376ab2da-0d66-11e8-9410-bb1806756cbd','in',11,'2018-02-09 > 17:55:30',4,4298) > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From brian.southworth at clocom.uk Tue Feb 13 10:35:35 2018 From: brian.southworth at clocom.uk (=?UTF-8?Q?Brian_Southworth?=) Date: Tue, 13 Feb 2018 15:35:35 +0000 Subject: [OpenSIPS-Users] migrating to opensips 2.3 from 2.2 References: Message-ID: <010201618fce9437-935d6432-bb78-4a2d-8a99-6566b28dd356-000000@eu-west-1.amazonses.com> Hi All,   I seem to be getting an error when running the migration. It is also saying the  following when I run the following command: opensipsdbctl migrate opensips_2_2 opensips_2_3   INFO: MySQL DB migration tool for OpenSIPS 2.1.x databases ---------------------------------------------------------------------- WARNING: We recommend using this tool ONLY in order to upgrade an existing OpenSIPS 2.1.x MySQL database to the 2.2 schema. Behaviour when automatically migrating earlier DB versions (1.8, 1.9, 1.10, 1.11) to 2.2 is undefined   my version is 2.2.3   also when I enter my password for mysql (this is an external DB not internal) I get this   ERROR 1045 (28000): Access denied for user 'opensips'@'local_ip.lightspeed.jcsnms.sbcglobal.net' (using password: YES) ERROR 1045 (28000): Access denied for user 'opensips'@'local_ip.lightspeed.jcsnms.sbcglobal.net' (using password: YES)   Any idea how I fix this ? I want to start trying to use the EBR module which was made available in opensips 2.3   Regards,   Brian Southworth Communications Developer -------------- next part -------------- An HTML attachment was scrubbed... URL: From rmundkowsky at ets.org Tue Feb 13 13:58:01 2018 From: rmundkowsky at ets.org (Mundkowsky, Robert) Date: Tue, 13 Feb 2018 18:58:01 +0000 Subject: [OpenSIPS-Users] Prefered Media Proxy or RTP Proxy In-Reply-To: <3800e743-bad4-c9c1-f1e1-d5b31acc0c9f@opensips.org> References: <3800e743-bad4-c9c1-f1e1-d5b31acc0c9f@opensips.org> Message-ID: I am curious about this too. I am guessing from this webpage (https://www.voip-info.org/wiki/view/MediaProxy+Comparison) and some others web pages that: - rtpproxy is the oldest - mediaproxy and rtpengine (formerly named mediaproxy-ng ) likely are newer and have more features and rtpengine seems to have the most documentation (that I can find) Robert -----Original Message----- From: Users [mailto:users-bounces at lists.opensips.org] On Behalf Of Bogdan-Andrei Iancu Sent: Tuesday, February 13, 2018 4:56 AM To: OpenSIPS users mailling list ; abisai matangira Subject: Re: [OpenSIPS-Users] Prefered Media Proxy or RTP Proxy Hi, All three engines are stable and properly work: rtpproxy mediaproxy rtpengine it is up to you to pick the one that does the job for you (in terms of fulfilling your needs). Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.opensips-solutions.com&data=02%7C01%7Crmundkowsky%40ets.org%7C3e67609ec71a453c4bf908d572c83017%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636541126500103944&sdata=Zt%2FAD0tfCYRra7POnZUKdDtzlE8EMvXP%2F5BFaTNjetA%3D&reserved=0 OpenSIPS Summit 2018 https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Fwww.opensips.org%2Fevents%2FSummit-2018Amsterdam&data=02%7C01%7Crmundkowsky%40ets.org%7C3e67609ec71a453c4bf908d572c83017%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636541126500103944&sdata=L15jQmVGf0%2FbLOlE604N7kvjeP99VNM0S6Q9OmGcEHY%3D&reserved=0 On 02/13/2018 08:51 AM, abisai matangira wrote: > Hi > > Which is the preferred or stable Media proxy for Opensips from below > > RTP Proxy > > Media Proxy > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Flists. > opensips.org%2Fcgi-bin%2Fmailman%2Flistinfo%2Fusers&data=02%7C01%7Crmu > ndkowsky%40ets.org%7C3e67609ec71a453c4bf908d572c83017%7C0ba6e9b760b34f > ae92f37e6ddd9e9b65%7C0%7C0%7C636541126500103944&sdata=55y1%2BpK%2B0Ku1 > hmghUTEQI4vfN32lpk4WSjig3Y9AT3U%3D&reserved=0 _______________________________________________ Users mailing list Users at lists.opensips.org https://na01.safelinks.protection.outlook.com/?url=http%3A%2F%2Flists.opensips.org%2Fcgi-bin%2Fmailman%2Flistinfo%2Fusers&data=02%7C01%7Crmundkowsky%40ets.org%7C3e67609ec71a453c4bf908d572c83017%7C0ba6e9b760b34fae92f37e6ddd9e9b65%7C0%7C0%7C636541126500103944&sdata=55y1%2BpK%2B0Ku1hmghUTEQI4vfN32lpk4WSjig3Y9AT3U%3D&reserved=0 ________________________________ This e-mail and any files transmitted with it may contain privileged or confidential information. It is solely for use by the individual for whom it is intended, even if addressed incorrectly. If you received this e-mail in error, please notify the sender; do not disclose, copy, distribute, or take any action in reliance on the contents of this information; and delete it from your system. Any other use of this e-mail is prohibited. Thank you for your compliance. ________________________________ From matangiraa at afri-com.net Tue Feb 13 14:18:37 2018 From: matangiraa at afri-com.net (Abisai Matangira) Date: Tue, 13 Feb 2018 19:18:37 +0000 Subject: [OpenSIPS-Users] Prefered Media Proxy or RTP Proxy Message-ID: <0d58cad5-9f3a-489f-98b5-a62c0c3dcef5@afri-com.net> Thanks , There seems to be limited documentation on setting If possible can you share one which has good documentation for Centos environment Sent from Nine ________________________________ From: Bogdan-Andrei Iancu Sent: Tuesday, 13 February 2018 11:57 am To: OpenSIPS users mailling list; Abisai Matangira Subject: Re: [OpenSIPS-Users] Prefered Media Proxy or RTP Proxy Hi, All three engines are stable and properly work: rtpproxy mediaproxy rtpengine it is up to you to pick the one that does the job for you (in terms of fulfilling your needs). Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Summit 2018 http://www.opensips.org/events/Summit-2018Amsterdam On 02/13/2018 08:51 AM, abisai matangira wrote: > Hi > > Which is the preferred or stable Media proxy for Opensips from below > > RTP Proxy > > Media Proxy > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From pasandev at ymail.com Tue Feb 13 21:06:58 2018 From: pasandev at ymail.com (Pasan Meemaduma) Date: Wed, 14 Feb 2018 02:06:58 +0000 (UTC) Subject: [OpenSIPS-Users] acc extra_fields param In-Reply-To: References: <978310918.155131.1518485998683.ref@mail.yahoo.com> <978310918.155131.1518485998683@mail.yahoo.com> Message-ID: <1536923565.539406.1518574018400@mail.yahoo.com> Hi Bogdan, Thanks for the response. I'll get my opensips rebuild , can you let me know the commit id so I know I got it correctly ? On Tuesday, 13 February 2018, 17:38, Bogdan-Andrei Iancu wrote: Hi Pasan, The "extra" support in ACC is always producing output as strings (when performing the DB query). Nevertheless this should not prevent you pushing integer or string values into the $acc_extra() variable. But, it seems you discovered a BUG here :D...if the pushed value is a static int (from the script), it is not properly converted :(. I just pushed a fix on GIT repo (master and 2.3) - so please update you checkout or if using packages, use the nightly build (of course, wait for the night :) first ) Best regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Summit 2018 http://www.opensips.org/events/Summit-2018Amsterdam On 02/13/2018 03:39 AM, Pasan Meemaduma via Users wrote: Hi List, I'm working on getting my opensips 1.10 -> 2.3.3 :) As db_extra param is not there anymore I tried to use extra_fields param in acc module and it appears it can hold string values only. Is it the case ? I tried to use it to hold an integer value and it comes up as empty string ''. Is it a bug or purpose set only to accept string data only ? cause my db column has integer type for it. following is my config modparam("acc", "extra_fields", "db: fu->from_uri; ru->to_uri; islocal->islocal; callid->\"exe-callid\"; direction->direction") $acc_extra(islocal) = 1; When call hangups it failed to insert the cdr record to acc table as islocal comes up as ''. Ex:- ERROR:db_postgres:db_postgres_submit_query: 0x7fd07fb45d18 PQsendQuery Error: ERROR:  invalid input syntax for integer: ""#012LINE 1:...sip:xx at xx;transport=UDP','','376ab2...#012                                                             ^#012 Query: insert into acc(method,from_tag,to_tag,callid,sip_code,sip_reason,time,from_uri,to_uri,islocal,"exe-callid",direction,setuptime,created,duration,ms_duration ) values('INVITE','101c267c','464b920d','6m1aTeUcif5ctoXI7ZSq9A..','200','OK','2018-02-0917:55:41','sip:xx at xx;transport=UDP','sip:xx at xxx;transport=UDP','','376ab2da-0d66-11e8-9410-bb1806756cbd','in',11,'2018-02-09 17:55:30',4,4298) _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From pasandev at ymail.com Wed Feb 14 01:51:55 2018 From: pasandev at ymail.com (Pasan Meemaduma) Date: Wed, 14 Feb 2018 06:51:55 +0000 (UTC) Subject: [OpenSIPS-Users] warning on postgres data type References: <1535234448.649098.1518591115262.ref@mail.yahoo.com> Message-ID: <1535234448.649098.1518591115262@mail.yahoo.com> Hi List, I got another issue where opensips issue a warning I'm trying to load a uuid from a postgres database as opensips doesn't have a built-in function to generate a uuid. How do I get rid of this warning ? WARNING:db_postgres:db_postgres_get_columns: unhandled data type column (uuid_generate_v1mc) type id (2950), use DB_STRING as default Its caused by following line in my config avp_db_query("SELECT uuid_generate_v1mc()","$avp(callid)"); does this mean I need to convert it to string before returning from postgres via the sql I use ? -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Wed Feb 14 03:42:09 2018 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Wed, 14 Feb 2018 10:42:09 +0200 Subject: [OpenSIPS-Users] acc extra_fields param In-Reply-To: <1536923565.539406.1518574018400@mail.yahoo.com> References: <978310918.155131.1518485998683.ref@mail.yahoo.com> <978310918.155131.1518485998683@mail.yahoo.com> <1536923565.539406.1518574018400@mail.yahoo.com> Message-ID: <31cf6622-bd3b-087e-f1b8-7ed2952b33bb@opensips.org> Hi Pasan, This is for OpenSIPS 2.3: https://github.com/OpenSIPS/opensips/commit/42cba2c0de9cb07e31c055bc82514c14735bd2e8 Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Summit 2018 http://www.opensips.org/events/Summit-2018Amsterdam On 02/14/2018 04:06 AM, Pasan Meemaduma wrote: > Hi Bogdan, > > Thanks for the response. I'll get my opensips rebuild , can you let me > know the commit id so I know I got it correctly ? > > > On Tuesday, 13 February 2018, 17:38, Bogdan-Andrei Iancu > wrote: > > > Hi Pasan, > > The "extra" support in ACC is always producing output as strings (when > performing the DB query). Nevertheless this should not prevent you > pushing integer or string values into the $acc_extra() variable. > > But, it seems you discovered a BUG here :D...if the pushed value is a > static int (from the script), it is not properly converted :(. I just > pushed a fix on GIT repo (master and 2.3) - so please update you > checkout or if using packages, use the nightly build (of course, wait > for the night :) first ) > > Best regards, > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > OpenSIPS Summit 2018 > http://www.opensips.org/events/Summit-2018Amsterdam > On 02/13/2018 03:39 AM, Pasan Meemaduma via Users wrote: >> Hi List, >> >> I'm working on getting my opensips 1.10 -> 2.3.3 :) >> >> As db_extra param is not there anymore I tried to use extra_fields >> param in acc module and it appears it can hold string values only. Is >> it the case ? >> >> I tried to use it to hold an integer value and it comes up as empty >> string ''. >> >> Is it a bug or purpose set only to accept string data only ? cause my >> db column has integer type for it. >> >> following is my config >> >> modparam("acc", "extra_fields", "db: fu->from_uri; ru->to_uri; >> islocal->islocal; callid->\"exe-callid\"; direction->direction") >> >> >> >> $acc_extra(islocal) = 1; >> >> >> When call hangups it failed to insert the cdr record to acc table as >> islocal comes up as ''. >> >> Ex:- >> ERROR:db_postgres:db_postgres_submit_query: 0x7fd07fb45d18 >> PQsendQuery Error: ERROR: invalid input syntax for integer: >> ""#012LINE 1: ...sip:xx at xx;transport=UDP','','376ab2...#012 ^#012 >> Query: insert into acc >> (method,from_tag,to_tag,callid,sip_code,sip_reason,time,from_uri,to_uri,islocal,"exe-callid",direction,setuptime,created,duration,ms_duration >> ) values >> ('INVITE','101c267c','464b920d','6m1aTeUcif5ctoXI7ZSq9A..','200','OK','2018-02-09 >> 17:55:41','sip:xx at xx;transport=UDP','sip:xx at xxx;transport=UDP','','376ab2da-0d66-11e8-9410-bb1806756cbd','in',11,'2018-02-09 >> 17:55:30',4,4298) >> >> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From pasandev at ymail.com Wed Feb 14 03:43:32 2018 From: pasandev at ymail.com (Pasan Meemaduma) Date: Wed, 14 Feb 2018 08:43:32 +0000 (UTC) Subject: [OpenSIPS-Users] acc extra_fields param In-Reply-To: <31cf6622-bd3b-087e-f1b8-7ed2952b33bb@opensips.org> References: <978310918.155131.1518485998683.ref@mail.yahoo.com> <978310918.155131.1518485998683@mail.yahoo.com> <1536923565.539406.1518574018400@mail.yahoo.com> <31cf6622-bd3b-087e-f1b8-7ed2952b33bb@opensips.org> Message-ID: <1313481756.681968.1518597812505@mail.yahoo.com> Thanks Bogdan, I found it. I'll let you know how it goes. On Wednesday, 14 February 2018, 14:12, Bogdan-Andrei Iancu wrote: Hi Pasan, This is for OpenSIPS 2.3:    https://github.com/OpenSIPS/opensips/commit/42cba2c0de9cb07e31c055bc82514c14735bd2e8 Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Summit 2018 http://www.opensips.org/events/Summit-2018Amsterdam On 02/14/2018 04:06 AM, Pasan Meemaduma wrote: Hi Bogdan, Thanks for the response. I'll get my opensips rebuild , can you let me know the commit id so I know I got it correctly ? On Tuesday, 13 February 2018, 17:38, Bogdan-Andrei Iancu wrote: Hi Pasan, The "extra" support in ACC is always producing output as strings (when performing the DB query). Nevertheless this should not prevent you pushing integer or string values into the $acc_extra() variable. But, it seems you discovered a BUG here :D...if the pushed value is a static int (from the script), it is not properly converted :(. I just pushed a fix on GIT repo (master and 2.3) - so please update you checkout or if using packages, use the nightly build (of course, wait for the night :) first ) Best regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Summit 2018 http://www.opensips.org/events/Summit-2018Amsterdam On 02/13/2018 03:39 AM, Pasan Meemaduma via Users wrote: Hi List, I'm working on getting my opensips 1.10 -> 2.3.3 :) As db_extra param is not there anymore I tried to use extra_fields param in acc module and it appears it can hold string values only. Is it the case ? I tried to use it to hold an integer value and it comes up as empty string ''. Is it a bug or purpose set only to accept string data only ? cause my db column has integer type for it. following is my config modparam("acc", "extra_fields", "db: fu->from_uri; ru->to_uri; islocal->islocal; callid->\"exe-callid\"; direction->direction") $acc_extra(islocal) = 1; When call hangups it failed to insert the cdr record to acc table as islocal comes up as ''. Ex:- ERROR:db_postgres:db_postgres_submit_query: 0x7fd07fb45d18 PQsendQuery Error: ERROR:  invalid input syntax for integer: ""#012LINE 1:...sip:xx at xx;transport=UDP','','376ab2...#012                                                             ^#012 Query: insert into acc(method,from_tag,to_tag,callid,sip_code,sip_reason,time,from_uri,to_uri,islocal,"exe-callid",direction,setuptime,created,duration,ms_duration ) values('INVITE','101c267c','464b920d','6m1aTeUcif5ctoXI7ZSq9A..','200','OK','2018-02-09 17:55:41','sip:xx at xx;transport=UDP','sip:xx at xxx;transport=UDP','','376ab2da-0d66-11e8-9410-bb1806756cbd','in',11,'2018-02-09 17:55:30',4,4298) _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Wed Feb 14 04:24:26 2018 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Wed, 14 Feb 2018 11:24:26 +0200 Subject: [OpenSIPS-Users] Port changes In-Reply-To: References: <648da842-8997-2c0a-6775-a57eb7622b7e@opensips.org> Message-ID: <265ee2f8-0308-5671-c8e0-70c50b0cd324@opensips.org> Hi Schneur, I see at initial INVITE you have the same issue - the contact advertises the 34063 port, but the source IP is 1090; and you do fix_nated_contact(). So, based on the initial contact, you learned that the caller device is behind a NAT (or with a bogus ALG). The idea is to remember this (caller requires help) during the whole dialog (maybe using some dialog flags) and each time you receive a request / reply from it -> do the fix_nated_contact(). This will trigger the fix on the 200 OK and to get a routable URI there. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Summit 2018 http://www.opensips.org/events/Summit-2018Amsterdam On 02/13/2018 12:16 PM, Schneur Rosenberg wrote: > Bogdan its only happening when the clients SIP ALG fixes the contact, > when the contact is broken I'm already using fix_nated_contact, do you > think I should use fix_nated_contact even when the contact was already > fixed by the SIP ALG? > > The clients router does not support disabling SIP ALG, I can try using > a non standard port, I've sen some routers ALG ignoring the packets > when sent to alternative port. > > On Tue, Feb 13, 2018 at 12:04 PM, Bogdan-Andrei Iancu > wrote: >> Hi Schneur, >> >> The ACK is routed back via the Route set (the Record Route headers from the >> initial Invite) and the Contact received in the final reply (in the 200 OK >> in your case). This is called in-dialog routing and is exclusively based on >> Route set (RR headers + Contact). >> >> The rport as part of via is used only for routing back the replies for a >> requests (in transaction routing) and has no impact on the in-dialog >> routing. >> >> So, if the RURI in ACK (which should be the Contact in the received 200 OK) >> is broken, maybe you should do a fix_nated_contact() for the Contact when >> handling the 200 OK reply. >> >> Best regards, >> >> Bogdan-Andrei Iancu >> >> OpenSIPS Founder and Developer >> http://www.opensips-solutions.com >> OpenSIPS Summit 2018 >> http://www.opensips.org/events/Summit-2018Amsterdam >> >> >> On 02/12/2018 05:05 PM, Schneur Rosenberg wrote: >>> I have this interesting scenario, caller sends call to our OpenSIPS >>> who actsd as a loadbalancer which sends the call to a gateway for >>> termination and the gateway sets Session-Expires: 1800;refresher=uas, >>> the caller sends call the call through port 1090 and the rport in the >>> Via shows rport=1090 and so far everything is ok, but after 15 minutes >>> the gateway sends a reinvite, OpenSIPS sends it to the client and the >>> client sends a OK and OpenSIPS properly sends the OK to the gateway, >>> the gateway sends a ACK to OpenSIPS, but here the problem starts >>> OpenSIPS sends the ACK to the port in the Contact header which is not >>> the same port as the rport, the clients router does not recognize the >>> packet and blocks it, the client sends multiple OK's to OpenSIPS who >>> ignores it because he has already moved on, as far as OpenSIPS is >>> concerned the ACK has already been sent to the next hop, and after >>> approx 30 seconds the client does not receive the ACK and sends a BYE >>> and the call terminates. >>> >>> Why is OpenSIPS changing the port to the Contact port? and what can I >>> do to fix it? >>> >>> _______________________________________________ >>> Users mailing list >>> Users at lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> From bogdan at opensips.org Wed Feb 14 04:25:06 2018 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Wed, 14 Feb 2018 11:25:06 +0200 Subject: [OpenSIPS-Users] warning on postgres data type In-Reply-To: <1535234448.649098.1518591115262@mail.yahoo.com> References: <1535234448.649098.1518591115262.ref@mail.yahoo.com> <1535234448.649098.1518591115262@mail.yahoo.com> Message-ID: <99dff800-d208-b262-347a-2ee36f5fa0c6@opensips.org> Hi Pasan, But from the DB point of view, what data type does your procedure return ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Summit 2018 http://www.opensips.org/events/Summit-2018Amsterdam On 02/14/2018 08:51 AM, Pasan Meemaduma via Users wrote: > Hi List, > > I got another issue where opensips issue a warning I'm trying to load > a uuid from a postgres database as opensips doesn't have a built-in > function to generate a uuid. How do I get rid of this warning ? > > > WARNING:db_postgres:db_postgres_get_columns: unhandled data type > column (uuid_generate_v1mc) type id (2950), use DB_STRING as default > > Its caused by following line in my config > > avp_db_query("SELECT uuid_generate_v1mc()","$avp(callid)"); > > > does this mean I need to convert it to string before returning from > postgres via the sql I use ? > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Wed Feb 14 04:32:47 2018 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Wed, 14 Feb 2018 11:32:47 +0200 Subject: [OpenSIPS-Users] migrating to opensips 2.3 from 2.2 In-Reply-To: <010201618fce9437-935d6432-bb78-4a2d-8a99-6566b28dd356-000000@eu-west-1.amazonses.com> References: <010201618fce9437-935d6432-bb78-4a2d-8a99-6566b28dd356-000000@eu-west-1.amazonses.com> Message-ID: <73df8814-fa56-ad6d-d08e-0759798e5a93@opensips.org> Hi Brian, Are you sure you are using the right opensipdbctl version ? The one provided by the 2.3 installation has the right info text - see: https://github.com/OpenSIPS/opensips/blob/2.3/scripts/opensipsdbctl#L309 Maybe due a version mixing, you use the opensipsdbctl from the 2.2 version (that ensure migration from 2.1 to 2.2). And you need to use the opensipsdbctl from 2.3 (to get a migration from 2.2 to 2.3). Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Summit 2018 http://www.opensips.org/events/Summit-2018Amsterdam On 02/13/2018 05:35 PM, Brian Southworth wrote: > > Hi All, > > I seem to be getting an error when running the migration. It is also > saying the following when I run the following command: opensipsdbctl > migrate opensips_2_2 opensips_2_3 > > INFO: MySQL DB migration tool for OpenSIPS 2.1.x databases > > ---------------------------------------------------------------------- > > WARNING: We recommend using this tool ONLY in order to upgrade an existing > > OpenSIPS 2.1.x MySQL database to the 2.2 schema. Behaviour when > automatically > > migrating earlier DB versions (1.8, 1.9, 1.10, 1.11) to 2.2 is undefined > > my version is 2.2.3 > > also when I enter my password for mysql (this is an external DB not > internal) I get this > > ERROR 1045 (28000): Access denied for user > 'opensips'@'local_ip.lightspeed.jcsnms.sbcglobal.net' (using password: > YES) > > ERROR 1045 (28000): Access denied for user > 'opensips'@'local_ip.lightspeed.jcsnms.sbcglobal.net' (using password: > YES) > > Any idea how I fix this ? I want to start trying to use the EBR module > which was made available in opensips 2.3 > > Regards, > > Brian Southworth > > Communications Developer > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From pasandev at ymail.com Wed Feb 14 05:03:10 2018 From: pasandev at ymail.com (Pasan Meemaduma) Date: Wed, 14 Feb 2018 10:03:10 +0000 (UTC) Subject: [OpenSIPS-Users] warning on postgres data type In-Reply-To: <99dff800-d208-b262-347a-2ee36f5fa0c6@opensips.org> References: <1535234448.649098.1518591115262.ref@mail.yahoo.com> <1535234448.649098.1518591115262@mail.yahoo.com> <99dff800-d208-b262-347a-2ee36f5fa0c6@opensips.org> Message-ID: <1848819946.702861.1518602590035@mail.yahoo.com> Hi Bogdan, It returns the postgres data type uuid. I put that in acc extra field to keep track calls. If I use following, I'm not getting the warning anymore. avp_db_query("SELECT uuid_generate_v1mc()::varchar","$avp(callid)") On Wednesday, 14 February 2018, 14:55, Bogdan-Andrei Iancu wrote: Hi Pasan, But from the DB point of view, what data type does your procedure return ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Summit 2018 http://www.opensips.org/events/Summit-2018Amsterdam On 02/14/2018 08:51 AM, Pasan Meemaduma via Users wrote: Hi List, I got another issue where opensips issue a warning I'm trying to load a uuid from a postgres database as opensips doesn't have a built-in function to generate a uuid. How do I get rid of this warning ? WARNING:db_postgres:db_postgres_get_columns: unhandled data type column (uuid_generate_v1mc) type id (2950), use DB_STRING as default Its caused by following line in my config avp_db_query("SELECT uuid_generate_v1mc()","$avp(callid)"); does this mean I need to convert it to string before returning from postgres via the sql I use ? _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Wed Feb 14 05:24:18 2018 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Wed, 14 Feb 2018 12:24:18 +0200 Subject: [OpenSIPS-Users] warning on postgres data type In-Reply-To: <1848819946.702861.1518602590035@mail.yahoo.com> References: <1535234448.649098.1518591115262.ref@mail.yahoo.com> <1535234448.649098.1518591115262@mail.yahoo.com> <99dff800-d208-b262-347a-2ee36f5fa0c6@opensips.org> <1848819946.702861.1518602590035@mail.yahoo.com> Message-ID: <943e8599-c1eb-83d7-01f4-52d962bc5471@opensips.org> Hi, OK, the uuid is custom data type (specific to postgres ??), so doing the cast at DB level is the proper way to address the warning. But, why not using the call-Id + from_tag as uuid for the calls ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Summit 2018 http://www.opensips.org/events/Summit-2018Amsterdam On 02/14/2018 12:03 PM, Pasan Meemaduma wrote: > Hi Bogdan, > > It returns the postgres data type uuid. I put that in acc extra field > to keep track calls. > > If I use following, I'm not getting the warning anymore. > > avp_db_query("SELECT uuid_generate_v1mc()::varchar","$avp(callid)") > > > > On Wednesday, 14 February 2018, 14:55, Bogdan-Andrei Iancu > wrote: > > > Hi Pasan, > > But from the DB point of view, what data type does your procedure return ? > > Regards, > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > OpenSIPS Summit 2018 > http://www.opensips.org/events/Summit-2018Amsterdam > On 02/14/2018 08:51 AM, Pasan Meemaduma via Users wrote: >> Hi List, >> >> I got another issue where opensips issue a warning I'm trying to load >> a uuid from a postgres database as opensips doesn't have a built-in >> function to generate a uuid. How do I get rid of this warning ? >> >> >> WARNING:db_postgres:db_postgres_get_columns: unhandled data type >> column (uuid_generate_v1mc) type id (2950), use DB_STRING as default >> >> Its caused by following line in my config >> >> avp_db_query("SELECT uuid_generate_v1mc()","$avp(callid)"); >> >> >> does this mean I need to convert it to string before returning from >> postgres via the sql I use ? >> >> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From rosenberg11219 at gmail.com Wed Feb 14 06:45:40 2018 From: rosenberg11219 at gmail.com (Schneur Rosenberg) Date: Wed, 14 Feb 2018 13:45:40 +0200 Subject: [OpenSIPS-Users] Port changes In-Reply-To: <265ee2f8-0308-5671-c8e0-70c50b0cd324@opensips.org> References: <648da842-8997-2c0a-6775-a57eb7622b7e@opensips.org> <265ee2f8-0308-5671-c8e0-70c50b0cd324@opensips.org> Message-ID: You're right, I noticed that the port was wrong at the first invite, but I didn't realize that it was fixed, I thought it left it untouched, I just checked again and you're correct it did fix it. Is there any downside (besides for obvious performance reasons) to call fix_nated_contact() on each packet? 99% of our customers are behind NAT anyway, I assume it cant hurt, I assume it uses the received IP and port to rewrite the contact, so even if the correct IP and port was sent with the initial invite it wont do any harm, is my assumption correct?

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On Wed, Feb 14, 2018 at 11:24 AM, Bogdan-Andrei Iancu wrote: > Hi Schneur, > > I see at initial INVITE you have the same issue - the contact advertises the > 34063 port, but the source IP is 1090; and you do fix_nated_contact(). So, > based on the initial contact, you learned that the caller device is behind a > NAT (or with a bogus ALG). The idea is to remember this (caller requires > help) during the whole dialog (maybe using some dialog flags) and each time > you receive a request / reply from it -> do the fix_nated_contact(). This > will trigger the fix on the 200 OK and to get a routable URI there. > > Regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > OpenSIPS Summit 2018 > http://www.opensips.org/events/Summit-2018Amsterdam > > On 02/13/2018 12:16 PM, Schneur Rosenberg wrote: >> >> Bogdan its only happening when the clients SIP ALG fixes the contact, >> when the contact is broken I'm already using fix_nated_contact, do you >> think I should use fix_nated_contact even when the contact was already >> fixed by the SIP ALG? >> >> The clients router does not support disabling SIP ALG, I can try using >> a non standard port, I've sen some routers ALG ignoring the packets >> when sent to alternative port. >> >> On Tue, Feb 13, 2018 at 12:04 PM, Bogdan-Andrei Iancu >> wrote: >>> >>> Hi Schneur, >>> >>> The ACK is routed back via the Route set (the Record Route headers from >>> the >>> initial Invite) and the Contact received in the final reply (in the 200 >>> OK >>> in your case). This is called in-dialog routing and is exclusively based >>> on >>> Route set (RR headers + Contact). >>> >>> The rport as part of via is used only for routing back the replies for a >>> requests (in transaction routing) and has no impact on the in-dialog >>> routing. >>> >>> So, if the RURI in ACK (which should be the Contact in the received 200 >>> OK) >>> is broken, maybe you should do a fix_nated_contact() for the Contact when >>> handling the 200 OK reply. >>> >>> Best regards, >>> >>> Bogdan-Andrei Iancu >>> >>> OpenSIPS Founder and Developer >>> http://www.opensips-solutions.com >>> OpenSIPS Summit 2018 >>> http://www.opensips.org/events/Summit-2018Amsterdam >>> >>> >>> On 02/12/2018 05:05 PM, Schneur Rosenberg wrote: >>>> >>>> I have this interesting scenario, caller sends call to our OpenSIPS >>>> who actsd as a loadbalancer which sends the call to a gateway for >>>> termination and the gateway sets Session-Expires: 1800;refresher=uas, >>>> the caller sends call the call through port 1090 and the rport in the >>>> Via shows rport=1090 and so far everything is ok, but after 15 minutes >>>> the gateway sends a reinvite, OpenSIPS sends it to the client and the >>>> client sends a OK and OpenSIPS properly sends the OK to the gateway, >>>> the gateway sends a ACK to OpenSIPS, but here the problem starts >>>> OpenSIPS sends the ACK to the port in the Contact header which is not >>>> the same port as the rport, the clients router does not recognize the >>>> packet and blocks it, the client sends multiple OK's to OpenSIPS who >>>> ignores it because he has already moved on, as far as OpenSIPS is >>>> concerned the ACK has already been sent to the next hop, and after >>>> approx 30 seconds the client does not receive the ACK and sends a BYE >>>> and the call terminates. >>>> >>>> Why is OpenSIPS changing the port to the Contact port? and what can I >>>> do to fix it? >>>> >>>> _______________________________________________ >>>> Users mailing list >>>> Users at lists.opensips.org >>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >>> > From john.quick at smartvox.co.uk Wed Feb 14 11:19:33 2018 From: john.quick at smartvox.co.uk (John Quick) Date: Wed, 14 Feb 2018 16:19:33 -0000 Subject: [OpenSIPS-Users] Modules in apt.opensips.org Message-ID: <001c01d3a5af$998b0100$cca10300$@smartvox.co.uk> I was hoping to simplify my installation by only using apt-get to install binaries on a Debian box, but I cannot see an apt package for the module cachedb_couchbase in the apt.opensips.org repository. Does this mean I have to make opensips from source if I want to use this module? John Quick Smartvox Limited From liviu at opensips.org Wed Feb 14 12:39:13 2018 From: liviu at opensips.org (Liviu Chircu) Date: Wed, 14 Feb 2018 19:39:13 +0200 Subject: [OpenSIPS-Users] Modules in apt.opensips.org In-Reply-To: <001c01d3a5af$998b0100$cca10300$@smartvox.co.uk> References: <001c01d3a5af$998b0100$cca10300$@smartvox.co.uk> Message-ID: Hi John, Yes, it would appear so, currently: root at localhost:~# apt-cache search opensips | grep mongo opensips-mongodb-module - Interface module to interact with a MongoDB server root at localhost:~# apt-cache search opensips | grep cassa root at localhost:~# apt-cache search opensips | grep redis opensips-redis-module - Interface module to interact with a Redis server root at localhost:~# apt-cache search opensips | grep couch root at localhost:~# Best regards, Liviu Chircu OpenSIPS Developer http://www.opensips-solutions.com On 14.02.2018 18:19, John Quick wrote: > I was hoping to simplify my installation by only using apt-get to install > binaries on a Debian box, but I > cannot see an apt package for the module cachedb_couchbase in the > apt.opensips.org repository. > > Does this mean I have to make opensips from source if I want to use this > module? > > John Quick > Smartvox Limited > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users From Ben.Newlin at genesys.com Wed Feb 14 13:55:03 2018 From: Ben.Newlin at genesys.com (Ben Newlin) Date: Wed, 14 Feb 2018 18:55:03 +0000 Subject: [OpenSIPS-Users] Proto_hep listeners Message-ID: Hi, I’m trying to upgrade from 1.11 to 2.3.3. We use the siptrace module to replicate messages via HEP, so I had to load in the proto_hep module as well. On startup, I’m getting the following errors: ERROR:proto_hep:mod_init: No HEP listener defined, neither TCP nor UDP! ERROR:core:init_mod: failed to initialize module proto_hep It is intentional that I have not defined any listeners as I do not wish to listen for HEP messages. I am only using the proto_hep module to enabling *sending* HEP messages via siptrace. Why are listeners required for the proto_hep module to start successfully? Thanks, Ben Newlin -------------- next part -------------- An HTML attachment was scrubbed... URL: From Ben.Newlin at genesys.com Wed Feb 14 15:32:17 2018 From: Ben.Newlin at genesys.com (Ben Newlin) Date: Wed, 14 Feb 2018 20:32:17 +0000 Subject: [OpenSIPS-Users] Dialog module requires clusterer Message-ID: <8B647828-CA5B-4ED1-BF59-0A57F1DF00C8@genesys.com> Hi, I’m trying to upgrade from 1.11 to 2.3.3. We use the dialog module but we do not use dialog or profile replication. When I try to run my server it fails with the following error: WARNING:core:solve_module_dependencies: module dialog depends on module clusterer, but it was not loaded! ERROR:core:main: failed to solve module dependencies This dependency is not documented [1] and also doesn’t seem right. Why is clusterer required if we are not using any of the cluster functionality? [1] - http://www.opensips.org/html/docs/modules/2.3.x/dialog.html#idp5554560 Thanks, Ben Newlin -------------- next part -------------- An HTML attachment was scrubbed... URL: From Ben.Newlin at genesys.com Wed Feb 14 19:36:04 2018 From: Ben.Newlin at genesys.com (Ben Newlin) Date: Thu, 15 Feb 2018 00:36:04 +0000 Subject: [OpenSIPS-Users] Avpops and db_virtual modules Message-ID: <27EDAC44-90A5-44CA-BE19-125246B0D65B@genesys.com> Hi, Sorry for all of the emails! I have found one more issue while trying to upgrade from OpenSIPS 1.11 to 2.x. We use the db_virtual module in front of our redundant Postgres DB instances to facilitate failover between the DBs. We also use the avpops module to perform raw queries against the DB. In both 2.2.6 and 2.3.3 I get the following error when trying to load this configuration: CRITICAL:avpops:avpops_db_bind: database modules (virtual://set1) does not provide all functions needed by avpops module ERROR:core:init_mod: failed to initialize module avpops My configuration is as follows: #// PostgreSQL Database module loadmodule "db_postgres.so" modparam("db_postgres", "exec_query_threshold", 500000) #// Virtual DB module loadmodule "db_virtual.so" modparam("db_virtual", "db_urls", "define set1 ROUND") modparam("db_virtual", "db_urls", "postgres://postgres:postgres at 127.0.0.1:5432/postgres") #// AVP OPerationS module loadmodule "avpops.so" modparam("avpops", "db_url", "virtual://set1") Is db_virtual simply not compatible with the new OpenSIPS versions or has something changed in the configuration that I am missing? Thanks, Ben Newlin -------------- next part -------------- An HTML attachment was scrubbed... URL: From pasandev at ymail.com Wed Feb 14 21:08:36 2018 From: pasandev at ymail.com (Pasan Meemaduma) Date: Thu, 15 Feb 2018 02:08:36 +0000 (UTC) Subject: [OpenSIPS-Users] warning on postgres data type In-Reply-To: <943e8599-c1eb-83d7-01f4-52d962bc5471@opensips.org> References: <1535234448.649098.1518591115262.ref@mail.yahoo.com> <1535234448.649098.1518591115262@mail.yahoo.com> <99dff800-d208-b262-347a-2ee36f5fa0c6@opensips.org> <1848819946.702861.1518602590035@mail.yahoo.com> <943e8599-c1eb-83d7-01f4-52d962bc5471@opensips.org> Message-ID: <91755228.1269832.1518660516921@mail.yahoo.com> Hi Bogdan, Thanks for the reply. I'll see if I can use call-Id + from_tag for it. But at the moment we are using uuid version 1 for this to make sure we get more accurate uniqueness. On Wednesday, 14 February 2018, 15:54, Bogdan-Andrei Iancu wrote: Hi, OK, the uuid is custom data type (specific to postgres ??), so doing the cast at DB level is the proper way to address the warning. But, why not using the call-Id + from_tag as uuid for the calls ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Summit 2018 http://www.opensips.org/events/Summit-2018Amsterdam On 02/14/2018 12:03 PM, Pasan Meemaduma wrote: Hi Bogdan, It returns the postgres data type uuid. I put that in acc extra field to keep track calls. If I use following, I'm not getting the warning anymore. avp_db_query("SELECT uuid_generate_v1mc()::varchar","$avp(callid)") On Wednesday, 14 February 2018, 14:55, Bogdan-Andrei Iancu wrote: Hi Pasan, But from the DB point of view, what data type does your procedure return ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Summit 2018 http://www.opensips.org/events/Summit-2018Amsterdam On 02/14/2018 08:51 AM, Pasan Meemaduma via Users wrote: Hi List, I got another issue where opensips issue a warning I'm trying to load a uuid from a postgres database as opensips doesn't have a built-in function to generate a uuid. How do I get rid of this warning ? WARNING:db_postgres:db_postgres_get_columns: unhandled data type column (uuid_generate_v1mc) type id (2950), use DB_STRING as default Its caused by following line in my config avp_db_query("SELECT uuid_generate_v1mc()","$avp(callid)"); does this mean I need to convert it to string before returning from postgres via the sql I use ? _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From pasandev at ymail.com Wed Feb 14 23:59:55 2018 From: pasandev at ymail.com (Pasan Meemaduma) Date: Thu, 15 Feb 2018 04:59:55 +0000 (UTC) Subject: [OpenSIPS-Users] acc extra_fields param In-Reply-To: <31cf6622-bd3b-087e-f1b8-7ed2952b33bb@opensips.org> References: <978310918.155131.1518485998683.ref@mail.yahoo.com> <978310918.155131.1518485998683@mail.yahoo.com> <1536923565.539406.1518574018400@mail.yahoo.com> <31cf6622-bd3b-087e-f1b8-7ed2952b33bb@opensips.org> Message-ID: <196599006.1340504.1518670795237@mail.yahoo.com> Hi Bogdan, your fix worked :). Thanks for that. it converts the int properly now. On Wednesday, 14 February 2018, 14:12, Bogdan-Andrei Iancu wrote: Hi Pasan, This is for OpenSIPS 2.3:    https://github.com/OpenSIPS/opensips/commit/42cba2c0de9cb07e31c055bc82514c14735bd2e8 Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Summit 2018 http://www.opensips.org/events/Summit-2018Amsterdam On 02/14/2018 04:06 AM, Pasan Meemaduma wrote: Hi Bogdan, Thanks for the response. I'll get my opensips rebuild , can you let me know the commit id so I know I got it correctly ? On Tuesday, 13 February 2018, 17:38, Bogdan-Andrei Iancu wrote: Hi Pasan, The "extra" support in ACC is always producing output as strings (when performing the DB query). Nevertheless this should not prevent you pushing integer or string values into the $acc_extra() variable. But, it seems you discovered a BUG here :D...if the pushed value is a static int (from the script), it is not properly converted :(. I just pushed a fix on GIT repo (master and 2.3) - so please update you checkout or if using packages, use the nightly build (of course, wait for the night :) first ) Best regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Summit 2018 http://www.opensips.org/events/Summit-2018Amsterdam On 02/13/2018 03:39 AM, Pasan Meemaduma via Users wrote: Hi List, I'm working on getting my opensips 1.10 -> 2.3.3 :) As db_extra param is not there anymore I tried to use extra_fields param in acc module and it appears it can hold string values only. Is it the case ? I tried to use it to hold an integer value and it comes up as empty string ''. Is it a bug or purpose set only to accept string data only ? cause my db column has integer type for it. following is my config modparam("acc", "extra_fields", "db: fu->from_uri; ru->to_uri; islocal->islocal; callid->\"exe-callid\"; direction->direction") $acc_extra(islocal) = 1; When call hangups it failed to insert the cdr record to acc table as islocal comes up as ''. Ex:- ERROR:db_postgres:db_postgres_submit_query: 0x7fd07fb45d18 PQsendQuery Error: ERROR:  invalid input syntax for integer: ""#012LINE 1:...sip:xx at xx;transport=UDP','','376ab2...#012                                                             ^#012 Query: insert into acc(method,from_tag,to_tag,callid,sip_code,sip_reason,time,from_uri,to_uri,islocal,"exe-callid",direction,setuptime,created,duration,ms_duration ) values('INVITE','101c267c','464b920d','6m1aTeUcif5ctoXI7ZSq9A..','200','OK','2018-02-09 17:55:41','sip:xx at xx;transport=UDP','sip:xx at xxx;transport=UDP','','376ab2da-0d66-11e8-9410-bb1806756cbd','in',11,'2018-02-09 17:55:30',4,4298) _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Thu Feb 15 02:11:01 2018 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Thu, 15 Feb 2018 09:11:01 +0200 Subject: [OpenSIPS-Users] Proto_hep listeners In-Reply-To: References: Message-ID: <45594797-a0d2-1759-0140-23536eec36df@opensips.org> Hi Ben, Even if you do not actually act a HEP traffic receiver (so no need for listening), you *need* to define a HEP listener in the config file. This constrain comes from the generic TCP stack in OpenSIPS (which also re-used to implement the HEP protocol) - this stack work only in a symmetric way (sending and receiving). So, just add a listener to make it happy :) Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Summit 2018 http://www.opensips.org/events/Summit-2018Amsterdam On 02/14/2018 08:55 PM, Ben Newlin wrote: > > Hi, > > I’m trying to upgrade from 1.11 to 2.3.3. We use the siptrace module > to replicate messages via HEP, so I had to load in the proto_hep > module as well. On startup, I’m getting the following errors: > > ERROR:proto_hep:mod_init: No HEP listener defined, neither TCP nor UDP! > > ERROR:core:init_mod: failed to initialize module proto_hep > > It is intentional that I have not defined any listeners as I do not > wish to listen for HEP messages. I am only using the proto_hep module > to enabling **sending** HEP messages via siptrace. Why are listeners > required for the proto_hep module to start successfully? > > Thanks, > > Ben Newlin > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Thu Feb 15 02:17:39 2018 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Thu, 15 Feb 2018 09:17:39 +0200 Subject: [OpenSIPS-Users] Dialog module requires clusterer In-Reply-To: <8B647828-CA5B-4ED1-BF59-0A57F1DF00C8@genesys.com> References: <8B647828-CA5B-4ED1-BF59-0A57F1DF00C8@genesys.com> Message-ID: Hi Ben, Do you have any of the following module parameters set in your cfg : * accept_replicated_dialogs * replicate_dialogs_to * accept_replicated_profiles * replicate_profiles_to Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Summit 2018 http://www.opensips.org/events/Summit-2018Amsterdam On 02/14/2018 10:32 PM, Ben Newlin wrote: > > Hi, > > I’m trying to upgrade from 1.11 to 2.3.3. We use the dialog module but > we do not use dialog or profile replication. When I try to run my > server it fails with the following error: > > WARNING:core:solve_module_dependencies: module dialog depends on > module clusterer, but it was not loaded! > > ERROR:core:main: failed to solve module dependencies > > This dependency is not documented [1] and also doesn’t seem right. Why > is clusterer required if we are not using any of the cluster > functionality? > > [1] - > http://www.opensips.org/html/docs/modules/2.3.x/dialog.html#idp5554560 > > Thanks, > > Ben Newlin > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Thu Feb 15 02:20:48 2018 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Thu, 15 Feb 2018 09:20:48 +0200 Subject: [OpenSIPS-Users] Avpops and db_virtual modules In-Reply-To: <27EDAC44-90A5-44CA-BE19-125246B0D65B@genesys.com> References: <27EDAC44-90A5-44CA-BE19-125246B0D65B@genesys.com> Message-ID: <23740ca4-b11e-604b-f76a-eb65834a88a6@opensips.org> Ben, emails are welcome, as they provide valuable feedback for us ;) Do you say that avpops + db_virtual combination was working in 1.11 ? I remember there were some fixes couple of months ago in regards to how the capabilities do propagate via the virtual module. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Summit 2018 http://www.opensips.org/events/Summit-2018Amsterdam On 02/15/2018 02:36 AM, Ben Newlin wrote: > > Hi, > > Sorry for all of the emails! I have found one more issue while trying > to upgrade from OpenSIPS 1.11 to 2.x. We use the db_virtual module in > front of our redundant Postgres DB instances to facilitate failover > between the DBs. We also use the avpops module to perform raw queries > against the DB. In both 2.2.6 and 2.3.3 I get the following error when > trying to load this configuration: > > CRITICAL:avpops:avpops_db_bind: database modules (virtual://set1) does > not provide all functions needed by avpops module > > ERROR:core:init_mod: failed to initialize module avpops > > My configuration is as follows: > > #// PostgreSQL Database module > > loadmodule "db_postgres.so" > > modparam("db_postgres", "exec_query_threshold", 500000) > > #// Virtual DB module > > loadmodule "db_virtual.so" > > modparam("db_virtual", "db_urls", "define set1 ROUND") > > modparam("db_virtual", "db_urls", > "postgres://postgres:postgres at 127.0.0.1:5432/postgres") > > #// AVP OPerationS module > > loadmodule "avpops.so" > > modparam("avpops", "db_url", "virtual://set1") > > Is db_virtual simply not compatible with the new OpenSIPS versions or > has something changed in the configuration that I am missing? > > Thanks, > > Ben Newlin > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Thu Feb 15 02:41:01 2018 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Thu, 15 Feb 2018 09:41:01 +0200 Subject: [OpenSIPS-Users] Port changes In-Reply-To: References: <648da842-8997-2c0a-6775-a57eb7622b7e@opensips.org> <265ee2f8-0308-5671-c8e0-70c50b0cd324@opensips.org> Message-ID: Hi Schneur, It is not 100% correct, as a sip message may come from a another SIP server and in this case the Contact (belong to the end point) should not be re-written with the IP OPenSIPS sees as source (which is the other SIP server). Shortly said, you can force the re-write as time as you know that you are dealing with traffic from a SIP endpoint (or acting as an endpoint), and not from a SIP proxy. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Summit 2018 http://www.opensips.org/events/Summit-2018Amsterdam On 02/14/2018 01:45 PM, Schneur Rosenberg wrote: > You're right, I noticed that the port was wrong at the first invite, > but I didn't realize that it was fixed, I thought it left it > untouched, I just checked again and you're correct it did fix it. > > Is there any downside (besides for obvious performance reasons) to > call fix_nated_contact() on each packet? 99% of our customers are > behind NAT anyway, I assume it cant hurt, I assume it uses the > received IP and port to rewrite the contact, so even if the correct > IP and port was sent with the initial invite it wont do any harm, is > my assumption correct? > > On Wed, Feb 14, 2018 at 11:24 AM, Bogdan-Andrei Iancu > wrote: >> Hi Schneur, >> >> I see at initial INVITE you have the same issue - the contact advertises the >> 34063 port, but the source IP is 1090; and you do fix_nated_contact(). So, >> based on the initial contact, you learned that the caller device is behind a >> NAT (or with a bogus ALG). The idea is to remember this (caller requires >> help) during the whole dialog (maybe using some dialog flags) and each time >> you receive a request / reply from it -> do the fix_nated_contact(). This >> will trigger the fix on the 200 OK and to get a routable URI there. >> >> Regards, >> >> Bogdan-Andrei Iancu >> >> OpenSIPS Founder and Developer >> http://www.opensips-solutions.com >> OpenSIPS Summit 2018 >> http://www.opensips.org/events/Summit-2018Amsterdam >> >> On 02/13/2018 12:16 PM, Schneur Rosenberg wrote: >>> Bogdan its only happening when the clients SIP ALG fixes the contact, >>> when the contact is broken I'm already using fix_nated_contact, do you >>> think I should use fix_nated_contact even when the contact was already >>> fixed by the SIP ALG? >>> >>> The clients router does not support disabling SIP ALG, I can try using >>> a non standard port, I've sen some routers ALG ignoring the packets >>> when sent to alternative port. >>> >>> On Tue, Feb 13, 2018 at 12:04 PM, Bogdan-Andrei Iancu >>> wrote: >>>> Hi Schneur, >>>> >>>> The ACK is routed back via the Route set (the Record Route headers from >>>> the >>>> initial Invite) and the Contact received in the final reply (in the 200 >>>> OK >>>> in your case). This is called in-dialog routing and is exclusively based >>>> on >>>> Route set (RR headers + Contact). >>>> >>>> The rport as part of via is used only for routing back the replies for a >>>> requests (in transaction routing) and has no impact on the in-dialog >>>> routing. >>>> >>>> So, if the RURI in ACK (which should be the Contact in the received 200 >>>> OK) >>>> is broken, maybe you should do a fix_nated_contact() for the Contact when >>>> handling the 200 OK reply. >>>> >>>> Best regards, >>>> >>>> Bogdan-Andrei Iancu >>>> >>>> OpenSIPS Founder and Developer >>>> http://www.opensips-solutions.com >>>> OpenSIPS Summit 2018 >>>> http://www.opensips.org/events/Summit-2018Amsterdam >>>> >>>> >>>> On 02/12/2018 05:05 PM, Schneur Rosenberg wrote: >>>>> I have this interesting scenario, caller sends call to our OpenSIPS >>>>> who actsd as a loadbalancer which sends the call to a gateway for >>>>> termination and the gateway sets Session-Expires: 1800;refresher=uas, >>>>> the caller sends call the call through port 1090 and the rport in the >>>>> Via shows rport=1090 and so far everything is ok, but after 15 minutes >>>>> the gateway sends a reinvite, OpenSIPS sends it to the client and the >>>>> client sends a OK and OpenSIPS properly sends the OK to the gateway, >>>>> the gateway sends a ACK to OpenSIPS, but here the problem starts >>>>> OpenSIPS sends the ACK to the port in the Contact header which is not >>>>> the same port as the rport, the clients router does not recognize the >>>>> packet and blocks it, the client sends multiple OK's to OpenSIPS who >>>>> ignores it because he has already moved on, as far as OpenSIPS is >>>>> concerned the ACK has already been sent to the next hop, and after >>>>> approx 30 seconds the client does not receive the ACK and sends a BYE >>>>> and the call terminates. >>>>> >>>>> Why is OpenSIPS changing the port to the Contact port? and what can I >>>>> do to fix it? From pasandev at ymail.com Thu Feb 15 04:53:44 2018 From: pasandev at ymail.com (Pasan Meemaduma) Date: Thu, 15 Feb 2018 09:53:44 +0000 (UTC) Subject: [OpenSIPS-Users] rest_append_hf References: <2131409490.1428017.1518688424855.ref@mail.yahoo.com> Message-ID: <2131409490.1428017.1518688424855@mail.yahoo.com> Hi List, I'm trying to use rest_append_hf function to append an extra header for my rest call. But it doesn't set it unless I call the same function twice. How do I get around this ? I tried async() statement as well as per docs but it didn't work either. I'm using opensips 2.3.3 route(xxxx); route[xxxx] {     rest_append_hf("Accept: application/vnd.xxx.vapi+json; version=2");     $var(rc) = rest_post("http://{{ API_USER }}:{{ API_SECRET }}@{{ API_HOST }}/xx/xx/xx", "{ \"xx\": { \"xx\": [ \"$var(auth_user)\" ] }}", "application/json", "$var(body)", "$var(ct)", "$var(rcode)");     $json(res) := $var(body);     if($json(res/success) != 1 ) {         xlog("L_ERR", "API: Failed to Blocked call request - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n");     } else {         xlog("L_WARN", "API: Blocked call request - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n");     } } Accept header will only send if I call route(xxxx) twice :/. -------------- next part -------------- An HTML attachment was scrubbed... URL: From Ben.Newlin at genesys.com Thu Feb 15 08:22:53 2018 From: Ben.Newlin at genesys.com (Ben Newlin) Date: Thu, 15 Feb 2018 13:22:53 +0000 Subject: [OpenSIPS-Users] Avpops and db_virtual modules In-Reply-To: <23740ca4-b11e-604b-f76a-eb65834a88a6@opensips.org> References: <27EDAC44-90A5-44CA-BE19-125246B0D65B@genesys.com> <23740ca4-b11e-604b-f76a-eb65834a88a6@opensips.org> Message-ID: <64CCB492-67CF-4D78-899B-630003E3FF51@genesys.com> Bogdan, Yes, this exact configuration was working in 1.11. None of these module required any configuration changes that I could see for the migration, so it is unchanged. Thanks, Ben Newlin From: Bogdan-Andrei Iancu Date: Thursday, February 15, 2018 at 2:21 AM To: OpenSIPS users mailling list , Ben Newlin Subject: Re: [OpenSIPS-Users] Avpops and db_virtual modules Ben, emails are welcome, as they provide valuable feedback for us ;) Do you say that avpops + db_virtual combination was working in 1.11 ? I remember there were some fixes couple of months ago in regards to how the capabilities do propagate via the virtual module. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Summit 2018 http://www.opensips.org/events/Summit-2018Amsterdam On 02/15/2018 02:36 AM, Ben Newlin wrote: Hi, Sorry for all of the emails! I have found one more issue while trying to upgrade from OpenSIPS 1.11 to 2.x. We use the db_virtual module in front of our redundant Postgres DB instances to facilitate failover between the DBs. We also use the avpops module to perform raw queries against the DB. In both 2.2.6 and 2.3.3 I get the following error when trying to load this configuration: CRITICAL:avpops:avpops_db_bind: database modules (virtual://set1) does not provide all functions needed by avpops module ERROR:core:init_mod: failed to initialize module avpops My configuration is as follows: #// PostgreSQL Database module loadmodule "db_postgres.so" modparam("db_postgres", "exec_query_threshold", 500000) #// Virtual DB module loadmodule "db_virtual.so" modparam("db_virtual", "db_urls", "define set1 ROUND") modparam("db_virtual", "db_urls", "postgres://postgres:postgres at 127.0.0.1:5432/postgres") #// AVP OPerationS module loadmodule "avpops.so" modparam("avpops", "db_url", "virtual://set1") Is db_virtual simply not compatible with the new OpenSIPS versions or has something changed in the configuration that I am missing? Thanks, Ben Newlin _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From Ben.Newlin at genesys.com Thu Feb 15 08:37:28 2018 From: Ben.Newlin at genesys.com (Ben Newlin) Date: Thu, 15 Feb 2018 13:37:28 +0000 Subject: [OpenSIPS-Users] Dialog module requires clusterer In-Reply-To: References: <8B647828-CA5B-4ED1-BF59-0A57F1DF00C8@genesys.com> Message-ID: <4869E154-5562-45F8-99E4-F95473E43123@genesys.com> Bogdan, I should have looked for that, but I didn’t because I knew we weren’t using it. Nevertheless, someone (probably me) had set accept_replicated_dialogs to true; probably in anticipation of using it one day. When I removed that it no longer required the clusterer module. Thanks for the quick response. Ben Newlin From: Bogdan-Andrei Iancu Date: Thursday, February 15, 2018 at 2:18 AM To: OpenSIPS users mailling list , Ben Newlin Subject: Re: [OpenSIPS-Users] Dialog module requires clusterer Hi Ben, Do you have any of the following module parameters set in your cfg : * accept_replicated_dialogs * replicate_dialogs_to * accept_replicated_profiles * replicate_profiles_to Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Summit 2018 http://www.opensips.org/events/Summit-2018Amsterdam On 02/14/2018 10:32 PM, Ben Newlin wrote: Hi, I’m trying to upgrade from 1.11 to 2.3.3. We use the dialog module but we do not use dialog or profile replication. When I try to run my server it fails with the following error: WARNING:core:solve_module_dependencies: module dialog depends on module clusterer, but it was not loaded! ERROR:core:main: failed to solve module dependencies This dependency is not documented [1] and also doesn’t seem right. Why is clusterer required if we are not using any of the cluster functionality? [1] - http://www.opensips.org/html/docs/modules/2.3.x/dialog.html#idp5554560 Thanks, Ben Newlin _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From liviu at opensips.org Thu Feb 15 12:07:24 2018 From: liviu at opensips.org (Liviu Chircu) Date: Thu, 15 Feb 2018 19:07:24 +0200 Subject: [OpenSIPS-Users] rest_append_hf In-Reply-To: <2131409490.1428017.1518688424855@mail.yahoo.com> References: <2131409490.1428017.1518688424855.ref@mail.yahoo.com> <2131409490.1428017.1518688424855@mail.yahoo.com> Message-ID: <74f16aa0-2a6e-d67a-1940-4d2ee14bdc7b@opensips.org> Hi Pasan, Thank you for the report - I have spotted and fixed [1] a bug in rest_append_hf() that could definitely lead to a world of problems. Can you try to pull the latest version of 2.3 and try again? If you're running off packages, the only option is to wait until tomorrow and deploy the nightly built ones. Cheers, [1]: https://github.com/OpenSIPS/opensips/commit/888d5e9b3f9 Liviu Chircu OpenSIPS Developer http://www.opensips-solutions.com On 15.02.2018 11:53, Pasan Meemaduma via Users wrote: > Hi List, > > I'm trying to use rest_append_hf function to append an extra header > for my rest call. But it doesn't set it unless I call the same > function twice. > > How do I get around this ? I tried async() statement as well as per > docs but it didn't work either. > > I'm using opensips 2.3.3 > > route(xxxx); > > route[xxxx] { >     rest_append_hf("Accept: application/vnd.xxx.vapi+json; version=2"); >     $var(rc) = rest_post("http://{{ API_USER }}:{{ API_SECRET }}@{{ > API_HOST }}/xx/xx/xx", "{ \"xx\": { \"xx\": [ \"$var(auth_user)\" ] > }}", "application/json", "$var(body)", "$var(ct)", "$var(rcode)"); >     $json(res) := $var(body); >     if($json(res/success) != 1 ) { >         xlog("L_ERR", "API: Failed to Blocked call request - M=$rm > RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); >     } else { >         xlog("L_WARN", "API: Blocked call request - M=$rm RURI=$ru > F=$fu T=$tu IP=$si ID=$ci\n"); >     } > } > > Accept header will only send if I call route(xxxx) twice :/. > > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From pasandev at ymail.com Thu Feb 15 20:50:29 2018 From: pasandev at ymail.com (Pasan Meemaduma) Date: Fri, 16 Feb 2018 01:50:29 +0000 (UTC) Subject: [OpenSIPS-Users] rest_append_hf In-Reply-To: <74f16aa0-2a6e-d67a-1940-4d2ee14bdc7b@opensips.org> References: <2131409490.1428017.1518688424855.ref@mail.yahoo.com> <2131409490.1428017.1518688424855@mail.yahoo.com> <74f16aa0-2a6e-d67a-1940-4d2ee14bdc7b@opensips.org> Message-ID: <518858603.2002449.1518745829896@mail.yahoo.com> HI Liviu, Thanks for the prompt reply as you guys always do :). I'll rebuild my opensips and test again. I'll get back to you with results. On Thursday, 15 February 2018, 22:38, Liviu Chircu wrote: Hi Pasan, Thank you for the report - I have spotted and fixed [1] a bug in rest_append_hf() that could definitely lead to a world of problems. Can you try to pull the latest version of 2.3 and try again? If you're running off packages, the only option is to wait until tomorrow and deploy the nightly built ones. Cheers, [1]: https://github.com/OpenSIPS/opensips/commit/888d5e9b3f9 Liviu Chircu OpenSIPS Developer http://www.opensips-solutions.com On 15.02.2018 11:53, Pasan Meemaduma via Users wrote: Hi List, I'm trying to use rest_append_hf function to append an extra header for my rest call. But it doesn't set it unless I call the same function twice. How do I get around this ? I tried async() statement as well as per docs but it didn't work either. I'm using opensips 2.3.3 route(xxxx); route[xxxx] {     rest_append_hf("Accept: application/vnd.xxx.vapi+json; version=2");     $var(rc) = rest_post("http://{{ API_USER }}:{{ API_SECRET }}@{{ API_HOST }}/xx/xx/xx", "{ \"xx\": { \"xx\": [ \"$var(auth_user)\" ] }}", "application/json", "$var(body)", "$var(ct)", "$var(rcode)");     $json(res) := $var(body);     if($json(res/success) != 1 ) {         xlog("L_ERR", "API: Failed to Blocked call request - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n");     } else {         xlog("L_WARN", "API: Blocked call request - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n");     } } Accept header will only send if I call route(xxxx) twice :/. _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From pimenta at inatel.br Fri Feb 16 11:06:41 2018 From: pimenta at inatel.br (Rodrigo Pimenta Carvalho) Date: Fri, 16 Feb 2018 16:06:41 +0000 Subject: [OpenSIPS-Users] How can we avoid log completely ? Message-ID: Hi. How can we avoid log completely? That is, how to configure the opensips.cfg file so that it will no more generate any log ? Should I remove something from the cfg file? Should I put something in such file? Can we avoid logs completely even when the cfg file still has xlog commands? Any hint will be very helpful! Best regards. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Mon Feb 19 06:11:25 2018 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Mon, 19 Feb 2018 13:11:25 +0200 Subject: [OpenSIPS-Users] Avpops and db_virtual modules In-Reply-To: <64CCB492-67CF-4D78-899B-630003E3FF51@genesys.com> References: <27EDAC44-90A5-44CA-BE19-125246B0D65B@genesys.com> <23740ca4-b11e-604b-f76a-eb65834a88a6@opensips.org> <64CCB492-67CF-4D78-899B-630003E3FF51@genesys.com> Message-ID: Hi Ben, Could you check if a 2.3 version prior this commit: https://github.com/OpenSIPS/opensips/commit/f25fbf631b6fb6eafea192f95f9a43566d68cc46#diff-91bbef43b346f11cbef768a1b01c74e0 will still have the 1.11 behavior? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Summit 2018 http://www.opensips.org/events/Summit-2018Amsterdam On 02/15/2018 03:22 PM, Ben Newlin wrote: > > Bogdan, > > Yes, this exact configuration was working in 1.11. None of these > module required any configuration changes that I could see for the > migration, so it is unchanged. > > Thanks, > > Ben Newlin > > *From: *Bogdan-Andrei Iancu > *Date: *Thursday, February 15, 2018 at 2:21 AM > *To: *OpenSIPS users mailling list , Ben > Newlin > *Subject: *Re: [OpenSIPS-Users] Avpops and db_virtual modules > > Ben, emails are welcome, as they provide valuable feedback for us ;) > > Do you say that avpops + db_virtual combination was working in 1.11 ? > I remember there were some fixes couple of months ago in regards to > how the capabilities do propagate via the virtual module. > > Regards, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > OpenSIPS Summit 2018 > http://www.opensips.org/events/Summit-2018Amsterdam > > On 02/15/2018 02:36 AM, Ben Newlin wrote: > > Hi, > > Sorry for all of the emails! I have found one more issue while > trying to upgrade from OpenSIPS 1.11 to 2.x. We use the db_virtual > module in front of our redundant Postgres DB instances to > facilitate failover between the DBs. We also use the avpops module > to perform raw queries against the DB. In both 2.2.6 and 2.3.3 I > get the following error when trying to load this configuration: > > CRITICAL:avpops:avpops_db_bind: database modules (virtual://set1) > does not provide all functions needed by avpops module > > ERROR:core:init_mod: failed to initialize module avpops > > My configuration is as follows: > > #// PostgreSQL Database module > > loadmodule "db_postgres.so" > > modparam("db_postgres", "exec_query_threshold", 500000) > > #// Virtual DB module > > loadmodule "db_virtual.so" > > modparam("db_virtual", "db_urls", "define set1 ROUND") > > modparam("db_virtual", "db_urls", > "postgres://postgres:postgres at 127.0.0.1:5432/postgres") > > #// AVP OPerationS module > > loadmodule "avpops.so" > > modparam("avpops", "db_url", "virtual://set1") > > Is db_virtual simply not compatible with the new OpenSIPS versions > or has something changed in the configuration that I am missing? > > Thanks, > > Ben Newlin > > > > > _______________________________________________ > > Users mailing list > > Users at lists.opensips.org > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Mon Feb 19 06:12:59 2018 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Mon, 19 Feb 2018 13:12:59 +0200 Subject: [OpenSIPS-Users] How can we avoid log completely ? In-Reply-To: References: Message-ID: Hi Rodrigo, Have your tried to set the logging level to a really low value like log_level=-5 ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Summit 2018 http://www.opensips.org/events/Summit-2018Amsterdam On 02/16/2018 06:06 PM, Rodrigo Pimenta Carvalho wrote: > > Hi. > > > How can we avoid log completely? That is, how to configure the > opensips.cfg file so that it will no more generate any log ? > > Should I remove something from the cfg file? Should I put something in > such file? > > > Can we avoid logs completely even when the cfg file still has xlog > commands? > > > Any hint will be very helpful! > > > Best regards. > > > RODRIGO PIMENTA CARVALHO > Inatel Competence Center > Software > Ph: +55 35 3471 9200 RAMAL 979 > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From mirko.csiky at gmail.com Mon Feb 19 07:32:25 2018 From: mirko.csiky at gmail.com (Mirko Csiky) Date: Mon, 19 Feb 2018 13:32:25 +0100 Subject: [OpenSIPS-Users] How to avoid UDP usage in load_balancer module Message-ID: Hi guys, i am playing with opensips and load balancer module, and it seems like inital TCP invites will be forwarded (opensips relay after load_balance(...) ) as UDP ? Can this behaviour be deactivated ? (i tried to find some configuration options, but without success) It is also possible that my config is not ok and it is so because of misconfiguration ... Can anybody help me here ? best regards Mirko -------------- next part -------------- An HTML attachment was scrubbed... URL: From xaled at web.de Mon Feb 19 07:39:46 2018 From: xaled at web.de (xaled) Date: Mon, 19 Feb 2018 13:39:46 +0100 Subject: [OpenSIPS-Users] history-Info header manipulation Message-ID: <014c01d3a97e$ba0f88b0$2e2e9a10$@web.de> Hi, Is there any other/better way for the history-Info header manipulation like adding or removing a hi-entry except subst? Thanks, xaled -------------- next part -------------- An HTML attachment was scrubbed... URL: From Ben.Newlin at genesys.com Mon Feb 19 09:20:47 2018 From: Ben.Newlin at genesys.com (Ben Newlin) Date: Mon, 19 Feb 2018 14:20:47 +0000 Subject: [OpenSIPS-Users] Avpops and db_virtual modules In-Reply-To: References: <27EDAC44-90A5-44CA-BE19-125246B0D65B@genesys.com> <23740ca4-b11e-604b-f76a-eb65834a88a6@opensips.org> <64CCB492-67CF-4D78-899B-630003E3FF51@genesys.com> Message-ID: Bogdan, Thank you for looking into this. I have confirmed that it works in 2.3.2, which was released in 8/17 before the commit you provided. So is there really an issue with the db_virtual capabilities that was masked by this bug, or is db_virtual simply not publishing the correct capabilities itself? Thanks, Ben Newlin From: Bogdan-Andrei Iancu Date: Monday, February 19, 2018 at 6:11 AM To: Ben Newlin , OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] Avpops and db_virtual modules Hi Ben, Could you check if a 2.3 version prior this commit: https://github.com/OpenSIPS/opensips/commit/f25fbf631b6fb6eafea192f95f9a43566d68cc46#diff-91bbef43b346f11cbef768a1b01c74e0 will still have the 1.11 behavior ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Summit 2018 http://www.opensips.org/events/Summit-2018Amsterdam On 02/15/2018 03:22 PM, Ben Newlin wrote: Bogdan, Yes, this exact configuration was working in 1.11. None of these module required any configuration changes that I could see for the migration, so it is unchanged. Thanks, Ben Newlin From: Bogdan-Andrei Iancu Date: Thursday, February 15, 2018 at 2:21 AM To: OpenSIPS users mailling list , Ben Newlin Subject: Re: [OpenSIPS-Users] Avpops and db_virtual modules Ben, emails are welcome, as they provide valuable feedback for us ;) Do you say that avpops + db_virtual combination was working in 1.11 ? I remember there were some fixes couple of months ago in regards to how the capabilities do propagate via the virtual module. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Summit 2018 http://www.opensips.org/events/Summit-2018Amsterdam On 02/15/2018 02:36 AM, Ben Newlin wrote: Hi, Sorry for all of the emails! I have found one more issue while trying to upgrade from OpenSIPS 1.11 to 2.x. We use the db_virtual module in front of our redundant Postgres DB instances to facilitate failover between the DBs. We also use the avpops module to perform raw queries against the DB. In both 2.2.6 and 2.3.3 I get the following error when trying to load this configuration: CRITICAL:avpops:avpops_db_bind: database modules (virtual://set1) does not provide all functions needed by avpops module ERROR:core:init_mod: failed to initialize module avpops My configuration is as follows: #// PostgreSQL Database module loadmodule "db_postgres.so" modparam("db_postgres", "exec_query_threshold", 500000) #// Virtual DB module loadmodule "db_virtual.so" modparam("db_virtual", "db_urls", "define set1 ROUND") modparam("db_virtual", "db_urls", "postgres://postgres:postgres at 127.0.0.1:5432/postgres") #// AVP OPerationS module loadmodule "avpops.so" modparam("avpops", "db_url", "virtual://set1") Is db_virtual simply not compatible with the new OpenSIPS versions or has something changed in the configuration that I am missing? Thanks, Ben Newlin _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From volga629 at networklab.ca Mon Feb 19 15:51:27 2018 From: volga629 at networklab.ca (volga629 at networklab.ca) Date: Mon, 19 Feb 2018 16:51:27 -0400 Subject: [OpenSIPS-Users] mid_registrar Message-ID: <1519073487.1873.6@smtp.networklab.ca> Hello Everyone, I am back to test mid registrar and having issue with modparam("mid_registrar", "contact_match_param", "regid") I so comment this parmater is obsolote what correct way to set it, because opensips reject calls with 404 on location lookup. I see that contact header is replaced with CallerID and when freeswitch send reply it can't locate the cantact. Original contact root at casbc00 ~> [/etc/opensips]# opensipsctl ul show Domain:: location table=512 records=2 AOR:: 4300 Contact:: sip:4300 at 192.168.89.227:34846;transport=tcp Q= ContactID:: 1205768431484279123 Expires:: 1478 Callid:: 567846761-21414-1 at BJC.BGI.IJ.CCH Cseq:: 2027 User-agent:: Grandstream GXP2140 1.0.8.47 State:: CS_NEW Flags:: 0 Cflags:: Socket:: tcp:158.69.151.95:5082 Methods:: 6015 Feb 19 15:45:20 casbc00 /usr/sbin/opensips[2049]: looking up [sip:1205768431484279123 at 10.18.130.27:5060] Feb 19 15:45:20 casbc00 /usr/sbin/opensips[2049]: ERROR:mid_registrar:mid_reg_lookup: no record found for sip:1205768431484279123 at 10.18.130.27:5060, ci: a413e68c-9058-1236-7e9c-5254003e39bb Github report https://github.com/OpenSIPS/opensips/issues/1111 From razvan at opensips.org Tue Feb 20 03:07:45 2018 From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=) Date: Tue, 20 Feb 2018 10:07:45 +0200 Subject: [OpenSIPS-Users] How to avoid UDP usage in load_balancer module In-Reply-To: References: Message-ID: <22ec289d-1a79-86fb-c190-581206520a6b@opensips.org> Hi, Mirko! If you provision a UDP URI in the database, then the outgoing INVITE will be sent as UDP. If you want to preserve the protocol, regardless the URI's protocol, you'll have to explicitely hack it in the script by altering the destination URI, something like: if ($rP == "TCP") $du = $du + ";transport=tcp"; Note that this hack presumes that the URIs in your database are always provisioned without a transport parameter; Hope this helps. Best regards, Răzvan On 02/19/2018 02:32 PM, Mirko Csiky wrote: > Hi guys, > i am playing with opensips and load balancer module, and it seems like > inital TCP invites will be forwarded (opensips relay after > load_balance(...) ) as UDP ? Can this behaviour be deactivated ? (i > tried to find some configuration options, but without success) > It is also possible that my config is not ok and it is so because of > misconfiguration ... > Can anybody help me here ? > best regards > Mirko > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -- Răzvan Crainea OpenSIPS Core Developer http://www.opensips-solutions.com OpenSIPS Summit 2018 http://www.opensips.org/events/Summit-2018Amsterdam From liviu at opensips.org Tue Feb 20 04:54:49 2018 From: liviu at opensips.org (Liviu Chircu) Date: Tue, 20 Feb 2018 11:54:49 +0200 Subject: [OpenSIPS-Users] mid_registrar In-Reply-To: <1519073487.1873.6@smtp.networklab.ca> References: <1519073487.1873.6@smtp.networklab.ca> Message-ID: <34b74726-9e54-9f99-5578-69942b89b6a8@opensips.org> Hi, volga! Can you send a SIP trace of this failed call to my mailbox? It's one of those "it's working for me" situations. The mid_reg_lookup() should be smart enough to locate a contact based on its unique ContactID, when the call comes in from FS. Regarding regid: it's completely gone now (even in 2.3), thanks for the heads up: I'll clean up the docs asap! Cheers, Liviu Chircu OpenSIPS Developer http://www.opensips-solutions.com On 19.02.2018 22:51, volga629 at networklab.ca wrote: > Hello Everyone, > I am back to test mid registrar and having issue with > > > modparam("mid_registrar", "contact_match_param", "regid") > I so comment this parmater is obsolote what correct way to set it, > because opensips reject calls with 404 on location lookup. I see that > contact header is replaced with CallerID and when freeswitch send > reply it can't locate the cantact. > Original contact > root at casbc00 ~> [/etc/opensips]# opensipsctl ul show > Domain:: location table=512 records=2 >     AOR:: 4300 >         Contact:: sip:4300 at 192.168.89.227:34846;transport=tcp Q= >             ContactID:: 1205768431484279123 >             Expires:: 1478 >             Callid:: 567846761-21414-1 at BJC.BGI.IJ.CCH >             Cseq:: 2027 >             User-agent:: Grandstream GXP2140 1.0.8.47 >             State:: CS_NEW >             Flags:: 0 >             Cflags:: >             Socket:: tcp:158.69.151.95:5082 >             Methods:: 6015 > > Feb 19 15:45:20 casbc00 /usr/sbin/opensips[2049]: looking up > [sip:1205768431484279123 at 10.18.130.27:5060] > Feb 19 15:45:20 casbc00 /usr/sbin/opensips[2049]: > ERROR:mid_registrar:mid_reg_lookup: no record found for > sip:1205768431484279123 at 10.18.130.27:5060, ci: > a413e68c-9058-1236-7e9c-5254003e39bb > Github report https://github.com/OpenSIPS/opensips/issues/1111 > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users From bogdan at opensips.org Tue Feb 20 07:14:54 2018 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Tue, 20 Feb 2018 14:14:54 +0200 Subject: [OpenSIPS-Users] history-Info header manipulation In-Reply-To: <014c01d3a97e$ba0f88b0$2e2e9a10$@web.de> References: <014c01d3a97e$ba0f88b0$2e2e9a10$@web.de> Message-ID: <640afee3-514b-d185-6c49-1b6d9cc1cc7c@opensips.org> Hi Xaled, You mean if a single History header contains multiple hi-entry instances ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Summit 2018 http://www.opensips.org/events/Summit-2018Amsterdam On 02/19/2018 02:39 PM, xaled wrote: > > Hi, > > Is there any other/better way for the history-Info header manipulation > like adding or removing a hi-entry except subst? > > Thanks, > > xaled > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From xaled at web.de Tue Feb 20 07:51:37 2018 From: xaled at web.de (xaled) Date: Tue, 20 Feb 2018 13:51:37 +0100 Subject: [OpenSIPS-Users] history-Info header manipulation In-Reply-To: <640afee3-514b-d185-6c49-1b6d9cc1cc7c@opensips.org> References: <014c01d3a97e$ba0f88b0$2e2e9a10$@web.de> <640afee3-514b-d185-6c49-1b6d9cc1cc7c@opensips.org> Message-ID: <023d01d3aa49$8c752ba0$a55f82e0$@web.de> Hi Bogdan, Mostly yes - single HI Header with multiple hi-entries. A general solution/advice that would work with both single and multiple HI headers would be much appreciated. Thanks, Xaled From: Bogdan-Andrei Iancu [mailto:bogdan at opensips.org] Sent: Tuesday, February 20, 2018 1:15 PM To: OpenSIPS users mailling list ; xaled Subject: Re: [OpenSIPS-Users] history-Info header manipulation Hi Xaled, You mean if a single History header contains multiple hi-entry instances ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Summit 2018 http://www.opensips.org/events/Summit-2018Amsterdam On 02/19/2018 02:39 PM, xaled wrote: Hi, Is there any other/better way for the history-Info header manipulation like adding or removing a hi-entry except subst? Thanks, xaled _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From mirko.csiky at gmail.com Tue Feb 20 07:55:24 2018 From: mirko.csiky at gmail.com (Mirko Csiky) Date: Tue, 20 Feb 2018 13:55:24 +0100 Subject: [OpenSIPS-Users] How to avoid UDP usage in load_balancer module In-Reply-To: <22ec289d-1a79-86fb-c190-581206520a6b@opensips.org> References: <22ec289d-1a79-86fb-c190-581206520a6b@opensips.org> Message-ID: Hi Razvan, in the meantime (yesterday), the problem was solved by using : force_send_socket(tcp:192.168.12.175); before t_relay() In my database (i have 3 rows in load_balancer table only) it is so: sip\:192.168.15.92 (\ as escape character for ":" DBTEXT syntax) so, i think, it doesnt provide the transport parameter. Thanks for hack idea ! It looks as better solution than mine: should i use it ? (or stay with force_send_socket(tcp ?) best regards Mirko On Tue, Feb 20, 2018 at 9:07 AM, Răzvan Crainea wrote: > Hi, Mirko! > > If you provision a UDP URI in the database, then the outgoing INVITE will > be sent as UDP. > If you want to preserve the protocol, regardless the URI's protocol, > you'll have to explicitely hack it in the script by altering the > destination URI, something like: > > if ($rP == "TCP") > $du = $du + ";transport=tcp"; > > Note that this hack presumes that the URIs in your database are always > provisioned without a transport parameter; > > Hope this helps. > > Best regards, > Răzvan > > > On 02/19/2018 02:32 PM, Mirko Csiky wrote: > >> Hi guys, >> i am playing with opensips and load balancer module, and it seems like >> inital TCP invites will be forwarded (opensips relay after >> load_balance(...) ) as UDP ? Can this behaviour be deactivated ? (i tried >> to find some configuration options, but without success) >> It is also possible that my config is not ok and it is so because of >> misconfiguration ... >> Can anybody help me here ? >> best regards >> Mirko >> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> > -- > Răzvan Crainea > OpenSIPS Core Developer > http://www.opensips-solutions.com > OpenSIPS Summit 2018 > http://www.opensips.org/events/Summit-2018Amsterdam > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -------------- next part -------------- An HTML attachment was scrubbed... URL: From razvan at opensips.org Tue Feb 20 08:56:54 2018 From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=) Date: Tue, 20 Feb 2018 15:56:54 +0200 Subject: [OpenSIPS-Users] How to avoid UDP usage in load_balancer module In-Reply-To: References: <22ec289d-1a79-86fb-c190-581206520a6b@opensips.org> Message-ID: <8b1d9555-5419-92be-5e45-0625e2a6589a@opensips.org> Hi, Mirko! Your solution works too, you can stick with it, it basically has the same effect. Best regards, Răzvan On 02/20/2018 02:55 PM, Mirko Csiky wrote: > Hi Razvan, > in the meantime (yesterday), the problem was solved by using : > force_send_socket(tcp:192.168.12.175); > before t_relay() > In my database (i have 3 rows in load_balancer table only) it is so: > sip\:192.168.15.92 > (\ as escape character for ":" DBTEXT syntax) > so, i think, it doesnt provide the transport parameter. > Thanks for hack idea ! It looks as better solution than mine: should i > use it ? (or stay with force_send_socket(tcp ?) > best regards > Mirko > > > > > On Tue, Feb 20, 2018 at 9:07 AM, Răzvan Crainea > wrote: > > Hi, Mirko! > > If you provision a UDP URI in the database, then the outgoing INVITE > will be sent as UDP. > If you want to preserve the protocol, regardless the URI's protocol, > you'll have to explicitely hack it in the script by altering the > destination URI, something like: > > if ($rP == "TCP") >     $du = $du + ";transport=tcp"; > > Note that this hack presumes that the URIs in your database are > always provisioned without a transport parameter; > > Hope this helps. > > Best regards, > Răzvan > > > On 02/19/2018 02:32 PM, Mirko Csiky wrote: > > Hi guys, > i am playing with opensips and load balancer module, and it > seems like inital TCP invites will be forwarded (opensips relay > after load_balance(...) ) as UDP ? Can this behaviour be > deactivated ? (i tried to find some configuration options, but > without success) > It is also possible that my config is not ok and it is so > because of misconfiguration ... > Can anybody help me here ? > best regards > Mirko > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > -- > Răzvan Crainea > OpenSIPS Core Developer > http://www.opensips-solutions.com > OpenSIPS Summit 2018 > http://www.opensips.org/events/Summit-2018Amsterdam > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -- Răzvan Crainea OpenSIPS Core Developer http://www.opensips-solutions.com OpenSIPS Summit 2018 http://www.opensips.org/events/Summit-2018Amsterdam From pimenta at inatel.br Tue Feb 20 10:07:35 2018 From: pimenta at inatel.br (Rodrigo Pimenta Carvalho) Date: Tue, 20 Feb 2018 15:07:35 +0000 Subject: [OpenSIPS-Users] How to avoid/solve TCP blocked connection? Message-ID: Hi. My softphone is registered with the following AOR: ------------------------------------------------------------------------------------- AOR:: g1r2u3p4o5 Contact:: sip:g1r2u3p4o5 at 127.0.0.1:50353;transport=TLS;ob Q= Expires:: 10 Callid:: 53e387dc-81fe-45f9-a6f1-8a5cf4248d62 Cseq:: 37190 User-agent:: n/a Received:: sip:127.0.0.1:49678;transport=TLS State:: CS_SYNC Flags:: 0 Cflags:: NAT Socket:: tls:127.0.0.1:5061 Methods:: 8063 Attr:: in_same_network SIP_instance:: ------------------------------------------------------------------------------------------------------------------------- Sometimes, when it receives a call and answers, the Opensips show the following error: [21532] INFO:core:probe_max_sock_buff: using snd buffer of 320 kb [21532] INFO:core:init_sock_keepalive: TCP keepalive enabled on socket 22 [21532] ERROR:core:tcp_connect_blocking: poll error: flags 28 - 4 8 16 32 [21532] ERROR:core:tcp_connect_blocking: failed to retrieve SO_ERROR [server=127.0.0.1:50353] (111) Connection refused [21532] ERROR:proto_tls:tls_sync_connect: tcp_blocking_connect failed [21532] ERROR:proto_tls:proto_tls_send: connect failed [21532] ERROR:core:msg_send: send() for proto 3 failed Is there a way to avoid this kind of problem? That is, can I configure the OpenSIPS to renew some TCP connection? Any hint will be very helpful! Best regards. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 -------------- next part -------------- An HTML attachment was scrubbed... URL: From Ben.Newlin at genesys.com Tue Feb 20 10:51:45 2018 From: Ben.Newlin at genesys.com (Ben Newlin) Date: Tue, 20 Feb 2018 15:51:45 +0000 Subject: [OpenSIPS-Users] dr_is_gw parameters wrong when not using partitions Message-ID: Hi, I am running into an issue migrating from 1.11 to 2.3. It appears the dr_is_gw function is broken in 2.3.x. When not using partitions, the function parameters are not being translated properly before being passed to the internal function. With this command: dr_is_gw("$avp(src_uri)", "2", "inc", "$avp(gw_attr)"); I get the following error: CRITICAL:core:fixup_get_svalue: bogus type 4 in gparam ERROR:drouting:_is_dr_gw_w_part: invalid flags parameter I believe the problem was introduced with this commit in master, which was pulled back into 2.3: https://github.com/OpenSIPS/opensips/commit/991b3ef4fabb4a21dd0f95ea18a5e934261be01c This commit changed the parameters being passed through so that the names matched what is expected, but when the partition is not present all arguments must be shifted over by one. As far as I can tell the logic was already correct prior to that commit. Thanks, Ben Newlin -------------- next part -------------- An HTML attachment was scrubbed... URL: From liviu at opensips.org Tue Feb 20 11:03:10 2018 From: liviu at opensips.org (Liviu Chircu) Date: Tue, 20 Feb 2018 18:03:10 +0200 Subject: [OpenSIPS-Users] dr_is_gw parameters wrong when not using partitions In-Reply-To: References: Message-ID: <39f023e3-4926-3e81-12cb-a511205f7a68@opensips.org> Hi Ben, I hope you're running off the release packages, because this 2.3 issue has been reported & fixed here [1], a month ago. Otherwise, we'll have to do some more digging. Best regards, [1]: https://github.com/OpenSIPS/opensips/commit/d448ab6e3599d Liviu Chircu OpenSIPS Developer http://www.opensips-solutions.com On 20.02.2018 17:51, Ben Newlin wrote: > > Hi, > > I am running into an issue migrating from 1.11 to 2.3. It appears the > dr_is_gw function is broken in 2.3.x. > > When not using partitions, the function parameters are not being > translated properly before being passed to the internal function. With > this command: > > dr_is_gw("$avp(src_uri)", "2", "inc", "$avp(gw_attr)"); > > I get the following error: > > CRITICAL:core:fixup_get_svalue: bogus type 4 in gparam > > ERROR:drouting:_is_dr_gw_w_part: invalid flags parameter > > I believe the problem was introduced with this commit in master, which > was pulled back into 2.3: > > https://github.com/OpenSIPS/opensips/commit/991b3ef4fabb4a21dd0f95ea18a5e934261be01c > > This commit changed the parameters being passed through so that the > names matched what is expected, but when the partition is not present > all arguments must be shifted over by one. As far as I can tell the > logic was already correct prior to that commit. > > Thanks, > > Ben Newlin > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From Ben.Newlin at genesys.com Tue Feb 20 11:23:41 2018 From: Ben.Newlin at genesys.com (Ben Newlin) Date: Tue, 20 Feb 2018 16:23:41 +0000 Subject: [OpenSIPS-Users] dr_is_gw parameters wrong when not using partitions In-Reply-To: <39f023e3-4926-3e81-12cb-a511205f7a68@opensips.org> References: <39f023e3-4926-3e81-12cb-a511205f7a68@opensips.org> Message-ID: <92A22F6B-4080-4044-83F5-07B7A016F946@genesys.com> My apologies. I tried to verify the change history for that file since I am not on the newest version and I didn’t see any commits after the one I reported. I must have been looking at it wrong. Was that commit included in 2.3.3? I am currently testing the migration with 2.3.2 due to an incompatibility between db_virtual and avpops in 2.3.3. I have reported to this list already last week. So unfortunately this makes me stuck as I cannot use the newer version with the fix for this issue as it will break my DB access. Thanks, Ben Newlin From: Users on behalf of Liviu Chircu Reply-To: OpenSIPS users mailling list Date: Tuesday, February 20, 2018 at 11:04 AM To: "users at lists.opensips.org" Subject: Re: [OpenSIPS-Users] dr_is_gw parameters wrong when not using partitions Hi Ben, I hope you're running off the release packages, because this 2.3 issue has been reported & fixed here [1], a month ago. Otherwise, we'll have to do some more digging. Best regards, [1]: https://github.com/OpenSIPS/opensips/commit/d448ab6e3599d Liviu Chircu OpenSIPS Developer http://www.opensips-solutions.com On 20.02.2018 17:51, Ben Newlin wrote: Hi, I am running into an issue migrating from 1.11 to 2.3. It appears the dr_is_gw function is broken in 2.3.x. When not using partitions, the function parameters are not being translated properly before being passed to the internal function. With this command: dr_is_gw("$avp(src_uri)", "2", "inc", "$avp(gw_attr)"); I get the following error: CRITICAL:core:fixup_get_svalue: bogus type 4 in gparam ERROR:drouting:_is_dr_gw_w_part: invalid flags parameter I believe the problem was introduced with this commit in master, which was pulled back into 2.3: https://github.com/OpenSIPS/opensips/commit/991b3ef4fabb4a21dd0f95ea18a5e934261be01c This commit changed the parameters being passed through so that the names matched what is expected, but when the partition is not present all arguments must be shifted over by one. As far as I can tell the logic was already correct prior to that commit. Thanks, Ben Newlin _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From liviu at opensips.org Tue Feb 20 11:41:02 2018 From: liviu at opensips.org (Liviu Chircu) Date: Tue, 20 Feb 2018 18:41:02 +0200 Subject: [OpenSIPS-Users] dr_is_gw parameters wrong when not using partitions In-Reply-To: <92A22F6B-4080-4044-83F5-07B7A016F946@genesys.com> References: <39f023e3-4926-3e81-12cb-a511205f7a68@opensips.org> <92A22F6B-4080-4044-83F5-07B7A016F946@genesys.com> Message-ID: <93e4733a-874d-338a-a6bb-3e08458157e8@opensips.org> Yup, it's included in 2.3.3. We'll look into the db_virtual issue asap. Cheers, Liviu Chircu OpenSIPS Developer http://www.opensips-solutions.com On 20.02.2018 18:23, Ben Newlin wrote: > > My apologies. I tried to verify the change history for that file since > I am not on the newest version and I didn’t see any commits after the > one I reported. I must have been looking at it wrong. Was that commit > included in 2.3.3? > > I am currently testing the migration with 2.3.2 due to an > incompatibility between db_virtual and avpops in 2.3.3. I have > reported to this list already last week. So unfortunately this makes > me stuck as I cannot use the newer version with the fix for this issue > as it will break my DB access. > > Thanks, > > Ben Newlin > > *From: *Users on behalf of Liviu > Chircu > *Reply-To: *OpenSIPS users mailling list > *Date: *Tuesday, February 20, 2018 at 11:04 AM > *To: *"users at lists.opensips.org" > *Subject: *Re: [OpenSIPS-Users] dr_is_gw parameters wrong when not > using partitions > > Hi Ben, > > I hope you're running off the release packages, because this 2.3 issue > has been reported & fixed here [1], a month ago. Otherwise, we'll have > to do some more digging. > > Best regards, > > [1]: https://github.com/OpenSIPS/opensips/commit/d448ab6e3599d > > Liviu Chircu > OpenSIPS Developer > http://www.opensips-solutions.com > > On 20.02.2018 17:51, Ben Newlin wrote: > > Hi, > > I am running into an issue migrating from 1.11 to 2.3. It appears > the dr_is_gw function is broken in 2.3.x. > > When not using partitions, the function parameters are not being > translated properly before being passed to the internal function. > With this command: > > dr_is_gw("$avp(src_uri)", "2", "inc", "$avp(gw_attr)"); > > I get the following error: > > CRITICAL:core:fixup_get_svalue: bogus type 4 in gparam > > ERROR:drouting:_is_dr_gw_w_part: invalid flags parameter > > I believe the problem was introduced with this commit in master, > which was pulled back into 2.3: > > https://github.com/OpenSIPS/opensips/commit/991b3ef4fabb4a21dd0f95ea18a5e934261be01c > > This commit changed the parameters being passed through so that > the names matched what is expected, but when the partition is not > present all arguments must be shifted over by one. As far as I can > tell the logic was already correct prior to that commit. > > Thanks, > > Ben Newlin > > > > > _______________________________________________ > > Users mailing list > > Users at lists.opensips.org > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From daniel.zanutti at gmail.com Tue Feb 20 11:43:54 2018 From: daniel.zanutti at gmail.com (Daniel Zanutti) Date: Tue, 20 Feb 2018 13:43:54 -0300 Subject: [OpenSIPS-Users] Problem with simultaneous CANCEL + 200 OK Message-ID: Hey I had a problem when receiving simultaneous CANCEL from customer and 200 OK from gateway. Seems that the first CANCEL was rejected, but the second CANCEL was accepted. This second CANCEL did NOT go to the gateway, just Opensips received and replied with 200 OK. This is the log of the first CANCEL: Feb 15 18:39:22 /sbin/opensips[28845]: SCRIPT:TRAFFIC:WARNING: method CANCEL ( 7Qbq3O3CReMfPflAtl8NY3ddTqPVBHO2785126 at 2.2.2.2/ XAeG2xj278512T2 / 1839212581509953 ) not validated and not fixed ( code=-1 ) code -1 is the return of validate_dialog() Second CANCEL didn't generated a log. Shouldn't all CANCELs be rejected? On this case, just the first one was rejected. I'm using version 1.9.11. -------------- next part -------------- An HTML attachment was scrubbed... URL: From liviu at opensips.org Tue Feb 20 14:22:00 2018 From: liviu at opensips.org (Liviu Chircu) Date: Tue, 20 Feb 2018 21:22:00 +0200 Subject: [OpenSIPS-Users] [Livestream][ClueCon Weekly] OpenSIPS-FreeSWITCH integration discussion - Feb 21st @ 12:00 PM CST Message-ID: Hi all, OpenSIPS is going to be live on the FreeSWITCH "ClueCon Weekly" conference tomorrow, February 21st @ 12 PM CST! On this edition, the duty of representing the project falls on my shoulders. The main topic of discussion will revolve around the latest FreeSWITCH integration features that we've pushed into the development branch. So if you also use FreeSWITCH but you haven't caught up with them yet [1], tomorrow would be a good time to tune in and hear all about it. Towards the end, there will also be a Q/A section. See you there! Best regards, [1]: https://blog.opensips.org/2018/01/17/how-to-script-advanced-freeswitch-integrations-with-opensips-2-4/ -- Liviu Chircu OpenSIPS Developer http://www.opensips-solutions.com From bogdan at opensips.org Wed Feb 21 09:52:00 2018 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Wed, 21 Feb 2018 16:52:00 +0200 Subject: [OpenSIPS-Users] Avpops and db_virtual modules In-Reply-To: References: <27EDAC44-90A5-44CA-BE19-125246B0D65B@genesys.com> <23740ca4-b11e-604b-f76a-eb65834a88a6@opensips.org> <64CCB492-67CF-4D78-899B-630003E3FF51@genesys.com> Message-ID: Hi Ben, OK, I reviewed the changes and the code and I identified the problem in the way db_virtual is computing the capabilities in ROUND ROBIN mode - more or less it was discarding the UPDATE DB capabilities, which was required by avpops. I pushed a fix for this, this is for 2.3: https://github.com/OpenSIPS/opensips/commit/edded2f1431cfaaf75ed2af0c5ff9ec28548e7c5 Please update and give it a try ! Thanks and Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Summit 2018 http://www.opensips.org/events/Summit-2018Amsterdam On 02/19/2018 04:20 PM, Ben Newlin wrote: > > Bogdan, > > Thank you for looking into this. I have confirmed that it works in > 2.3.2, which was released in 8/17 before the commit you provided. > > So is there really an issue with the db_virtual capabilities that was > masked by this bug, or is db_virtual simply not publishing the correct > capabilities itself? > > Thanks, > > Ben Newlin > > *From: *Bogdan-Andrei Iancu > *Date: *Monday, February 19, 2018 at 6:11 AM > *To: *Ben Newlin , OpenSIPS users mailling > list > *Subject: *Re: [OpenSIPS-Users] Avpops and db_virtual modules > > Hi Ben, > > Could you check if a 2.3 version prior this commit: > https://github.com/OpenSIPS/opensips/commit/f25fbf631b6fb6eafea192f95f9a43566d68cc46#diff-91bbef43b346f11cbef768a1b01c74e0 > will still have the 1.11 behavior ? > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Wed Feb 21 09:57:03 2018 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Wed, 21 Feb 2018 16:57:03 +0200 Subject: [OpenSIPS-Users] history-Info header manipulation In-Reply-To: <023d01d3aa49$8c752ba0$a55f82e0$@web.de> References: <014c01d3a97e$ba0f88b0$2e2e9a10$@web.de> <640afee3-514b-d185-6c49-1b6d9cc1cc7c@opensips.org> <023d01d3aa49$8c752ba0$a55f82e0$@web.de> Message-ID: <62fcdd37-750f-c204-9e0e-40079e207eb0@opensips.org> Hi Xaled, There were recently some fixes in regards to such headers - multi body name-addr like. Internally the parser is able to handle it (the fixes I mentioned were related to PAI/PPI hdr parsing). Still, from script level there is no way (yet) to get access to each body part, via variables or transformations:( Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Summit 2018 http://www.opensips.org/events/Summit-2018Amsterdam On 02/20/2018 02:51 PM, xaled wrote: > > Hi Bogdan, > > Mostly yes - single HI Header with multiple hi-entries. > > A general solution/advice that would work with both single and > multiple HI headers would be much appreciated. > > Thanks, > > Xaled > > *From:*Bogdan-Andrei Iancu [mailto:bogdan at opensips.org] > *Sent:* Tuesday, February 20, 2018 1:15 PM > *To:* OpenSIPS users mailling list ; xaled > > *Subject:* Re: [OpenSIPS-Users] history-Info header manipulation > > Hi Xaled, > > You mean if a single History header contains multiple hi-entry instances ? > > Regards, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > OpenSIPS Summit 2018 > http://www.opensips.org/events/Summit-2018Amsterdam > > On 02/19/2018 02:39 PM, xaled wrote: > > Hi, > > Is there any other/better way for the history-Info header > manipulation like adding or removing a hi-entry except subst? > > Thanks, > > xaled > > > > > _______________________________________________ > > Users mailing list > > Users at lists.opensips.org > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -------------- next part -------------- An HTML attachment was scrubbed... URL: From xaled at web.de Wed Feb 21 11:11:25 2018 From: xaled at web.de (xaled) Date: Wed, 21 Feb 2018 17:11:25 +0100 Subject: [OpenSIPS-Users] history-Info header manipulation In-Reply-To: <62fcdd37-750f-c204-9e0e-40079e207eb0@opensips.org> References: <014c01d3a97e$ba0f88b0$2e2e9a10$@web.de> <640afee3-514b-d185-6c49-1b6d9cc1cc7c@opensips.org> <023d01d3aa49$8c752ba0$a55f82e0$@web.de> <62fcdd37-750f-c204-9e0e-40079e207eb0@opensips.org> Message-ID: <041901d3ab2e$9fdd92a0$df98b7e0$@web.de> Hi Bogdan, ok, thank you for the info. It would be great to have a generic method of accessing such multipart headers. Greetings, Xaled From: Bogdan-Andrei Iancu [mailto:bogdan at opensips.org] Sent: Wednesday, February 21, 2018 3:57 PM To: xaled ; 'OpenSIPS users mailling list' Subject: Re: [OpenSIPS-Users] history-Info header manipulation Hi Xaled, There were recently some fixes in regards to such headers - multi body name-addr like. Internally the parser is able to handle it (the fixes I mentioned were related to PAI/PPI hdr parsing). Still, from script level there is no way (yet) to get access to each body part, via variables or transformations :( Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Summit 2018 http://www.opensips.org/events/Summit-2018Amsterdam On 02/20/2018 02:51 PM, xaled wrote: Hi Bogdan, Mostly yes - single HI Header with multiple hi-entries. A general solution/advice that would work with both single and multiple HI headers would be much appreciated. Thanks, Xaled From: Bogdan-Andrei Iancu [mailto:bogdan at opensips.org] Sent: Tuesday, February 20, 2018 1:15 PM To: OpenSIPS users mailling list ; xaled Subject: Re: [OpenSIPS-Users] history-Info header manipulation Hi Xaled, You mean if a single History header contains multiple hi-entry instances ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Summit 2018 http://www.opensips.org/events/Summit-2018Amsterdam On 02/19/2018 02:39 PM, xaled wrote: Hi, Is there any other/better way for the history-Info header manipulation like adding or removing a hi-entry except subst? Thanks, xaled _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From xaled at web.de Wed Feb 21 11:19:06 2018 From: xaled at web.de (xaled) Date: Wed, 21 Feb 2018 17:19:06 +0100 Subject: [OpenSIPS-Users] Per phone number statistics Message-ID: <043b01d3ab2f$b2afe1c0$180fa540$@web.de> Hello, What is a good way of collecting and accessing per phone number (A or B side) statistics? For example current number of simultaneous calls to a particular phone number. The best I could come up is to use radius accounting and then do SQL queries on the accounting data. SNMP stats don't seem to allow per number statistic collection. Thanks, Xaled -------------- next part -------------- An HTML attachment was scrubbed... URL: From Ben.Newlin at genesys.com Wed Feb 21 11:33:49 2018 From: Ben.Newlin at genesys.com (Ben Newlin) Date: Wed, 21 Feb 2018 16:33:49 +0000 Subject: [OpenSIPS-Users] Per phone number statistics In-Reply-To: <043b01d3ab2f$b2afe1c0$180fa540$@web.de> References: <043b01d3ab2f$b2afe1c0$180fa540$@web.de> Message-ID: <447AD8EC-AD0A-4E01-A99B-FF036F34FE18@genesys.com> I would recommend using Dialog profiling. http://www.opensips.org/html/docs/modules/2.3.x/dialog.html#idp158208 Ben Newlin From: Users on behalf of xaled Reply-To: OpenSIPS users mailling list Date: Wednesday, February 21, 2018 at 11:13 AM To: 'OpenSIPS users mailling list' Subject: [OpenSIPS-Users] Per phone number statistics Hello, What is a good way of collecting and accessing per phone number (A or B side) statistics? For example current number of simultaneous calls to a particular phone number. The best I could come up is to use radius accounting and then do SQL queries on the accounting data. SNMP stats don’t seem to allow per number statistic collection. Thanks, Xaled -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Wed Feb 21 11:43:21 2018 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Wed, 21 Feb 2018 18:43:21 +0200 Subject: [OpenSIPS-Users] history-Info header manipulation In-Reply-To: <041901d3ab2e$9fdd92a0$df98b7e0$@web.de> References: <014c01d3a97e$ba0f88b0$2e2e9a10$@web.de> <640afee3-514b-d185-6c49-1b6d9cc1cc7c@opensips.org> <023d01d3aa49$8c752ba0$a55f82e0$@web.de> <62fcdd37-750f-c204-9e0e-40079e207eb0@opensips.org> <041901d3ab2e$9fdd92a0$df98b7e0$@web.de> Message-ID: Hi Xaled, Indeed - please open a feature request on the OpenSIPS tracker (https://github.com/OpenSIPS/opensips/issues) so we can have this on the table for future work. Thanks & Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Summit 2018 http://www.opensips.org/events/Summit-2018Amsterdam On 02/21/2018 06:11 PM, xaled wrote: > > Hi Bogdan, > > ok, thank you for the info. > > It would be great to have a generic method of accessing such multipart > headers. > > Greetings, > > Xaled > > *From:*Bogdan-Andrei Iancu [mailto:bogdan at opensips.org] > *Sent:* Wednesday, February 21, 2018 3:57 PM > *To:* xaled ; 'OpenSIPS users mailling list' > > *Subject:* Re: [OpenSIPS-Users] history-Info header manipulation > > Hi Xaled, > > There were recently some fixes in regards to such headers - multi body > name-addr like. Internally the parser is able to handle it (the fixes > I mentioned were related to PAI/PPI hdr parsing). > Still, from script level there is no way (yet) to get access to each > body part, via variables or transformations :( > > Regards, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > OpenSIPS Summit 2018 > http://www.opensips.org/events/Summit-2018Amsterdam > > On 02/20/2018 02:51 PM, xaled wrote: > > Hi Bogdan, > > Mostly yes - single HI Header with multiple hi-entries. > > A general solution/advice that would work with both single and > multiple HI headers would be much appreciated. > > Thanks, > > Xaled > > *From:*Bogdan-Andrei Iancu [mailto:bogdan at opensips.org] > *Sent:* Tuesday, February 20, 2018 1:15 PM > *To:* OpenSIPS users mailling list > ; xaled > > *Subject:* Re: [OpenSIPS-Users] history-Info header manipulation > > Hi Xaled, > > You mean if a single History header contains multiple hi-entry > instances ? > > Regards, > > > Bogdan-Andrei Iancu > > > > OpenSIPS Founder and Developer > > http://www.opensips-solutions.com > > OpenSIPS Summit 2018 > > http://www.opensips.org/events/Summit-2018Amsterdam > > On 02/19/2018 02:39 PM, xaled wrote: > > Hi, > > Is there any other/better way for the history-Info header > manipulation like adding or removing a hi-entry except subst? > > Thanks, > > xaled > > > > > > _______________________________________________ > > Users mailing list > > Users at lists.opensips.org > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -------------- next part -------------- An HTML attachment was scrubbed... URL: From xaled at web.de Wed Feb 21 13:10:27 2018 From: xaled at web.de (xaled) Date: Wed, 21 Feb 2018 19:10:27 +0100 Subject: [OpenSIPS-Users] Per phone number statistics In-Reply-To: <447AD8EC-AD0A-4E01-A99B-FF036F34FE18@genesys.com> References: <043b01d3ab2f$b2afe1c0$180fa540$@web.de> <447AD8EC-AD0A-4E01-A99B-FF036F34FE18@genesys.com> Message-ID: <047e01d3ab3f$40d6e1b0$c284a510$@web.de> Hello Ben, thank you for the advice. I will take a look at the module. Never actually used it bevor. Thanks, Xaled From: Users [mailto:users-bounces at lists.opensips.org] On Behalf Of Ben Newlin Sent: Wednesday, February 21, 2018 5:34 PM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] Per phone number statistics I would recommend using Dialog profiling. http://www.opensips.org/html/docs/modules/2.3.x/dialog.html#idp158208 Ben Newlin From: Users > on behalf of xaled > Reply-To: OpenSIPS users mailling list > Date: Wednesday, February 21, 2018 at 11:13 AM To: 'OpenSIPS users mailling list' > Subject: [OpenSIPS-Users] Per phone number statistics Hello, What is a good way of collecting and accessing per phone number (A or B side) statistics? For example current number of simultaneous calls to a particular phone number. The best I could come up is to use radius accounting and then do SQL queries on the accounting data. SNMP stats don’t seem to allow per number statistic collection. Thanks, Xaled -------------- next part -------------- An HTML attachment was scrubbed... URL: From Ben.Newlin at genesys.com Wed Feb 21 14:57:38 2018 From: Ben.Newlin at genesys.com (Ben Newlin) Date: Wed, 21 Feb 2018 19:57:38 +0000 Subject: [OpenSIPS-Users] Avpops and db_virtual modules In-Reply-To: References: <27EDAC44-90A5-44CA-BE19-125246B0D65B@genesys.com> <23740ca4-b11e-604b-f76a-eb65834a88a6@opensips.org> <64CCB492-67CF-4D78-899B-630003E3FF51@genesys.com> Message-ID: Bogdan, I have tested with this fix and I was able to start up with these modules properly. I am getting another error now, but I will report that in a new thread so as not to confuse unrelated issues. Thanks, Ben Newlin From: Bogdan-Andrei Iancu Date: Wednesday, February 21, 2018 at 9:53 AM To: Ben Newlin , OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] Avpops and db_virtual modules Hi Ben, OK, I reviewed the changes and the code and I identified the problem in the way db_virtual is computing the capabilities in ROUND ROBIN mode - more or less it was discarding the UPDATE DB capabilities, which was required by avpops. I pushed a fix for this, this is for 2.3: https://github.com/OpenSIPS/opensips/commit/edded2f1431cfaaf75ed2af0c5ff9ec28548e7c5 Please update and give it a try ! Thanks and Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Summit 2018 http://www.opensips.org/events/Summit-2018Amsterdam On 02/19/2018 04:20 PM, Ben Newlin wrote: Bogdan, Thank you for looking into this. I have confirmed that it works in 2.3.2, which was released in 8/17 before the commit you provided. So is there really an issue with the db_virtual capabilities that was masked by this bug, or is db_virtual simply not publishing the correct capabilities itself? Thanks, Ben Newlin From: Bogdan-Andrei Iancu Date: Monday, February 19, 2018 at 6:11 AM To: Ben Newlin , OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] Avpops and db_virtual modules Hi Ben, Could you check if a 2.3 version prior this commit: https://github.com/OpenSIPS/opensips/commit/f25fbf631b6fb6eafea192f95f9a43566d68cc46#diff-91bbef43b346f11cbef768a1b01c74e0 will still have the 1.11 behavior ? -------------- next part -------------- An HTML attachment was scrubbed... URL: From Ben.Newlin at genesys.com Wed Feb 21 17:38:14 2018 From: Ben.Newlin at genesys.com (Ben Newlin) Date: Wed, 21 Feb 2018 22:38:14 +0000 Subject: [OpenSIPS-Users] OpenSIPS crashing on 200 OK Message-ID: <530E1611-2A4D-424F-938D-2E409E1A78A2@genesys.com> Hi, I am currently upgrading to 2.3.3 and am running off the HEAD of the 2.3 branch. My OpenSIPS is consistently crashing when trying to process the ACK to an initial INVITE request. I am testing with SIPp using a simple single call scenario. I see the following in the logs: Feb 21 21:32:27 [383] DBG:core:io_wait_loop_epoll: [TCP_main] EPOLLHUP on IN ->connection closed by the remote peer! Feb 21 21:32:27 [383] CRITICAL:core:receive_fd: EOF on 33 Feb 21 21:32:27 [383] DBG:core:handle_worker: dead child 9, pid 370 (shutting down?) Feb 21 21:32:27 [383] DBG:core:io_watch_del: [TCP_main] io_watch_del op on index 8 33 (0x8d4e80, 33, 8, 0x0,0x1) fd_no=29 called Feb 21 21:32:27 [361] DBG:core:handle_sigs: status = 139 Feb 21 21:32:27 [361] INFO:core:handle_sigs: child process 370 exited by a signal 11 Feb 21 21:32:27 [361] INFO:core:handle_sigs: core was generated Feb 21 21:32:27 [361] INFO:core:handle_sigs: terminating due to SIGCHLD I also notice that OpenSIPS is starting up a number of TCP processes even though I do not use TCP in my script. I do not even have the proto_tcp module loaded. I used to use the “disable_tcp” global parameter in 1.11, but that has been removed in 2.3. Feb 21 22:03:56 [361] DBG:core:internal_fork: forking new process "SIP receiver TCP" Feb 21 22:03:56 [361] DBG:core:internal_fork: forking new process "SIP receiver TCP" Feb 21 22:03:56 [361] DBG:core:internal_fork: forking new process "SIP receiver TCP" Feb 21 22:03:56 [361] DBG:core:internal_fork: forking new process "SIP receiver TCP" Feb 21 22:03:56 [361] DBG:core:internal_fork: forking new process "SIP receiver TCP" Feb 21 22:03:56 [375] DBG:core:io_watch_add: [TCP_worker] io_watch_add op (3 on 53) (0x8d4e80, 3, 1, (nil),1), fd_no=0/209715 Feb 21 22:03:56 [361] DBG:core:internal_fork: forking new process "SIP receiver TCP" Feb 21 22:03:56 [361] DBG:core:internal_fork: forking new process "SIP receiver TCP" Feb 21 22:03:56 [361] DBG:core:internal_fork: forking new process "SIP receiver TCP" Initialization logs: https://pastebin.com/nsJQGkdy Call logs: https://pastebin.com/vY5crK8U I also have the core dump if any more information is needed from that. Thanks, Ben Newlin -------------- next part -------------- An HTML attachment was scrubbed... URL: From Ben.Newlin at genesys.com Wed Feb 21 18:07:38 2018 From: Ben.Newlin at genesys.com (Ben Newlin) Date: Wed, 21 Feb 2018 23:07:38 +0000 Subject: [OpenSIPS-Users] del_uri_param failure Message-ID: Hi, I am very glad to have the new del_uri_param function. This was a common problem of mine and it is great not to have to use regex to do this. However, while implementing this I have run into some strange behavior by the function when the URI param being deleted does not exist. In my case I am using the dialog module and attempting to remove a URI param just after the dialog creation. When the function does not find the URI param, it causes the dialog to immediately be destroyed and all message processing stops, including exiting the script. Feb 21 22:47:42 [371] DBG:uri:del_uri_param: requested key not found in RURI Feb 21 22:47:42 [371] DBG:dialog:next_state_dlg: unref dlg 0x7f460780a8c8 with 1 -> 2 in entry 0x7f46077fbfa8 Feb 21 22:47:42 [371] DBG:core:evi_param_set: adding string param Feb 21 22:47:42 [371] DBG:core:evi_param_set: adding string param Feb 21 22:47:42 [371] DBG:core:evi_param_set: adding int param Feb 21 22:47:42 [371] DBG:core:evi_param_set: adding int param Feb 21 22:47:42 [371] DBG:core:destroy_avp_list: destroying list (nil) Feb 21 22:47:42 [371] DBG:dialog:next_state_dlg: dialog 0x7f460780a8c8 changed from state 1 to state 5, due event 1 Feb 21 22:47:42 [371] DBG:dialog:dlg_onreply: dialog 0x7f460780a8c8 failed (negative reply) Feb 21 22:47:42 [371] DBG:dialog:unref_dlg: unref dlg 0x7f460780a8c8 with 1 -> 1 in entry 0x7f46077fbfa8 Feb 21 22:47:42 [371] DBG:dialog:unref_dlg: unref dlg 0x7f460780a8c8 with 1 -> 0 in entry 0x7f46077fbfa8 Feb 21 22:47:42 [371] DBG:dialog:unref_dlg: ref <=0 for dialog 0x7f460780a8c8 Feb 21 22:47:42 [371] DBG:dialog:destroy_dlg: destroying dialog 0x7f460780a8c8 Feb 21 22:47:42 [371] DBG:dialog:destroy_dlg: dlg expired or not in list - dlg 0x7f460780a8c8 [3710:1818203549] with clid '2-185 at 127.0.0.1' and tags '185SIPpTag002' 'NULL' Feb 21 22:47:42 [371] DBG:core:destroy_avp_list: destroying list 0x7f460780c048 Feb 21 22:47:42 [371] DBG:core:receive_msg: cleaning up The logs indicate that a DLG_EVENT_TDEL is being raised which, when the dialog is still in UNCONFIRMED state, causes the dialog to be destroyed. It’s not clear to me how or why the del_uri_param function could be doing this, especially as a transaction hasn’t even been created for the message yet in this case. I’m not sure what effect this would have if the dialog is in other states or at other times during the call. It took me a while to realize it was the del_uri_param function causing this, as it seems so strange. But I have verified that when I remove the function, or when I verify the URI param exists before calling the function, everything is fine. That workaround works perfectly well, but it seemed such strange and catastrophic error behavior to drop the entire message that I wanted to report it anyway to see if anything needed to be addressed. Call traces can be found here: https://pastebin.com/9FnmJCD9 You will see the same INVITE is offered multiple times as OpenSIPS is not responding after dropping the previous requests and dialog. Thanks, Ben Newlin -------------- next part -------------- An HTML attachment was scrubbed... URL: From liviu at opensips.org Thu Feb 22 04:23:14 2018 From: liviu at opensips.org (Liviu Chircu) Date: Thu, 22 Feb 2018 11:23:14 +0200 Subject: [OpenSIPS-Users] [Livestream][ClueCon Weekly] OpenSIPS-FreeSWITCH integration discussion - Feb 21st @ 12:00 PM CST In-Reply-To: References: Message-ID: <3d776c43-9392-1f18-5267-34ca5db7129f@opensips.org> Thank you to everyone who took part in the livestream yesterday! You may find its recording here [1]. As a final note, we would once again like to give proper credits to Mr. Giovanni Maruzzelli, who provided both the initial idea and some proof-of-concept code samples for the extended OpenSIPS-FreeSWITCH integration. Best regards, [1]: https://www.youtube.com/watch?v=9a2Cd1AV7V0 Liviu Chircu OpenSIPS Developer http://www.opensips-solutions.com On 20.02.2018 21:22, Liviu Chircu wrote: > Hi all, > > OpenSIPS is going to be live on the FreeSWITCH "ClueCon Weekly" > conference tomorrow, February 21st @ 12 PM CST! > > On this edition, the duty of representing the project falls on my > shoulders. The main topic of discussion will revolve around the latest > FreeSWITCH integration features that we've pushed into the development > branch. So if you also use FreeSWITCH but you haven't caught up with > them yet [1], tomorrow would be a good time to tune in and hear all > about it. Towards the end, there will also be a Q/A section. > > See you there! > > Best regards, > > [1]: > https://blog.opensips.org/2018/01/17/how-to-script-advanced-freeswitch-integrations-with-opensips-2-4/ > From bogdan at opensips.org Thu Feb 22 04:45:35 2018 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Thu, 22 Feb 2018 11:45:35 +0200 Subject: [OpenSIPS-Users] Avpops and db_virtual modules In-Reply-To: References: <27EDAC44-90A5-44CA-BE19-125246B0D65B@genesys.com> <23740ca4-b11e-604b-f76a-eb65834a88a6@opensips.org> <64CCB492-67CF-4D78-899B-630003E3FF51@genesys.com> Message-ID: Cool, Ihappy this issue was fixed. Waiting for your new report :) Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Summit 2018 http://www.opensips.org/events/Summit-2018Amsterdam On 02/21/2018 09:57 PM, Ben Newlin wrote: > > Bogdan, > > I have tested with this fix and I was able to start up with these > modules properly. I am getting another error now, but I will report > that in a new thread so as not to confuse unrelated issues. > > Thanks, > > Ben Newlin > > *From: *Bogdan-Andrei Iancu > *Date: *Wednesday, February 21, 2018 at 9:53 AM > *To: *Ben Newlin , OpenSIPS users mailling > list > *Subject: *Re: [OpenSIPS-Users] Avpops and db_virtual modules > > Hi Ben, > > OK, I reviewed the changes and the code and I identified the problem > in the way db_virtual is computing the capabilities in ROUND ROBIN > mode - more or less it was discarding the UPDATE DB capabilities, > which was required by avpops. > > I pushed a fix for this, this is for 2.3: > https://github.com/OpenSIPS/opensips/commit/edded2f1431cfaaf75ed2af0c5ff9ec28548e7c5 > > Please update and give it a try ! > > Thanks and Regards, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > OpenSIPS Summit 2018 > http://www.opensips.org/events/Summit-2018Amsterdam > > On 02/19/2018 04:20 PM, Ben Newlin wrote: > > Bogdan, > > Thank you for looking into this. I have confirmed that it works in > 2.3.2, which was released in 8/17 before the commit you provided. > > So is there really an issue with the db_virtual capabilities that > was masked by this bug, or is db_virtual simply not publishing the > correct capabilities itself? > > Thanks, > > Ben Newlin > > *From: *Bogdan-Andrei Iancu > > *Date: *Monday, February 19, 2018 at 6:11 AM > *To: *Ben Newlin > , OpenSIPS users mailling list > > *Subject: *Re: [OpenSIPS-Users] Avpops and db_virtual modules > > Hi Ben, > > Could you check if a 2.3 version prior this commit: > https://github.com/OpenSIPS/opensips/commit/f25fbf631b6fb6eafea192f95f9a43566d68cc46#diff-91bbef43b346f11cbef768a1b01c74e0 > will still have the 1.11 behavior ? > > > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Thu Feb 22 05:20:16 2018 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Thu, 22 Feb 2018 12:20:16 +0200 Subject: [OpenSIPS-Users] del_uri_param failure In-Reply-To: References: Message-ID: <6df5d699-fb1f-795f-29bb-9ed346a8a436@opensips.org> Hi Ben, What OpensSIPS version and revision are you working with ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Summit 2018 http://www.opensips.org/events/Summit-2018Amsterdam On 02/22/2018 01:07 AM, Ben Newlin wrote: > > Hi, > > I am very glad to have the new del_uri_param function. This was a > common problem of mine and it is great not to have to use regex to do > this. However, while implementing this I have run into some strange > behavior by the function when the URI param being deleted does not exist. > > In my case I am using the dialog module and attempting to remove a URI > param just after the dialog creation. When the function does not find > the URI param, it causes the dialog to immediately be destroyed and > all message processing stops, including exiting the script. > > Feb 21 22:47:42 [371] DBG:uri:del_uri_param: requested key not found > in RURI > > Feb 21 22:47:42 [371] DBG:dialog:next_state_dlg: unref dlg > 0x7f460780a8c8 with 1 -> 2 in entry 0x7f46077fbfa8 > > Feb 21 22:47:42 [371] DBG:core:evi_param_set: adding string param > > Feb 21 22:47:42 [371] DBG:core:evi_param_set: adding string param > > Feb 21 22:47:42 [371] DBG:core:evi_param_set: adding int param > > Feb 21 22:47:42 [371] DBG:core:evi_param_set: adding int param > > Feb 21 22:47:42 [371] DBG:core:destroy_avp_list: destroying list (nil) > > Feb 21 22:47:42 [371] DBG:dialog:next_state_dlg: dialog 0x7f460780a8c8 > changed from state 1 to state 5, due event 1 > > Feb 21 22:47:42 [371] DBG:dialog:dlg_onreply: dialog 0x7f460780a8c8 > failed (negative reply) > > Feb 21 22:47:42 [371] DBG:dialog:unref_dlg: unref dlg 0x7f460780a8c8 > with 1 -> 1 in entry 0x7f46077fbfa8 > > Feb 21 22:47:42 [371] DBG:dialog:unref_dlg: unref dlg 0x7f460780a8c8 > with 1 -> 0 in entry 0x7f46077fbfa8 > > Feb 21 22:47:42 [371] DBG:dialog:unref_dlg: ref <=0 for dialog > 0x7f460780a8c8 > > Feb 21 22:47:42 [371] DBG:dialog:destroy_dlg: destroying dialog > 0x7f460780a8c8 > > Feb 21 22:47:42 [371] DBG:dialog:destroy_dlg: dlg expired or not in > list - dlg 0x7f460780a8c8 [3710:1818203549] with clid > '2-185 at 127.0.0.1' and tags '185SIPpTag002' 'NULL' > > Feb 21 22:47:42 [371] DBG:core:destroy_avp_list: destroying list > 0x7f460780c048 > > Feb 21 22:47:42 [371] DBG:core:receive_msg: cleaning up > > The logs indicate that a DLG_EVENT_TDEL is being raised which, when > the dialog is still in UNCONFIRMED state, causes the dialog to be > destroyed. It’s not clear to me how or why the del_uri_param function > could be doing this, especially as a transaction hasn’t even been > created for the message yet in this case. I’m not sure what effect > this would have if the dialog is in other states or at other times > during the call. > > It took me a while to realize it was the del_uri_param function > causing this, as it seems so strange. But I have verified that when I > remove the function, or when I verify the URI param exists before > calling the function, everything is fine. That workaround works > perfectly well, but it seemed such strange and catastrophic error > behavior to drop the entire message that I wanted to report it anyway > to see if anything needed to be addressed. > > Call traces can be found here: https://pastebin.com/9FnmJCD9 > > You will see the same INVITE is offered multiple times as OpenSIPS is > not responding after dropping the previous requests and dialog. > > Thanks, > > Ben Newlin > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Thu Feb 22 05:25:25 2018 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Thu, 22 Feb 2018 12:25:25 +0200 Subject: [OpenSIPS-Users] OpenSIPS crashing on 200 OK In-Reply-To: <530E1611-2A4D-424F-938D-2E409E1A78A2@genesys.com> References: <530E1611-2A4D-424F-938D-2E409E1A78A2@genesys.com> Message-ID: <7cf6bb3d-967a-b8e2-d4b3-8b42c8dbf0b8@opensips.org> Hi Ben, Do you have a full backtrace extracted from the core ? In regards to the TCP processes, the TCP layer will automatically enabled itself depending on what protocols (tcp-based) you are using in script. Maybe you do not use SIP with TCP, but do you use BIN or HEP ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Summit 2018 http://www.opensips.org/events/Summit-2018Amsterdam On 02/22/2018 12:38 AM, Ben Newlin wrote: > > Hi, > > I am currently upgrading to 2.3.3 and am running off the HEAD of the > 2.3 branch. My OpenSIPS is consistently crashing when trying to > process the ACK to an initial INVITE request. I am testing with SIPp > using a simple single call scenario. I see the following in the logs: > > Feb 21 21:32:27 [383] DBG:core:io_wait_loop_epoll: [TCP_main] EPOLLHUP > on IN ->connection closed by the remote peer! > > Feb 21 21:32:27 [383] CRITICAL:core:receive_fd: EOF on 33 > > Feb 21 21:32:27 [383] DBG:core:handle_worker: dead child 9, pid 370 > (shutting down?) > > Feb 21 21:32:27 [383] DBG:core:io_watch_del: [TCP_main] io_watch_del > op on index 8 33 (0x8d4e80, 33, 8, 0x0,0x1) fd_no=29 called > > Feb 21 21:32:27 [361] DBG:core:handle_sigs: status = 139 > > Feb 21 21:32:27 [361] INFO:core:handle_sigs: child process 370 exited > by a signal 11 > > Feb 21 21:32:27 [361] INFO:core:handle_sigs: core was generated > > Feb 21 21:32:27 [361] INFO:core:handle_sigs: terminating due to SIGCHLD > > I also notice that OpenSIPS is starting up a number of TCP processes > even though I do not use TCP in my script. I do not even have the > proto_tcp module loaded. I used to use the “disable_tcp” global > parameter in 1.11, but that has been removed in 2.3. > > Feb 21 22:03:56 [361] DBG:core:internal_fork: forking new process "SIP > receiver TCP" > > Feb 21 22:03:56 [361] DBG:core:internal_fork: forking new process "SIP > receiver TCP" > > Feb 21 22:03:56 [361] DBG:core:internal_fork: forking new process "SIP > receiver TCP" > > Feb 21 22:03:56 [361] DBG:core:internal_fork: forking new process "SIP > receiver TCP" > > Feb 21 22:03:56 [361] DBG:core:internal_fork: forking new process "SIP > receiver TCP" > > Feb 21 22:03:56 [375] DBG:core:io_watch_add: [TCP_worker] io_watch_add > op (3 on 53) (0x8d4e80, 3, 1, (nil),1), fd_no=0/209715 > > Feb 21 22:03:56 [361] DBG:core:internal_fork: forking new process "SIP > receiver TCP" > > Feb 21 22:03:56 [361] DBG:core:internal_fork: forking new process "SIP > receiver TCP" > > Feb 21 22:03:56 [361] DBG:core:internal_fork: forking new process "SIP > receiver TCP" > > Initialization logs: https://pastebin.com/nsJQGkdy > > Call logs: https://pastebin.com/vY5crK8U > > I also have the core dump if any more information is needed from that. > > Thanks, > > Ben Newlin > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Thu Feb 22 05:27:28 2018 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Thu, 22 Feb 2018 12:27:28 +0200 Subject: [OpenSIPS-Users] How to avoid/solve TCP blocked connection? In-Reply-To: References: Message-ID: Hi Rodrigo, If I understand correctly, you get these errors (due failed connect attempt) when trying to deliver (from OpenSIPS) a call to a registered users who lost his TCP connection, right ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Summit 2018 http://www.opensips.org/events/Summit-2018Amsterdam On 02/20/2018 05:07 PM, Rodrigo Pimenta Carvalho wrote: > > Hi. > > > My softphone is registered with the following AOR: > > > ------------------------------------------------------------------------------------- > > AOR:: g1r2u3p4o5 > Contact:: sip:g1r2u3p4o5 at 127.0.0.1:50353;transport=TLS;ob Q= > Expires:: 10 > Callid:: 53e387dc-81fe-45f9-a6f1-8a5cf4248d62 > Cseq:: 37190 > User-agent:: n/a > Received:: sip:127.0.0.1:49678;transport=TLS > State:: CS_SYNC > Flags:: 0 > Cflags:: NAT > Socket:: tls:127.0.0.1:5061 > Methods:: 8063 > Attr:: in_same_network > SIP_instance:: > ------------------------------------------------------------------------------------------------------------------------- > > > > Sometimes, when it receives a call and answers, the Opensips show the > following error: > > > [21532] INFO:core:probe_max_sock_buff: using snd buffer of 320 kb > [21532] INFO:core:init_sock_keepalive: TCP keepalive enabled on socket 22 > [21532] ERROR:core:tcp_connect_blocking: poll error: flags 28 - 4 8 16 32 > [21532] ERROR:core:tcp_connect_blocking: failed to retrieve SO_ERROR > [server=127.0.0.1:50353] (111) Connection refused > [21532] ERROR:proto_tls:tls_sync_connect: tcp_blocking_connect failed > [21532] ERROR:proto_tls:proto_tls_send: connect failed > [21532] ERROR:core:msg_send: send() for proto 3 failed > > Is there a way to avoid this kind of problem? That is, can I configure > the OpenSIPS to renew some TCP connection? > > Any hint will be very helpful! > > Best regards. > > > RODRIGO PIMENTA CARVALHO > Inatel Competence Center > Software > Ph: +55 35 3471 9200 RAMAL 979 > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Thu Feb 22 05:30:48 2018 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Thu, 22 Feb 2018 12:30:48 +0200 Subject: [OpenSIPS-Users] Problem with simultaneous CANCEL + 200 OK In-Reply-To: References: Message-ID: <9f9681b1-1d92-18f3-5df0-91f032bf5e4e@opensips.org> Hi Daniel, Without a pcap showing the signaling is hard to understand (not to mention helping) your scenario. Please provide a link to the pcap or ngrep. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Summit 2018 http://www.opensips.org/events/Summit-2018Amsterdam On 02/20/2018 06:43 PM, Daniel Zanutti wrote: > Hey > > I had a problem when receiving simultaneous CANCEL from customer and > 200 OK from gateway. > > Seems that the first CANCEL was rejected, but the second CANCEL was > accepted. This second CANCEL did NOT go to the gateway, just Opensips > received and replied with 200 OK. > > This is the log of the first CANCEL: > Feb 15 18:39:22 /sbin/opensips[28845]: SCRIPT:TRAFFIC:WARNING: method > CANCEL (7Qbq3O3CReMfPflAtl8NY3ddTqPVBHO2785126 at 2.2.2.2/ > XAeG2xj278512T2 > / 1839212581509953 ) not validated and not fixed ( code=-1 ) > > code -1 is the return of validate_dialog() > > Second CANCEL didn't generated a log. > > Shouldn't all CANCELs be rejected? On this case, just the first one > was rejected. > > I'm using version 1.9.11. > > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From Ben.Newlin at genesys.com Thu Feb 22 08:18:40 2018 From: Ben.Newlin at genesys.com (Ben Newlin) Date: Thu, 22 Feb 2018 13:18:40 +0000 Subject: [OpenSIPS-Users] del_uri_param failure In-Reply-To: <6df5d699-fb1f-795f-29bb-9ed346a8a436@opensips.org> References: <6df5d699-fb1f-795f-29bb-9ed346a8a436@opensips.org> Message-ID: I saw this behavior in 2.3.2, 2.3.3, and I am now running on HEAD of the 2.3 branch. Ben Newlin From: Bogdan-Andrei Iancu Date: Thursday, February 22, 2018 at 5:20 AM To: OpenSIPS users mailling list , Ben Newlin Subject: Re: [OpenSIPS-Users] del_uri_param failure Hi Ben, What OpensSIPS version and revision are you working with ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Summit 2018 http://www.opensips.org/events/Summit-2018Amsterdam On 02/22/2018 01:07 AM, Ben Newlin wrote: Hi, I am very glad to have the new del_uri_param function. This was a common problem of mine and it is great not to have to use regex to do this. However, while implementing this I have run into some strange behavior by the function when the URI param being deleted does not exist. In my case I am using the dialog module and attempting to remove a URI param just after the dialog creation. When the function does not find the URI param, it causes the dialog to immediately be destroyed and all message processing stops, including exiting the script. Feb 21 22:47:42 [371] DBG:uri:del_uri_param: requested key not found in RURI Feb 21 22:47:42 [371] DBG:dialog:next_state_dlg: unref dlg 0x7f460780a8c8 with 1 -> 2 in entry 0x7f46077fbfa8 Feb 21 22:47:42 [371] DBG:core:evi_param_set: adding string param Feb 21 22:47:42 [371] DBG:core:evi_param_set: adding string param Feb 21 22:47:42 [371] DBG:core:evi_param_set: adding int param Feb 21 22:47:42 [371] DBG:core:evi_param_set: adding int param Feb 21 22:47:42 [371] DBG:core:destroy_avp_list: destroying list (nil) Feb 21 22:47:42 [371] DBG:dialog:next_state_dlg: dialog 0x7f460780a8c8 changed from state 1 to state 5, due event 1 Feb 21 22:47:42 [371] DBG:dialog:dlg_onreply: dialog 0x7f460780a8c8 failed (negative reply) Feb 21 22:47:42 [371] DBG:dialog:unref_dlg: unref dlg 0x7f460780a8c8 with 1 -> 1 in entry 0x7f46077fbfa8 Feb 21 22:47:42 [371] DBG:dialog:unref_dlg: unref dlg 0x7f460780a8c8 with 1 -> 0 in entry 0x7f46077fbfa8 Feb 21 22:47:42 [371] DBG:dialog:unref_dlg: ref <=0 for dialog 0x7f460780a8c8 Feb 21 22:47:42 [371] DBG:dialog:destroy_dlg: destroying dialog 0x7f460780a8c8 Feb 21 22:47:42 [371] DBG:dialog:destroy_dlg: dlg expired or not in list - dlg 0x7f460780a8c8 [3710:1818203549] with clid '2-185 at 127.0.0.1' and tags '185SIPpTag002' 'NULL' Feb 21 22:47:42 [371] DBG:core:destroy_avp_list: destroying list 0x7f460780c048 Feb 21 22:47:42 [371] DBG:core:receive_msg: cleaning up The logs indicate that a DLG_EVENT_TDEL is being raised which, when the dialog is still in UNCONFIRMED state, causes the dialog to be destroyed. It’s not clear to me how or why the del_uri_param function could be doing this, especially as a transaction hasn’t even been created for the message yet in this case. I’m not sure what effect this would have if the dialog is in other states or at other times during the call. It took me a while to realize it was the del_uri_param function causing this, as it seems so strange. But I have verified that when I remove the function, or when I verify the URI param exists before calling the function, everything is fine. That workaround works perfectly well, but it seemed such strange and catastrophic error behavior to drop the entire message that I wanted to report it anyway to see if anything needed to be addressed. Call traces can be found here: https://pastebin.com/9FnmJCD9 You will see the same INVITE is offered multiple times as OpenSIPS is not responding after dropping the previous requests and dialog. Thanks, Ben Newlin _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From daniel.zanutti at gmail.com Thu Feb 22 09:22:52 2018 From: daniel.zanutti at gmail.com (Daniel Zanutti) Date: Thu, 22 Feb 2018 11:22:52 -0300 Subject: [OpenSIPS-Users] Problem with simultaneous CANCEL + 200 OK In-Reply-To: <9f9681b1-1d92-18f3-5df0-91f032bf5e4e@opensips.org> References: <9f9681b1-1d92-18f3-5df0-91f032bf5e4e@opensips.org> Message-ID: Hi Bogdan Thanks for replying. Here is the PCAP, please take a look: https://drive.google.com/file/d/1e7SKjxDtdVYmN-7fCHSEqNEsNHjPsaKo/view?usp=sharing Thanks On Thu, Feb 22, 2018 at 7:30 AM, Bogdan-Andrei Iancu wrote: > Hi Daniel, > > Without a pcap showing the signaling is hard to understand (not to mention > helping) your scenario. Please provide a link to the pcap or ngrep. > > Regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > OpenSIPS Summit 2018 > http://www.opensips.org/events/Summit-2018Amsterdam > > On 02/20/2018 06:43 PM, Daniel Zanutti wrote: > > Hey > > I had a problem when receiving simultaneous CANCEL from customer and 200 > OK from gateway. > > Seems that the first CANCEL was rejected, but the second CANCEL was > accepted. This second CANCEL did NOT go to the gateway, just Opensips > received and replied with 200 OK. > > This is the log of the first CANCEL: > Feb 15 18:39:22 /sbin/opensips[28845]: SCRIPT:TRAFFIC:WARNING: method > CANCEL ( 7Qbq3O3CReMfPflAtl8NY3ddTqPVBHO2785126 at 2.2.2.2/ XAeG2xj278512T2 > / 1839212581509953 ) not validated and not fixed ( code=-1 ) > > code -1 is the return of validate_dialog() > > Second CANCEL didn't generated a log. > > Shouldn't all CANCELs be rejected? On this case, just the first one was > rejected. > > I'm using version 1.9.11. > > > > > _______________________________________________ > Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Thu Feb 22 10:02:34 2018 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Thu, 22 Feb 2018 17:02:34 +0200 Subject: [OpenSIPS-Users] del_uri_param failure In-Reply-To: References: <6df5d699-fb1f-795f-29bb-9ed346a8a436@opensips.org> Message-ID: <1a21f833-e2e1-7f5f-5c63-87d100449f23@opensips.org> Hi Ben, I just pushed the fix on git, it was areally stupid error. Thanks for reporting, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Summit 2018 http://www.opensips.org/events/Summit-2018Amsterdam On 02/22/2018 03:18 PM, Ben Newlin wrote: > > I saw this behavior in 2.3.2, 2.3.3, and I am now running on HEAD of > the 2.3 branch. > > Ben Newlin > > *From: *Bogdan-Andrei Iancu > *Date: *Thursday, February 22, 2018 at 5:20 AM > *To: *OpenSIPS users mailling list , Ben > Newlin > *Subject: *Re: [OpenSIPS-Users] del_uri_param failure > > Hi Ben, > > What OpensSIPS version and revision are you working with ? > > Regards, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > OpenSIPS Summit 2018 > http://www.opensips.org/events/Summit-2018Amsterdam > > On 02/22/2018 01:07 AM, Ben Newlin wrote: > > Hi, > > I am very glad to have the new del_uri_param function. This was a > common problem of mine and it is great not to have to use regex to > do this. However, while implementing this I have run into some > strange behavior by the function when the URI param being deleted > does not exist. > > In my case I am using the dialog module and attempting to remove a > URI param just after the dialog creation. When the function does > not find the URI param, it causes the dialog to immediately be > destroyed and all message processing stops, including exiting the > script. > > Feb 21 22:47:42 [371] DBG:uri:del_uri_param: requested key not > found in RURI > > Feb 21 22:47:42 [371] DBG:dialog:next_state_dlg: unref dlg > 0x7f460780a8c8 with 1 -> 2 in entry 0x7f46077fbfa8 > > Feb 21 22:47:42 [371] DBG:core:evi_param_set: adding string param > > Feb 21 22:47:42 [371] DBG:core:evi_param_set: adding string param > > Feb 21 22:47:42 [371] DBG:core:evi_param_set: adding int param > > Feb 21 22:47:42 [371] DBG:core:evi_param_set: adding int param > > Feb 21 22:47:42 [371] DBG:core:destroy_avp_list: destroying list (nil) > > Feb 21 22:47:42 [371] DBG:dialog:next_state_dlg: dialog > 0x7f460780a8c8 changed from state 1 to state 5, due event 1 > > Feb 21 22:47:42 [371] DBG:dialog:dlg_onreply: dialog > 0x7f460780a8c8 failed (negative reply) > > Feb 21 22:47:42 [371] DBG:dialog:unref_dlg: unref dlg > 0x7f460780a8c8 with 1 -> 1 in entry 0x7f46077fbfa8 > > Feb 21 22:47:42 [371] DBG:dialog:unref_dlg: unref dlg > 0x7f460780a8c8 with 1 -> 0 in entry 0x7f46077fbfa8 > > Feb 21 22:47:42 [371] DBG:dialog:unref_dlg: ref <=0 for dialog > 0x7f460780a8c8 > > Feb 21 22:47:42 [371] DBG:dialog:destroy_dlg: destroying dialog > 0x7f460780a8c8 > > Feb 21 22:47:42 [371] DBG:dialog:destroy_dlg: dlg expired or not > in list - dlg 0x7f460780a8c8 [3710:1818203549] with clid > '2-185 at 127.0.0.1' and tags '185SIPpTag002' > 'NULL' > > Feb 21 22:47:42 [371] DBG:core:destroy_avp_list: destroying list > 0x7f460780c048 > > Feb 21 22:47:42 [371] DBG:core:receive_msg: cleaning up > > The logs indicate that a DLG_EVENT_TDEL is being raised which, > when the dialog is still in UNCONFIRMED state, causes the dialog > to be destroyed. It’s not clear to me how or why the del_uri_param > function could be doing this, especially as a transaction hasn’t > even been created for the message yet in this case. I’m not sure > what effect this would have if the dialog is in other states or at > other times during the call. > > It took me a while to realize it was the del_uri_param function > causing this, as it seems so strange. But I have verified that > when I remove the function, or when I verify the URI param exists > before calling the function, everything is fine. That workaround > works perfectly well, but it seemed such strange and catastrophic > error behavior to drop the entire message that I wanted to report > it anyway to see if anything needed to be addressed. > > Call traces can be found here: https://pastebin.com/9FnmJCD9 > > You will see the same INVITE is offered multiple times as OpenSIPS > is not responding after dropping the previous requests and dialog. > > Thanks, > > Ben Newlin > > > > > _______________________________________________ > > Users mailing list > > Users at lists.opensips.org > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Thu Feb 22 11:24:01 2018 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Thu, 22 Feb 2018 18:24:01 +0200 Subject: [OpenSIPS-Users] Problem with simultaneous CANCEL + 200 OK In-Reply-To: References: <9f9681b1-1d92-18f3-5df0-91f032bf5e4e@opensips.org> Message-ID: <41b30a10-3743-7938-aeaa-f81547c71109@opensips.org> Daniel, The pcap shows only one leg of the the communication (I guess caller versus OpenSIPS). So I cannot see what happens on the callee side. Still, I see the first cancel is rejected with 400 (I guess by OpenSIPS - check the logs to see the reason) and the second one is accepted by OpenSIPS (no idea if relayed to callee or not). Still, if there is a race between the cancel from caller and the answer from callee, it is up to the caller to sort it out....according to RFC, the caller must ACK the received 200 OK (even if CANCEL was sent) and if it really wants to terminate the call, it has to fire a BYE. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Summit 2018 http://www.opensips.org/events/Summit-2018Amsterdam On 02/22/2018 04:22 PM, Daniel Zanutti wrote: > Hi Bogdan > > Thanks for replying. > > Here is the PCAP, please take a look: > https://drive.google.com/file/d/1e7SKjxDtdVYmN-7fCHSEqNEsNHjPsaKo/view?usp=sharing > > Thanks > > On Thu, Feb 22, 2018 at 7:30 AM, Bogdan-Andrei Iancu > > wrote: > > Hi Daniel, > > Without a pcap showing the signaling is hard to understand (not to > mention helping) your scenario. Please provide a link to the pcap > or ngrep. > > Regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > OpenSIPS Summit 2018 > http://www.opensips.org/events/Summit-2018Amsterdam > > > On 02/20/2018 06:43 PM, Daniel Zanutti wrote: >> Hey >> >> I had a problem when receiving simultaneous CANCEL from customer >> and 200 OK from gateway. >> >> Seems that the first CANCEL was rejected, but the second CANCEL >> was accepted. This second CANCEL did NOT go to the gateway, just >> Opensips received and replied with 200 OK. >> >> This is the log of the first CANCEL: >> Feb 15 18:39:22 /sbin/opensips[28845]: SCRIPT:TRAFFIC:WARNING: >> method CANCEL (7Qbq3O3CReMfPflAtl8NY3ddTqPVBHO2785126 at 2.2.2.2/ >> XAeG2xj278512T2 >> / 1839212581509953 ) not validated and not fixed ( code=-1 ) >> >> code -1 is the return of validate_dialog() >> >> Second CANCEL didn't generated a log. >> >> Shouldn't all CANCELs be rejected? On this case, just the first >> one was rejected. >> >> I'm using version 1.9.11. >> >> >> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From Ben.Newlin at genesys.com Thu Feb 22 11:59:45 2018 From: Ben.Newlin at genesys.com (Ben Newlin) Date: Thu, 22 Feb 2018 16:59:45 +0000 Subject: [OpenSIPS-Users] del_uri_param failure In-Reply-To: <1a21f833-e2e1-7f5f-5c63-87d100449f23@opensips.org> References: <6df5d699-fb1f-795f-29bb-9ed346a8a436@opensips.org> <1a21f833-e2e1-7f5f-5c63-87d100449f23@opensips.org> Message-ID: <8CA33A8B-DFF4-41CF-9526-9F3A6632E4F2@genesys.com> Bogdan, Thanks for the quick response. I have verified the fix works in 2.3 branch head. Thanks, Ben Newlin From: Bogdan-Andrei Iancu Date: Thursday, February 22, 2018 at 10:03 AM To: Ben Newlin , OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] del_uri_param failure Hi Ben, I just pushed the fix on git, it was a really stupid error. Thanks for reporting, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Summit 2018 http://www.opensips.org/events/Summit-2018Amsterdam On 02/22/2018 03:18 PM, Ben Newlin wrote: I saw this behavior in 2.3.2, 2.3.3, and I am now running on HEAD of the 2.3 branch. Ben Newlin From: Bogdan-Andrei Iancu Date: Thursday, February 22, 2018 at 5:20 AM To: OpenSIPS users mailling list , Ben Newlin Subject: Re: [OpenSIPS-Users] del_uri_param failure Hi Ben, What OpensSIPS version and revision are you working with ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Summit 2018 http://www.opensips.org/events/Summit-2018Amsterdam On 02/22/2018 01:07 AM, Ben Newlin wrote: Hi, I am very glad to have the new del_uri_param function. This was a common problem of mine and it is great not to have to use regex to do this. However, while implementing this I have run into some strange behavior by the function when the URI param being deleted does not exist. In my case I am using the dialog module and attempting to remove a URI param just after the dialog creation. When the function does not find the URI param, it causes the dialog to immediately be destroyed and all message processing stops, including exiting the script. Feb 21 22:47:42 [371] DBG:uri:del_uri_param: requested key not found in RURI Feb 21 22:47:42 [371] DBG:dialog:next_state_dlg: unref dlg 0x7f460780a8c8 with 1 -> 2 in entry 0x7f46077fbfa8 Feb 21 22:47:42 [371] DBG:core:evi_param_set: adding string param Feb 21 22:47:42 [371] DBG:core:evi_param_set: adding string param Feb 21 22:47:42 [371] DBG:core:evi_param_set: adding int param Feb 21 22:47:42 [371] DBG:core:evi_param_set: adding int param Feb 21 22:47:42 [371] DBG:core:destroy_avp_list: destroying list (nil) Feb 21 22:47:42 [371] DBG:dialog:next_state_dlg: dialog 0x7f460780a8c8 changed from state 1 to state 5, due event 1 Feb 21 22:47:42 [371] DBG:dialog:dlg_onreply: dialog 0x7f460780a8c8 failed (negative reply) Feb 21 22:47:42 [371] DBG:dialog:unref_dlg: unref dlg 0x7f460780a8c8 with 1 -> 1 in entry 0x7f46077fbfa8 Feb 21 22:47:42 [371] DBG:dialog:unref_dlg: unref dlg 0x7f460780a8c8 with 1 -> 0 in entry 0x7f46077fbfa8 Feb 21 22:47:42 [371] DBG:dialog:unref_dlg: ref <=0 for dialog 0x7f460780a8c8 Feb 21 22:47:42 [371] DBG:dialog:destroy_dlg: destroying dialog 0x7f460780a8c8 Feb 21 22:47:42 [371] DBG:dialog:destroy_dlg: dlg expired or not in list - dlg 0x7f460780a8c8 [3710:1818203549] with clid '2-185 at 127.0.0.1' and tags '185SIPpTag002' 'NULL' Feb 21 22:47:42 [371] DBG:core:destroy_avp_list: destroying list 0x7f460780c048 Feb 21 22:47:42 [371] DBG:core:receive_msg: cleaning up The logs indicate that a DLG_EVENT_TDEL is being raised which, when the dialog is still in UNCONFIRMED state, causes the dialog to be destroyed. It’s not clear to me how or why the del_uri_param function could be doing this, especially as a transaction hasn’t even been created for the message yet in this case. I’m not sure what effect this would have if the dialog is in other states or at other times during the call. It took me a while to realize it was the del_uri_param function causing this, as it seems so strange. But I have verified that when I remove the function, or when I verify the URI param exists before calling the function, everything is fine. That workaround works perfectly well, but it seemed such strange and catastrophic error behavior to drop the entire message that I wanted to report it anyway to see if anything needed to be addressed. Call traces can be found here: https://pastebin.com/9FnmJCD9 You will see the same INVITE is offered multiple times as OpenSIPS is not responding after dropping the previous requests and dialog. Thanks, Ben Newlin _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From Ben.Newlin at genesys.com Thu Feb 22 12:33:05 2018 From: Ben.Newlin at genesys.com (Ben Newlin) Date: Thu, 22 Feb 2018 17:33:05 +0000 Subject: [OpenSIPS-Users] OpenSIPS crashing on 200 OK In-Reply-To: <7cf6bb3d-967a-b8e2-d4b3-8b42c8dbf0b8@opensips.org> References: <530E1611-2A4D-424F-938D-2E409E1A78A2@genesys.com> <7cf6bb3d-967a-b8e2-d4b3-8b42c8dbf0b8@opensips.org> Message-ID: <755F831E-90E4-4105-80C5-42C7A2E2050A@genesys.com> Bogdan, Here is the backtrace: https://pastebin.com/6tZhrWek I am not using TCP, BIN, or HEP in this configuration. I am not using anything, to my knowledge, which should be causing TCP worker threads to be spawned. Ben Newlin From: Bogdan-Andrei Iancu Date: Thursday, February 22, 2018 at 5:26 AM To: OpenSIPS users mailling list , Ben Newlin Subject: Re: [OpenSIPS-Users] OpenSIPS crashing on 200 OK Hi Ben, Do you have a full backtrace extracted from the core ? In regards to the TCP processes, the TCP layer will automatically enabled itself depending on what protocols (tcp-based) you are using in script. Maybe you do not use SIP with TCP, but do you use BIN or HEP ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Summit 2018 http://www.opensips.org/events/Summit-2018Amsterdam On 02/22/2018 12:38 AM, Ben Newlin wrote: Hi, I am currently upgrading to 2.3.3 and am running off the HEAD of the 2.3 branch. My OpenSIPS is consistently crashing when trying to process the ACK to an initial INVITE request. I am testing with SIPp using a simple single call scenario. I see the following in the logs: Feb 21 21:32:27 [383] DBG:core:io_wait_loop_epoll: [TCP_main] EPOLLHUP on IN ->connection closed by the remote peer! Feb 21 21:32:27 [383] CRITICAL:core:receive_fd: EOF on 33 Feb 21 21:32:27 [383] DBG:core:handle_worker: dead child 9, pid 370 (shutting down?) Feb 21 21:32:27 [383] DBG:core:io_watch_del: [TCP_main] io_watch_del op on index 8 33 (0x8d4e80, 33, 8, 0x0,0x1) fd_no=29 called Feb 21 21:32:27 [361] DBG:core:handle_sigs: status = 139 Feb 21 21:32:27 [361] INFO:core:handle_sigs: child process 370 exited by a signal 11 Feb 21 21:32:27 [361] INFO:core:handle_sigs: core was generated Feb 21 21:32:27 [361] INFO:core:handle_sigs: terminating due to SIGCHLD I also notice that OpenSIPS is starting up a number of TCP processes even though I do not use TCP in my script. I do not even have the proto_tcp module loaded. I used to use the “disable_tcp” global parameter in 1.11, but that has been removed in 2.3. Feb 21 22:03:56 [361] DBG:core:internal_fork: forking new process "SIP receiver TCP" Feb 21 22:03:56 [361] DBG:core:internal_fork: forking new process "SIP receiver TCP" Feb 21 22:03:56 [361] DBG:core:internal_fork: forking new process "SIP receiver TCP" Feb 21 22:03:56 [361] DBG:core:internal_fork: forking new process "SIP receiver TCP" Feb 21 22:03:56 [361] DBG:core:internal_fork: forking new process "SIP receiver TCP" Feb 21 22:03:56 [375] DBG:core:io_watch_add: [TCP_worker] io_watch_add op (3 on 53) (0x8d4e80, 3, 1, (nil),1), fd_no=0/209715 Feb 21 22:03:56 [361] DBG:core:internal_fork: forking new process "SIP receiver TCP" Feb 21 22:03:56 [361] DBG:core:internal_fork: forking new process "SIP receiver TCP" Feb 21 22:03:56 [361] DBG:core:internal_fork: forking new process "SIP receiver TCP" Initialization logs: https://pastebin.com/nsJQGkdy Call logs: https://pastebin.com/vY5crK8U I also have the core dump if any more information is needed from that. Thanks, Ben Newlin _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From govoiper at gmail.com Thu Feb 22 23:13:42 2018 From: govoiper at gmail.com (SamyGo) Date: Thu, 22 Feb 2018 23:13:42 -0500 Subject: [OpenSIPS-Users] ACC module with JSON events Message-ID: Hi, I'm using ACC module to trigger accounting events which land up in event_xmlrpc and ultimately shipped to remote CDR collection server. The sent data is obviously an XML - now, can I have an event_json like module so data is sent as JSON or is there any other way I can achieve the same ? Regards Sammy -------------- next part -------------- An HTML attachment was scrubbed... URL: From brian.southworth at clocom.uk Fri Feb 23 04:12:37 2018 From: brian.southworth at clocom.uk (=?UTF-8?Q?Brian_Southworth?=) Date: Fri, 23 Feb 2018 09:12:37 +0000 Subject: [OpenSIPS-Users] On hold References: Message-ID: <01020161c1ef8de9-1fa3116b-b84a-4c0c-a3a9-f3419d72813d-000000@eu-west-1.amazonses.com> Hi All,   I am trying to get opensips to forward the on hold request to asterisk I have done this using an if statement for SDP =~ “sendonly”   How ever when I go to take them off hold the call is dropped, is there something special I need to do ?   I have tried and if statement for is_method(“invite”) && $rb =~ “sendrecv” ß this loops until opensips give a 500 internal error And I have also tried is_method(“invite”) && $rb =~ “inactive” ß this doesn’t seem to work at all   The initial hold works perfect puts them on hold just cant take them off or transfer the call (asterisk will handle all the B2B stuff)   Any help would be appreciated.   Regards,   Brian Southworth Communications Developer 111 Wilmslow Road Handforth Wilmslow SK9 3ER   T: 03333 446677 W: www.clocom.uk           Like us on Facebook Follow us on Twitter           Clocom is a green company. Think, do you need to print this email?   This message contains confidential information and is intended only for the individual named. If you are not the named addressee you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately by e-mail if you have received this e-mail by mistake and delete this e-mail from your system. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version. Registered in England & Wales: 07081192. Registered name and address: Clocom UK Ltd, 111 Wilmslow Road, Handforth, Cheshire, SK9 3ER  www.clocom.uk   -------------- next part -------------- An HTML attachment was scrubbed... URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/png Size: 9352 bytes Desc: not available URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/png Size: 2708 bytes Desc: not available URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/png Size: 234 bytes Desc: not available URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/png Size: 2999 bytes Desc: not available URL: From xaled at web.de Fri Feb 23 07:25:31 2018 From: xaled at web.de (xaled) Date: Fri, 23 Feb 2018 13:25:31 +0100 Subject: [OpenSIPS-Users] Wrong number of children Message-ID: <027501d3aca1$66551340$32ff39c0$@web.de> Hi I have configured 5 children in opensips.cfg, but 16 get logged and 18 initiated what could be the cause of it? I search for word children in opensips directory and it only comes ones in opensips.conf. children=5 Feb 23 13:07:09 srv01 opensips: DBG:core:daemonize: waiting for status code from children Feb 23 13:07:09 srv01 opensips: DBG:core:count_init_children: 16 children are going to be inited root at dus-appsrv01:/var/log/opensips# ps ax|grep sips 14854 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14856 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14857 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14858 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14861 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14862 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14863 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14864 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14865 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14867 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14869 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14870 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14871 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14872 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14874 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14875 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14876 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14877 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 With commended children parameter I get 19 children in the log and 21 in reality: Feb 23 13:13:24 dus-appsrv01 opensips: DBG:core:daemonize: waiting for status code from children Feb 23 13:13:24 dus-appsrv01 opensips: DBG:core:count_init_children: 19 children are going to be inited root at srv01:/var/log/opensips# ps ax|grep sips 14723 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14725 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14726 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14727 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14730 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14731 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14732 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14733 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14734 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14736 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14737 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14738 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14739 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14741 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14742 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14743 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14744 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14745 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14748 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14749 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14750 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 root at srv01:/etc/opensips# opensips -V version: opensips 2.3.3 (x86_64/linux) flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535 poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. main.c compiled on with gcc 6.3.0 Thanks, Xaled -------------- next part -------------- An HTML attachment was scrubbed... URL: From osas at voipembedded.com Fri Feb 23 08:09:55 2018 From: osas at voipembedded.com (Ovidiu Sas) Date: Fri, 23 Feb 2018 08:09:55 -0500 Subject: [OpenSIPS-Users] On hold In-Reply-To: <01020161c1ef8de9-1fa3116b-b84a-4c0c-a3a9-f3419d72813d-000000@eu-west-1.amazonses.com> References: <01020161c1ef8de9-1fa3116b-b84a-4c0c-a3a9-f3419d72813d-000000@eu-west-1.amazonses.com> Message-ID: Hello Brian, You can detect hold using is_audio_on_hold(): http://www.opensips.org/html/docs/modules/2.3.x/sipmsgops#idp5575408 Try to use it and see if this will solve your issue. Regards, Ovidiu Sas On Feb 23, 2018 4:13 AM, "Brian Southworth" wrote: Hi All, I am trying to get opensips to forward the on hold request to asterisk I have done this using an if statement for SDP =~ “sendonly” How ever when I go to take them off hold the call is dropped, is there something special I need to do ? I have tried and if statement for is_method(“invite”) && $rb =~ “sendrecv” ß this loops until opensips give a 500 internal error And I have also tried is_method(“invite”) && $rb =~ “inactive” ß this doesn’t seem to work at all The initial hold works perfect puts them on hold just cant take them off or transfer the call (asterisk will handle all the B2B stuff) Any help would be appreciated. Regards, Brian Southworth Communications Developer [image: cid:image001.png at 01D22CAC.1DCB8580] 111 Wilmslow Road Handforth Wilmslow SK9 3ER T: 03333 446677 W: www.clocom.uk [image: cid:image002.png at 01CDDC62.D8483910] Like us on Facebook [image: cid:image003.png at 01CDDC62.D8483910] Follow us on Twitter [image: cid:image004.png at 01CDDC62.D8483910] Clocom is a *green* company. Think, do you need to print this email? This message contains confidential information and is intended only for the individual named. If you are not the named addressee you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately by e-mail if you have received this e-mail by mistake and delete this e-mail from your system. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version. Registered in England & Wales: 07081192. Registered name and address: Clocom UK Ltd, 111 Wilmslow Road, Handforth, Cheshire, SK9 3ER www.clocom.uk _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/png Size: 234 bytes Desc: not available URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/png Size: 2999 bytes Desc: not available URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/png Size: 9352 bytes Desc: not available URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/png Size: 2708 bytes Desc: not available URL: From daniel.zanutti at gmail.com Fri Feb 23 08:33:40 2018 From: daniel.zanutti at gmail.com (Daniel Zanutti) Date: Fri, 23 Feb 2018 10:33:40 -0300 Subject: [OpenSIPS-Users] Wrong number of children In-Reply-To: <027501d3aca1$66551340$32ff39c0$@web.de> References: <027501d3aca1$66551340$32ff39c0$@web.de> Message-ID: Hi You should have 5 per interface + some internal control threads. I'm not sure exactly. Regards On Fri, Feb 23, 2018 at 9:25 AM, xaled wrote: > Hi > > > > I have configured 5 children in opensips.cfg, but 16 get logged and 18 > initiated what could be the cause of it? > > I search for word children in opensips directory and it only comes ones in > opensips.conf. > > > > children=5 > > > > Feb 23 13:07:09 srv01 opensips: DBG:core:daemonize: waiting for status > code from children > > Feb 23 13:07:09 srv01 opensips: DBG:core:count_init_children: 16 children > are going to be inited > > > > root at dus-appsrv01:/var/log/opensips# ps ax|grep sips > > 14854 ? S 0:00 /usr/sbin/opensips -P > /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 > > 14856 ? S 0:00 /usr/sbin/opensips -P > /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 > > 14857 ? S 0:00 /usr/sbin/opensips -P > /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 > > 14858 ? S 0:00 /usr/sbin/opensips -P > /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 > > 14861 ? S 0:00 /usr/sbin/opensips -P > /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 > > 14862 ? S 0:00 /usr/sbin/opensips -P > /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 > > 14863 ? S 0:00 /usr/sbin/opensips -P > /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 > > 14864 ? S 0:00 /usr/sbin/opensips -P > /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 > > 14865 ? S 0:00 /usr/sbin/opensips -P > /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 > > 14867 ? S 0:00 /usr/sbin/opensips -P > /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 > > 14869 ? S 0:00 /usr/sbin/opensips -P > /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 > > 14870 ? S 0:00 /usr/sbin/opensips -P > /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 > > 14871 ? S 0:00 /usr/sbin/opensips -P > /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 > > 14872 ? S 0:00 /usr/sbin/opensips -P > /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 > > 14874 ? S 0:00 /usr/sbin/opensips -P > /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 > > 14875 ? S 0:00 /usr/sbin/opensips -P > /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 > > 14876 ? S 0:00 /usr/sbin/opensips -P > /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 > > 14877 ? S 0:00 /usr/sbin/opensips -P > /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 > > > > With commended children parameter I get 19 children in the log and 21 in > reality: > > > > Feb 23 13:13:24 dus-appsrv01 opensips: DBG:core:daemonize: waiting for > status code from children > > Feb 23 13:13:24 dus-appsrv01 opensips: DBG:core:count_init_children: 19 > children are going to be inited > > > > root at srv01:/var/log/opensips# ps ax|grep sips > > 14723 ? S 0:00 /usr/sbin/opensips -P > /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 > > 14725 ? S 0:00 /usr/sbin/opensips -P > /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 > > 14726 ? S 0:00 /usr/sbin/opensips -P > /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 > > 14727 ? S 0:00 /usr/sbin/opensips -P > /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 > > 14730 ? S 0:00 /usr/sbin/opensips -P > /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 > > 14731 ? S 0:00 /usr/sbin/opensips -P > /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 > > 14732 ? S 0:00 /usr/sbin/opensips -P > /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 > > 14733 ? S 0:00 /usr/sbin/opensips -P > /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 > > 14734 ? S 0:00 /usr/sbin/opensips -P > /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 > > 14736 ? S 0:00 /usr/sbin/opensips -P > /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 > > 14737 ? S 0:00 /usr/sbin/opensips -P > /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 > > 14738 ? S 0:00 /usr/sbin/opensips -P > /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 > > 14739 ? S 0:00 /usr/sbin/opensips -P > /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 > > 14741 ? S 0:00 /usr/sbin/opensips -P > /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 > > 14742 ? S 0:00 /usr/sbin/opensips -P > /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 > > 14743 ? S 0:00 /usr/sbin/opensips -P > /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 > > 14744 ? S 0:00 /usr/sbin/opensips -P > /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 > > 14745 ? S 0:00 /usr/sbin/opensips -P > /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 > > 14748 ? S 0:00 /usr/sbin/opensips -P > /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 > > 14749 ? S 0:00 /usr/sbin/opensips -P > /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 > > 14750 ? S 0:00 /usr/sbin/opensips -P > /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 > > > > root at srv01:/etc/opensips# opensips -V > > version: opensips 2.3.3 (x86_64/linux) > > flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, > F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT > > ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, > MAX_URI_SIZE 1024, BUF_SIZE 65535 > > poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. > > main.c compiled on with gcc 6.3.0 > > > > Thanks, > > Xaled > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From callum.guy at x-on.co.uk Fri Feb 23 08:41:52 2018 From: callum.guy at x-on.co.uk (Callum Guy) Date: Fri, 23 Feb 2018 13:41:52 +0000 Subject: [OpenSIPS-Users] Wrong number of children In-Reply-To: References: <027501d3aca1$66551340$32ff39c0$@web.de> Message-ID: Hi Xaled, If you have the FIFO module loaded you can run "opensipsctl fifo ps" on the instance to see full details of each process. https://www.opensips.org/Documentation/Interface-CoreMI-2-2#toc7 Hope that helps, Callum On Fri, Feb 23, 2018 at 1:35 PM Daniel Zanutti wrote: > Hi > > You should have 5 per interface + some internal control threads. I'm not > sure exactly. > > Regards > > On Fri, Feb 23, 2018 at 9:25 AM, xaled wrote: > >> Hi >> >> >> >> I have configured 5 children in opensips.cfg, but 16 get logged and 18 >> initiated what could be the cause of it? >> >> I search for word children in opensips directory and it only comes ones >> in opensips.conf. >> >> >> >> children=5 >> >> >> >> Feb 23 13:07:09 srv01 opensips: DBG:core:daemonize: waiting for status >> code from children >> >> Feb 23 13:07:09 srv01 opensips: DBG:core:count_init_children: 16 children >> are going to be inited >> >> >> >> root at dus-appsrv01:/var/log/opensips# ps ax|grep sips >> >> 14854 ? S 0:00 /usr/sbin/opensips -P >> /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 >> >> 14856 ? S 0:00 /usr/sbin/opensips -P >> /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 >> >> 14857 ? S 0:00 /usr/sbin/opensips -P >> /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 >> >> 14858 ? S 0:00 /usr/sbin/opensips -P >> /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 >> >> 14861 ? S 0:00 /usr/sbin/opensips -P >> /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 >> >> 14862 ? S 0:00 /usr/sbin/opensips -P >> /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 >> >> 14863 ? S 0:00 /usr/sbin/opensips -P >> /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 >> >> 14864 ? S 0:00 /usr/sbin/opensips -P >> /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 >> >> 14865 ? S 0:00 /usr/sbin/opensips -P >> /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 >> >> 14867 ? S 0:00 /usr/sbin/opensips -P >> /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 >> >> 14869 ? S 0:00 /usr/sbin/opensips -P >> /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 >> >> 14870 ? S 0:00 /usr/sbin/opensips -P >> /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 >> >> 14871 ? S 0:00 /usr/sbin/opensips -P >> /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 >> >> 14872 ? S 0:00 /usr/sbin/opensips -P >> /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 >> >> 14874 ? S 0:00 /usr/sbin/opensips -P >> /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 >> >> 14875 ? S 0:00 /usr/sbin/opensips -P >> /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 >> >> 14876 ? S 0:00 /usr/sbin/opensips -P >> /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 >> >> 14877 ? S 0:00 /usr/sbin/opensips -P >> /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 >> >> >> >> With commended children parameter I get 19 children in the log and 21 in >> reality: >> >> >> >> Feb 23 13:13:24 dus-appsrv01 opensips: DBG:core:daemonize: waiting for >> status code from children >> >> Feb 23 13:13:24 dus-appsrv01 opensips: DBG:core:count_init_children: 19 >> children are going to be inited >> >> >> >> root at srv01:/var/log/opensips# ps ax|grep sips >> >> 14723 ? S 0:00 /usr/sbin/opensips -P >> /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 >> >> 14725 ? S 0:00 /usr/sbin/opensips -P >> /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 >> >> 14726 ? S 0:00 /usr/sbin/opensips -P >> /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 >> >> 14727 ? S 0:00 /usr/sbin/opensips -P >> /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 >> >> 14730 ? S 0:00 /usr/sbin/opensips -P >> /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 >> >> 14731 ? S 0:00 /usr/sbin/opensips -P >> /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 >> >> 14732 ? S 0:00 /usr/sbin/opensips -P >> /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 >> >> 14733 ? S 0:00 /usr/sbin/opensips -P >> /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 >> >> 14734 ? S 0:00 /usr/sbin/opensips -P >> /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 >> >> 14736 ? S 0:00 /usr/sbin/opensips -P >> /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 >> >> 14737 ? S 0:00 /usr/sbin/opensips -P >> /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 >> >> 14738 ? S 0:00 /usr/sbin/opensips -P >> /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 >> >> 14739 ? S 0:00 /usr/sbin/opensips -P >> /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 >> >> 14741 ? S 0:00 /usr/sbin/opensips -P >> /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 >> >> 14742 ? S 0:00 /usr/sbin/opensips -P >> /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 >> >> 14743 ? S 0:00 /usr/sbin/opensips -P >> /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 >> >> 14744 ? S 0:00 /usr/sbin/opensips -P >> /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 >> >> 14745 ? S 0:00 /usr/sbin/opensips -P >> /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 >> >> 14748 ? S 0:00 /usr/sbin/opensips -P >> /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 >> >> 14749 ? S 0:00 /usr/sbin/opensips -P >> /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 >> >> 14750 ? S 0:00 /usr/sbin/opensips -P >> /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 >> >> >> >> root at srv01:/etc/opensips# opensips -V >> >> version: opensips 2.3.3 (x86_64/linux) >> >> flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, >> F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT >> >> ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, >> MAX_URI_SIZE 1024, BUF_SIZE 65535 >> >> poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. >> >> main.c compiled on with gcc 6.3.0 >> >> >> >> Thanks, >> >> Xaled >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -- Callum Guy Head of Information Security X-on -- *0333 332 0000 | www.x-on.co.uk | ** * X-on is a trading name of Storacall Technology Ltd a limited company registered in England and Wales. Registered Office : Avaland House, 110 London Road, Apsley, Hemel Hempstead, Herts, HP3 9SD. Company Registration No. 2578478. The information in this e-mail is confidential and for use by the addressee(s) only. If you are not the intended recipient, please notify X-on immediately on +44(0)333 332 0000 and delete the message from your computer. If you are not a named addressee you must not use, disclose, disseminate, distribute, copy, print or reply to this email. Views or opinions expressed by an individual within this email may not necessarily reflect the views of X-on or its associated companies. Although X-on routinely screens for viruses, addressees should scan this email and any attachments for viruses. X-on makes no representation or warranty as to the absence of viruses in this email or any attachments. -------------- next part -------------- An HTML attachment was scrubbed... URL: From razvan at opensips.org Fri Feb 23 10:40:33 2018 From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=) Date: Fri, 23 Feb 2018 17:40:33 +0200 Subject: [OpenSIPS-Users] ACC module with JSON events In-Reply-To: References: Message-ID: Hi, Samy! Unfortunately there currently is no way (or at least no nice way) to do that. Please open a feature request so we can take this into account[1]. [1] https://github.com/OpenSIPS/opensips/issues Best regards, Răzvan On 02/23/2018 06:13 AM, SamyGo wrote: > Hi, > > I'm using ACC module to trigger accounting events which land up in > event_xmlrpc and ultimately shipped to remote CDR collection server. The > sent data is obviously an XML - now, can I have an event_json like > module so data is sent as JSON or is there any other way I can achieve > the same ? > > Regards > Sammy > > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -- Răzvan Crainea OpenSIPS Core Developer http://www.opensips-solutions.com OpenSIPS Summit 2018 http://www.opensips.org/events/Summit-2018Amsterdam From jarrod at unixc.org Fri Feb 23 10:49:10 2018 From: jarrod at unixc.org (Jarrod Baumann) Date: Fri, 23 Feb 2018 15:49:10 +0000 Subject: [OpenSIPS-Users] ACC module with JSON events In-Reply-To: References: Message-ID: I build the JSON data with cdr and convert to a string for throwing events. Then whatever collects your CDR can serialize it again. This works well and is probably similar to what XML is doing. On Thu, Feb 22, 2018 at 10:14 PM SamyGo wrote: > Hi, > > I'm using ACC module to trigger accounting events which land up in > event_xmlrpc and ultimately shipped to remote CDR collection server. The > sent data is obviously an XML - now, can I have an event_json like module > so data is sent as JSON or is there any other way I can achieve the same ? > > Regards > Sammy > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -------------- next part -------------- An HTML attachment was scrubbed... URL: From govoiper at gmail.com Fri Feb 23 22:26:58 2018 From: govoiper at gmail.com (SamyGo) Date: Fri, 23 Feb 2018 22:26:58 -0500 Subject: [OpenSIPS-Users] ACC module with JSON events In-Reply-To: References: Message-ID: Thanks Razvan, will open a feature request. Jarrod Can you elaborate the steps involved please? So, I don't have anything against XML right now but my XML receiver side is a bit of difficulty in terms of parsing the XML structure (Its done but not in a way I like it) - JSON will make it far easier. Regards, Sammy On Fri, Feb 23, 2018 at 10:49 AM, Jarrod Baumann wrote: > I build the JSON data with cdr and convert to a string for throwing > events. Then whatever collects your CDR can serialize it again. This works > well and is probably similar to what XML is doing. > > On Thu, Feb 22, 2018 at 10:14 PM SamyGo wrote: > >> Hi, >> >> I'm using ACC module to trigger accounting events which land up in >> event_xmlrpc and ultimately shipped to remote CDR collection server. The >> sent data is obviously an XML - now, can I have an event_json like module >> so data is sent as JSON or is there any other way I can achieve the same ? >> >> Regards >> Sammy >> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From jarrod at unixc.org Fri Feb 23 23:01:37 2018 From: jarrod at unixc.org (Jarrod Baumann) Date: Fri, 23 Feb 2018 22:01:37 -0600 Subject: [OpenSIPS-Users] ACC module with JSON events In-Reply-To: References: Message-ID: modparam("acc", "evi_extra", "json=$avp(json_acc)") startup_route { if (!subscribe_event("E_ACC_CDR", "rabbitmq:user:pass at host/queue")) { xlog("L_ERR","Cannot subscribe RabbitMQ to E_ACC_CDR\n"); } } route["set_acc_variables"] { $json(acc) := "{}"; $json(acc/from_user) = $avp(ani); $json(acc/to_user) = $avp(dnis); ... $avp(json_acc) = $json(acc); #deserialize for raising the event } Obviously there are other parameters involved for the acc module but this should provide the gist for using aqmp or any other method you want once you deserialize the json object. You can catch this in a worker and serialize it to json using any language you want. On Friday, February 23, 2018, SamyGo wrote: > Thanks Razvan, will open a feature request. > Jarrod Can you elaborate the steps involved please? > > So, I don't have anything against XML right now but my XML receiver side > is a bit of difficulty in terms of parsing the XML structure (Its done but > not in a way I like it) - JSON will make it far easier. > > Regards, > Sammy > > On Fri, Feb 23, 2018 at 10:49 AM, Jarrod Baumann wrote: > >> I build the JSON data with cdr and convert to a string for throwing >> events. Then whatever collects your CDR can serialize it again. This works >> well and is probably similar to what XML is doing. >> >> On Thu, Feb 22, 2018 at 10:14 PM SamyGo wrote: >> >>> Hi, >>> >>> I'm using ACC module to trigger accounting events which land up in >>> event_xmlrpc and ultimately shipped to remote CDR collection server. The >>> sent data is obviously an XML - now, can I have an event_json like module >>> so data is sent as JSON or is there any other way I can achieve the same ? >>> >>> Regards >>> Sammy >>> >>> >>> _______________________________________________ >>> Users mailing list >>> Users at lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: From govoiper at gmail.com Sun Feb 25 13:14:35 2018 From: govoiper at gmail.com (SamyGo) Date: Sun, 25 Feb 2018 13:14:35 -0500 Subject: [OpenSIPS-Users] ACC module with JSON events In-Reply-To: References: Message-ID: Hi Jarrod, So I see that you're creating your own CDR like variables, exactly what I want to avoid, how about call duration? I want to avoid manual intervention and the way I'm thinking is that it'll be "event_JSONRPC" job to fetch params from E_ACC_CDR and sent a JSON out. Simple. Thanks for your help, much appreciated. Regards, Sammy On Fri, Feb 23, 2018 at 11:01 PM, Jarrod Baumann wrote: > modparam("acc", "evi_extra", "json=$avp(json_acc)") > > startup_route { > if (!subscribe_event("E_ACC_CDR", "rabbitmq:user:pass at host/queue")) { > xlog("L_ERR","Cannot subscribe RabbitMQ to E_ACC_CDR\n"); > } > } > > route["set_acc_variables"] { > $json(acc) := "{}"; > $json(acc/from_user) = $avp(ani); > $json(acc/to_user) = $avp(dnis); > ... > $avp(json_acc) = $json(acc); #deserialize for raising the > event > } > > Obviously there are other parameters involved for the acc module but this > should provide the gist for using aqmp or any other method you want once > you deserialize the json object. You can catch this in a worker and > serialize it to json using any language you want. > > On Friday, February 23, 2018, SamyGo wrote: > >> Thanks Razvan, will open a feature request. >> Jarrod Can you elaborate the steps involved please? >> >> So, I don't have anything against XML right now but my XML receiver side >> is a bit of difficulty in terms of parsing the XML structure (Its done but >> not in a way I like it) - JSON will make it far easier. >> >> Regards, >> Sammy >> >> On Fri, Feb 23, 2018 at 10:49 AM, Jarrod Baumann >> wrote: >> >>> I build the JSON data with cdr and convert to a string for throwing >>> events. Then whatever collects your CDR can serialize it again. This works >>> well and is probably similar to what XML is doing. >>> >>> On Thu, Feb 22, 2018 at 10:14 PM SamyGo wrote: >>> >>>> Hi, >>>> >>>> I'm using ACC module to trigger accounting events which land up in >>>> event_xmlrpc and ultimately shipped to remote CDR collection server. The >>>> sent data is obviously an XML - now, can I have an event_json like module >>>> so data is sent as JSON or is there any other way I can achieve the same ? >>>> >>>> Regards >>>> Sammy >>>> >>>> >>>> _______________________________________________ >>>> Users mailing list >>>> Users at lists.opensips.org >>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>> >>> >>> _______________________________________________ >>> Users mailing list >>> Users at lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >>> >> > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From pasandev at ymail.com Mon Feb 26 01:38:05 2018 From: pasandev at ymail.com (Pasan Meemaduma) Date: Mon, 26 Feb 2018 06:38:05 +0000 (UTC) Subject: [OpenSIPS-Users] dialog replication References: <311827884.3546251.1519627085908.ref@mail.yahoo.com> Message-ID: <311827884.3546251.1519627085908@mail.yahoo.com> Hi Guys, Its me again :). I'm using dialog replication in opensips 2.3.3 and it appears it doesn't work after a node goes down and come back online. The recovered nodedoesn't seems to receiving dialog info via binary interface. I have node1 and node2 with dialog replication on. everything works fine and if I shutdown node1 and bring it back online after a whilenode2 doesn't send new call dialog info via binary interface. I have attach the debug trace from both nodes, if you need anything else let me know. I'm also using an anycast ip as the service ip. on node2 for new call Feb 26 17:28:35 voip2-sip23b /usr/sbin/opensips[4703]: DBG:dialog:build_new_dlg: new dialog 0x7f188afd3bf8 (c=ktQ0Pkdwz50qGYjED6Brpw..,f=sip:XXXXXXXXX at somedomain;transport=UDP,t=sip:+YYYYYYYYYYY at somedomain;transport=UDP,ft=2b161508) on hash 2317 Feb 26 17:28:35 voip2-sip23b /usr/sbin/opensips[4703]: DBG:dialog:init_leg_info: route_set , contact sip:XXXXXXXXX at 192.168.27.11:56419;transport=UDP, cseq 2 and bind_addr udp:10.3.3.1:5060 Feb 26 17:28:35 voip2-sip23b /usr/sbin/opensips[4703]: DBG:dialog:dlg_add_leg_info: set leg 0 for 0x7f188afd3bf8: tag=<2b161508> rcseq=<0> Feb 26 17:28:35 voip2-sip23b /usr/sbin/opensips[4703]: DBG:dialog:link_dlg: ref dlg 0x7f188afd3bf8 with 3 -> 3 in h_entry 0x7f188afc3828 - 2317 Feb 26 17:28:35 voip2-sip23b /usr/sbin/opensips[4703]: DBG:dialog:new_dlg_val: inserting =<45ad4c76-1abe-11e8-9410-831894b67d0c> Feb 26 17:28:35 voip2-sip23b /usr/sbin/opensips[4703]: DBG:dialog:dlg_onreq: t hash_index = 47425, t label = 2069082013 Feb 26 17:28:35 voip2-sip23b /usr/sbin/opensips[4703]: DBG:dialog:dlg_update_contact: Updated dialog 0x7f188afd3bf8 contact to Feb 26 17:28:35 voip2-sip23b /usr/sbin/opensips[4703]: DBG:dialog:unref_dlg: unref dlg 0x7f188afd3bf8 with 1 -> 2 in entry 0x7f188afc3828 Feb 26 17:28:52 voip2-sip23b /usr/sbin/opensips[4702]: DBG:dialog:ref_dlg: ref dlg 0x7f188afd3bf8 with 1 -> 3 Feb 26 17:28:52 voip2-sip23b /usr/sbin/opensips[4702]: DBG:dialog:ref_dlg: ref dlg 0x7f188afd3bf8 with 1 -> 4 Feb 26 17:28:52 voip2-sip23b /usr/sbin/opensips[4702]: DBG:dialog:unref_dlg: unref dlg 0x7f188afd3bf8 with 1 -> 3 in entry 0x7f188afc3828 Feb 26 17:28:56 voip2-sip23b /usr/sbin/opensips[4711]: DBG:dialog:unref_dlg: unref dlg 0x7f188afd3bf8 with 1 -> 2 in entry 0x7f188afc3828 Feb 26 17:29:05 voip2-sip23b /usr/sbin/opensips[4711]: DBG:dialog:next_state_dlg: dialog 0x7f188afd3bf8 changed from state 1 to state 5, due event 4 Feb 26 17:29:05 voip2-sip23b /usr/sbin/opensips[4711]: DBG:dialog:dlg_onreply: dialog 0x7f188afd3bf8 failed (negative reply) Feb 26 17:29:05 voip2-sip23b /usr/sbin/opensips[4711]: DBG:dialog:unref_dlg: unref dlg 0x7f188afd3bf8 with 1 -> 1 in entry 0x7f188afc3828 Feb 26 17:29:11 voip2-sip23b /usr/sbin/opensips[4711]: DBG:dialog:next_state_dlg: dialog 0x7f188afd3bf8 changed from state 5 to state 5, due event 1 Feb 26 17:29:11 voip2-sip23b /usr/sbin/opensips[4711]: DBG:dialog:unref_dlg: unref dlg 0x7f188afd3bf8 with 1 -> 0 in entry 0x7f188afc3828 Feb 26 17:29:11 voip2-sip23b /usr/sbin/opensips[4711]: DBG:dialog:unref_dlg: ref <=0 for dialog 0x7f188afd3bf8 Feb 26 17:29:11 voip2-sip23b /usr/sbin/opensips[4711]: DBG:dialog:destroy_dlg: destroying dialog 0x7f188afd3bf8 Feb 26 17:29:11 voip2-sip23b /usr/sbin/opensips[4711]: DBG:dialog:destroy_dlg: dlg expired or not in list - dlg 0x7f188afd3bf8 [2317:677053409] with clid 'ktQ0Pkdwz50qGYjED6Brpw..' and tags '2b161508' 'NULL' on node1 Feb 26 17:28:41 voip2-sip23a /usr/sbin/opensips[15820]: DBG:dialog:dlg_onroute: route param is 'd09.1e30b582' (len=12) Feb 26 17:28:41 voip2-sip23a /usr/sbin/opensips[15820]: DBG:dialog:lookup_dlg: no dialog id=677053409 found on entry 2317 Feb 26 17:28:41 voip2-sip23a /usr/sbin/opensips[15820]: DBG:dialog:dlg_onroute: unable to find dialog for BYE with route param 'd09.1e30b582' Feb 26 17:28:41 voip2-sip23a /usr/sbin/opensips[15822]: DBG:dialog:dlg_onroute: route param is 'd09.1e30b582' (len=12) Feb 26 17:28:41 voip2-sip23a /usr/sbin/opensips[15822]: DBG:dialog:lookup_dlg: no dialog id=677053409 found on entry 2317 Feb 26 17:28:41 voip2-sip23a /usr/sbin/opensips[15822]: DBG:dialog:dlg_onroute: unable to find dialog for BYE with route param 'd09.1e30b582' Feb 26 17:28:41 voip2-sip23a /usr/sbin/opensips[15822]: DBG:dialog:dlg_onroute: route param is 'd09.1e30b582' (len=12) Feb 26 17:28:41 voip2-sip23a /usr/sbin/opensips[15822]: DBG:dialog:lookup_dlg: no dialog id=677053409 found on entry 2317 Feb 26 17:28:41 voip2-sip23a /usr/sbin/opensips[15822]: DBG:dialog:dlg_onroute: unable to find dialog for BYE with route param 'd09.1e30b582' Feb 26 17:28:42 voip2-sip23a /usr/sbin/opensips[15820]: DBG:dialog:dlg_onroute: route param is 'd09.1e30b582' (len=12) Feb 26 17:28:42 voip2-sip23a /usr/sbin/opensips[15820]: DBG:dialog:lookup_dlg: no dialog id=677053409 found on entry 2317 Feb 26 17:28:42 voip2-sip23a /usr/sbin/opensips[15820]: DBG:dialog:dlg_onroute: unable to find dialog for BYE with route param 'd09.1e30b582' Feb 26 17:28:42 voip2-sip23a /usr/sbin/opensips[15820]: DBG:dialog:dlg_onroute: route param is 'd09.1e30b582' (len=12) Feb 26 17:28:42 voip2-sip23a /usr/sbin/opensips[15820]: DBG:dialog:lookup_dlg: no dialog id=677053409 found on entry 2317 If I purposely take down node1 again and try to make a call I can see that it indicates the failure to replicate dialog info on the cluster as below, Feb 26 17:27:01 voip2-sip23b /usr/sbin/opensips[4701]: ERROR:dialog:replicate_dialog_updated: All destinations in cluster: 1 are down or probing Feb 26 17:27:01 voip2-sip23b /usr/sbin/opensips[4701]: ERROR:dialog:replicate_dialog_updated: Failed to replicate updated dialog -------------- next part -------------- An HTML attachment was scrubbed... URL: From danb.lists at gmail.com Mon Feb 26 02:51:51 2018 From: danb.lists at gmail.com (DanB) Date: Mon, 26 Feb 2018 08:51:51 +0100 Subject: [OpenSIPS-Users] ACC module with JSON events In-Reply-To: References: Message-ID: <1519631511.3267.4.camel@gmail.com> Sammy, Another option on short term until the new feature will be implemented in OpenSIPS would be to use CGRateS as CDR format converter: receive CDRs from "cgrates" module in OpenSIPS and use online export of CGRateS to further export the CDR in the JSON over http (customizable fields). You don't need to configure much on CGRateS side in this case since no billing needs to be involved. DanB From khamlichi.khalil at gmail.com Mon Feb 26 06:13:33 2018 From: khamlichi.khalil at gmail.com (Khalil Khamlichi) Date: Mon, 26 Feb 2018 11:13:33 +0000 Subject: [OpenSIPS-Users] ACC module with JSON events In-Reply-To: <1519631511.3267.4.camel@gmail.com> References: <1519631511.3267.4.camel@gmail.com> Message-ID: Hi Dan, Can you show us how to do that with some step by step tutorial, we had a lot of trouble trying to figure out how to use CGRates, with this functionality of json export, we will at least use some of CGRates functionalities and hopefully slowly get familiar with it. Thanks in advance. On Mon, Feb 26, 2018 at 7:51 AM, DanB wrote: > Sammy, > > Another option on short term until the new feature will be implemented > in OpenSIPS would be to use CGRateS as CDR format converter: receive > CDRs from "cgrates" module in OpenSIPS and use online export of CGRateS > to further export the CDR in the JSON over http (customizable fields). > You don't need to configure much on CGRateS side in this case since no > billing needs to be involved. > > DanB > > > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users From xaled at web.de Mon Feb 26 06:36:14 2018 From: xaled at web.de (xaled) Date: Mon, 26 Feb 2018 12:36:14 +0100 Subject: [OpenSIPS-Users] Wrong number of children In-Reply-To: References: <027501d3aca1$66551340$32ff39c0$@web.de> Message-ID: <010001d3aef6$02f70f90$08e52eb0$@web.de> Hi Callum, that helped thanks. I have 8 TCP receivers and one TCP main process that I did not expected as I only listen on udp. Will have to understand the TCP spawn logic. Greetings, Xaled From: Users [mailto:users-bounces at lists.opensips.org] On Behalf Of Callum Guy Sent: Friday, February 23, 2018 2:42 PM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] Wrong number of children Hi Xaled, If you have the FIFO module loaded you can run "opensipsctl fifo ps" on the instance to see full details of each process. https://www.opensips.org/Documentation/Interface-CoreMI-2-2#toc7 Hope that helps, Callum On Fri, Feb 23, 2018 at 1:35 PM Daniel Zanutti > wrote: Hi You should have 5 per interface + some internal control threads. I'm not sure exactly. Regards On Fri, Feb 23, 2018 at 9:25 AM, xaled > wrote: Hi I have configured 5 children in opensips.cfg, but 16 get logged and 18 initiated what could be the cause of it? I search for word children in opensips directory and it only comes ones in opensips.conf. children=5 Feb 23 13:07:09 srv01 opensips: DBG:core:daemonize: waiting for status code from children Feb 23 13:07:09 srv01 opensips: DBG:core:count_init_children: 16 children are going to be inited root at dus-appsrv01:/var/log/opensips# ps ax|grep sips 14854 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14856 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14857 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14858 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14861 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14862 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14863 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14864 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14865 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14867 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14869 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14870 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14871 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14872 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14874 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14875 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14876 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14877 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 With commended children parameter I get 19 children in the log and 21 in reality: Feb 23 13:13:24 dus-appsrv01 opensips: DBG:core:daemonize: waiting for status code from children Feb 23 13:13:24 dus-appsrv01 opensips: DBG:core:count_init_children: 19 children are going to be inited root at srv01:/var/log/opensips# ps ax|grep sips 14723 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14725 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14726 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14727 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14730 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14731 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14732 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14733 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14734 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14736 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14737 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14738 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14739 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14741 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14742 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14743 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14744 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14745 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14748 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14749 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14750 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 root at srv01:/etc/opensips# opensips -V version: opensips 2.3.3 (x86_64/linux) flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535 poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. main.c compiled on with gcc 6.3.0 Thanks, Xaled _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Callum Guy Head of Information Security X-on 0333 332 0000 | www.x-on.co.uk | X-on is a trading name of Storacall Technology Ltd a limited company registered in England and Wales. 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URL: From vladp at opensips.org Mon Feb 26 07:31:36 2018 From: vladp at opensips.org (Vlad Patrascu) Date: Mon, 26 Feb 2018 14:31:36 +0200 Subject: [OpenSIPS-Users] dialog replication In-Reply-To: <311827884.3546251.1519627085908@mail.yahoo.com> References: <311827884.3546251.1519627085908.ref@mail.yahoo.com> <311827884.3546251.1519627085908@mail.yahoo.com> Message-ID: <704df948-6873-5fde-fe25-193f1cf2a420@opensips.org> Hi, Can you send the full logs for both nodes from the time that node1 restarts onwards? Also, what is the output of 'clusterer_list' mi command on the instances? Regards, Vlad Patrascu OpenSIPS Developer http://www.opensips-solutions.com On 26.02.2018 08:38, Pasan Meemaduma via Users wrote: > Hi Guys, > > Its me again :). I'm using dialog replication in opensips 2.3.3 and it > appears it doesn't work after a node goes down and come back online. > The recovered node > doesn't seems to receiving dialog info via binary interface. > > I have node1 and node2 with dialog replication on. everything works > fine and if I shutdown node1 and bring it back online after a while > node2 doesn't send new call dialog info via binary interface. > > I have attach the debug trace from both nodes, if you need anything > else let me know. I'm also using an anycast ip as the service ip. > > > on node2 for new call > > Feb 26 17:28:35 voip2-sip23b /usr/sbin/opensips[4703]: > DBG:dialog:build_new_dlg: new dialog 0x7f188afd3bf8 > (c=ktQ0Pkdwz50qGYjED6Brpw..,f=sip:XXXXXXXXX at somedomain;transport=UDP,t=sip:+YYYYYYYYYYY at somedomain;transport=UDP,ft=2b161508) > on hash 2317 > Feb 26 17:28:35 voip2-sip23b /usr/sbin/opensips[4703]: > DBG:dialog:init_leg_info: route_set , contact > sip:XXXXXXXXX at 192.168.27.11:56419;transport=UDP, cseq 2 and bind_addr > udp:10.3.3.1:5060 > Feb 26 17:28:35 voip2-sip23b /usr/sbin/opensips[4703]: > DBG:dialog:dlg_add_leg_info: set leg 0 for 0x7f188afd3bf8: > tag=<2b161508> rcseq=<0> > Feb 26 17:28:35 voip2-sip23b /usr/sbin/opensips[4703]: > DBG:dialog:link_dlg: ref dlg 0x7f188afd3bf8 with 3 -> 3 in h_entry > 0x7f188afc3828 - 2317 > Feb 26 17:28:35 voip2-sip23b /usr/sbin/opensips[4703]: > DBG:dialog:new_dlg_val: inserting > =<45ad4c76-1abe-11e8-9410-831894b67d0c> > Feb 26 17:28:35 voip2-sip23b /usr/sbin/opensips[4703]: > DBG:dialog:dlg_onreq: t hash_index = 47425, t label = 2069082013 > Feb 26 17:28:35 voip2-sip23b /usr/sbin/opensips[4703]: > DBG:dialog:dlg_update_contact: Updated dialog 0x7f188afd3bf8 contact > to > Feb 26 17:28:35 voip2-sip23b /usr/sbin/opensips[4703]: > DBG:dialog:unref_dlg: unref dlg 0x7f188afd3bf8 with 1 -> 2 in entry > 0x7f188afc3828 > Feb 26 17:28:52 voip2-sip23b /usr/sbin/opensips[4702]: > DBG:dialog:ref_dlg: ref dlg 0x7f188afd3bf8 with 1 -> 3 > Feb 26 17:28:52 voip2-sip23b /usr/sbin/opensips[4702]: > DBG:dialog:ref_dlg: ref dlg 0x7f188afd3bf8 with 1 -> 4 > Feb 26 17:28:52 voip2-sip23b /usr/sbin/opensips[4702]: > DBG:dialog:unref_dlg: unref dlg 0x7f188afd3bf8 with 1 -> 3 in entry > 0x7f188afc3828 > Feb 26 17:28:56 voip2-sip23b /usr/sbin/opensips[4711]: > DBG:dialog:unref_dlg: unref dlg 0x7f188afd3bf8 with 1 -> 2 in entry > 0x7f188afc3828 > Feb 26 17:29:05 voip2-sip23b /usr/sbin/opensips[4711]: > DBG:dialog:next_state_dlg: dialog 0x7f188afd3bf8 changed from state 1 > to state 5, due event 4 > Feb 26 17:29:05 voip2-sip23b /usr/sbin/opensips[4711]: > DBG:dialog:dlg_onreply: dialog 0x7f188afd3bf8 failed (negative reply) > Feb 26 17:29:05 voip2-sip23b /usr/sbin/opensips[4711]: > DBG:dialog:unref_dlg: unref dlg 0x7f188afd3bf8 with 1 -> 1 in entry > 0x7f188afc3828 > Feb 26 17:29:11 voip2-sip23b /usr/sbin/opensips[4711]: > DBG:dialog:next_state_dlg: dialog 0x7f188afd3bf8 changed from state 5 > to state 5, due event 1 > Feb 26 17:29:11 voip2-sip23b /usr/sbin/opensips[4711]: > DBG:dialog:unref_dlg: unref dlg 0x7f188afd3bf8 with 1 -> 0 in entry > 0x7f188afc3828 > Feb 26 17:29:11 voip2-sip23b /usr/sbin/opensips[4711]: > DBG:dialog:unref_dlg: ref <=0 for dialog 0x7f188afd3bf8 > Feb 26 17:29:11 voip2-sip23b /usr/sbin/opensips[4711]: > DBG:dialog:destroy_dlg: destroying dialog 0x7f188afd3bf8 > Feb 26 17:29:11 voip2-sip23b /usr/sbin/opensips[4711]: > DBG:dialog:destroy_dlg: dlg expired or not in list - dlg > 0x7f188afd3bf8 [2317:677053409] with clid 'ktQ0Pkdwz50qGYjED6Brpw..' > and tags '2b161508' 'NULL' > > > on node1 > > Feb 26 17:28:41 voip2-sip23a /usr/sbin/opensips[15820]: > DBG:dialog:dlg_onroute: route param is 'd09.1e30b582' (len=12) > Feb 26 17:28:41 voip2-sip23a /usr/sbin/opensips[15820]: > DBG:dialog:lookup_dlg: no dialog id=677053409 found on entry 2317 > Feb 26 17:28:41 voip2-sip23a /usr/sbin/opensips[15820]: > DBG:dialog:dlg_onroute: unable to find dialog for BYE with route param > 'd09.1e30b582' > Feb 26 17:28:41 voip2-sip23a /usr/sbin/opensips[15822]: > DBG:dialog:dlg_onroute: route param is 'd09.1e30b582' (len=12) > Feb 26 17:28:41 voip2-sip23a /usr/sbin/opensips[15822]: > DBG:dialog:lookup_dlg: no dialog id=677053409 found on entry 2317 > Feb 26 17:28:41 voip2-sip23a /usr/sbin/opensips[15822]: > DBG:dialog:dlg_onroute: unable to find dialog for BYE with route param > 'd09.1e30b582' > Feb 26 17:28:41 voip2-sip23a /usr/sbin/opensips[15822]: > DBG:dialog:dlg_onroute: route param is 'd09.1e30b582' (len=12) > Feb 26 17:28:41 voip2-sip23a /usr/sbin/opensips[15822]: > DBG:dialog:lookup_dlg: no dialog id=677053409 found on entry 2317 > Feb 26 17:28:41 voip2-sip23a /usr/sbin/opensips[15822]: > DBG:dialog:dlg_onroute: unable to find dialog for BYE with route param > 'd09.1e30b582' > Feb 26 17:28:42 voip2-sip23a /usr/sbin/opensips[15820]: > DBG:dialog:dlg_onroute: route param is 'd09.1e30b582' (len=12) > Feb 26 17:28:42 voip2-sip23a /usr/sbin/opensips[15820]: > DBG:dialog:lookup_dlg: no dialog id=677053409 found on entry 2317 > Feb 26 17:28:42 voip2-sip23a /usr/sbin/opensips[15820]: > DBG:dialog:dlg_onroute: unable to find dialog for BYE with route param > 'd09.1e30b582' > Feb 26 17:28:42 voip2-sip23a /usr/sbin/opensips[15820]: > DBG:dialog:dlg_onroute: route param is 'd09.1e30b582' (len=12) > Feb 26 17:28:42 voip2-sip23a /usr/sbin/opensips[15820]: > DBG:dialog:lookup_dlg: no dialog id=677053409 found on entry 2317 > > > If I purposely take down node1 again and try to make a call I can see > that it indicates the failure to replicate dialog info on the cluster > as below, > > Feb 26 17:27:01 voip2-sip23b /usr/sbin/opensips[4701]: > ERROR:dialog:replicate_dialog_updated: All destinations in cluster: 1 > are down or probing > Feb 26 17:27:01 voip2-sip23b /usr/sbin/opensips[4701]: > ERROR:dialog:replicate_dialog_updated: Failed to replicate updated dialog > > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From Ben.Newlin at genesys.com Mon Feb 26 08:49:57 2018 From: Ben.Newlin at genesys.com (Ben Newlin) Date: Mon, 26 Feb 2018 13:49:57 +0000 Subject: [OpenSIPS-Users] OpenSIPS crashing on 200 OK In-Reply-To: <755F831E-90E4-4105-80C5-42C7A2E2050A@genesys.com> References: <530E1611-2A4D-424F-938D-2E409E1A78A2@genesys.com> <7cf6bb3d-967a-b8e2-d4b3-8b42c8dbf0b8@opensips.org> <755F831E-90E4-4105-80C5-42C7A2E2050A@genesys.com> Message-ID: <266D520B-6381-4AB8-8CFD-EF98AD78914B@genesys.com> Any progress on this? I’m happy to provide more information if needed. Ben Newlin From: Ben Newlin Date: Thursday, February 22, 2018 at 12:33 PM To: Bogdan-Andrei Iancu , OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] OpenSIPS crashing on 200 OK Bogdan, Here is the backtrace: https://pastebin.com/6tZhrWek I am not using TCP, BIN, or HEP in this configuration. I am not using anything, to my knowledge, which should be causing TCP worker threads to be spawned. Ben Newlin From: Bogdan-Andrei Iancu Date: Thursday, February 22, 2018 at 5:26 AM To: OpenSIPS users mailling list , Ben Newlin Subject: Re: [OpenSIPS-Users] OpenSIPS crashing on 200 OK Hi Ben, Do you have a full backtrace extracted from the core ? In regards to the TCP processes, the TCP layer will automatically enabled itself depending on what protocols (tcp-based) you are using in script. Maybe you do not use SIP with TCP, but do you use BIN or HEP ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Summit 2018 http://www.opensips.org/events/Summit-2018Amsterdam On 02/22/2018 12:38 AM, Ben Newlin wrote: Hi, I am currently upgrading to 2.3.3 and am running off the HEAD of the 2.3 branch. My OpenSIPS is consistently crashing when trying to process the ACK to an initial INVITE request. I am testing with SIPp using a simple single call scenario. I see the following in the logs: Feb 21 21:32:27 [383] DBG:core:io_wait_loop_epoll: [TCP_main] EPOLLHUP on IN ->connection closed by the remote peer! Feb 21 21:32:27 [383] CRITICAL:core:receive_fd: EOF on 33 Feb 21 21:32:27 [383] DBG:core:handle_worker: dead child 9, pid 370 (shutting down?) Feb 21 21:32:27 [383] DBG:core:io_watch_del: [TCP_main] io_watch_del op on index 8 33 (0x8d4e80, 33, 8, 0x0,0x1) fd_no=29 called Feb 21 21:32:27 [361] DBG:core:handle_sigs: status = 139 Feb 21 21:32:27 [361] INFO:core:handle_sigs: child process 370 exited by a signal 11 Feb 21 21:32:27 [361] INFO:core:handle_sigs: core was generated Feb 21 21:32:27 [361] INFO:core:handle_sigs: terminating due to SIGCHLD I also notice that OpenSIPS is starting up a number of TCP processes even though I do not use TCP in my script. I do not even have the proto_tcp module loaded. I used to use the “disable_tcp” global parameter in 1.11, but that has been removed in 2.3. Feb 21 22:03:56 [361] DBG:core:internal_fork: forking new process "SIP receiver TCP" Feb 21 22:03:56 [361] DBG:core:internal_fork: forking new process "SIP receiver TCP" Feb 21 22:03:56 [361] DBG:core:internal_fork: forking new process "SIP receiver TCP" Feb 21 22:03:56 [361] DBG:core:internal_fork: forking new process "SIP receiver TCP" Feb 21 22:03:56 [361] DBG:core:internal_fork: forking new process "SIP receiver TCP" Feb 21 22:03:56 [375] DBG:core:io_watch_add: [TCP_worker] io_watch_add op (3 on 53) (0x8d4e80, 3, 1, (nil),1), fd_no=0/209715 Feb 21 22:03:56 [361] DBG:core:internal_fork: forking new process "SIP receiver TCP" Feb 21 22:03:56 [361] DBG:core:internal_fork: forking new process "SIP receiver TCP" Feb 21 22:03:56 [361] DBG:core:internal_fork: forking new process "SIP receiver TCP" Initialization logs: https://pastebin.com/nsJQGkdy Call logs: https://pastebin.com/vY5crK8U I also have the core dump if any more information is needed from that. Thanks, Ben Newlin _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Mon Feb 26 09:15:26 2018 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Mon, 26 Feb 2018 16:15:26 +0200 Subject: [OpenSIPS-Users] OpenSIPS Summit 2018 - Speakers and Birds Message-ID: <5e45325e-68bf-cf22-c946-db12f1eae852@opensips.org> OpenSIPS Summit 2018 May 1-4, 2018 Amsterdam, The Netherlands *The Speakers List is almost done!* During the last 2 months we worked hard to bring you valuables speakers with interesting topics from different areas of VoIP & RTC, topics covering or related to OpenSIPS, topics on Open Source (or not) projects related to VoIP & RTC. Even if the process is still ongoing, we are proud to present you our awesome list of speakers and topics: *http://www.opensips.org/events/Summit-2018Amsterdam/#mu-schedule** * ** *Be wise, be an early bird!* Is the speaker list interesting for you? Then reserve your seat with the OpenSIPS Summit in the next 15 days with a 50% discount for the Conference Days. *Register now * with the Early Birds token and secure your participation: *5b48a939-61d9-4bc1-b8b3-008aba92c1a4* While registering, remember that the Summit is more than conference. It is also about Interactive Demos, about OpenSIPS and FreeSWITCH trainings , about Design Clinics and many more.. and available seats / slots are limited ! *Event Location* * * *Radisson Blu** **Rusland 17, 1012CK Amsterdam, The Netherlands* Meet us again at our familiar Venue, with even more space and comfort than ever! This year the OpenSIPS Summit expands in size and will accommodate more participants and speakers. ** -- Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Mon Feb 26 10:39:38 2018 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Mon, 26 Feb 2018 17:39:38 +0200 Subject: [OpenSIPS-Users] OpenSIPS crashing on 200 OK In-Reply-To: <755F831E-90E4-4105-80C5-42C7A2E2050A@genesys.com> References: <530E1611-2A4D-424F-938D-2E409E1A78A2@genesys.com> <7cf6bb3d-967a-b8e2-d4b3-8b42c8dbf0b8@opensips.org> <755F831E-90E4-4105-80C5-42C7A2E2050A@genesys.com> Message-ID: Hi Ben, The backtrace points to the script interpreter :D....Please do the followings (assuming you are a bit faimilar with GDB:) ) : in frame 0, please do "print *s1" and in frame 5 do "print *a" Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Summit 2018 http://www.opensips.org/events/Summit-2018Amsterdam On 02/22/2018 07:33 PM, Ben Newlin wrote: > > Bogdan, > > Here is the backtrace: https://pastebin.com/6tZhrWek > > > I am not using TCP, BIN, or HEP in this configuration. I am not using > anything, to my knowledge, which should be causing TCP worker threads > to be spawned. > > Ben Newlin > > *From: *Bogdan-Andrei Iancu > *Date: *Thursday, February 22, 2018 at 5:26 AM > *To: *OpenSIPS users mailling list , Ben > Newlin > *Subject: *Re: [OpenSIPS-Users] OpenSIPS crashing on 200 OK > > Hi Ben, > > Do you have a full backtrace extracted from the core ? > > In regards to the TCP processes, the TCP layer will automatically > enabled itself depending on what protocols (tcp-based) you are using > in script. Maybe you do not use SIP with TCP, but do you use BIN or HEP ? > > Regards, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > OpenSIPS Summit 2018 > http://www.opensips.org/events/Summit-2018Amsterdam > > On 02/22/2018 12:38 AM, Ben Newlin wrote: > > Hi, > > I am currently upgrading to 2.3.3 and am running off the HEAD of > the 2.3 branch. My OpenSIPS is consistently crashing when trying > to process the ACK to an initial INVITE request. I am testing with > SIPp using a simple single call scenario. I see the following in > the logs: > > Feb 21 21:32:27 [383] DBG:core:io_wait_loop_epoll: [TCP_main] > EPOLLHUP on IN ->connection closed by the remote peer! > > Feb 21 21:32:27 [383] CRITICAL:core:receive_fd: EOF on 33 > > Feb 21 21:32:27 [383] DBG:core:handle_worker: dead child 9, pid > 370 (shutting down?) > > Feb 21 21:32:27 [383] DBG:core:io_watch_del: [TCP_main] > io_watch_del op on index 8 33 (0x8d4e80, 33, 8, 0x0,0x1) fd_no=29 > called > > Feb 21 21:32:27 [361] DBG:core:handle_sigs: status = 139 > > Feb 21 21:32:27 [361] INFO:core:handle_sigs: child process 370 > exited by a signal 11 > > Feb 21 21:32:27 [361] INFO:core:handle_sigs: core was generated > > Feb 21 21:32:27 [361] INFO:core:handle_sigs: terminating due to > SIGCHLD > > I also notice that OpenSIPS is starting up a number of TCP > processes even though I do not use TCP in my script. I do not even > have the proto_tcp module loaded. I used to use the “disable_tcp” > global parameter in 1.11, but that has been removed in 2.3. > > Feb 21 22:03:56 [361] DBG:core:internal_fork: forking new process > "SIP receiver TCP" > > Feb 21 22:03:56 [361] DBG:core:internal_fork: forking new process > "SIP receiver TCP" > > Feb 21 22:03:56 [361] DBG:core:internal_fork: forking new process > "SIP receiver TCP" > > Feb 21 22:03:56 [361] DBG:core:internal_fork: forking new process > "SIP receiver TCP" > > Feb 21 22:03:56 [361] DBG:core:internal_fork: forking new process > "SIP receiver TCP" > > Feb 21 22:03:56 [375] DBG:core:io_watch_add: [TCP_worker] > io_watch_add op (3 on 53) (0x8d4e80, 3, 1, (nil),1), fd_no=0/209715 > > Feb 21 22:03:56 [361] DBG:core:internal_fork: forking new process > "SIP receiver TCP" > > Feb 21 22:03:56 [361] DBG:core:internal_fork: forking new process > "SIP receiver TCP" > > Feb 21 22:03:56 [361] DBG:core:internal_fork: forking new process > "SIP receiver TCP" > > Initialization logs: https://pastebin.com/nsJQGkdy > > Call logs: https://pastebin.com/vY5crK8U > > I also have the core dump if any more information is needed from that. > > Thanks, > > Ben Newlin > > > > > _______________________________________________ > > Users mailing list > > Users at lists.opensips.org > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From Ben.Newlin at genesys.com Mon Feb 26 10:43:10 2018 From: Ben.Newlin at genesys.com (Ben Newlin) Date: Mon, 26 Feb 2018 15:43:10 +0000 Subject: [OpenSIPS-Users] OpenSIPS crashing on 200 OK In-Reply-To: References: <530E1611-2A4D-424F-938D-2E409E1A78A2@genesys.com> <7cf6bb3d-967a-b8e2-d4b3-8b42c8dbf0b8@opensips.org> <755F831E-90E4-4105-80C5-42C7A2E2050A@genesys.com> Message-ID: <6D9EBB18-914E-49ED-988F-7D582007B88E@genesys.com> Bogdan, Unfortunately I am not at all familiar with gdb. How do I change frames? Ben Newlin From: Bogdan-Andrei Iancu Date: Monday, February 26, 2018 at 10:40 AM To: Ben Newlin , OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] OpenSIPS crashing on 200 OK Hi Ben, The backtrace points to the script interpreter :D....Please do the followings (assuming you are a bit faimilar with GDB :) ) : in frame 0, please do "print *s1" and in frame 5 do "print *a" Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Summit 2018 http://www.opensips.org/events/Summit-2018Amsterdam On 02/22/2018 07:33 PM, Ben Newlin wrote: Bogdan, Here is the backtrace: https://pastebin.com/6tZhrWek I am not using TCP, BIN, or HEP in this configuration. I am not using anything, to my knowledge, which should be causing TCP worker threads to be spawned. Ben Newlin From: Bogdan-Andrei Iancu Date: Thursday, February 22, 2018 at 5:26 AM To: OpenSIPS users mailling list , Ben Newlin Subject: Re: [OpenSIPS-Users] OpenSIPS crashing on 200 OK Hi Ben, Do you have a full backtrace extracted from the core ? In regards to the TCP processes, the TCP layer will automatically enabled itself depending on what protocols (tcp-based) you are using in script. Maybe you do not use SIP with TCP, but do you use BIN or HEP ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Summit 2018 http://www.opensips.org/events/Summit-2018Amsterdam On 02/22/2018 12:38 AM, Ben Newlin wrote: Hi, I am currently upgrading to 2.3.3 and am running off the HEAD of the 2.3 branch. My OpenSIPS is consistently crashing when trying to process the ACK to an initial INVITE request. I am testing with SIPp using a simple single call scenario. I see the following in the logs: Feb 21 21:32:27 [383] DBG:core:io_wait_loop_epoll: [TCP_main] EPOLLHUP on IN ->connection closed by the remote peer! Feb 21 21:32:27 [383] CRITICAL:core:receive_fd: EOF on 33 Feb 21 21:32:27 [383] DBG:core:handle_worker: dead child 9, pid 370 (shutting down?) Feb 21 21:32:27 [383] DBG:core:io_watch_del: [TCP_main] io_watch_del op on index 8 33 (0x8d4e80, 33, 8, 0x0,0x1) fd_no=29 called Feb 21 21:32:27 [361] DBG:core:handle_sigs: status = 139 Feb 21 21:32:27 [361] INFO:core:handle_sigs: child process 370 exited by a signal 11 Feb 21 21:32:27 [361] INFO:core:handle_sigs: core was generated Feb 21 21:32:27 [361] INFO:core:handle_sigs: terminating due to SIGCHLD I also notice that OpenSIPS is starting up a number of TCP processes even though I do not use TCP in my script. I do not even have the proto_tcp module loaded. I used to use the “disable_tcp” global parameter in 1.11, but that has been removed in 2.3. Feb 21 22:03:56 [361] DBG:core:internal_fork: forking new process "SIP receiver TCP" Feb 21 22:03:56 [361] DBG:core:internal_fork: forking new process "SIP receiver TCP" Feb 21 22:03:56 [361] DBG:core:internal_fork: forking new process "SIP receiver TCP" Feb 21 22:03:56 [361] DBG:core:internal_fork: forking new process "SIP receiver TCP" Feb 21 22:03:56 [361] DBG:core:internal_fork: forking new process "SIP receiver TCP" Feb 21 22:03:56 [375] DBG:core:io_watch_add: [TCP_worker] io_watch_add op (3 on 53) (0x8d4e80, 3, 1, (nil),1), fd_no=0/209715 Feb 21 22:03:56 [361] DBG:core:internal_fork: forking new process "SIP receiver TCP" Feb 21 22:03:56 [361] DBG:core:internal_fork: forking new process "SIP receiver TCP" Feb 21 22:03:56 [361] DBG:core:internal_fork: forking new process "SIP receiver TCP" Initialization logs: https://pastebin.com/nsJQGkdy Call logs: https://pastebin.com/vY5crK8U I also have the core dump if any more information is needed from that. Thanks, Ben Newlin _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From cadre5000 at gmail.com Thu Feb 22 15:35:55 2018 From: cadre5000 at gmail.com (Tiaki Cadre) Date: Fri, 23 Feb 2018 09:35:55 +1300 Subject: [OpenSIPS-Users] Parallel forking - use mid_registrar module question Message-ID: I have been using Opensips now for a few years, and in the last month or so I needed to use Opensips for another application. We need to put Opensips in front of our Asterisk servers to handle better the register, notify, invite and subscribe messages (basically the SIP messaging). My goal initially, is to have Opensips as a transparent SIP proxy for this. I have already got this working mostly using the mid_registrar module - just been looking at it further. I can send and receive calls from a device that is using the proxy and the SIP is sent to the main registrar (the asterisk). The audio works fine since at this stage I am using the RTP on the asterisk. The issue I need to solve is multiple registered devices being able to be called when a user on asterisk makes a call. Outbound calls from the extension via the proxy work with no issue. Also the notify (BLF) works ok. I am working on how to get the branch/forking working correctly and the single call to ring all the contacts in that have "registered" via the mid_registrar. I have been trying the AOR for in the mid_registrar request, and then using mid_registrar_lookup but not having much success. I am using opensips 2.3.3. After some reading yesterday, there was suggestion to move to opensips 2.4 branch I thought I would post here before proceeding too much further in case I am missing something else. What is working: - I can make a call from a phone that is registered to the main registrar fyi the proxy to an external number (handled by the main registrar) - I can make a call between phones that are registered to the main registrar fyi the proxy or that are registered directly to the main registrar - I can see the BLF/Notify and subscribe messages working from the main registrar correctly Questions: - Some of the registered devices may be behind NAT and some may be on Public IPs. How do I best cater for this? What I need to work: - When a phone, or incoming call from the main registrar to the proxy occurs, I need the proxy to ring all phones that are registered through it. Here are some technical details: version: opensips 2.3.3 (x86_64/linux) flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535 poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. main.c compiled on 00:04:06 Jan 19 2018 with gcc 4.4.7 >From my opensips.cfg I have some other modules loaded -but here is the mid_registrar section: loadmodule "mid_registrar.so" modparam("mid_registrar", "mode", 1) /* 0 = mirror / 1 = ct / 2 = AoR */ modparam("mid_registrar", "outgoing_expires", 900) modparam("mid_registrar", "insertion_mode", 1) /* 0 = contact; 1 = path */ modparam("mid_registrar", "received_avp", "$avp(rcv)") modparam("mid_registrar", "contact_match_param", "regid") modparam("mid_registrar", "max_contacts", 4) modparam("mid_registrar", "tcp_persistent_flag", 10) modparam("mid_registrar", "outgoing_expires", 900) Snippet from my routing section: if (is_method("REGISTER")) { #fix_nated_register(); xlog("L_INFO","Saving location to mid_registrar"); mid_registrar_save("location"); switch ($retcode) { case 1: xlog("forwarding REGISTER to main registrar ($$ci=$ci)\n"); $ru = "sip:x.x.x.x:5060"; t_relay(); break; case 2: xlog("absorbing REGISTER! ($$ci=$ci)\n"); break; default: xlog("failed to save registration! ($$ci=$ci)\n"); } exit; } remove_hf("Route"); # record routing if (is_method("INVITE|REFER|SUBSCRIBE|NOTIFY")) record_route(); # account only INVITEs if (is_method("INVITE")) { do_accounting("log"); xlog("L_INFO", "[LOG] INVITE PACKET [rm:$rm] [fu:$fu] [ou:$ou] [ru:$ru] [si:$si]"); } # initial requests from main registrar, need to look them up! if (is_method("INVITE|MESSAGE") && $si == "" && $sp == 5060) { xlog("looking up $ru!\n"); fix_nated_register(); if (!mid_registrar_lookup("location")) { xlog("looked up main registrar - not found"); t_reply("404", "Not Found"); exit; } xlog("L_INFO","MYLOG INVITE mid_registrar after looking up $ru $ci"); t_relay(); exit; } route[relay] { # for INVITEs enable some additional helper routes if (is_method("INVITE|NOTIFY|SUBSCRIBE|PUBLISH")) { fix_nated_contact(); t_on_branch("per_branch_ops"); t_on_reply("handle_nat"); t_on_failure("missed_call"); } if (!t_relay()) { send_reply("500","Internal Error"); }; exit; } branch_route[per_branch_ops] { xlog("new branch at $ru\n"); } onreply_route[handle_nat] { if (nat_uac_test("1")) { xlog("FIXING THE NAT!"); fix_nated_contact(); } xlog("incoming reply\n"); } failure_route[missed_call] { if (t_was_cancelled()) { exit; } # uncomment the following lines if you want to block client # redirect based on 3xx replies. ##if (t_check_status("3[0-9][0-9]")) { ##t_reply("404","Not found"); ## exit; ##} } -------------- next part -------------- An HTML attachment was scrubbed... URL: From brian.southworth at clocom.uk Fri Feb 23 08:27:33 2018 From: brian.southworth at clocom.uk (=?UTF-8?Q?Brian_Southworth?=) Date: Fri, 23 Feb 2018 13:27:33 +0000 Subject: [OpenSIPS-Users] On hold In-Reply-To: References: <01020161c1ef8de9-1fa3116b-b84a-4c0c-a3a9-f3419d72813d-000000@eu-west-1.amazonses.com> Message-ID: <01020161c2d8f516-0b3641f3-8c66-4e63-aff5-f57150cb24fe-000000@eu-west-1.amazonses.com> Hi Ovidiu,   I tried this in a if statement it works but same thing happens I go to take the call off hold and it just drops the call.   Regards,   Brian Southworth Communications Developer 111 Wilmslow Road Handforth Wilmslow SK9 3ER   T: 03333 446677 W: www.clocom.uk           Like us on Facebook Follow us on Twitter           Clocom is a green company. Think, do you need to print this email?   This message contains confidential information and is intended only for the individual named. If you are not the named addressee you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately by e-mail if you have received this e-mail by mistake and delete this e-mail from your system. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version. Registered in England & Wales: 07081192. Registered name and address: Clocom UK Ltd, 111 Wilmslow Road, Handforth, Cheshire, SK9 3ER  www.clocom.uk   From: Users [mailto:users-bounces at lists.opensips.org] On Behalf Of Ovidiu Sas Sent: 23 February 2018 13:10 To: users at lists.opensips.org users Subject: Re: [OpenSIPS-Users] On hold   Hello Brian,   You can detect hold using is_audio_on_hold(): http://www.opensips.org/html/docs/modules/2.3.x/sipmsgops#idp5575408   Try to use it and see if this will solve your issue.   Regards, Ovidiu Sas   On Feb 23, 2018 4:13 AM, "Brian Southworth" > wrote: Hi All,   I am trying to get opensips to forward the on hold request to asterisk I have done this using an if statement for SDP =~ “sendonly”   How ever when I go to take them off hold the call is dropped, is there something special I need to do ?   I have tried and if statement for is_method(“invite”) && $rb =~ “sendrecv” ß this loops until opensips give a 500 internal error And I have also tried is_method(“invite”) && $rb =~ “inactive” ß this doesn’t seem to work at all   The initial hold works perfect puts them on hold just cant take them off or transfer the call (asterisk will handle all the B2B stuff)   Any help would be appreciated.   Regards,   Brian Southworth Communications Developer 111 Wilmslow Road Handforth Wilmslow SK9 3ER   T: 03333 446677 W: www.clocom.uk           Like us on Facebook Follow us on Twitter           Clocom is a green company. Think, do you need to print this email?   This message contains confidential information and is intended only for the individual named. If you are not the named addressee you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately by e-mail if you have received this e-mail by mistake and delete this e-mail from your system. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version. Registered in England & Wales: 07081192. Registered name and address: Clocom UK Ltd, 111 Wilmslow Road, Handforth, Cheshire, SK9 3ER  www.clocom.uk   _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users   _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/png Size: 9352 bytes Desc: not available URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/png Size: 2708 bytes Desc: not available URL: -------------- next part -------------- A non-text attachment was scrubbed... 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Name: not available Type: image/png Size: 2999 bytes Desc: not available URL: From brian.southworth at clocom.uk Fri Feb 23 08:28:38 2018 From: brian.southworth at clocom.uk (=?UTF-8?Q?Brian_Southworth?=) Date: Fri, 23 Feb 2018 13:28:38 +0000 Subject: [OpenSIPS-Users] On hold In-Reply-To: References: <01020161c1ef8de9-1fa3116b-b84a-4c0c-a3a9-f3419d72813d-000000@eu-west-1.amazonses.com> Message-ID: <01020161c2d9f00c-aa3eda14-ed9e-4eb0-ade8-20abbb41560c-000000@eu-west-1.amazonses.com> Hi Ovidiu,   I tried this in a if statement it works but same thing happens I go to take the call off hold and it just drops the call.   I am also using v2.2 but I did check those docs too.   Regards,   Brian Southworth   From: Users [mailto:users-bounces at lists.opensips.org] On Behalf Of Ovidiu Sas Sent: 23 February 2018 13:10 To: users at lists.opensips.org users Subject: Re: [OpenSIPS-Users] On hold   Hello Brian,   You can detect hold using is_audio_on_hold(): http://www.opensips.org/html/docs/modules/2.3.x/sipmsgops#idp5575408   Try to use it and see if this will solve your issue.   Regards, Ovidiu Sas   On Feb 23, 2018 4:13 AM, "Brian Southworth" > wrote: Hi All,   I am trying to get opensips to forward the on hold request to asterisk I have done this using an if statement for SDP =~ “sendonly”   How ever when I go to take them off hold the call is dropped, is there something special I need to do ?   I have tried and if statement for is_method(“invite”) && $rb =~ “sendrecv” ß this loops until opensips give a 500 internal error And I have also tried is_method(“invite”) && $rb =~ “inactive” ß this doesn’t seem to work at all   The initial hold works perfect puts them on hold just cant take them off or transfer the call (asterisk will handle all the B2B stuff)   Any help would be appreciated.   Regards,   Brian Southworth Communications Developer 111 Wilmslow Road Handforth Wilmslow SK9 3ER   T: 03333 446677 W: www.clocom.uk           Like us on Facebook Follow us on Twitter           Clocom is a green company. Think, do you need to print this email?   This message contains confidential information and is intended only for the individual named. If you are not the named addressee you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately by e-mail if you have received this e-mail by mistake and delete this e-mail from your system. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version. Registered in England & Wales: 07081192. Registered name and address: Clocom UK Ltd, 111 Wilmslow Road, Handforth, Cheshire, SK9 3ER  www.clocom.uk   _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users   _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/png Size: 9352 bytes Desc: not available URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/png Size: 2708 bytes Desc: not available URL: -------------- next part -------------- A non-text attachment was scrubbed... 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Name: not available Type: image/png Size: 2999 bytes Desc: not available URL: From brian.southworth at clocom.uk Fri Feb 23 08:41:52 2018 From: brian.southworth at clocom.uk (=?UTF-8?Q?Brian_Southworth?=) Date: Fri, 23 Feb 2018 13:41:52 +0000 Subject: [OpenSIPS-Users] On hold In-Reply-To: References: <01020161c1ef8de9-1fa3116b-b84a-4c0c-a3a9-f3419d72813d-000000@eu-west-1.amazonses.com> Message-ID: <01020161c2e60d3d-dedd2037-775c-45d2-adf1-5c7ae8ef3dfa-000000@eu-west-1.amazonses.com> Please ignore my last email.   Both ways work it was an error in my code I wasn’t using t_relay();   Thanks for your help 😊   Regards,   Brian Southworth   From: Users [mailto:users-bounces at lists.opensips.org] On Behalf Of Ovidiu Sas Sent: 23 February 2018 13:10 To: users at lists.opensips.org users Subject: Re: [OpenSIPS-Users] On hold   Hello Brian,   You can detect hold using is_audio_on_hold(): http://www.opensips.org/html/docs/modules/2.3.x/sipmsgops#idp5575408   Try to use it and see if this will solve your issue.   Regards, Ovidiu Sas   On Feb 23, 2018 4:13 AM, "Brian Southworth" > wrote: Hi All,   I am trying to get opensips to forward the on hold request to asterisk I have done this using an if statement for SDP =~ “sendonly”   How ever when I go to take them off hold the call is dropped, is there something special I need to do ?   I have tried and if statement for is_method(“invite”) && $rb =~ “sendrecv” ß this loops until opensips give a 500 internal error And I have also tried is_method(“invite”) && $rb =~ “inactive” ß this doesn’t seem to work at all   The initial hold works perfect puts them on hold just cant take them off or transfer the call (asterisk will handle all the B2B stuff)   Any help would be appreciated.   Regards,   Brian Southworth Communications Developer 111 Wilmslow Road Handforth Wilmslow SK9 3ER   T: 03333 446677 W: www.clocom.uk           Like us on Facebook Follow us on Twitter           Clocom is a green company. 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Registered name and address: Clocom UK Ltd, 111 Wilmslow Road, Handforth, Cheshire, SK9 3ER  www.clocom.uk   _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users   _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/png Size: 9352 bytes Desc: not available URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/png Size: 2708 bytes Desc: not available URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/png Size: 234 bytes Desc: not available URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/png Size: 2999 bytes Desc: not available URL: From contact at 35belowsoks.com Sun Feb 25 05:02:34 2018 From: contact at 35belowsoks.com (Buy 35 Below Socks) Date: Sun, 25 Feb 2018 05:02:34 -0500 Subject: [OpenSIPS-Users] Socks Keep Your Feet Just The Right Temperature Message-ID: Socks Keep Your Feet Just The Right Temperature http://paleocoockbok.bid/EJHmm4aej9PJoLgltGyNESjTZ_JhFjUSBcIOzr3YOZlum-Hf http://paleocoockbok.bid/WM7dFAyrmc57NyiTrfaRUQ4PlgaxDbx9j-vnYMBKQnP24gUl refer to commensal or even parasitic interactions. The term myrmecophile is used mainly for animals that associate with ants. There are an estimated 10,000 species of ants (Formicidae), with a higher diversity in the tropics. 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URL: From cadre5000 at gmail.com Mon Feb 26 05:17:41 2018 From: cadre5000 at gmail.com (Tiaki Cadre) Date: Mon, 26 Feb 2018 23:17:41 +1300 Subject: [OpenSIPS-Users] Parallel forking - use mid_registrar module question In-Reply-To: References: Message-ID: In addition to this last email, this is now working. Changes made: - upgraded to Opensips 2.4-dev The mid_registrar module now works as expected with parallel forking. A bit more work yet, but much better result. Thank you On Fri, Feb 23, 2018 at 9:35 AM, Tiaki Cadre wrote: > I have been using Opensips now for a few years, and in the last month or > so I needed to use Opensips for another application. > > > We need to put Opensips in front of our Asterisk servers to handle better > the register, notify, invite and subscribe messages (basically the SIP > messaging). > > My goal initially, is to have Opensips as a transparent SIP proxy for > this. > > I have already got this working mostly using the mid_registrar module - > just been looking at it further. > > I can send and receive calls from a device that is using the proxy and the > SIP is sent to the main registrar (the asterisk). > > The audio works fine since at this stage I am using the RTP on the > asterisk. > > > The issue I need to solve is multiple registered devices being able to be > called when a user on asterisk makes a call. Outbound calls from the > extension via the proxy work with no issue. Also the notify (BLF) works ok. > > I am working on how to get the branch/forking working correctly and the > single call to ring all the contacts in that have "registered" via the > mid_registrar. > I have been trying the AOR for in the mid_registrar request, and then > using mid_registrar_lookup but not having much success. > I am using opensips 2.3.3. After some reading yesterday, there was > suggestion to move to opensips 2.4 branch > I thought I would post here before proceeding too much further in case I > am missing something else. > > > > What is working: > > - I can make a call from a phone that is registered to the main registrar > fyi the proxy to an external number (handled by the main registrar) > > - I can make a call between phones that are registered to the main > registrar fyi the proxy or that are registered directly to the main > registrar > > - I can see the BLF/Notify and subscribe messages working from the main > registrar correctly > > > Questions: > > - Some of the registered devices may be behind NAT and some may be on > Public IPs. How do I best cater for this? > > > What I need to work: > > - When a phone, or incoming call from the main registrar to the proxy > occurs, I need the proxy to ring all phones that are registered through it. > > > Here are some technical details: > > version: opensips 2.3.3 (x86_64/linux) > flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, > F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT > ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, > MAX_URI_SIZE 1024, BUF_SIZE 65535 > poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. > main.c compiled on 00:04:06 Jan 19 2018 with gcc 4.4.7 > > > From my opensips.cfg > > > I have some other modules loaded -but here is the mid_registrar section: > > loadmodule "mid_registrar.so" > modparam("mid_registrar", "mode", 1) /* 0 = mirror / 1 = ct / 2 = AoR */ > modparam("mid_registrar", "outgoing_expires", 900) > modparam("mid_registrar", "insertion_mode", 1) /* 0 = contact; 1 = path */ > modparam("mid_registrar", "received_avp", "$avp(rcv)") > modparam("mid_registrar", "contact_match_param", "regid") > modparam("mid_registrar", "max_contacts", 4) > modparam("mid_registrar", "tcp_persistent_flag", 10) > modparam("mid_registrar", "outgoing_expires", 900) > > > Snippet from my routing section: > > if (is_method("REGISTER")) { > #fix_nated_register(); > xlog("L_INFO","Saving location to mid_registrar"); > mid_registrar_save("location"); > switch ($retcode) { > case 1: > xlog("forwarding REGISTER to main registrar ($$ci=$ci)\n"); > $ru = "sip:x.x.x.x:5060"; > t_relay(); > break; > case 2: > xlog("absorbing REGISTER! ($$ci=$ci)\n"); > break; > default: > xlog("failed to save registration! ($$ci=$ci)\n"); > } > exit; > } > > > remove_hf("Route"); > # record routing > if (is_method("INVITE|REFER|SUBSCRIBE|NOTIFY")) > record_route(); > > # account only INVITEs > if (is_method("INVITE")) { > do_accounting("log"); > xlog("L_INFO", "[LOG] INVITE PACKET [rm:$rm] [fu:$fu] [ou:$ou] > [ru:$ru] [si:$si]"); > } > > > # initial requests from main registrar, need to look them up! > if (is_method("INVITE|MESSAGE") && $si == "" > && $sp == 5060) { > xlog("looking up $ru!\n"); > fix_nated_register(); > if (!mid_registrar_lookup("location")) { > xlog("looked up main registrar - not found"); > t_reply("404", "Not Found"); > exit; > } > xlog("L_INFO","MYLOG INVITE mid_registrar after looking up $ru > $ci"); > t_relay(); > exit; > } > > > > > route[relay] { > > # for INVITEs enable some additional helper routes > if (is_method("INVITE|NOTIFY|SUBSCRIBE|PUBLISH")) { > fix_nated_contact(); > t_on_branch("per_branch_ops"); > t_on_reply("handle_nat"); > t_on_failure("missed_call"); > } > > if (!t_relay()) { > send_reply("500","Internal Error"); > }; > exit; > } > > > branch_route[per_branch_ops] { > xlog("new branch at $ru\n"); > } > > > onreply_route[handle_nat] { > if (nat_uac_test("1")) { > xlog("FIXING THE NAT!"); > fix_nated_contact(); > } > xlog("incoming reply\n"); > } > > > failure_route[missed_call] { > if (t_was_cancelled()) { > exit; > } > > # uncomment the following lines if you want to block client > # redirect based on 3xx replies. > ##if (t_check_status("3[0-9][0-9]")) { > ##t_reply("404","Not found"); > ## exit; > ##} > > } > -------------- next part -------------- An HTML attachment was scrubbed... URL: From yuval.dinari at vonage.com Mon Feb 26 05:31:14 2018 From: yuval.dinari at vonage.com (Dinari, Yuval) Date: Mon, 26 Feb 2018 12:31:14 +0200 Subject: [OpenSIPS-Users] textops module replace functions don't work Message-ID: Hi, I tried using textops module functions to replace a part of the message body, but the body remains unchanged. A search() with the regex to be replaced succeeds. I tried all replacement functions (replace*, subst, subst_body). I see that the body is unchanged by printing $rb, and also ngreping the SDP that opensips sends out to rtpengine via rtpengine_offer(). opensips version - 2.2.5 Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: From Ben.Newlin at genesys.com Mon Feb 26 08:48:26 2018 From: Ben.Newlin at genesys.com (Ben Newlin) Date: Mon, 26 Feb 2018 13:48:26 +0000 Subject: [OpenSIPS-Users] Wrong number of children In-Reply-To: <010001d3aef6$02f70f90$08e52eb0$@web.de> References: <027501d3aca1$66551340$32ff39c0$@web.de> <010001d3aef6$02f70f90$08e52eb0$@web.de> Message-ID: <7C9C26CF-D163-4E9F-A428-72D1746731F1@genesys.com> I am seeing this as well while testing out my migration to 2.3.3. In 1.11, I used the disable_tcp parameter since I don’t use TCP for anything. That parameter is no longer available in 2.x and now I get a large number of TCP receiver threads starting automatically with OpenSIPS even though I am not using TCP and have no modules loaded that use TCP, to my knowledge. This is very concerning for me that there no longer seems to be any way to stop OpenSIPS from starting these TCP receivers in 2.3.3. Ben Newlin From: Users on behalf of xaled Reply-To: OpenSIPS users mailling list Date: Monday, February 26, 2018 at 6:29 AM To: 'OpenSIPS users mailling list' Subject: Re: [OpenSIPS-Users] Wrong number of children Hi Callum, that helped thanks. I have 8 TCP receivers and one TCP main process that I did not expected as I only listen on udp. Will have to understand the TCP spawn logic. Greetings, Xaled From: Users [mailto:users-bounces at lists.opensips.org] On Behalf Of Callum Guy Sent: Friday, February 23, 2018 2:42 PM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] Wrong number of children Hi Xaled, If you have the FIFO module loaded you can run "opensipsctl fifo ps" on the instance to see full details of each process. https://www.opensips.org/Documentation/Interface-CoreMI-2-2#toc7 Hope that helps, Callum On Fri, Feb 23, 2018 at 1:35 PM Daniel Zanutti > wrote: Hi You should have 5 per interface + some internal control threads. I'm not sure exactly. Regards On Fri, Feb 23, 2018 at 9:25 AM, xaled > wrote: Hi I have configured 5 children in opensips.cfg, but 16 get logged and 18 initiated what could be the cause of it? I search for word children in opensips directory and it only comes ones in opensips.conf. children=5 Feb 23 13:07:09 srv01 opensips: DBG:core:daemonize: waiting for status code from children Feb 23 13:07:09 srv01 opensips: DBG:core:count_init_children: 16 children are going to be inited root at dus-appsrv01:/var/log/opensips# ps ax|grep sips 14854 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14856 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14857 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14858 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14861 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14862 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14863 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14864 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14865 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14867 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14869 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14870 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14871 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14872 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14874 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14875 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14876 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14877 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 With commended children parameter I get 19 children in the log and 21 in reality: Feb 23 13:13:24 dus-appsrv01 opensips: DBG:core:daemonize: waiting for status code from children Feb 23 13:13:24 dus-appsrv01 opensips: DBG:core:count_init_children: 19 children are going to be inited root at srv01:/var/log/opensips# ps ax|grep sips 14723 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14725 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14726 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14727 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14730 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14731 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14732 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14733 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14734 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14736 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14737 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14738 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14739 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14741 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14742 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14743 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14744 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14745 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14748 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14749 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14750 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 root at srv01:/etc/opensips# opensips -V version: opensips 2.3.3 (x86_64/linux) flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535 poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. main.c compiled on with gcc 6.3.0 Thanks, Xaled _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Callum Guy Head of Information Security X-on [Image removed by sender.] 0333 332 0000 | www.x-on.co.uk | [Image removed by sender.] [Image removed by sender.] [Image removed by sender.] X-on is a trading name of Storacall Technology Ltd a limited company registered in England and Wales. Registered Office : Avaland House, 110 London Road, Apsley, Hemel Hempstead, Herts, HP3 9SD. Company Registration No. 2578478. The information in this e-mail is confidential and for use by the addressee(s) only. If you are not the intended recipient, please notify X-on immediately on +44(0)333 332 0000 and delete the message from your computer. If you are not a named addressee you must not use, disclose, disseminate, distribute, copy, print or reply to this email. Views or opinions expressed by an individual within this email may not necessarily reflect the views of X-on or its associated companies. Although X-on routinely screens for viruses, addressees should scan this email and any attachments for viruses. X-on makes no representation or warranty as to the absence of viruses in this email or any attachments. -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Mon Feb 26 10:41:24 2018 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Mon, 26 Feb 2018 17:41:24 +0200 Subject: [OpenSIPS-Users] Wrong number of children In-Reply-To: <010001d3aef6$02f70f90$08e52eb0$@web.de> References: <027501d3aca1$66551340$32ff39c0$@web.de> <010001d3aef6$02f70f90$08e52eb0$@web.de> Message-ID: <4e50d0e9-d0c8-5317-4f21-433cbc5cde13@opensips.org> Hi Xaled, Do you use HEP for tracing or BIN for replication ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Summit 2018 http://www.opensips.org/events/Summit-2018Amsterdam On 02/26/2018 01:36 PM, xaled wrote: > > Hi Callum, > > that helped thanks. > > I have 8 TCP receivers and one TCP main process that I did not > expected as I only listen on udp. > > Will have to understand the TCP spawn logic. > > Greetings, > > Xaled > > *From:*Users [mailto:users-bounces at lists.opensips.org] *On Behalf Of > *Callum Guy > *Sent:* Friday, February 23, 2018 2:42 PM > *To:* OpenSIPS users mailling list > *Subject:* Re: [OpenSIPS-Users] Wrong number of children > > Hi Xaled, > > If you have the FIFO module loaded you can run "opensipsctl fifo ps" > on the instance to see full details of each process. > > https://www.opensips.org/Documentation/Interface-CoreMI-2-2#toc7 > > Hope that helps, > > Callum > > On Fri, Feb 23, 2018 at 1:35 PM Daniel Zanutti > > wrote: > > Hi > > You should have 5 per interface + some internal control threads. > I'm not sure exactly. > > Regards > > On Fri, Feb 23, 2018 at 9:25 AM, xaled > wrote: > > Hi > > I have configured 5 children in opensips.cfg, but 16 get > logged and 18 initiated what could be the cause of it? > > I search for word children in opensips directory and it only > comes ones in opensips.conf. > > children=5 > > Feb 23 13:07:09 srv01 opensips: DBG:core:daemonize: waiting > for status code from children > > Feb 23 13:07:09 srv01 opensips: DBG:core:count_init_children: > 16 children are going to be inited > > root at dus-appsrv01:/var/log/opensips# ps ax|grep sips > > 14854 ? S 0:00 /usr/sbin/opensips -P > /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 > -M 4 > > 14856 ? S 0:00 /usr/sbin/opensips -P > /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 > -M 4 > > 14857 ? S 0:00 /usr/sbin/opensips -P > /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 > -M 4 > > 14858 ? S 0:00 /usr/sbin/opensips -P > /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 > -M 4 > > 14861 ? S 0:00 /usr/sbin/opensips -P > /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 > -M 4 > > 14862 ? S 0:00 /usr/sbin/opensips -P > /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 > -M 4 > > 14863 ? S 0:00 /usr/sbin/opensips -P > /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 > -M 4 > > 14864 ? S 0:00 /usr/sbin/opensips -P > /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 > -M 4 > > 14865 ? S 0:00 /usr/sbin/opensips -P > /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 > -M 4 > > 14867 ? S 0:00 /usr/sbin/opensips -P > /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 > -M 4 > > 14869 ? S 0:00 /usr/sbin/opensips -P > /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 > -M 4 > > 14870 ? S 0:00 /usr/sbin/opensips -P > /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 > -M 4 > > 14871 ? S 0:00 /usr/sbin/opensips -P > /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 > -M 4 > > 14872 ? S 0:00 /usr/sbin/opensips -P > /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 > -M 4 > > 14874 ? S 0:00 /usr/sbin/opensips -P > /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 > -M 4 > > 14875 ? S 0:00 /usr/sbin/opensips -P > /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 > -M 4 > > 14876 ? S 0:00 /usr/sbin/opensips -P > /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 > -M 4 > > 14877 ? S 0:00 /usr/sbin/opensips -P > /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 > -M 4 > > With commended children parameter I get 19 children in the log > and 21 in reality: > > Feb 23 13:13:24 dus-appsrv01 opensips: DBG:core:daemonize: > waiting for status code from children > > Feb 23 13:13:24 dus-appsrv01 opensips: > DBG:core:count_init_children: 19 children are going to be inited > > root at srv01:/var/log/opensips# ps ax|grep sips > > 14723 ? S 0:00 /usr/sbin/opensips -P > /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 > -M 4 > > 14725 ? S 0:00 /usr/sbin/opensips -P > /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 > -M 4 > > 14726 ? S 0:00 /usr/sbin/opensips -P > /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 > -M 4 > > 14727 ? S 0:00 /usr/sbin/opensips -P > /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 > -M 4 > > 14730 ? S 0:00 /usr/sbin/opensips -P > /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 > -M 4 > > 14731 ? S 0:00 /usr/sbin/opensips -P > /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 > -M 4 > > 14732 ? S 0:00 /usr/sbin/opensips -P > /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 > -M 4 > > 14733 ? S 0:00 /usr/sbin/opensips -P > /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 > -M 4 > > 14734 ? S 0:00 /usr/sbin/opensips -P > /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 > -M 4 > > 14736 ? S 0:00 /usr/sbin/opensips -P > /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 > -M 4 > > 14737 ? S 0:00 /usr/sbin/opensips -P > /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 > -M 4 > > 14738 ? S 0:00 /usr/sbin/opensips -P > /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 > -M 4 > > 14739 ? S 0:00 /usr/sbin/opensips -P > /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 > -M 4 > > 14741 ? S 0:00 /usr/sbin/opensips -P > /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 > -M 4 > > 14742 ? S 0:00 /usr/sbin/opensips -P > /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 > -M 4 > > 14743 ? S 0:00 /usr/sbin/opensips -P > /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 > -M 4 > > 14744 ? S 0:00 /usr/sbin/opensips -P > /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 > -M 4 > > 14745 ? S 0:00 /usr/sbin/opensips -P > /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 > -M 4 > > 14748 ? S 0:00 /usr/sbin/opensips -P > /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 > -M 4 > > 14749 ? S 0:00 /usr/sbin/opensips -P > /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 > -M 4 > > 14750 ? S 0:00 /usr/sbin/opensips -P > /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 > -M 4 > > root at srv01:/etc/opensips# opensips -V > > version: opensips 2.3.3 (x86_64/linux) > > flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, > PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT > > ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, > MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535 > > poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. > > main.c compiled on with gcc 6.3.0 > > Thanks, > > Xaled > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > -- > > Callum Guy > > Head of Information Security > > X-on > > *^0333 332 0000 | www.x-on.co.uk | > **_**^ > *** > X-on is a trading name of Storacall Technology Ltd a limited company > registered in England and Wales. > Registered Office : Avaland House, 110 London Road, Apsley, Hemel > Hempstead, Herts, HP3 9SD. Company Registration No. 2578478. > The information in this e-mail is confidential and for use by the > addressee(s) only. If you are not the intended recipient, please > notify X-on immediately on +44(0)333 332 0000 and delete the > message from your computer. If you are not a named addressee you must > not use, disclose, disseminate, distribute, copy, print or reply to > this email. Views or opinions expressed by an individual > within this email may not necessarily reflect the views of X-on or its > associated companies. Although X-on routinely screens for viruses, > addressees should scan this email and any attachments > for viruses. X-on makes no representation or warranty as to the > absence of viruses in this email or any attachments. > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Mon Feb 26 10:44:48 2018 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Mon, 26 Feb 2018 17:44:48 +0200 Subject: [OpenSIPS-Users] OpenSIPS crashing on 200 OK In-Reply-To: <6D9EBB18-914E-49ED-988F-7D582007B88E@genesys.com> References: <530E1611-2A4D-424F-938D-2E409E1A78A2@genesys.com> <7cf6bb3d-967a-b8e2-d4b3-8b42c8dbf0b8@opensips.org> <755F831E-90E4-4105-80C5-42C7A2E2050A@genesys.com> <6D9EBB18-914E-49ED-988F-7D582007B88E@genesys.com> Message-ID: <085c6c6b-75ca-0557-aa5e-0fcb9e3eb07f@opensips.org> OK, once you started GDB like "gdb /path/to/opensips /path/to/core", run step by step: f 0 p *s1 f 5 p *a Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Summit 2018 http://www.opensips.org/events/Summit-2018Amsterdam On 02/26/2018 05:43 PM, Ben Newlin wrote: > > Bogdan, > > Unfortunately I am not at all familiar with gdb. How do I change frames? > > Ben Newlin > > *From: *Bogdan-Andrei Iancu > *Date: *Monday, February 26, 2018 at 10:40 AM > *To: *Ben Newlin , OpenSIPS users mailling > list > *Subject: *Re: [OpenSIPS-Users] OpenSIPS crashing on 200 OK > > Hi Ben, > > The backtrace points to the script interpreter :D....Please do the > followings (assuming you are a bit faimilar with GDB :) ) : > > in frame 0, please do "print *s1" > and in frame 5 do "print *a" > > Regards, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > OpenSIPS Summit 2018 > http://www.opensips.org/events/Summit-2018Amsterdam > > On 02/22/2018 07:33 PM, Ben Newlin wrote: > > Bogdan, > > Here is the backtrace: https://pastebin.com/6tZhrWek > > > I am not using TCP, BIN, or HEP in this configuration. I am not > using anything, to my knowledge, which should be causing TCP > worker threads to be spawned. > > Ben Newlin > > *From: *Bogdan-Andrei Iancu > > *Date: *Thursday, February 22, 2018 at 5:26 AM > *To: *OpenSIPS users mailling list > , Ben Newlin > > *Subject: *Re: [OpenSIPS-Users] OpenSIPS crashing on 200 OK > > Hi Ben, > > Do you have a full backtrace extracted from the core ? > > In regards to the TCP processes, the TCP layer will automatically > enabled itself depending on what protocols (tcp-based) you are > using in script. Maybe you do not use SIP with TCP, but do you use > BIN or HEP ? > > Regards, > > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > > http://www.opensips-solutions.com > > OpenSIPS Summit 2018 > > http://www.opensips.org/events/Summit-2018Amsterdam > > On 02/22/2018 12:38 AM, Ben Newlin wrote: > > Hi, > > I am currently upgrading to 2.3.3 and am running off the HEAD > of the 2.3 branch. My OpenSIPS is consistently crashing when > trying to process the ACK to an initial INVITE request. I am > testing with SIPp using a simple single call scenario. I see > the following in the logs: > > Feb 21 21:32:27 [383] DBG:core:io_wait_loop_epoll: [TCP_main] > EPOLLHUP on IN ->connection closed by the remote peer! > > Feb 21 21:32:27 [383] CRITICAL:core:receive_fd: EOF on 33 > > Feb 21 21:32:27 [383] DBG:core:handle_worker: dead child 9, > pid 370 (shutting down?) > > Feb 21 21:32:27 [383] DBG:core:io_watch_del: [TCP_main] > io_watch_del op on index 8 33 (0x8d4e80, 33, 8, 0x0,0x1) > fd_no=29 called > > Feb 21 21:32:27 [361] DBG:core:handle_sigs: status = 139 > > Feb 21 21:32:27 [361] INFO:core:handle_sigs: child process 370 > exited by a signal 11 > > Feb 21 21:32:27 [361] INFO:core:handle_sigs: core was generated > > Feb 21 21:32:27 [361] INFO:core:handle_sigs: terminating due > to SIGCHLD > > I also notice that OpenSIPS is starting up a number of TCP > processes even though I do not use TCP in my script. I do not > even have the proto_tcp module loaded. I used to use the > “disable_tcp” global parameter in 1.11, but that has been > removed in 2.3. > > Feb 21 22:03:56 [361] DBG:core:internal_fork: forking new > process "SIP receiver TCP" > > Feb 21 22:03:56 [361] DBG:core:internal_fork: forking new > process "SIP receiver TCP" > > Feb 21 22:03:56 [361] DBG:core:internal_fork: forking new > process "SIP receiver TCP" > > Feb 21 22:03:56 [361] DBG:core:internal_fork: forking new > process "SIP receiver TCP" > > Feb 21 22:03:56 [361] DBG:core:internal_fork: forking new > process "SIP receiver TCP" > > Feb 21 22:03:56 [375] DBG:core:io_watch_add: [TCP_worker] > io_watch_add op (3 on 53) (0x8d4e80, 3, 1, (nil),1), > fd_no=0/209715 > > Feb 21 22:03:56 [361] DBG:core:internal_fork: forking new > process "SIP receiver TCP" > > Feb 21 22:03:56 [361] DBG:core:internal_fork: forking new > process "SIP receiver TCP" > > Feb 21 22:03:56 [361] DBG:core:internal_fork: forking new > process "SIP receiver TCP" > > Initialization logs: https://pastebin.com/nsJQGkdy > > Call logs: https://pastebin.com/vY5crK8U > > I also have the core dump if any more information is needed > from that. > > Thanks, > > Ben Newlin > > > > > > _______________________________________________ > > Users mailing list > > Users at lists.opensips.org > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From osas at voipembedded.com Mon Feb 26 10:49:14 2018 From: osas at voipembedded.com (Ovidiu Sas) Date: Mon, 26 Feb 2018 10:49:14 -0500 Subject: [OpenSIPS-Users] On hold In-Reply-To: <01020161c2e60d3d-dedd2037-775c-45d2-adf1-5c7ae8ef3dfa-000000@eu-west-1.amazonses.com> References: <01020161c1ef8de9-1fa3116b-b84a-4c0c-a3a9-f3419d72813d-000000@eu-west-1.amazonses.com> <01020161c2e60d3d-dedd2037-775c-45d2-adf1-5c7ae8ef3dfa-000000@eu-west-1.amazonses.com> Message-ID: You need to take a look at the signalling and figure out who is dropping the call and why. -ovidiu On Feb 26, 2018 10:45, "Brian Southworth" wrote: > Please ignore my last email. > > > > Both ways work it was an error in my code I wasn’t using t_relay(); > > > > Thanks for your help 😊 > > > > Regards, > > > > Brian Southworth > > > > *From:* Users [mailto:users-bounces at lists.opensips.org] *On Behalf Of *Ovidiu > Sas > *Sent:* 23 February 2018 13:10 > *To:* users at lists.opensips.org users > *Subject:* Re: [OpenSIPS-Users] On hold > > > > Hello Brian, > > > > You can detect hold using is_audio_on_hold(): > > http://www.opensips.org/html/docs/modules/2.3.x/sipmsgops#idp5575408 > > > > Try to use it and see if this will solve your issue. > > > > Regards, > > Ovidiu Sas > > > > On Feb 23, 2018 4:13 AM, "Brian Southworth" > wrote: > > Hi All, > > > > I am trying to get opensips to forward the on hold request to asterisk I > have done this using an if statement for SDP =~ “sendonly” > > > > How ever when I go to take them off hold the call is dropped, is there > something special I need to do ? > > > > I have tried and if statement for is_method(“invite”) && $rb =~ “sendrecv” > ß this loops until opensips give a 500 internal error > > And I have also tried is_method(“invite”) && $rb =~ “inactive” ß this > doesn’t seem to work at all > > > > The initial hold works perfect puts them on hold just cant take them off > or transfer the call (asterisk will handle all the B2B stuff) > > > > Any help would be appreciated. > > > > Regards, > > > > Brian Southworth > > Communications Developer > > [image: cid:image001.png at 01D22CAC.1DCB8580] > > 111 Wilmslow Road > > Handforth > > Wilmslow > > SK9 3ER > > > > T: 03333 446677 > > W: www.clocom.uk > > > > > > > > > > > > [image: cid:image002.png at 01CDDC62.D8483910] > > > Like us on Facebook > > [image: cid:image003.png at 01CDDC62.D8483910] > > Follow us on Twitter > > [image: cid:image004.png at 01CDDC62.D8483910] > > > > > > > > > > > > > Clocom is a *green* company. Think, do you need to print this email? > > > > This message contains confidential information and is intended only for > the individual named. If you are not the named addressee you should not > disseminate, distribute or copy this e-mail. Please notify the sender > immediately by e-mail if you have received this e-mail by mistake and > delete this e-mail from your system. E-mail transmission cannot be > guaranteed to be secure or error-free as information could be intercepted, > corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. > The sender therefore does not accept liability for any errors or omissions > in the contents of this message, which arise as a result of e-mail > transmission. If verification is required please request a hard-copy > version. Registered in England & Wales: 07081192. Registered name and > address: Clocom UK Ltd, 111 Wilmslow Road, Handforth, Cheshire, SK9 3ER > www.clocom.uk > > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > _______________________________________________ > > Users mailing list > > Users at lists.opensips.org > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > -------------- next part -------------- An HTML attachment was scrubbed... URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/png Size: 2708 bytes Desc: not available URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/png Size: 9352 bytes Desc: not available URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/png Size: 234 bytes Desc: not available URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/png Size: 2999 bytes Desc: not available URL: From bogdan at opensips.org Mon Feb 26 10:54:38 2018 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Mon, 26 Feb 2018 17:54:38 +0200 Subject: [OpenSIPS-Users] textops module replace functions don't work In-Reply-To: References: Message-ID: Hi, Have you used replace() or replace_body() ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Summit 2018 http://www.opensips.org/events/Summit-2018Amsterdam On 02/26/2018 12:31 PM, Dinari, Yuval via Users wrote: > Hi, > I tried using textops module functions to replace a part of the > message body, but the body remains unchanged. A search() with the > regex to be replaced succeeds. > I tried all replacement functions (replace*, subst, subst_body). > I see that the body is unchanged by printing $rb, and also ngreping > the SDP that opensips sends out to rtpengine via rtpengine_offer(). > opensips version - 2.2.5 > Thanks > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From xaled at web.de Mon Feb 26 10:53:48 2018 From: xaled at web.de (xaled) Date: Mon, 26 Feb 2018 16:53:48 +0100 Subject: [OpenSIPS-Users] Wrong number of children In-Reply-To: <4e50d0e9-d0c8-5317-4f21-433cbc5cde13@opensips.org> References: <027501d3aca1$66551340$32ff39c0$@web.de> <010001d3aef6$02f70f90$08e52eb0$@web.de> <4e50d0e9-d0c8-5317-4f21-433cbc5cde13@opensips.org> Message-ID: <01ff01d3af19$fda10db0$f8e32910$@web.de> Hi Bogdan, nope, non of them. Greetings, Xaled From: Bogdan-Andrei Iancu [mailto:bogdan at opensips.org] Sent: Monday, February 26, 2018 4:41 PM To: OpenSIPS users mailling list ; xaled Subject: Re: [OpenSIPS-Users] Wrong number of children Hi Xaled, Do you use HEP for tracing or BIN for replication ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Summit 2018 http://www.opensips.org/events/Summit-2018Amsterdam On 02/26/2018 01:36 PM, xaled wrote: Hi Callum, that helped thanks. I have 8 TCP receivers and one TCP main process that I did not expected as I only listen on udp. Will have to understand the TCP spawn logic. Greetings, Xaled From: Users [mailto:users-bounces at lists.opensips.org] On Behalf Of Callum Guy Sent: Friday, February 23, 2018 2:42 PM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] Wrong number of children Hi Xaled, If you have the FIFO module loaded you can run "opensipsctl fifo ps" on the instance to see full details of each process. https://www.opensips.org/Documentation/Interface-CoreMI-2-2#toc7 Hope that helps, Callum On Fri, Feb 23, 2018 at 1:35 PM Daniel Zanutti > wrote: Hi You should have 5 per interface + some internal control threads. I'm not sure exactly. Regards On Fri, Feb 23, 2018 at 9:25 AM, xaled > wrote: Hi I have configured 5 children in opensips.cfg, but 16 get logged and 18 initiated what could be the cause of it? I search for word children in opensips directory and it only comes ones in opensips.conf. children=5 Feb 23 13:07:09 srv01 opensips: DBG:core:daemonize: waiting for status code from children Feb 23 13:07:09 srv01 opensips: DBG:core:count_init_children: 16 children are going to be inited root at dus-appsrv01:/var/log/opensips# ps ax|grep sips 14854 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14856 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14857 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14858 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14861 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14862 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14863 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14864 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14865 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14867 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14869 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14870 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14871 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14872 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14874 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14875 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14876 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14877 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 With commended children parameter I get 19 children in the log and 21 in reality: Feb 23 13:13:24 dus-appsrv01 opensips: DBG:core:daemonize: waiting for status code from children Feb 23 13:13:24 dus-appsrv01 opensips: DBG:core:count_init_children: 19 children are going to be inited root at srv01:/var/log/opensips# ps ax|grep sips 14723 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14725 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14726 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14727 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14730 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14731 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14732 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14733 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14734 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14736 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14737 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14738 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14739 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14741 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14742 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14743 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14744 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14745 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14748 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14749 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14750 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 root at srv01:/etc/opensips# opensips -V version: opensips 2.3.3 (x86_64/linux) flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535 poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. main.c compiled on with gcc 6.3.0 Thanks, Xaled _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Callum Guy Head of Information Security X-on 0333 332 0000 | www.x-on.co.uk | X-on is a trading name of Storacall Technology Ltd a limited company registered in England and Wales. Registered Office : Avaland House, 110 London Road, Apsley, Hemel Hempstead, Herts, HP3 9SD. Company Registration No. 2578478. The information in this e-mail is confidential and for use by the addressee(s) only. If you are not the intended recipient, please notify X-on immediately on +44(0)333 332 0000 and delete the message from your computer. If you are not a named addressee you must not use, disclose, disseminate, distribute, copy, print or reply to this email. Views or opinions expressed by an individual within this email may not necessarily reflect the views of X-on or its associated companies. Although X-on routinely screens for viruses, addressees should scan this email and any attachments for viruses. X-on makes no representation or warranty as to the absence of viruses in this email or any attachments. _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: ~WRD084.jpg Type: image/jpeg Size: 823 bytes Desc: not available URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: image001.jpg Type: image/jpeg Size: 338 bytes Desc: not available URL: From brian.southworth at clocom.uk Mon Feb 26 10:55:38 2018 From: brian.southworth at clocom.uk (=?UTF-8?Q?Brian_Southworth?=) Date: Mon, 26 Feb 2018 15:55:38 +0000 Subject: [OpenSIPS-Users] On hold In-Reply-To: References: <01020161c1ef8de9-1fa3116b-b84a-4c0c-a3a9-f3419d72813d-000000@eu-west-1.amazonses.com> <01020161c2e60d3d-dedd2037-775c-45d2-adf1-5c7ae8ef3dfa-000000@eu-west-1.amazonses.com> Message-ID: <01020161d2d39b50-ecde506f-9a4e-430e-9d46-6cdee6b8c2ab-000000@eu-west-1.amazonses.com> Seems to be an issue with it sending the on hold invite to get the call back to the wrong place on inbound calls.   Example caller calls in --- agent 1 picks up --- agent 1 places them on hold all is fine --- agent 1 takes call off hold call is dropped   Seems that the invite to get the call back gets sent to .192 when it should go back to .193   This doesn’t happen is I use rewritehostport but then on hold calls only work for calls coming from that media server t_relay works across the board but doesn’t actually work for inbound.   Regards,   Brian Southworth   From: Users [mailto:users-bounces at lists.opensips.org] On Behalf Of Ovidiu Sas Sent: 26 February 2018 15:49 To: users at lists.opensips.org users Subject: Re: [OpenSIPS-Users] On hold   You need to take a look at the signalling and figure out who is dropping the call and why.   -ovidiu   On Feb 26, 2018 10:45, "Brian Southworth" > wrote: Please ignore my last email.   Both ways work it was an error in my code I wasn’t using t_relay();   Thanks for your help 😊   Regards,   Brian Southworth   From: Users [mailto:users-bounces at lists.opensips.org ] On Behalf Of Ovidiu Sas Sent: 23 February 2018 13:10 To: users at lists.opensips.org users > Subject: Re: [OpenSIPS-Users] On hold   Hello Brian,   You can detect hold using is_audio_on_hold(): http://www.opensips.org/html/docs/modules/2.3.x/sipmsgops#idp5575408   Try to use it and see if this will solve your issue.   Regards, Ovidiu Sas   On Feb 23, 2018 4:13 AM, "Brian Southworth" > wrote: Hi All,   I am trying to get opensips to forward the on hold request to asterisk I have done this using an if statement for SDP =~ “sendonly”   How ever when I go to take them off hold the call is dropped, is there something special I need to do ?   I have tried and if statement for is_method(“invite”) && $rb =~ “sendrecv” ß this loops until opensips give a 500 internal error And I have also tried is_method(“invite”) && $rb =~ “inactive” ß this doesn’t seem to work at all   The initial hold works perfect puts them on hold just cant take them off or transfer the call (asterisk will handle all the B2B stuff)   Any help would be appreciated.   Regards,   Brian Southworth Communications Developer 111 Wilmslow Road Handforth Wilmslow SK9 3ER   T: 03333 446677 W: www.clocom.uk           Like us on Facebook Follow us on Twitter           Clocom is a green company. 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Registered name and address: Clocom UK Ltd, 111 Wilmslow Road, Handforth, Cheshire, SK9 3ER  www.clocom.uk   _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users   _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: -------------- next part -------------- A non-text attachment was scrubbed... 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Name: not available Type: image/png Size: 2999 bytes Desc: not available URL: From bogdan at opensips.org Mon Feb 26 10:57:25 2018 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Mon, 26 Feb 2018 17:57:25 +0200 Subject: [OpenSIPS-Users] Wrong number of children In-Reply-To: <01ff01d3af19$fda10db0$f8e32910$@web.de> References: <027501d3aca1$66551340$32ff39c0$@web.de> <010001d3aef6$02f70f90$08e52eb0$@web.de> <4e50d0e9-d0c8-5317-4f21-433cbc5cde13@opensips.org> <01ff01d3af19$fda10db0$f8e32910$@web.de> Message-ID: <15812914-24ca-8dfa-9371-732ce42c91bd@opensips.org> If you do "netstat -tlnp | grep opensips", what do you get ? The TCP layer automatically enable itself if aTCP-based protocol is used from the OpenSIPS config (as listener), like SIP TCP/TLS/WS/WSS , HEP or BIN. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Summit 2018 http://www.opensips.org/events/Summit-2018Amsterdam On 02/26/2018 05:53 PM, xaled wrote: > > Hi Bogdan, > > nope, non of them. > > Greetings, > > Xaled > > *From:*Bogdan-Andrei Iancu [mailto:bogdan at opensips.org] > *Sent:* Monday, February 26, 2018 4:41 PM > *To:* OpenSIPS users mailling list ; xaled > > *Subject:* Re: [OpenSIPS-Users] Wrong number of children > > Hi Xaled, > > Do you use HEP for tracing or BIN for replication ? > > Regards, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > OpenSIPS Summit 2018 > http://www.opensips.org/events/Summit-2018Amsterdam > > On 02/26/2018 01:36 PM, xaled wrote: > > Hi Callum, > > that helped thanks. > > I have 8 TCP receivers and one TCP main process that I did not > expected as I only listen on udp. > > Will have to understand the TCP spawn logic. > > Greetings, > > Xaled > > *From:*Users [mailto:users-bounces at lists.opensips.org] *On Behalf > Of *Callum Guy > *Sent:* Friday, February 23, 2018 2:42 PM > *To:* OpenSIPS users mailling list > > *Subject:* Re: [OpenSIPS-Users] Wrong number of children > > Hi Xaled, > > If you have the FIFO module loaded you can run "opensipsctl fifo > ps" on the instance to see full details of each process. > > https://www.opensips.org/Documentation/Interface-CoreMI-2-2#toc7 > > Hope that helps, > > Callum > > On Fri, Feb 23, 2018 at 1:35 PM Daniel Zanutti > > wrote: > > Hi > > You should have 5 per interface + some internal control > threads. I'm not sure exactly. > > Regards > > On Fri, Feb 23, 2018 at 9:25 AM, xaled > wrote: > > Hi > > I have configured 5 children in opensips.cfg, but 16 get > logged and 18 initiated what could be the cause of it? > > I search for word children in opensips directory and it > only comes ones in opensips.conf. > > children=5 > > Feb 23 13:07:09 srv01 opensips: DBG:core:daemonize: > waiting for status code from children > > Feb 23 13:07:09 srv01 opensips: > DBG:core:count_init_children: 16 children are going to be > inited > > root at dus-appsrv01:/var/log/opensips# ps ax|grep sips > > 14854 ? S 0:00 /usr/sbin/opensips -P > /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg > -m 64 -M 4 > > 14856 ? S 0:00 /usr/sbin/opensips -P > /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg > -m 64 -M 4 > > 14857 ? S 0:00 /usr/sbin/opensips -P > /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg > -m 64 -M 4 > > 14858 ? S 0:00 /usr/sbin/opensips -P > /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg > -m 64 -M 4 > > 14861 ? S 0:00 /usr/sbin/opensips -P > /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg > -m 64 -M 4 > > 14862 ? S 0:00 /usr/sbin/opensips -P > /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg > -m 64 -M 4 > > 14863 ? S 0:00 /usr/sbin/opensips -P > /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg > -m 64 -M 4 > > 14864 ? S 0:00 /usr/sbin/opensips -P > /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg > -m 64 -M 4 > > 14865 ? S 0:00 /usr/sbin/opensips -P > /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg > -m 64 -M 4 > > 14867 ? S 0:00 /usr/sbin/opensips -P > /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg > -m 64 -M 4 > > 14869 ? S 0:00 /usr/sbin/opensips -P > /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg > -m 64 -M 4 > > 14870 ? S 0:00 /usr/sbin/opensips -P > /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg > -m 64 -M 4 > > 14871 ? S 0:00 /usr/sbin/opensips -P > /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg > -m 64 -M 4 > > 14872 ? S 0:00 /usr/sbin/opensips -P > /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg > -m 64 -M 4 > > 14874 ? S 0:00 /usr/sbin/opensips -P > /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg > -m 64 -M 4 > > 14875 ? S 0:00 /usr/sbin/opensips -P > /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg > -m 64 -M 4 > > 14876 ? S 0:00 /usr/sbin/opensips -P > /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg > -m 64 -M 4 > > 14877 ? S 0:00 /usr/sbin/opensips -P > /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg > -m 64 -M 4 > > With commended children parameter I get 19 children in the > log and 21 in reality: > > Feb 23 13:13:24 dus-appsrv01 opensips: DBG:core:daemonize: > waiting for status code from children > > Feb 23 13:13:24 dus-appsrv01 opensips: > DBG:core:count_init_children: 19 children are going to be > inited > > root at srv01:/var/log/opensips# ps ax|grep sips > > 14723 ? S 0:00 /usr/sbin/opensips -P > /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg > -m 64 -M 4 > > 14725 ? S 0:00 /usr/sbin/opensips -P > /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg > -m 64 -M 4 > > 14726 ? S 0:00 /usr/sbin/opensips -P > /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg > -m 64 -M 4 > > 14727 ? S 0:00 /usr/sbin/opensips -P > /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg > -m 64 -M 4 > > 14730 ? S 0:00 /usr/sbin/opensips -P > /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg > -m 64 -M 4 > > 14731 ? S 0:00 /usr/sbin/opensips -P > /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg > -m 64 -M 4 > > 14732 ? S 0:00 /usr/sbin/opensips -P > /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg > -m 64 -M 4 > > 14733 ? S 0:00 /usr/sbin/opensips -P > /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg > -m 64 -M 4 > > 14734 ? S 0:00 /usr/sbin/opensips -P > /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg > -m 64 -M 4 > > 14736 ? S 0:00 /usr/sbin/opensips -P > /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg > -m 64 -M 4 > > 14737 ? S 0:00 /usr/sbin/opensips -P > /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg > -m 64 -M 4 > > 14738 ? S 0:00 /usr/sbin/opensips -P > /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg > -m 64 -M 4 > > 14739 ? S 0:00 /usr/sbin/opensips -P > /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg > -m 64 -M 4 > > 14741 ? S 0:00 /usr/sbin/opensips -P > /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg > -m 64 -M 4 > > 14742 ? S 0:00 /usr/sbin/opensips -P > /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg > -m 64 -M 4 > > 14743 ? S 0:00 /usr/sbin/opensips -P > /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg > -m 64 -M 4 > > 14744 ? S 0:00 /usr/sbin/opensips -P > /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg > -m 64 -M 4 > > 14745 ? S 0:00 /usr/sbin/opensips -P > /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg > -m 64 -M 4 > > 14748 ? S 0:00 /usr/sbin/opensips -P > /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg > -m 64 -M 4 > > 14749 ? S 0:00 /usr/sbin/opensips -P > /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg > -m 64 -M 4 > > 14750 ? S 0:00 /usr/sbin/opensips -P > /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg > -m 64 -M 4 > > root at srv01:/etc/opensips# opensips -V > > version: opensips 2.3.3 (x86_64/linux) > > flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, > PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT > > ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, > MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535 > > poll method support: poll, epoll_lt, epoll_et, sigio_rt, > select. > > main.c compiled on with gcc 6.3.0 > > Thanks, > > Xaled > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > -- > > Callum Guy > > Head of Information Security > > X-on > > Image removed by sender. > > *^0333 332 0000 | www.x-on.co.uk | > **_**^Image removed by sender. > Image removed by sender. > Image removed by sender. > *** > X-on is a trading name of Storacall Technology Ltd a limited > company registered in England and Wales. > Registered Office : Avaland House, 110 London Road, Apsley, Hemel > Hempstead, Herts, HP3 9SD. Company Registration No. 2578478. > The information in this e-mail is confidential and for use by the > addressee(s) only. If you are not the intended recipient, please > notify X-on immediately on +44(0)333 332 0000 and delete the > message from your computer. If you are not a named addressee you > must not use, disclose, disseminate, distribute, copy, print or > reply to this email. Views or opinions expressed by an individual > within this email may not necessarily reflect the views of X-on or > its associated companies. Although X-on routinely screens for > viruses, addressees should scan this email and any attachments > for viruses. X-on makes no representation or warranty as to the > absence of viruses in this email or any attachments. > > > > > _______________________________________________ > > Users mailing list > > Users at lists.opensips.org > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -------------- next part -------------- An HTML attachment was scrubbed... URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: ~WRD084.jpg Type: image/jpeg Size: 823 bytes Desc: not available URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: image001.jpg Type: image/jpeg Size: 338 bytes Desc: not available URL: From bogdan at opensips.org Mon Feb 26 11:25:19 2018 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Mon, 26 Feb 2018 18:25:19 +0200 Subject: [OpenSIPS-Users] Wrong number of children In-Reply-To: <026201d3af1e$a7be42a0$f73ac7e0$@web.de> References: <027501d3aca1$66551340$32ff39c0$@web.de> <010001d3aef6$02f70f90$08e52eb0$@web.de> <4e50d0e9-d0c8-5317-4f21-433cbc5cde13@opensips.org> <01ff01d3af19$fda10db0$f8e32910$@web.de> <15812914-24ca-8dfa-9371-732ce42c91bd@opensips.org> <026201d3af1e$a7be42a0$f73ac7e0$@web.de> Message-ID: Hmm.... that is strange. Could you send me (off list) your cfg to give it a try - I see you use the 2.3.3 version , right ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Summit 2018 http://www.opensips.org/events/Summit-2018Amsterdam On 02/26/2018 06:27 PM, xaled wrote: > > Nothing > > root at dus-appsrv01:/etc/opensips# opensipsctl fifo ps > > Process:: ID=0 PID=14966 Type=attendant > > Process:: ID=1 PID=14967 Type=MI FIFO > > Process:: ID=2 PID=14968 Type=time_keeper > > Process:: ID=3 PID=14969 Type=timer > > Process:: ID=4 PID=14970 Type=SIP receiver udp:1.2.3.4:5060 > > Process:: ID=5 PID=14975 Type=SIP receiver udp:1.2.3.4:5060 > > Process:: ID=6 PID=14976 Type=SIP receiver udp:1.2.3.4:5060 > > Process:: ID=7 PID=14977 Type=SIP receiver udp:1.2.3.4:5060 > > Process:: ID=8 PID=14978 Type=SIP receiver udp:1.2.3.4:5060 > > Process:: ID=9 PID=14979 Type=TCP receiver > > Process:: ID=10 PID=14982 Type=TCP receiver > > Process:: ID=11 PID=14984 Type=TCP receiver > > Process:: ID=12 PID=14985 Type=TCP receiver > > Process:: ID=13 PID=14986 Type=TCP receiver > > Process:: ID=14 PID=14987 Type=TCP receiver > > Process:: ID=15 PID=14988 Type=TCP receiver > > Process:: ID=16 PID=14989 Type=TCP receiver > > Process:: ID=17 PID=14991 Type=Timer handler > > Process:: ID=18 PID=14992 Type=TCP main > > root at srv01:/etc/opensips# netstat -tlnp | grep opensips > > root at srv01:/etc/opensips# > > Thanks, > > Xaled > > *From:*Bogdan-Andrei Iancu [mailto:bogdan at opensips.org] > *Sent:* Monday, February 26, 2018 4:57 PM > *To:* xaled ; 'OpenSIPS users mailling list' > > *Subject:* Re: [OpenSIPS-Users] Wrong number of children > > If you do "netstat -tlnp | grep opensips", what do you get ? > > The TCP layer automatically enable itself if a TCP-based protocol is > used from the OpenSIPS config (as listener), like SIP TCP/TLS/WS/WSS , > HEP or BIN. > > Regards, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > OpenSIPS Summit 2018 > http://www.opensips.org/events/Summit-2018Amsterdam > > On 02/26/2018 05:53 PM, xaled wrote: > > Hi Bogdan, > > nope, non of them. > > Greetings, > > Xaled > > *From:*Bogdan-Andrei Iancu [mailto:bogdan at opensips.org] > *Sent:* Monday, February 26, 2018 4:41 PM > *To:* OpenSIPS users mailling list > ; xaled > > *Subject:* Re: [OpenSIPS-Users] Wrong number of children > > Hi Xaled, > > Do you use HEP for tracing or BIN for replication ? > > Regards, > > > Bogdan-Andrei Iancu > > > > OpenSIPS Founder and Developer > > http://www.opensips-solutions.com > > OpenSIPS Summit 2018 > > http://www.opensips.org/events/Summit-2018Amsterdam > > On 02/26/2018 01:36 PM, xaled wrote: > > Hi Callum, > > that helped thanks. > > I have 8 TCP receivers and one TCP main process that I did not > expected as I only listen on udp. > > Will have to understand the TCP spawn logic. > > Greetings, > > Xaled > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From xaled at web.de Mon Feb 26 11:27:11 2018 From: xaled at web.de (xaled) Date: Mon, 26 Feb 2018 17:27:11 +0100 Subject: [OpenSIPS-Users] Wrong number of children In-Reply-To: <15812914-24ca-8dfa-9371-732ce42c91bd@opensips.org> References: <027501d3aca1$66551340$32ff39c0$@web.de> <010001d3aef6$02f70f90$08e52eb0$@web.de> <4e50d0e9-d0c8-5317-4f21-433cbc5cde13@opensips.org> <01ff01d3af19$fda10db0$f8e32910$@web.de> <15812914-24ca-8dfa-9371-732ce42c91bd@opensips.org> Message-ID: <026201d3af1e$a7be42a0$f73ac7e0$@web.de> Nothing root at dus-appsrv01:/etc/opensips# opensipsctl fifo ps Process:: ID=0 PID=14966 Type=attendant Process:: ID=1 PID=14967 Type=MI FIFO Process:: ID=2 PID=14968 Type=time_keeper Process:: ID=3 PID=14969 Type=timer Process:: ID=4 PID=14970 Type=SIP receiver udp:1.2.3.4:5060 Process:: ID=5 PID=14975 Type=SIP receiver udp:1.2.3.4:5060 Process:: ID=6 PID=14976 Type=SIP receiver udp:1.2.3.4:5060 Process:: ID=7 PID=14977 Type=SIP receiver udp:1.2.3.4:5060 Process:: ID=8 PID=14978 Type=SIP receiver udp:1.2.3.4:5060 Process:: ID=9 PID=14979 Type=TCP receiver Process:: ID=10 PID=14982 Type=TCP receiver Process:: ID=11 PID=14984 Type=TCP receiver Process:: ID=12 PID=14985 Type=TCP receiver Process:: ID=13 PID=14986 Type=TCP receiver Process:: ID=14 PID=14987 Type=TCP receiver Process:: ID=15 PID=14988 Type=TCP receiver Process:: ID=16 PID=14989 Type=TCP receiver Process:: ID=17 PID=14991 Type=Timer handler Process:: ID=18 PID=14992 Type=TCP main root at srv01:/etc/opensips# netstat -tlnp | grep opensips root at srv01:/etc/opensips# Thanks, Xaled From: Bogdan-Andrei Iancu [mailto:bogdan at opensips.org] Sent: Monday, February 26, 2018 4:57 PM To: xaled ; 'OpenSIPS users mailling list' Subject: Re: [OpenSIPS-Users] Wrong number of children If you do "netstat -tlnp | grep opensips", what do you get ? The TCP layer automatically enable itself if a TCP-based protocol is used from the OpenSIPS config (as listener), like SIP TCP/TLS/WS/WSS , HEP or BIN. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Summit 2018 http://www.opensips.org/events/Summit-2018Amsterdam On 02/26/2018 05:53 PM, xaled wrote: Hi Bogdan, nope, non of them. Greetings, Xaled From: Bogdan-Andrei Iancu [mailto:bogdan at opensips.org] Sent: Monday, February 26, 2018 4:41 PM To: OpenSIPS users mailling list ; xaled Subject: Re: [OpenSIPS-Users] Wrong number of children Hi Xaled, Do you use HEP for tracing or BIN for replication ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Summit 2018 http://www.opensips.org/events/Summit-2018Amsterdam On 02/26/2018 01:36 PM, xaled wrote: Hi Callum, that helped thanks. I have 8 TCP receivers and one TCP main process that I did not expected as I only listen on udp. Will have to understand the TCP spawn logic. Greetings, Xaled From: Users [mailto:users-bounces at lists.opensips.org] On Behalf Of Callum Guy Sent: Friday, February 23, 2018 2:42 PM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] Wrong number of children Hi Xaled, If you have the FIFO module loaded you can run "opensipsctl fifo ps" on the instance to see full details of each process. https://www.opensips.org/Documentation/Interface-CoreMI-2-2#toc7 Hope that helps, Callum On Fri, Feb 23, 2018 at 1:35 PM Daniel Zanutti > wrote: Hi You should have 5 per interface + some internal control threads. I'm not sure exactly. Regards On Fri, Feb 23, 2018 at 9:25 AM, xaled > wrote: Hi I have configured 5 children in opensips.cfg, but 16 get logged and 18 initiated what could be the cause of it? I search for word children in opensips directory and it only comes ones in opensips.conf. children=5 Feb 23 13:07:09 srv01 opensips: DBG:core:daemonize: waiting for status code from children Feb 23 13:07:09 srv01 opensips: DBG:core:count_init_children: 16 children are going to be inited root at dus-appsrv01:/var/log/opensips# ps ax|grep sips 14854 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14856 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14857 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14858 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14861 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14862 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14863 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14864 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14865 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14867 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14869 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14870 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14871 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14872 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14874 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14875 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14876 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14877 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 With commended children parameter I get 19 children in the log and 21 in reality: Feb 23 13:13:24 dus-appsrv01 opensips: DBG:core:daemonize: waiting for status code from children Feb 23 13:13:24 dus-appsrv01 opensips: DBG:core:count_init_children: 19 children are going to be inited root at srv01:/var/log/opensips# ps ax|grep sips 14723 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14725 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14726 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14727 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14730 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14731 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14732 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14733 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14734 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14736 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14737 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14738 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14739 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14741 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14742 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14743 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14744 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14745 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14748 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14749 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 14750 ? S 0:00 /usr/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 64 -M 4 root at srv01:/etc/opensips# opensips -V version: opensips 2.3.3 (x86_64/linux) flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535 poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. main.c compiled on with gcc 6.3.0 Thanks, Xaled _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Callum Guy Head of Information Security X-on 0333 332 0000 | www.x-on.co.uk | X-on is a trading name of Storacall Technology Ltd a limited company registered in England and Wales. Registered Office : Avaland House, 110 London Road, Apsley, Hemel Hempstead, Herts, HP3 9SD. Company Registration No. 2578478. The information in this e-mail is confidential and for use by the addressee(s) only. If you are not the intended recipient, please notify X-on immediately on +44(0)333 332 0000 and delete the message from your computer. If you are not a named addressee you must not use, disclose, disseminate, distribute, copy, print or reply to this email. Views or opinions expressed by an individual within this email may not necessarily reflect the views of X-on or its associated companies. Although X-on routinely screens for viruses, addressees should scan this email and any attachments for viruses. X-on makes no representation or warranty as to the absence of viruses in this email or any attachments. _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: image001.jpg Type: image/jpeg Size: 823 bytes Desc: not available URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: image002.jpg Type: image/jpeg Size: 338 bytes Desc: not available URL: From sip.nslu at gmail.com Mon Feb 26 11:21:21 2018 From: sip.nslu at gmail.com (Ovidiu Sas) Date: Mon, 26 Feb 2018 11:21:21 -0500 Subject: [OpenSIPS-Users] On hold In-Reply-To: <01020161d2d39b50-ecde506f-9a4e-430e-9d46-6cdee6b8c2ab-000000@eu-west-1.amazonses.com> References: <01020161c1ef8de9-1fa3116b-b84a-4c0c-a3a9-f3419d72813d-000000@eu-west-1.amazonses.com> <01020161c2e60d3d-dedd2037-775c-45d2-adf1-5c7ae8ef3dfa-000000@eu-west-1.amazonses.com> <01020161d2d39b50-ecde506f-9a4e-430e-9d46-6cdee6b8c2ab-000000@eu-west-1.amazonses.com> Message-ID: You need to make sure that in your config you are not trying to re-route in dialog requests (like reINVITE). In dialog requests are routed according to the loose routing mechanism. -ovidiu On Feb 26, 2018 11:01, "Brian Southworth" wrote: Seems to be an issue with it sending the on hold invite to get the call back to the wrong place on inbound calls. Example caller calls in --- agent 1 picks up --- agent 1 places them on hold all is fine --- agent 1 takes call off hold call is dropped Seems that the invite to get the call back gets sent to .192 when it should go back to .193 This doesn’t happen is I use rewritehostport but then on hold calls only work for calls coming from that media server t_relay works across the board but doesn’t actually work for inbound. Regards, Brian Southworth *From:* Users [mailto:users-bounces at lists.opensips.org] *On Behalf Of *Ovidiu Sas *Sent:* 26 February 2018 15:49 *To:* users at lists.opensips.org users *Subject:* Re: [OpenSIPS-Users] On hold You need to take a look at the signalling and figure out who is dropping the call and why. -ovidiu On Feb 26, 2018 10:45, "Brian Southworth" wrote: Please ignore my last email. Both ways work it was an error in my code I wasn’t using t_relay(); Thanks for your help 😊 Regards, Brian Southworth *From:* Users [mailto:users-bounces at lists.opensips.org] *On Behalf Of *Ovidiu Sas *Sent:* 23 February 2018 13:10 *To:* users at lists.opensips.org users *Subject:* Re: [OpenSIPS-Users] On hold Hello Brian, You can detect hold using is_audio_on_hold(): http://www.opensips.org/html/docs/modules/2.3.x/sipmsgops#idp5575408 Try to use it and see if this will solve your issue. Regards, Ovidiu Sas On Feb 23, 2018 4:13 AM, "Brian Southworth" wrote: Hi All, I am trying to get opensips to forward the on hold request to asterisk I have done this using an if statement for SDP =~ “sendonly” How ever when I go to take them off hold the call is dropped, is there something special I need to do ? I have tried and if statement for is_method(“invite”) && $rb =~ “sendrecv” ß this loops until opensips give a 500 internal error And I have also tried is_method(“invite”) && $rb =~ “inactive” ß this doesn’t seem to work at all The initial hold works perfect puts them on hold just cant take them off or transfer the call (asterisk will handle all the B2B stuff) Any help would be appreciated. Regards, Brian Southworth Communications Developer [image: cid:image001.png at 01D22CAC.1DCB8580] 111 Wilmslow Road Handforth Wilmslow SK9 3ER T: 03333 446677 W: www.clocom.uk [image: cid:image002.png at 01CDDC62.D8483910] Like us on Facebook [image: cid:image003.png at 01CDDC62.D8483910] Follow us on Twitter [image: cid:image004.png at 01CDDC62.D8483910] Clocom is a *green* company. Think, do you need to print this email? This message contains confidential information and is intended only for the individual named. 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Registered name and address: Clocom UK Ltd, 111 Wilmslow Road, Handforth, Cheshire, SK9 3ER www.clocom.uk _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... 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Name: not available Type: image/png Size: 234 bytes Desc: not available URL: From Ben.Newlin at genesys.com Mon Feb 26 12:04:13 2018 From: Ben.Newlin at genesys.com (Ben Newlin) Date: Mon, 26 Feb 2018 17:04:13 +0000 Subject: [OpenSIPS-Users] OpenSIPS crashing on 200 OK In-Reply-To: <085c6c6b-75ca-0557-aa5e-0fcb9e3eb07f@opensips.org> References: <530E1611-2A4D-424F-938D-2E409E1A78A2@genesys.com> <7cf6bb3d-967a-b8e2-d4b3-8b42c8dbf0b8@opensips.org> <755F831E-90E4-4105-80C5-42C7A2E2050A@genesys.com> <6D9EBB18-914E-49ED-988F-7D582007B88E@genesys.com> <085c6c6b-75ca-0557-aa5e-0fcb9e3eb07f@opensips.org> Message-ID: Here you go: https://pastebin.com/rpUjaHBG Thanks, Ben Newlin From: Bogdan-Andrei Iancu Date: Monday, February 26, 2018 at 10:45 AM To: Ben Newlin , OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] OpenSIPS crashing on 200 OK OK, once you started GDB like "gdb /path/to/opensips /path/to/core", run step by step: f 0 p *s1 f 5 p *a Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Summit 2018 http://www.opensips.org/events/Summit-2018Amsterdam On 02/26/2018 05:43 PM, Ben Newlin wrote: Bogdan, Unfortunately I am not at all familiar with gdb. How do I change frames? Ben Newlin From: Bogdan-Andrei Iancu Date: Monday, February 26, 2018 at 10:40 AM To: Ben Newlin , OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] OpenSIPS crashing on 200 OK Hi Ben, The backtrace points to the script interpreter :D....Please do the followings (assuming you are a bit faimilar with GDB :) ) : in frame 0, please do "print *s1" and in frame 5 do "print *a" Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Summit 2018 http://www.opensips.org/events/Summit-2018Amsterdam On 02/22/2018 07:33 PM, Ben Newlin wrote: Bogdan, Here is the backtrace: https://pastebin.com/6tZhrWek I am not using TCP, BIN, or HEP in this configuration. I am not using anything, to my knowledge, which should be causing TCP worker threads to be spawned. Ben Newlin From: Bogdan-Andrei Iancu Date: Thursday, February 22, 2018 at 5:26 AM To: OpenSIPS users mailling list , Ben Newlin Subject: Re: [OpenSIPS-Users] OpenSIPS crashing on 200 OK Hi Ben, Do you have a full backtrace extracted from the core ? In regards to the TCP processes, the TCP layer will automatically enabled itself depending on what protocols (tcp-based) you are using in script. Maybe you do not use SIP with TCP, but do you use BIN or HEP ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Summit 2018 http://www.opensips.org/events/Summit-2018Amsterdam On 02/22/2018 12:38 AM, Ben Newlin wrote: Hi, I am currently upgrading to 2.3.3 and am running off the HEAD of the 2.3 branch. My OpenSIPS is consistently crashing when trying to process the ACK to an initial INVITE request. I am testing with SIPp using a simple single call scenario. I see the following in the logs: Feb 21 21:32:27 [383] DBG:core:io_wait_loop_epoll: [TCP_main] EPOLLHUP on IN ->connection closed by the remote peer! Feb 21 21:32:27 [383] CRITICAL:core:receive_fd: EOF on 33 Feb 21 21:32:27 [383] DBG:core:handle_worker: dead child 9, pid 370 (shutting down?) Feb 21 21:32:27 [383] DBG:core:io_watch_del: [TCP_main] io_watch_del op on index 8 33 (0x8d4e80, 33, 8, 0x0,0x1) fd_no=29 called Feb 21 21:32:27 [361] DBG:core:handle_sigs: status = 139 Feb 21 21:32:27 [361] INFO:core:handle_sigs: child process 370 exited by a signal 11 Feb 21 21:32:27 [361] INFO:core:handle_sigs: core was generated Feb 21 21:32:27 [361] INFO:core:handle_sigs: terminating due to SIGCHLD I also notice that OpenSIPS is starting up a number of TCP processes even though I do not use TCP in my script. I do not even have the proto_tcp module loaded. I used to use the “disable_tcp” global parameter in 1.11, but that has been removed in 2.3. Feb 21 22:03:56 [361] DBG:core:internal_fork: forking new process "SIP receiver TCP" Feb 21 22:03:56 [361] DBG:core:internal_fork: forking new process "SIP receiver TCP" Feb 21 22:03:56 [361] DBG:core:internal_fork: forking new process "SIP receiver TCP" Feb 21 22:03:56 [361] DBG:core:internal_fork: forking new process "SIP receiver TCP" Feb 21 22:03:56 [361] DBG:core:internal_fork: forking new process "SIP receiver TCP" Feb 21 22:03:56 [375] DBG:core:io_watch_add: [TCP_worker] io_watch_add op (3 on 53) (0x8d4e80, 3, 1, (nil),1), fd_no=0/209715 Feb 21 22:03:56 [361] DBG:core:internal_fork: forking new process "SIP receiver TCP" Feb 21 22:03:56 [361] DBG:core:internal_fork: forking new process "SIP receiver TCP" Feb 21 22:03:56 [361] DBG:core:internal_fork: forking new process "SIP receiver TCP" Initialization logs: https://pastebin.com/nsJQGkdy Call logs: https://pastebin.com/vY5crK8U I also have the core dump if any more information is needed from that. Thanks, Ben Newlin _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From govoiper at gmail.com Mon Feb 26 12:07:17 2018 From: govoiper at gmail.com (SamyGo) Date: Mon, 26 Feb 2018 12:07:17 -0500 Subject: [OpenSIPS-Users] ACC module with JSON events In-Reply-To: References: <1519631511.3267.4.camel@gmail.com> Message-ID: Hi DanB, I can certainly try CGRates, but...surely enough its an overkill for simple CDRs. Anyways, my request is for an event_JSONRPC which maybe very helpful in events outside of ACC. Best Regards, Sammy On Mon, Feb 26, 2018 at 6:13 AM, Khalil Khamlichi < khamlichi.khalil at gmail.com> wrote: > Hi Dan, > > Can you show us how to do that with some step by step tutorial, we had > a lot of trouble trying to figure out how to use CGRates, with this > functionality of json export, we will at least use some of CGRates > functionalities and hopefully slowly get familiar with it. > > Thanks in advance. > > On Mon, Feb 26, 2018 at 7:51 AM, DanB wrote: > > Sammy, > > > > Another option on short term until the new feature will be implemented > > in OpenSIPS would be to use CGRateS as CDR format converter: receive > > CDRs from "cgrates" module in OpenSIPS and use online export of CGRateS > > to further export the CDR in the JSON over http (customizable fields). > > You don't need to configure much on CGRateS side in this case since no > > billing needs to be involved. > > > > DanB > > > > > > > > > > > > _______________________________________________ > > Users mailing list > > Users at lists.opensips.org > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Mon Feb 26 12:15:53 2018 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Mon, 26 Feb 2018 19:15:53 +0200 Subject: [OpenSIPS-Users] OpenSIPS crashing on 200 OK In-Reply-To: References: <530E1611-2A4D-424F-938D-2E409E1A78A2@genesys.com> <7cf6bb3d-967a-b8e2-d4b3-8b42c8dbf0b8@opensips.org> <755F831E-90E4-4105-80C5-42C7A2E2050A@genesys.com> <6D9EBB18-914E-49ED-988F-7D582007B88E@genesys.com> <085c6c6b-75ca-0557-aa5e-0fcb9e3eb07f@opensips.org> Message-ID: Could you off-list pastme the line 435 of your cfg - I guess you use some $DLG_dir there ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Summit 2018 http://www.opensips.org/events/Summit-2018Amsterdam On 02/26/2018 07:04 PM, Ben Newlin wrote: > > Here you go: https://pastebin.com/rpUjaHBG > > Thanks, > > Ben Newlin > > *From: *Bogdan-Andrei Iancu > *Date: *Monday, February 26, 2018 at 10:45 AM > *To: *Ben Newlin , OpenSIPS users mailling > list > *Subject: *Re: [OpenSIPS-Users] OpenSIPS crashing on 200 OK > > OK, once you started GDB like "gdb /path/to/opensips /path/to/core", > run step by step: > > f 0 > p *s1 > f 5 > p *a > > Regards, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > OpenSIPS Summit 2018 > http://www.opensips.org/events/Summit-2018Amsterdam > > On 02/26/2018 05:43 PM, Ben Newlin wrote: > > Bogdan, > > Unfortunately I am not at all familiar with gdb. How do I change > frames? > > Ben Newlin > > *From: *Bogdan-Andrei Iancu > > *Date: *Monday, February 26, 2018 at 10:40 AM > *To: *Ben Newlin > , OpenSIPS users mailling list > > *Subject: *Re: [OpenSIPS-Users] OpenSIPS crashing on 200 OK > > Hi Ben, > > The backtrace points to the script interpreter :D....Please do the > followings (assuming you are a bit faimilar with GDB :) ) : > > in frame 0, please do "print *s1" > and in frame 5 do "print *a" > > Regards, > -------------- next part -------------- An HTML attachment was scrubbed... URL: From Ben.Newlin at genesys.com Mon Feb 26 13:10:56 2018 From: Ben.Newlin at genesys.com (Ben Newlin) Date: Mon, 26 Feb 2018 18:10:56 +0000 Subject: [OpenSIPS-Users] OpenSIPS crashing on 200 OK In-Reply-To: References: <530E1611-2A4D-424F-938D-2E409E1A78A2@genesys.com> <7cf6bb3d-967a-b8e2-d4b3-8b42c8dbf0b8@opensips.org> <755F831E-90E4-4105-80C5-42C7A2E2050A@genesys.com> <6D9EBB18-914E-49ED-988F-7D582007B88E@genesys.com> <085c6c6b-75ca-0557-aa5e-0fcb9e3eb07f@opensips.org> Message-ID: <4FD028FE-6607-496C-8B3A-EF4293FB6155@genesys.com> Hmm, no. Line 435 of my config is a return statement. return(-1); Ben Newlin From: Bogdan-Andrei Iancu Date: Monday, February 26, 2018 at 12:16 PM To: Ben Newlin , OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] OpenSIPS crashing on 200 OK Could you off-list past me the line 435 of your cfg - I guess you use some $DLG_dir there ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Summit 2018 http://www.opensips.org/events/Summit-2018Amsterdam On 02/26/2018 07:04 PM, Ben Newlin wrote: Here you go: https://pastebin.com/rpUjaHBG Thanks, Ben Newlin From: Bogdan-Andrei Iancu Date: Monday, February 26, 2018 at 10:45 AM To: Ben Newlin , OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] OpenSIPS crashing on 200 OK OK, once you started GDB like "gdb /path/to/opensips /path/to/core", run step by step: f 0 p *s1 f 5 p *a Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Summit 2018 http://www.opensips.org/events/Summit-2018Amsterdam On 02/26/2018 05:43 PM, Ben Newlin wrote: Bogdan, Unfortunately I am not at all familiar with gdb. How do I change frames? Ben Newlin From: Bogdan-Andrei Iancu Date: Monday, February 26, 2018 at 10:40 AM To: Ben Newlin , OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] OpenSIPS crashing on 200 OK Hi Ben, The backtrace points to the script interpreter :D....Please do the followings (assuming you are a bit faimilar with GDB :) ) : in frame 0, please do "print *s1" and in frame 5 do "print *a" Regards, -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Mon Feb 26 16:12:56 2018 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Mon, 26 Feb 2018 23:12:56 +0200 Subject: [OpenSIPS-Users] OpenSIPS crashing on 200 OK In-Reply-To: <4FD028FE-6607-496C-8B3A-EF4293FB6155@genesys.com> References: <530E1611-2A4D-424F-938D-2E409E1A78A2@genesys.com> <7cf6bb3d-967a-b8e2-d4b3-8b42c8dbf0b8@opensips.org> <755F831E-90E4-4105-80C5-42C7A2E2050A@genesys.com> <6D9EBB18-914E-49ED-988F-7D582007B88E@genesys.com> <085c6c6b-75ca-0557-aa5e-0fcb9e3eb07f@opensips.org> <4FD028FE-6607-496C-8B3A-EF4293FB6155@genesys.com> Message-ID: OK,but check the exact file you pass to Opensips - according to the debug info is /opensips/control.cfg like 435. Also, do you use the $DLG_dir variable in your cfg ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Summit 2018 http://www.opensips.org/events/Summit-2018Amsterdam On 02/26/2018 08:10 PM, Ben Newlin wrote: > > Hmm, no. Line 435 of my config is a return statement. > > return(-1); > > Ben Newlin > > *From: *Bogdan-Andrei Iancu > *Date: *Monday, February 26, 2018 at 12:16 PM > *To: *Ben Newlin , OpenSIPS users mailling > list > *Subject: *Re: [OpenSIPS-Users] OpenSIPS crashing on 200 OK > > Could you off-list past me the line 435 of your cfg - I guess you use > some $DLG_dir there ? > > Regards, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > OpenSIPS Summit 2018 > http://www.opensips.org/events/Summit-2018Amsterdam > > On 02/26/2018 07:04 PM, Ben Newlin wrote: > > Here you go: https://pastebin.com/rpUjaHBG > > > Thanks, > > Ben Newlin > > *From: *Bogdan-Andrei Iancu > > *Date: *Monday, February 26, 2018 at 10:45 AM > *To: *Ben Newlin > , OpenSIPS users mailling list > > *Subject: *Re: [OpenSIPS-Users] OpenSIPS crashing on 200 OK > > OK, once you started GDB like "gdb /path/to/opensips > /path/to/core", run step by step: > > f 0 > p *s1 > f 5 > p *a > > Regards, > > > Bogdan-Andrei Iancu > > > > OpenSIPS Founder and Developer > > http://www.opensips-solutions.com > > OpenSIPS Summit 2018 > > http://www.opensips.org/events/Summit-2018Amsterdam > > On 02/26/2018 05:43 PM, Ben Newlin wrote: > > Bogdan, > > Unfortunately I am not at all familiar with gdb. How do I > change frames? > > Ben Newlin > > *From: *Bogdan-Andrei Iancu > > *Date: *Monday, February 26, 2018 at 10:40 AM > *To: *Ben Newlin > , OpenSIPS users mailling list > > *Subject: *Re: [OpenSIPS-Users] OpenSIPS crashing on 200 OK > > Hi Ben, > > The backtrace points to the script interpreter :D....Please do > the followings (assuming you are a bit faimilar with GDB :) ) : > > in frame 0, please do "print *s1" > and in frame 5 do "print *a" > > Regards, > > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From Ben.Newlin at genesys.com Mon Feb 26 16:32:27 2018 From: Ben.Newlin at genesys.com (Ben Newlin) Date: Mon, 26 Feb 2018 21:32:27 +0000 Subject: [OpenSIPS-Users] OpenSIPS crashing on 200 OK In-Reply-To: References: <530E1611-2A4D-424F-938D-2E409E1A78A2@genesys.com> <7cf6bb3d-967a-b8e2-d4b3-8b42c8dbf0b8@opensips.org> <755F831E-90E4-4105-80C5-42C7A2E2050A@genesys.com> <6D9EBB18-914E-49ED-988F-7D582007B88E@genesys.com> <085c6c6b-75ca-0557-aa5e-0fcb9e3eb07f@opensips.org> <4FD028FE-6607-496C-8B3A-EF4293FB6155@genesys.com> Message-ID: Yes, that is from that exact file, line 435. It’s just a return statement. I can send you the file off-list if you’d like. I do use the $DLG_dir in my script in a few places, but the only one this crash would have hit is in some logging. Ben Newlin From: Bogdan-Andrei Iancu Date: Monday, February 26, 2018 at 4:14 PM To: Ben Newlin , OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] OpenSIPS crashing on 200 OK OK, but check the exact file you pass to Opensips - according to the debug info is /opensips/control.cfg like 435. Also, do you use the $DLG_dir variable in your cfg ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Summit 2018 http://www.opensips.org/events/Summit-2018Amsterdam On 02/26/2018 08:10 PM, Ben Newlin wrote: Hmm, no. Line 435 of my config is a return statement. return(-1); Ben Newlin From: Bogdan-Andrei Iancu Date: Monday, February 26, 2018 at 12:16 PM To: Ben Newlin , OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] OpenSIPS crashing on 200 OK Could you off-list past me the line 435 of your cfg - I guess you use some $DLG_dir there ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Summit 2018 http://www.opensips.org/events/Summit-2018Amsterdam On 02/26/2018 07:04 PM, Ben Newlin wrote: Here you go: https://pastebin.com/rpUjaHBG Thanks, Ben Newlin From: Bogdan-Andrei Iancu Date: Monday, February 26, 2018 at 10:45 AM To: Ben Newlin , OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] OpenSIPS crashing on 200 OK OK, once you started GDB like "gdb /path/to/opensips /path/to/core", run step by step: f 0 p *s1 f 5 p *a Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Summit 2018 http://www.opensips.org/events/Summit-2018Amsterdam On 02/26/2018 05:43 PM, Ben Newlin wrote: Bogdan, Unfortunately I am not at all familiar with gdb. How do I change frames? Ben Newlin From: Bogdan-Andrei Iancu Date: Monday, February 26, 2018 at 10:40 AM To: Ben Newlin , OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] OpenSIPS crashing on 200 OK Hi Ben, The backtrace points to the script interpreter :D....Please do the followings (assuming you are a bit faimilar with GDB :) ) : in frame 0, please do "print *s1" and in frame 5 do "print *a" Regards, -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Mon Feb 26 17:12:08 2018 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Tue, 27 Feb 2018 00:12:08 +0200 Subject: [OpenSIPS-Users] OpenSIPS crashing on 200 OK In-Reply-To: References: <530E1611-2A4D-424F-938D-2E409E1A78A2@genesys.com> <7cf6bb3d-967a-b8e2-d4b3-8b42c8dbf0b8@opensips.org> <755F831E-90E4-4105-80C5-42C7A2E2050A@genesys.com> <6D9EBB18-914E-49ED-988F-7D582007B88E@genesys.com> <085c6c6b-75ca-0557-aa5e-0fcb9e3eb07f@opensips.org> <4FD028FE-6607-496C-8B3A-EF4293FB6155@genesys.com> Message-ID: OK, to be more precise, the crash is around an "If" statement where you have in the condition an REGEXP test (=~) with a left operand translated into "downstream" (maybe a $DLG_dir ??). Does it sound familiar to you :) ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Summit 2018 http://www.opensips.org/events/Summit-2018Amsterdam On 02/26/2018 11:32 PM, Ben Newlin wrote: > > Yes, that is from that exact file, line 435. It’s just a return > statement. I can send you the file off-list if you’d like. > > I do use the $DLG_dir in my script in a few places, but the only one > this crash would have hit is in some logging. > > Ben Newlin > > *From: *Bogdan-Andrei Iancu > *Date: *Monday, February 26, 2018 at 4:14 PM > *To: *Ben Newlin , OpenSIPS users mailling > list > *Subject: *Re: [OpenSIPS-Users] OpenSIPS crashing on 200 OK > > OK, but check the exact file you pass to Opensips - according to the > debug info is /opensips/control.cfg like 435. > > Also, do you use the $DLG_dir variable in your cfg ? > > Regards, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > OpenSIPS Summit 2018 > http://www.opensips.org/events/Summit-2018Amsterdam > > On 02/26/2018 08:10 PM, Ben Newlin wrote: > > Hmm, no. Line 435 of my config is a return statement. > > return(-1); > > Ben Newlin > > *From: *Bogdan-Andrei Iancu > > *Date: *Monday, February 26, 2018 at 12:16 PM > *To: *Ben Newlin > , OpenSIPS users mailling list > > *Subject: *Re: [OpenSIPS-Users] OpenSIPS crashing on 200 OK > > Could you off-list past me the line 435 of your cfg - I guess you > use some $DLG_dir there ? > > Regards, > > > Bogdan-Andrei Iancu > > > > OpenSIPS Founder and Developer > > http://www.opensips-solutions.com > > OpenSIPS Summit 2018 > > http://www.opensips.org/events/Summit-2018Amsterdam > > On 02/26/2018 07:04 PM, Ben Newlin wrote: > > Here you go: https://pastebin.com/rpUjaHBG > > > Thanks, > > Ben Newlin > > *From: *Bogdan-Andrei Iancu > > *Date: *Monday, February 26, 2018 at 10:45 AM > *To: *Ben Newlin > , OpenSIPS users mailling list > > *Subject: *Re: [OpenSIPS-Users] OpenSIPS crashing on 200 OK > > OK, once you started GDB like "gdb /path/to/opensips > /path/to/core", run step by step: > > f 0 > p *s1 > f 5 > p *a > > Regards, > > > > Bogdan-Andrei Iancu > > > > OpenSIPS Founder and Developer > > http://www.opensips-solutions.com > > OpenSIPS Summit 2018 > > http://www.opensips.org/events/Summit-2018Amsterdam > > On 02/26/2018 05:43 PM, Ben Newlin wrote: > > Bogdan, > > Unfortunately I am not at all familiar with gdb. How do I > change frames? > > Ben Newlin > > *From: *Bogdan-Andrei Iancu > > *Date: *Monday, February 26, 2018 at 10:40 AM > *To: *Ben Newlin > , OpenSIPS users mailling > list > > *Subject: *Re: [OpenSIPS-Users] OpenSIPS crashing on 200 OK > > Hi Ben, > > The backtrace points to the script interpreter > :D....Please do the followings (assuming you are a bit > faimilar with GDB :) ) : > > in frame 0, please do "print *s1" > and in frame 5 do "print *a" > > Regards, > > > > > > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From pasandev at ymail.com Mon Feb 26 21:15:41 2018 From: pasandev at ymail.com (Pasan Meemaduma) Date: Tue, 27 Feb 2018 02:15:41 +0000 (UTC) Subject: [OpenSIPS-Users] dialog replication In-Reply-To: <704df948-6873-5fde-fe25-193f1cf2a420@opensips.org> References: <311827884.3546251.1519627085908.ref@mail.yahoo.com> <311827884.3546251.1519627085908@mail.yahoo.com> <704df948-6873-5fde-fe25-193f1cf2a420@opensips.org> Message-ID: <2103041839.6602135.1519697741901@mail.yahoo.com> Hi Vlad, I have sent you the full debug logs as requested. Also clusterer_list mi command gives following output when dialog replication stopped working. node1>>opensipsctl fifo clusterer_list Cluster:: 1         Node:: 2 DB_ID=5 URL=bin:10.3.1.137:5566 Enabled=1 Link_state=Up      Next_hop=2 Description=Node 2  node2>> opensipsctl fifo clusterer_list Cluster:: 1         Node:: 1 DB_ID=4 URL=bin:10.3.1.136:5566 Enabled=1 Link_state=Up      Next_hop=1 Description=Node 1 1 On Monday, 26 February 2018, 18:02, Vlad Patrascu wrote: Hi, Can you send the full logs for both nodes from the time that node1 restarts onwards? Also, what is the output of 'clusterer_list' mi command on the instances? Regards, Vlad Patrascu OpenSIPS Developer http://www.opensips-solutions.com On 26.02.2018 08:38, Pasan Meemaduma via Users wrote: Hi Guys, Its me again :). I'm using dialog replication in opensips 2.3.3 and it appears it doesn't work after a node goes down and come back online. The recovered node doesn't seems to receiving dialog info via binary interface. I have node1 and node2 with dialog replication on. everything works fine and if I shutdown node1 and bring it back online after a while node2 doesn't send new call dialog info via binary interface. I have attach the debug trace from both nodes, if you need anything else let me know. I'm also using an anycast ip as the service ip. on node2 for new call Feb 26 17:28:35 voip2-sip23b /usr/sbin/opensips[4703]: DBG:dialog:build_new_dlg: new dialog 0x7f188afd3bf8(c=ktQ0Pkdwz50qGYjED6Brpw..,f=sip:XXXXXXXXX at somedomain;transport=UDP,t=sip:+YYYYYYYYYYY at somedomain;transport=UDP,ft=2b161508) on hash 2317 Feb 26 17:28:35 voip2-sip23b /usr/sbin/opensips[4703]: DBG:dialog:init_leg_info: route_set , contact sip:XXXXXXXXX at 192.168.27.11:56419;transport=UDP, cseq 2 and bind_addr udp:10.3.3.1:5060 Feb 26 17:28:35 voip2-sip23b /usr/sbin/opensips[4703]: DBG:dialog:dlg_add_leg_info: set leg 0 for 0x7f188afd3bf8: tag=<2b161508> rcseq=<0> Feb 26 17:28:35 voip2-sip23b /usr/sbin/opensips[4703]: DBG:dialog:link_dlg: ref dlg 0x7f188afd3bf8 with 3 -> 3 in h_entry 0x7f188afc3828 - 2317 Feb 26 17:28:35 voip2-sip23b /usr/sbin/opensips[4703]: DBG:dialog:new_dlg_val: inserting =<45ad4c76-1abe-11e8-9410-831894b67d0c> Feb 26 17:28:35 voip2-sip23b /usr/sbin/opensips[4703]: DBG:dialog:dlg_onreq: t hash_index = 47425, t label = 2069082013 Feb 26 17:28:35 voip2-sip23b /usr/sbin/opensips[4703]: DBG:dialog:dlg_update_contact: Updated dialog 0x7f188afd3bf8 contact to Feb 26 17:28:35 voip2-sip23b /usr/sbin/opensips[4703]: DBG:dialog:unref_dlg: unref dlg 0x7f188afd3bf8 with 1 -> 2 in entry 0x7f188afc3828 Feb 26 17:28:52 voip2-sip23b /usr/sbin/opensips[4702]: DBG:dialog:ref_dlg: ref dlg 0x7f188afd3bf8 with 1 -> 3 Feb 26 17:28:52 voip2-sip23b /usr/sbin/opensips[4702]: DBG:dialog:ref_dlg: ref dlg 0x7f188afd3bf8 with 1 -> 4 Feb 26 17:28:52 voip2-sip23b /usr/sbin/opensips[4702]: DBG:dialog:unref_dlg: unref dlg 0x7f188afd3bf8 with 1 -> 3 in entry 0x7f188afc3828 Feb 26 17:28:56 voip2-sip23b /usr/sbin/opensips[4711]: DBG:dialog:unref_dlg: unref dlg 0x7f188afd3bf8 with 1 -> 2 in entry 0x7f188afc3828 Feb 26 17:29:05 voip2-sip23b /usr/sbin/opensips[4711]: DBG:dialog:next_state_dlg: dialog 0x7f188afd3bf8 changed from state 1 to state 5, due event 4 Feb 26 17:29:05 voip2-sip23b /usr/sbin/opensips[4711]: DBG:dialog:dlg_onreply: dialog 0x7f188afd3bf8 failed (negative reply) Feb 26 17:29:05 voip2-sip23b /usr/sbin/opensips[4711]: DBG:dialog:unref_dlg: unref dlg 0x7f188afd3bf8 with 1 -> 1 in entry 0x7f188afc3828 Feb 26 17:29:11 voip2-sip23b /usr/sbin/opensips[4711]: DBG:dialog:next_state_dlg: dialog 0x7f188afd3bf8 changed from state 5 to state 5, due event 1 Feb 26 17:29:11 voip2-sip23b /usr/sbin/opensips[4711]: DBG:dialog:unref_dlg: unref dlg 0x7f188afd3bf8 with 1 -> 0 in entry 0x7f188afc3828 Feb 26 17:29:11 voip2-sip23b /usr/sbin/opensips[4711]: DBG:dialog:unref_dlg: ref <=0 for dialog 0x7f188afd3bf8 Feb 26 17:29:11 voip2-sip23b /usr/sbin/opensips[4711]: DBG:dialog:destroy_dlg: destroying dialog 0x7f188afd3bf8 Feb 26 17:29:11 voip2-sip23b /usr/sbin/opensips[4711]: DBG:dialog:destroy_dlg: dlg expired or not in list - dlg 0x7f188afd3bf8 [2317:677053409] with clid 'ktQ0Pkdwz50qGYjED6Brpw..' and tags '2b161508' 'NULL' on node1 Feb 26 17:28:41 voip2-sip23a /usr/sbin/opensips[15820]: DBG:dialog:dlg_onroute: route param is 'd09.1e30b582' (len=12) Feb 26 17:28:41 voip2-sip23a /usr/sbin/opensips[15820]: DBG:dialog:lookup_dlg: no dialog id=677053409 found on entry 2317 Feb 26 17:28:41 voip2-sip23a /usr/sbin/opensips[15820]: DBG:dialog:dlg_onroute: unable to find dialog for BYE with route param 'd09.1e30b582' Feb 26 17:28:41 voip2-sip23a /usr/sbin/opensips[15822]: DBG:dialog:dlg_onroute: route param is 'd09.1e30b582' (len=12) Feb 26 17:28:41 voip2-sip23a /usr/sbin/opensips[15822]: DBG:dialog:lookup_dlg: no dialog id=677053409 found on entry 2317 Feb 26 17:28:41 voip2-sip23a /usr/sbin/opensips[15822]: DBG:dialog:dlg_onroute: unable to find dialog for BYE with route param 'd09.1e30b582' Feb 26 17:28:41 voip2-sip23a /usr/sbin/opensips[15822]: DBG:dialog:dlg_onroute: route param is 'd09.1e30b582' (len=12) Feb 26 17:28:41 voip2-sip23a /usr/sbin/opensips[15822]: DBG:dialog:lookup_dlg: no dialog id=677053409 found on entry 2317 Feb 26 17:28:41 voip2-sip23a /usr/sbin/opensips[15822]: DBG:dialog:dlg_onroute: unable to find dialog for BYE with route param 'd09.1e30b582' Feb 26 17:28:42 voip2-sip23a /usr/sbin/opensips[15820]: DBG:dialog:dlg_onroute: route param is 'd09.1e30b582' (len=12) Feb 26 17:28:42 voip2-sip23a /usr/sbin/opensips[15820]: DBG:dialog:lookup_dlg: no dialog id=677053409 found on entry 2317 Feb 26 17:28:42 voip2-sip23a /usr/sbin/opensips[15820]: DBG:dialog:dlg_onroute: unable to find dialog for BYE with route param 'd09.1e30b582' Feb 26 17:28:42 voip2-sip23a /usr/sbin/opensips[15820]: DBG:dialog:dlg_onroute: route param is 'd09.1e30b582' (len=12) Feb 26 17:28:42 voip2-sip23a /usr/sbin/opensips[15820]: DBG:dialog:lookup_dlg: no dialog id=677053409 found on entry 2317 If I purposely take down node1 again and try to make a call I can see that it indicates the failure to replicate dialog info on the cluster as below, Feb 26 17:27:01 voip2-sip23b /usr/sbin/opensips[4701]: ERROR:dialog:replicate_dialog_updated: All destinations in cluster: 1 are down or probing Feb 26 17:27:01 voip2-sip23b /usr/sbin/opensips[4701]: ERROR:dialog:replicate_dialog_updated: Failed to replicate updated dialog _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From rutu.patel at ecosmob.com Tue Feb 27 02:16:37 2018 From: rutu.patel at ecosmob.com (Rutu Patel) Date: Tue, 27 Feb 2018 12:46:37 +0530 Subject: [OpenSIPS-Users] TLS calls not working Message-ID: Hi All, I have installed opensips-2.3.2 on centos and followed https://www.opensips.org/Documentation/Tutorials-TLS-2-1 AND http://www.opensips.org/html/docs/modules/2.3.x/tls_mgm.html to generate self-signed certificates and TLS setup. I want to achieve scenario asterisk(TLS) -> opensips(TLS) -> asterisk(TLS) What should be the certificate settings for this kind of setup? My opensips.cfg configurations for TLS are as below: modparam("proto_udp", "udp_port", 5060) modparam("tls_mgm", "verify_cert", "1") modparam("tls_mgm", "require_cert", "0") modparam("tls_mgm", "tls_method", "TLSv1") modparam("tls_mgm", "certificate", "/usr/local/etc/opensips/tls_cnf/tls/rootCA/cacert.pem") modparam("tls_mgm", "private_key", "/usr/local/etc/opensips/tls_cnf/tls/rootCA/private/cakey1.pem") modparam("tls_mgm", "certificate", "/usr/local/etc/opensips/tls_cnf/tls/rootCA/cacert.pem") modparam("tls_mgm", "ca_list", "/usr/local/etc/opensips/tls_cnf/tls/rootCA/cacert.pem") modparam("tls_mgm", "client_domain", "dom1=172.16.16.149:5080") modparam("tls_mgm", "private_key", "[dom1]/usr/local/etc/opensips/tls_cnf/asterisk_149.pem") modparam("tls_mgm", "certificate", "[dom1]/usr/local/etc/opensips/tls_cnf/asterisk_149.pem") modparam("tls_mgm", "ca_list", "[dom1]/usr/local/etc/opensips/tls_cnf/asterisk_149.pem") *asterisk_149.pem file is imported from asterisk server.Asterisk server is listening on port 5080 for TLS. When I am setting "verify_cert" to "0",calls working but on setting its value to "1", opensips gives below errors: ERROR:proto_tls:tls_accept: New TLS connection from 172.16.16.149:34678 failed to accept ERROR:proto_tls:tls_print_errstack: TLS errstack: error:14089086:SSL routines:ssl3_get_client_certificate:certificate verify failed ERROR:proto_tls:tls_read_req: failed to do pre-tls reading Please provide guidance to solve this issue. Thank you, Rutu Patel -------------- next part -------------- An HTML attachment was scrubbed... URL: From rinorhoxha at gmail.com Tue Feb 27 10:23:04 2018 From: rinorhoxha at gmail.com (Rinor Hoxha) Date: Tue, 27 Feb 2018 16:23:04 +0100 Subject: [OpenSIPS-Users] Wrong number of children In-Reply-To: <026201d3af1e$a7be42a0$f73ac7e0$@web.de> References: <027501d3aca1$66551340$32ff39c0$@web.de> <010001d3aef6$02f70f90$08e52eb0$@web.de> <4e50d0e9-d0c8-5317-4f21-433cbc5cde13@opensips.org> <01ff01d3af19$fda10db0$f8e32910$@web.de> <15812914-24ca-8dfa-9371-732ce42c91bd@opensips.org> <026201d3af1e$a7be42a0$f73ac7e0$@web.de> Message-ID: <3aabaca1-36a8-9229-714e-6d34767c0b9e@gmail.com> Hi Xaled, Try adding *tcp_children=0* to the config file. Br, Rinor -------------- next part -------------- An HTML attachment was scrubbed... URL: From vladp at opensips.org Tue Feb 27 11:11:15 2018 From: vladp at opensips.org (Vlad Patrascu) Date: Tue, 27 Feb 2018 18:11:15 +0200 Subject: [OpenSIPS-Users] dialog replication In-Reply-To: <2103041839.6602135.1519697741901@mail.yahoo.com> References: <311827884.3546251.1519627085908.ref@mail.yahoo.com> <311827884.3546251.1519627085908@mail.yahoo.com> <704df948-6873-5fde-fe25-193f1cf2a420@opensips.org> <2103041839.6602135.1519697741901@mail.yahoo.com> Message-ID: <48b7be0c-d5d9-b6f3-a81b-42d768116f07@opensips.org> Hi Pasan, I don't see anything in the logs or your cluster configuration which could indicate that the dialogs don't replicate. Are you sure that the problem is consistently reproducible and that the INVITE for the call in question receives a final reply? Dialogs are replicated when they get to the confirmed state, and judging from the logs, this doesn't appear to be happening for that call. Regards, Vlad Patrascu OpenSIPS Developer http://www.opensips-solutions.com On 27.02.2018 04:15, Pasan Meemaduma via Users wrote: > Hi Vlad, > > I have sent you the full debug logs as requested. Also clusterer_list > mi command gives following output when dialog replication stopped working. > > node1>>opensipsctl fifo clusterer_list > Cluster:: 1 >         Node:: 2 DB_ID=5 URL=bin:10.3.1.137:5566 Enabled=1 > Link_state=Up      Next_hop=2 Description=Node 2 > > >  node2>> opensipsctl fifo clusterer_list > Cluster:: 1 >         Node:: 1 DB_ID=4 URL=bin:10.3.1.136:5566 Enabled=1 > Link_state=Up      Next_hop=1 Description=Node 1 > 1 > > > > On Monday, 26 February 2018, 18:02, Vlad Patrascu > wrote: > > > Hi, > Can you send the full logs for both nodes from the time that node1 > restarts onwards? Also, what is the output of 'clusterer_list' mi > command on the instances? > Regards, > Vlad Patrascu > OpenSIPS Developer > http://www.opensips-solutions.com > On 26.02.2018 08:38, Pasan Meemaduma via Users wrote: >> Hi Guys, >> >> Its me again :). I'm using dialog replication in opensips 2.3.3 and >> it appears it doesn't work after a node goes down and come back >> online. The recovered node >> doesn't seems to receiving dialog info via binary interface. >> >> I have node1 and node2 with dialog replication on. everything works >> fine and if I shutdown node1 and bring it back online after a while >> node2 doesn't send new call dialog info via binary interface. >> >> I have attach the debug trace from both nodes, if you need anything >> else let me know. I'm also using an anycast ip as the service ip. >> >> >> on node2 for new call >> >> Feb 26 17:28:35 voip2-sip23b /usr/sbin/opensips[4703]: >> DBG:dialog:build_new_dlg: new dialog 0x7f188afd3bf8 >> (c=ktQ0Pkdwz50qGYjED6Brpw..,f=sip:XXXXXXXXX at somedomain;transport=UDP,t=sip:+YYYYYYYYYYY at somedomain;transport=UDP,ft=2b161508) >> on hash 2317 >> Feb 26 17:28:35 voip2-sip23b /usr/sbin/opensips[4703]: >> DBG:dialog:init_leg_info: route_set , contact >> sip:XXXXXXXXX at 192.168.27.11:56419;transport=UDP >> , cseq 2 and >> bind_addr udp:10.3.3.1:5060 >> Feb 26 17:28:35 voip2-sip23b /usr/sbin/opensips[4703]: >> DBG:dialog:dlg_add_leg_info: set leg 0 for 0x7f188afd3bf8: >> tag=<2b161508> rcseq=<0> >> Feb 26 17:28:35 voip2-sip23b /usr/sbin/opensips[4703]: >> DBG:dialog:link_dlg: ref dlg 0x7f188afd3bf8 with 3 -> 3 in h_entry >> 0x7f188afc3828 - 2317 >> Feb 26 17:28:35 voip2-sip23b /usr/sbin/opensips[4703]: >> DBG:dialog:new_dlg_val: inserting >> =<45ad4c76-1abe-11e8-9410-831894b67d0c> >> Feb 26 17:28:35 voip2-sip23b /usr/sbin/opensips[4703]: >> DBG:dialog:dlg_onreq: t hash_index = 47425, t label = 2069082013 >> Feb 26 17:28:35 voip2-sip23b /usr/sbin/opensips[4703]: >> DBG:dialog:dlg_update_contact: Updated dialog 0x7f188afd3bf8 contact >> to >> >> Feb 26 17:28:35 voip2-sip23b /usr/sbin/opensips[4703]: >> DBG:dialog:unref_dlg: unref dlg 0x7f188afd3bf8 with 1 -> 2 in entry >> 0x7f188afc3828 >> Feb 26 17:28:52 voip2-sip23b /usr/sbin/opensips[4702]: >> DBG:dialog:ref_dlg: ref dlg 0x7f188afd3bf8 with 1 -> 3 >> Feb 26 17:28:52 voip2-sip23b /usr/sbin/opensips[4702]: >> DBG:dialog:ref_dlg: ref dlg 0x7f188afd3bf8 with 1 -> 4 >> Feb 26 17:28:52 voip2-sip23b /usr/sbin/opensips[4702]: >> DBG:dialog:unref_dlg: unref dlg 0x7f188afd3bf8 with 1 -> 3 in entry >> 0x7f188afc3828 >> Feb 26 17:28:56 voip2-sip23b /usr/sbin/opensips[4711]: >> DBG:dialog:unref_dlg: unref dlg 0x7f188afd3bf8 with 1 -> 2 in entry >> 0x7f188afc3828 >> Feb 26 17:29:05 voip2-sip23b /usr/sbin/opensips[4711]: >> DBG:dialog:next_state_dlg: dialog 0x7f188afd3bf8 changed from state 1 >> to state 5, due event 4 >> Feb 26 17:29:05 voip2-sip23b /usr/sbin/opensips[4711]: >> DBG:dialog:dlg_onreply: dialog 0x7f188afd3bf8 failed (negative reply) >> Feb 26 17:29:05 voip2-sip23b /usr/sbin/opensips[4711]: >> DBG:dialog:unref_dlg: unref dlg 0x7f188afd3bf8 with 1 -> 1 in entry >> 0x7f188afc3828 >> Feb 26 17:29:11 voip2-sip23b /usr/sbin/opensips[4711]: >> DBG:dialog:next_state_dlg: dialog 0x7f188afd3bf8 changed from state 5 >> to state 5, due event 1 >> Feb 26 17:29:11 voip2-sip23b /usr/sbin/opensips[4711]: >> DBG:dialog:unref_dlg: unref dlg 0x7f188afd3bf8 with 1 -> 0 in entry >> 0x7f188afc3828 >> Feb 26 17:29:11 voip2-sip23b /usr/sbin/opensips[4711]: >> DBG:dialog:unref_dlg: ref <=0 for dialog 0x7f188afd3bf8 >> Feb 26 17:29:11 voip2-sip23b /usr/sbin/opensips[4711]: >> DBG:dialog:destroy_dlg: destroying dialog 0x7f188afd3bf8 >> Feb 26 17:29:11 voip2-sip23b /usr/sbin/opensips[4711]: >> DBG:dialog:destroy_dlg: dlg expired or not in list - dlg >> 0x7f188afd3bf8 [2317:677053409] with clid 'ktQ0Pkdwz50qGYjED6Brpw..' >> and tags '2b161508' 'NULL' >> >> >> on node1 >> >> Feb 26 17:28:41 voip2-sip23a /usr/sbin/opensips[15820]: >> DBG:dialog:dlg_onroute: route param is 'd09.1e30b582' (len=12) >> Feb 26 17:28:41 voip2-sip23a /usr/sbin/opensips[15820]: >> DBG:dialog:lookup_dlg: no dialog id=677053409 found on entry 2317 >> Feb 26 17:28:41 voip2-sip23a /usr/sbin/opensips[15820]: >> DBG:dialog:dlg_onroute: unable to find dialog for BYE with route >> param 'd09.1e30b582' >> Feb 26 17:28:41 voip2-sip23a /usr/sbin/opensips[15822]: >> DBG:dialog:dlg_onroute: route param is 'd09.1e30b582' (len=12) >> Feb 26 17:28:41 voip2-sip23a /usr/sbin/opensips[15822]: >> DBG:dialog:lookup_dlg: no dialog id=677053409 found on entry 2317 >> Feb 26 17:28:41 voip2-sip23a /usr/sbin/opensips[15822]: >> DBG:dialog:dlg_onroute: unable to find dialog for BYE with route >> param 'd09.1e30b582' >> Feb 26 17:28:41 voip2-sip23a /usr/sbin/opensips[15822]: >> DBG:dialog:dlg_onroute: route param is 'd09.1e30b582' (len=12) >> Feb 26 17:28:41 voip2-sip23a /usr/sbin/opensips[15822]: >> DBG:dialog:lookup_dlg: no dialog id=677053409 found on entry 2317 >> Feb 26 17:28:41 voip2-sip23a /usr/sbin/opensips[15822]: >> DBG:dialog:dlg_onroute: unable to find dialog for BYE with route >> param 'd09.1e30b582' >> Feb 26 17:28:42 voip2-sip23a /usr/sbin/opensips[15820]: >> DBG:dialog:dlg_onroute: route param is 'd09.1e30b582' (len=12) >> Feb 26 17:28:42 voip2-sip23a /usr/sbin/opensips[15820]: >> DBG:dialog:lookup_dlg: no dialog id=677053409 found on entry 2317 >> Feb 26 17:28:42 voip2-sip23a /usr/sbin/opensips[15820]: >> DBG:dialog:dlg_onroute: unable to find dialog for BYE with route >> param 'd09.1e30b582' >> Feb 26 17:28:42 voip2-sip23a /usr/sbin/opensips[15820]: >> DBG:dialog:dlg_onroute: route param is 'd09.1e30b582' (len=12) >> Feb 26 17:28:42 voip2-sip23a /usr/sbin/opensips[15820]: >> DBG:dialog:lookup_dlg: no dialog id=677053409 found on entry 2317 >> >> >> If I purposely take down node1 again and try to make a call I can see >> that it indicates the failure to replicate dialog info on the cluster >> as below, >> >> Feb 26 17:27:01 voip2-sip23b /usr/sbin/opensips[4701]: >> ERROR:dialog:replicate_dialog_updated: All destinations in cluster: 1 >> are down or probing >> Feb 26 17:27:01 voip2-sip23b /usr/sbin/opensips[4701]: >> ERROR:dialog:replicate_dialog_updated: Failed to replicate updated dialog >> >> >> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Tue Feb 27 11:19:19 2018 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Tue, 27 Feb 2018 18:19:19 +0200 Subject: [OpenSIPS-Users] OpenSIPS crashing on 200 OK In-Reply-To: References: <530E1611-2A4D-424F-938D-2E409E1A78A2@genesys.com> <7cf6bb3d-967a-b8e2-d4b3-8b42c8dbf0b8@opensips.org> <755F831E-90E4-4105-80C5-42C7A2E2050A@genesys.com> <6D9EBB18-914E-49ED-988F-7D582007B88E@genesys.com> <085c6c6b-75ca-0557-aa5e-0fcb9e3eb07f@opensips.org> <4FD028FE-6607-496C-8B3A-EF4293FB6155@genesys.com> Message-ID: Thanks Benfor the support in investigating this. The bug was identified and confirm, let me work on a fix right now. I will shortly ping you for testing ;) Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Summit 2018 http://www.opensips.org/events/Summit-2018Amsterdam On 02/27/2018 12:12 AM, Bogdan-Andrei Iancu wrote: > OK, to be more precise, the crash is around an "If" statement where > you have in the condition an REGEXP test (=~) with a left operand > translated into "downstream" (maybe a $DLG_dir ??). > Does it sound familiar to you :) ? > > Regards, > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > OpenSIPS Summit 2018 > http://www.opensips.org/events/Summit-2018Amsterdam > On 02/26/2018 11:32 PM, Ben Newlin wrote: >> >> Yes, that is from that exact file, line 435. It’s just a return >> statement. I can send you the file off-list if you’d like. >> >> I do use the $DLG_dir in my script in a few places, but the only one >> this crash would have hit is in some logging. >> >> Ben Newlin >> >> *From: *Bogdan-Andrei Iancu >> *Date: *Monday, February 26, 2018 at 4:14 PM >> *To: *Ben Newlin , OpenSIPS users mailling >> list >> *Subject: *Re: [OpenSIPS-Users] OpenSIPS crashing on 200 OK >> >> OK, but check the exact file you pass to Opensips - according to the >> debug info is /opensips/control.cfg like 435. >> >> Also, do you use the $DLG_dir variable in your cfg ? >> >> Regards, >> >> Bogdan-Andrei Iancu >> OpenSIPS Founder and Developer >> http://www.opensips-solutions.com >> OpenSIPS Summit 2018 >> http://www.opensips.org/events/Summit-2018Amsterdam >> >> On 02/26/2018 08:10 PM, Ben Newlin wrote: >> >> Hmm, no. Line 435 of my config is a return statement. >> >> return(-1); >> >> Ben Newlin >> >> *From: *Bogdan-Andrei Iancu >> >> *Date: *Monday, February 26, 2018 at 12:16 PM >> *To: *Ben Newlin >> , OpenSIPS users mailling list >> >> *Subject: *Re: [OpenSIPS-Users] OpenSIPS crashing on 200 OK >> >> Could you off-list past me the line 435 of your cfg - I guess you >> use some $DLG_dir there ? >> >> Regards, >> >> >> Bogdan-Andrei Iancu >> >> >> >> OpenSIPS Founder and Developer >> >> http://www.opensips-solutions.com >> >> OpenSIPS Summit 2018 >> >> http://www.opensips.org/events/Summit-2018Amsterdam >> >> On 02/26/2018 07:04 PM, Ben Newlin wrote: >> >> Here you go: https://pastebin.com/rpUjaHBG >> >> >> Thanks, >> >> Ben Newlin >> >> *From: *Bogdan-Andrei Iancu >> >> *Date: *Monday, February 26, 2018 at 10:45 AM >> *To: *Ben Newlin >> , OpenSIPS users mailling list >> >> *Subject: *Re: [OpenSIPS-Users] OpenSIPS crashing on 200 OK >> >> OK, once you started GDB like "gdb /path/to/opensips >> /path/to/core", run step by step: >> >> f 0 >> p *s1 >> f 5 >> p *a >> >> Regards, >> >> >> >> Bogdan-Andrei Iancu >> >> >> >> OpenSIPS Founder and Developer >> >> http://www.opensips-solutions.com >> >> OpenSIPS Summit 2018 >> >> http://www.opensips.org/events/Summit-2018Amsterdam >> >> On 02/26/2018 05:43 PM, Ben Newlin wrote: >> >> Bogdan, >> >> Unfortunately I am not at all familiar with gdb. How do I >> change frames? >> >> Ben Newlin >> >> *From: *Bogdan-Andrei Iancu >> >> *Date: *Monday, February 26, 2018 at 10:40 AM >> *To: *Ben Newlin >> , OpenSIPS users mailling >> list >> >> *Subject: *Re: [OpenSIPS-Users] OpenSIPS crashing on 200 OK >> >> Hi Ben, >> >> The backtrace points to the script interpreter >> :D....Please do the followings (assuming you are a bit >> faimilar with GDB :) ) : >> >> in frame 0, please do "print *s1" >> and in frame 5 do "print *a" >> >> Regards, >> >> >> >> >> >> >> >> > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Tue Feb 27 11:38:55 2018 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Tue, 27 Feb 2018 18:38:55 +0200 Subject: [OpenSIPS-Users] Wrong number of children In-Reply-To: <00a601d3afe8$3c0a0740$b41e15c0$@web.de> References: <027501d3aca1$66551340$32ff39c0$@web.de> <010001d3aef6$02f70f90$08e52eb0$@web.de> <4e50d0e9-d0c8-5317-4f21-433cbc5cde13@opensips.org> <01ff01d3af19$fda10db0$f8e32910$@web.de> <15812914-24ca-8dfa-9371-732ce42c91bd@opensips.org> <026201d3af1e$a7be42a0$f73ac7e0$@web.de> <02dd01d3af2d$b3380b20$19a82160$@web.de> <00a601d3afe8$3c0a0740$b41e15c0$@web.de> Message-ID: <3f4b7059-74f6-e2ca-0697-123c3ccac637@opensips.org> Thank you Xaled, Based on the output you provided I managed to identify the issue. There is a tricky internal test that actually forces the TCP layer to start more or less all the time. Let us have this fixed !! I will update you Best regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Summit 2018 http://www.opensips.org/events/Summit-2018Amsterdam On 02/27/2018 06:30 PM, xaled wrote: > > Hi Bogdan, > > It’s a different server, but the same tcp issue: > > 2018-02-27T17:19:29.939207+01:00 dus-ivr01 opensips: DBG:core:main: > Loaded 1 transport protocols > > 2018-02-27T17:19:29.939411+01:00 dus-ivr01 opensips: > DBG:core:socket2str: > > 2018-02-27T17:19:29.939630+01:00 dus-ivr01 opensips: > INFO:core:tcp_init: enabling TCP for proto 5 > > 2018-02-27T17:19:29.939842+01:00 dus-ivr01 opensips: > DBG:core:ipc_register_handler: IPC type 0 [TCP reporting] registered with > > handler 0x5f9fee > > 2018-02-27T17:19:29.940120+01:00 dus-ivr01 opensips: > DBG:core:create_status_pipe: pipe created ? rc = 0, errno = Success > > 2018-02-27T17:19:29.940326+01:00 dus-ivr01 opensips: > DBG:core:daemonize: waiting for status code from children > > 2018-02-27T17:19:29.940526+01:00 dus-ivr01 opensips: > DBG:core:count_init_children: 15 children are going to be inited > > 2018-02-27T17:19:29.940724+01:00 dus-ivr01 > /usr/local/sbin/opensips[13076]: DBG:core:set_core_dump: core dump > limits set to 18 > > 446744073709551615 > > 2018-02-27T17:19:29.940951+01:00 dus-ivr01 > /usr/local/sbin/opensips[13076]: NOTICE:core:main: version: opensips > 2.3.3 (x86_64/ > > linux) > > Do you need full logs, or is this enough? > > Thanks, > > Xale > > *From:*Bogdan-Andrei Iancu [mailto:bogdan at opensips.org] > *Sent:* Monday, February 26, 2018 10:38 PM > *To:* xaled > *Subject:* Re: [OpenSIPS-Users] Wrong number of children > > Please apply the below patch, recompile and send again the startup logs: > > diff --git a/net/net_tcp.c b/net/net_tcp.c > index 2dbea49..2d63940 100644 > --- a/net/net_tcp.c > +++ b/net/net_tcp.c > @@ -1624,7 +1624,7 @@ int tcp_init(void) > /* first we do auto-detection to see if there are any TCP based > * protocols loaded */ > for ( i=PROTO_FIRST ; i - if (is_tcp_based_proto(i)) {tcp_disabled=0;break;} > + if (is_tcp_based_proto(i)) {LM_INFO("enabling TCP for > proto %d\n",i);tcp_disabled=0;break;} > > if (tcp_disabled) > return 0; > > > Regards, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > OpenSIPS Summit 2018 > http://www.opensips.org/events/Summit-2018Amsterdam > > On 02/26/2018 08:14 PM, xaled wrote: > > Hi Bogdan, > > would a full debug log will be sufficient? > > The config is spread across quite a few files. > > Thanks, > > Xaled > > *From:*Bogdan-Andrei Iancu [mailto:bogdan at opensips.org] > *Sent:* Monday, February 26, 2018 5:25 PM > *To:* xaled ; 'OpenSIPS users > mailling list' > > *Subject:* Re: [OpenSIPS-Users] Wrong number of children > > Hmm.... that is strange. > > Could you send me (off list) your cfg to give it a try - I see you > use the 2.3.3 version , right ? > > Regards, > > > Bogdan-Andrei Iancu > > > > OpenSIPS Founder and Developer > > http://www.opensips-solutions.com > > OpenSIPS Summit 2018 > > http://www.opensips.org/events/Summit-2018Amsterdam > > On 02/26/2018 06:27 PM, xaled wrote: > > Nothing > > root at dus-appsrv01:/etc/opensips# opensipsctl fifo ps > > Process:: ID=0 PID=14966 Type=attendant > > Process:: ID=1 PID=14967 Type=MI FIFO > > Process:: ID=2 PID=14968 Type=time_keeper > > Process:: ID=3 PID=14969 Type=timer > > Process:: ID=4 PID=14970 Type=SIP receiver udp:1.2.3.4:5060 > > Process:: ID=5 PID=14975 Type=SIP receiver udp:1.2.3.4:5060 > > Process:: ID=6 PID=14976 Type=SIP receiver udp:1.2.3.4:5060 > > Process:: ID=7 PID=14977 Type=SIP receiver udp:1.2.3.4:5060 > > Process:: ID=8 PID=14978 Type=SIP receiver udp:1.2.3.4:5060 > > Process:: ID=9 PID=14979 Type=TCP receiver > > Process:: ID=10 PID=14982 Type=TCP receiver > > Process:: ID=11 PID=14984 Type=TCP receiver > > Process:: ID=12 PID=14985 Type=TCP receiver > > Process:: ID=13 PID=14986 Type=TCP receiver > > Process:: ID=14 PID=14987 Type=TCP receiver > > Process:: ID=15 PID=14988 Type=TCP receiver > > Process:: ID=16 PID=14989 Type=TCP receiver > > Process:: ID=17 PID=14991 Type=Timer handler > > Process:: ID=18 PID=14992 Type=TCP main > > root at srv01:/etc/opensips# netstat -tlnp | grep opensips > > root at srv01:/etc/opensips# > > Thanks, > > Xaled > > *From:*Bogdan-Andrei Iancu [mailto:bogdan at opensips.org] > *Sent:* Monday, February 26, 2018 4:57 PM > *To:* xaled ; 'OpenSIPS > users mailling list' > > *Subject:* Re: [OpenSIPS-Users] Wrong number of children > > If you do "netstat -tlnp | grep opensips", what do you get ? > > The TCP layer automatically enable itself if a TCP-based > protocol is used from the OpenSIPS config (as listener), like > SIP TCP/TLS/WS/WSS , HEP or BIN. > > Regards, > > > > Bogdan-Andrei Iancu > > > > OpenSIPS Founder and Developer > > http://www.opensips-solutions.com > > OpenSIPS Summit 2018 > > http://www.opensips.org/events/Summit-2018Amsterdam > > On 02/26/2018 05:53 PM, xaled wrote: > > Hi Bogdan, > > nope, non of them. > > Greetings, > > Xaled > > *From:*Bogdan-Andrei Iancu [mailto:bogdan at opensips.org] > *Sent:* Monday, February 26, 2018 4:41 PM > *To:* OpenSIPS users mailling list > > ; xaled > > *Subject:* Re: [OpenSIPS-Users] Wrong number of children > > Hi Xaled, > > Do you use HEP for tracing or BIN for replication ? > > Regards, > > > > > Bogdan-Andrei Iancu > > > > OpenSIPS Founder and Developer > > http://www.opensips-solutions.com > > OpenSIPS Summit 2018 > > http://www.opensips.org/events/Summit-2018Amsterdam > > On 02/26/2018 01:36 PM, xaled wrote: > > Hi Callum, > > that helped thanks. > > I have 8 TCP receivers and one TCP main process that I > did not expected as I only listen on udp. > > Will have to understand the TCP spawn logic. > > Greetings, > > Xaled > -------------- next part -------------- An HTML attachment was scrubbed... URL: From brian.southworth at clocom.uk Mon Feb 26 11:44:25 2018 From: brian.southworth at clocom.uk (=?UTF-8?Q?Brian_Southworth?=) Date: Mon, 26 Feb 2018 16:44:25 +0000 Subject: [OpenSIPS-Users] On hold In-Reply-To: References: <01020161c1ef8de9-1fa3116b-b84a-4c0c-a3a9-f3419d72813d-000000@eu-west-1.amazonses.com> <01020161c2e60d3d-dedd2037-775c-45d2-adf1-5c7ae8ef3dfa-000000@eu-west-1.amazonses.com> <01020161d2d39b50-ecde506f-9a4e-430e-9d46-6cdee6b8c2ab-000000@eu-west-1.amazonses.com> Message-ID: <01020161d3004320-980e8eda-67f2-4e18-b40a-46ea00699437-000000@eu-west-1.amazonses.com> I don’t think im trying to re reroute re invites not that ive noticed would it help if you took a look on my cfg I am still new to opensips and learning as I go along. But I will also take another look thanks.   Regards,   Brian Southworth From: Users [mailto:users-bounces at lists.opensips.org] On Behalf Of Ovidiu Sas Sent: 26 February 2018 16:21 To: users at lists.opensips.org users Subject: Re: [OpenSIPS-Users] On hold   You need to make sure that in your config you are not trying to re-route in dialog requests (like reINVITE). In dialog requests are routed according to the loose routing mechanism.   -ovidiu     On Feb 26, 2018 11:01, "Brian Southworth" > wrote: Seems to be an issue with it sending the on hold invite to get the call back to the wrong place on inbound calls.   Example caller calls in --- agent 1 picks up --- agent 1 places them on hold all is fine --- agent 1 takes call off hold call is dropped   Seems that the invite to get the call back gets sent to .192 when it should go back to .193   This doesn’t happen is I use rewritehostport but then on hold calls only work for calls coming from that media server t_relay works across the board but doesn’t actually work for inbound.   Regards,   Brian Southworth   From: Users [mailto:users-bounces at lists.opensips.org ] On Behalf Of Ovidiu Sas Sent: 26 February 2018 15:49 To: users at lists.opensips.org users > Subject: Re: [OpenSIPS-Users] On hold   You need to take a look at the signalling and figure out who is dropping the call and why.   -ovidiu   On Feb 26, 2018 10:45, "Brian Southworth" > wrote: Please ignore my last email.   Both ways work it was an error in my code I wasn’t using t_relay();   Thanks for your help 😊   Regards,   Brian Southworth   From: Users [mailto:users-bounces at lists.opensips.org ] On Behalf Of Ovidiu Sas Sent: 23 February 2018 13:10 To: users at lists.opensips.org users > Subject: Re: [OpenSIPS-Users] On hold   Hello Brian,   You can detect hold using is_audio_on_hold(): http://www.opensips.org/html/docs/modules/2.3.x/sipmsgops#idp5575408   Try to use it and see if this will solve your issue.   Regards, Ovidiu Sas   On Feb 23, 2018 4:13 AM, "Brian Southworth" > wrote: Hi All,   I am trying to get opensips to forward the on hold request to asterisk I have done this using an if statement for SDP =~ “sendonly”   How ever when I go to take them off hold the call is dropped, is there something special I need to do ?   I have tried and if statement for is_method(“invite”) && $rb =~ “sendrecv” ß this loops until opensips give a 500 internal error And I have also tried is_method(“invite”) && $rb =~ “inactive” ß this doesn’t seem to work at all   The initial hold works perfect puts them on hold just cant take them off or transfer the call (asterisk will handle all the B2B stuff)   Any help would be appreciated.   Regards,   Brian Southworth Communications Developer 111 Wilmslow Road Handforth Wilmslow SK9 3ER   T: 03333 446677 W: www.clocom.uk           Like us on Facebook Follow us on Twitter           Clocom is a green company. Think, do you need to print this email?   This message contains confidential information and is intended only for the individual named. If you are not the named addressee you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately by e-mail if you have received this e-mail by mistake and delete this e-mail from your system. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. The sender therefore does not accept liability for any errors or omissions in the contents of this message, which arise as a result of e-mail transmission. If verification is required please request a hard-copy version. Registered in England & Wales: 07081192. Registered name and address: Clocom UK Ltd, 111 Wilmslow Road, Handforth, Cheshire, SK9 3ER  www.clocom.uk   _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users   _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users   _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... 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Name: not available Type: image/png Size: 2999 bytes Desc: not available URL: From yuval.dinari at vonage.com Mon Feb 26 11:45:44 2018 From: yuval.dinari at vonage.com (Dinari, Yuval) Date: Mon, 26 Feb 2018 18:45:44 +0200 Subject: [OpenSIPS-Users] textops module replace functions don't work In-Reply-To: References: Message-ID: Hi, I tried all of them, not just those two but also replace_all(), replace_body_all(), replace_body_atonce(), subst(), and subst_body(). Thanks On Mon, Feb 26, 2018 at 5:54 PM, Bogdan-Andrei Iancu wrote: > Hi, > > Have you used replace() or replace_body() ? > > Regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > OpenSIPS Summit 2018 > http://www.opensips.org/events/Summit-2018Amsterdam > > On 02/26/2018 12:31 PM, Dinari, Yuval via Users wrote: > > Hi, > I tried using textops module functions to replace a part of the message > body, but the body remains unchanged. A search() with the regex to be > replaced succeeds. > I tried all replacement functions (replace*, subst, subst_body). > I see that the body is unchanged by printing $rb, and also ngreping the > SDP that opensips sends out to rtpengine via rtpengine_offer(). > opensips version - 2.2.5 > Thanks > > > _______________________________________________ > Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From osas at voipembedded.com Mon Feb 26 11:57:16 2018 From: osas at voipembedded.com (Ovidiu Sas) Date: Mon, 26 Feb 2018 11:57:16 -0500 Subject: [OpenSIPS-Users] On hold In-Reply-To: <01020161d2d39b50-ecde506f-9a4e-430e-9d46-6cdee6b8c2ab-000000@eu-west-1.amazonses.com> References: <01020161c1ef8de9-1fa3116b-b84a-4c0c-a3a9-f3419d72813d-000000@eu-west-1.amazonses.com> <01020161c2e60d3d-dedd2037-775c-45d2-adf1-5c7ae8ef3dfa-000000@eu-west-1.amazonses.com> <01020161d2d39b50-ecde506f-9a4e-430e-9d46-6cdee6b8c2ab-000000@eu-west-1.amazonses.com> Message-ID: You need to make sure that in your config you are not trying to re-route in dialog requests (like reINVITE). In dialog requests are routed according to the loose routing mechanism. -ovidiu On Feb 26, 2018 11:01 AM, "Brian Southworth" wrote: > Seems to be an issue with it sending the on hold invite to get the call > back to the wrong place on inbound calls. > > > > Example caller calls in --- agent 1 picks up --- agent 1 places them on > hold all is fine --- agent 1 takes call off hold call is dropped > > > > Seems that the invite to get the call back gets sent to .192 when it > should go back to .193 > > > > This doesn’t happen is I use rewritehostport but then on hold calls only > work for calls coming from that media server t_relay works across the board > but doesn’t actually work for inbound. > > > > Regards, > > > > Brian Southworth > > > > *From:* Users [mailto:users-bounces at lists.opensips.org] *On Behalf Of *Ovidiu > Sas > *Sent:* 26 February 2018 15:49 > *To:* users at lists.opensips.org users > *Subject:* Re: [OpenSIPS-Users] On hold > > > > You need to take a look at the signalling and figure out who is dropping > the call and why. > > > > -ovidiu > > > > On Feb 26, 2018 10:45, "Brian Southworth" > wrote: > > Please ignore my last email. > > > > Both ways work it was an error in my code I wasn’t using t_relay(); > > > > Thanks for your help 😊 > > > > Regards, > > > > Brian Southworth > > > > *From:* Users [mailto:users-bounces at lists.opensips.org] *On Behalf Of *Ovidiu > Sas > *Sent:* 23 February 2018 13:10 > *To:* users at lists.opensips.org users > *Subject:* Re: [OpenSIPS-Users] On hold > > > > Hello Brian, > > > > You can detect hold using is_audio_on_hold(): > > http://www.opensips.org/html/docs/modules/2.3.x/sipmsgops#idp5575408 > > > > Try to use it and see if this will solve your issue. > > > > Regards, > > Ovidiu Sas > > > > On Feb 23, 2018 4:13 AM, "Brian Southworth" > wrote: > > Hi All, > > > > I am trying to get opensips to forward the on hold request to asterisk I > have done this using an if statement for SDP =~ “sendonly” > > > > How ever when I go to take them off hold the call is dropped, is there > something special I need to do ? > > > > I have tried and if statement for is_method(“invite”) && $rb =~ “sendrecv” > ß this loops until opensips give a 500 internal error > > And I have also tried is_method(“invite”) && $rb =~ “inactive” ß this > doesn’t seem to work at all > > > > The initial hold works perfect puts them on hold just cant take them off > or transfer the call (asterisk will handle all the B2B stuff) > > > > Any help would be appreciated. > > > > Regards, > > > > Brian Southworth > > Communications Developer > > [image: cid:image001.png at 01D22CAC.1DCB8580] > > 111 Wilmslow Road > > Handforth > > Wilmslow > > SK9 3ER > > > > T: 03333 446677 > > W: www.clocom.uk > > > > > > > > > > > > [image: cid:image002.png at 01CDDC62.D8483910] > > > Like us on Facebook > > [image: cid:image003.png at 01CDDC62.D8483910] > > Follow us on Twitter > > [image: cid:image004.png at 01CDDC62.D8483910] > > > > > > > > > > > > > Clocom is a *green* company. Think, do you need to print this email? > > > > This message contains confidential information and is intended only for > the individual named. If you are not the named addressee you should not > disseminate, distribute or copy this e-mail. Please notify the sender > immediately by e-mail if you have received this e-mail by mistake and > delete this e-mail from your system. E-mail transmission cannot be > guaranteed to be secure or error-free as information could be intercepted, > corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. > The sender therefore does not accept liability for any errors or omissions > in the contents of this message, which arise as a result of e-mail > transmission. If verification is required please request a hard-copy > version. Registered in England & Wales: 07081192. Registered name and > address: Clocom UK Ltd, 111 Wilmslow Road, Handforth, Cheshire, SK9 3ER > www.clocom.uk > > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > _______________________________________________ > > Users mailing list > > Users at lists.opensips.org > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > _______________________________________________ > > Users mailing list > > Users at lists.opensips.org > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > -------------- next part -------------- An HTML attachment was scrubbed... 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Name: not available Type: image/png Size: 234 bytes Desc: not available URL: From bogdan at opensips.org Tue Feb 27 12:39:28 2018 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Tue, 27 Feb 2018 19:39:28 +0200 Subject: [OpenSIPS-Users] Wrong number of children In-Reply-To: <3f4b7059-74f6-e2ca-0697-123c3ccac637@opensips.org> References: <027501d3aca1$66551340$32ff39c0$@web.de> <010001d3aef6$02f70f90$08e52eb0$@web.de> <4e50d0e9-d0c8-5317-4f21-433cbc5cde13@opensips.org> <01ff01d3af19$fda10db0$f8e32910$@web.de> <15812914-24ca-8dfa-9371-732ce42c91bd@opensips.org> <026201d3af1e$a7be42a0$f73ac7e0$@web.de> <02dd01d3af2d$b3380b20$19a82160$@web.de> <00a601d3afe8$3c0a0740$b41e15c0$@web.de> <3f4b7059-74f6-e2ca-0697-123c3ccac637@opensips.org> Message-ID: Ok, thanks to Razvan, we have an fix for this: https://github.com/OpenSIPS/opensips/commit/b022ad5a7c1c24aa9cf3ddf22d5f50510807c9f4 Please update from GIT and give it a try. Many Thanks, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Summit 2018 http://www.opensips.org/events/Summit-2018Amsterdam On 02/27/2018 06:38 PM, Bogdan-Andrei Iancu wrote: > Thank you Xaled, > > Based on the output you provided I managed to identify the issue. > There is a tricky internal test that actually forces the TCP layer to > start more or less all the time. > Let us have this fixed !! I will update you > > Best regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > OpenSIPS Summit 2018 > http://www.opensips.org/events/Summit-2018Amsterdam > On 02/27/2018 06:30 PM, xaled wrote: >> >> Hi Bogdan, >> >> It’s a different server, but the same tcp issue: >> >> 2018-02-27T17:19:29.939207+01:00 dus-ivr01 opensips: DBG:core:main: >> Loaded 1 transport protocols >> >> 2018-02-27T17:19:29.939411+01:00 dus-ivr01 opensips: >> DBG:core:socket2str: >> >> 2018-02-27T17:19:29.939630+01:00 dus-ivr01 opensips: >> INFO:core:tcp_init: enabling TCP for proto 5 >> >> 2018-02-27T17:19:29.939842+01:00 dus-ivr01 opensips: >> DBG:core:ipc_register_handler: IPC type 0 [TCP reporting] registered with >> >> handler 0x5f9fee >> >> 2018-02-27T17:19:29.940120+01:00 dus-ivr01 opensips: >> DBG:core:create_status_pipe: pipe created ? rc = 0, errno = Success >> >> 2018-02-27T17:19:29.940326+01:00 dus-ivr01 opensips: >> DBG:core:daemonize: waiting for status code from children >> >> 2018-02-27T17:19:29.940526+01:00 dus-ivr01 opensips: >> DBG:core:count_init_children: 15 children are going to be inited >> >> 2018-02-27T17:19:29.940724+01:00 dus-ivr01 >> /usr/local/sbin/opensips[13076]: DBG:core:set_core_dump: core dump >> limits set to 18 >> >> 446744073709551615 >> >> 2018-02-27T17:19:29.940951+01:00 dus-ivr01 >> /usr/local/sbin/opensips[13076]: NOTICE:core:main: version: opensips >> 2.3.3 (x86_64/ >> >> linux) >> >> Do you need full logs, or is this enough? >> >> Thanks, >> >> Xale >> >> *From:*Bogdan-Andrei Iancu [mailto:bogdan at opensips.org] >> *Sent:* Monday, February 26, 2018 10:38 PM >> *To:* xaled >> *Subject:* Re: [OpenSIPS-Users] Wrong number of children >> >> Please apply the below patch, recompile and send again the startup logs: >> >> diff --git a/net/net_tcp.c b/net/net_tcp.c >> index 2dbea49..2d63940 100644 >> --- a/net/net_tcp.c >> +++ b/net/net_tcp.c >> @@ -1624,7 +1624,7 @@ int tcp_init(void) >> /* first we do auto-detection to see if there are any TCP based >> * protocols loaded */ >> for ( i=PROTO_FIRST ; i> - if (is_tcp_based_proto(i)) {tcp_disabled=0;break;} >> + if (is_tcp_based_proto(i)) {LM_INFO("enabling TCP for >> proto %d\n",i);tcp_disabled=0;break;} >> >> if (tcp_disabled) >> return 0; >> >> >> Regards, >> >> Bogdan-Andrei Iancu >> OpenSIPS Founder and Developer >> http://www.opensips-solutions.com >> OpenSIPS Summit 2018 >> http://www.opensips.org/events/Summit-2018Amsterdam >> >> On 02/26/2018 08:14 PM, xaled wrote: >> >> Hi Bogdan, >> >> would a full debug log will be sufficient? >> >> The config is spread across quite a few files. >> >> Thanks, >> >> Xaled >> >> *From:*Bogdan-Andrei Iancu [mailto:bogdan at opensips.org] >> *Sent:* Monday, February 26, 2018 5:25 PM >> *To:* xaled ; 'OpenSIPS users >> mailling list' >> >> *Subject:* Re: [OpenSIPS-Users] Wrong number of children >> >> Hmm.... that is strange. >> >> Could you send me (off list) your cfg to give it a try - I see >> you use the 2.3.3 version , right ? >> >> Regards, >> >> >> Bogdan-Andrei Iancu >> >> >> >> OpenSIPS Founder and Developer >> >> http://www.opensips-solutions.com >> >> OpenSIPS Summit 2018 >> >> http://www.opensips.org/events/Summit-2018Amsterdam >> >> On 02/26/2018 06:27 PM, xaled wrote: >> >> Nothing >> >> root at dus-appsrv01:/etc/opensips# opensipsctl fifo ps >> >> Process:: ID=0 PID=14966 Type=attendant >> >> Process:: ID=1 PID=14967 Type=MI FIFO >> >> Process:: ID=2 PID=14968 Type=time_keeper >> >> Process:: ID=3 PID=14969 Type=timer >> >> Process:: ID=4 PID=14970 Type=SIP receiver udp:1.2.3.4:5060 >> >> Process:: ID=5 PID=14975 Type=SIP receiver udp:1.2.3.4:5060 >> >> Process:: ID=6 PID=14976 Type=SIP receiver udp:1.2.3.4:5060 >> >> Process:: ID=7 PID=14977 Type=SIP receiver udp:1.2.3.4:5060 >> >> Process:: ID=8 PID=14978 Type=SIP receiver udp:1.2.3.4:5060 >> >> Process:: ID=9 PID=14979 Type=TCP receiver >> >> Process:: ID=10 PID=14982 Type=TCP receiver >> >> Process:: ID=11 PID=14984 Type=TCP receiver >> >> Process:: ID=12 PID=14985 Type=TCP receiver >> >> Process:: ID=13 PID=14986 Type=TCP receiver >> >> Process:: ID=14 PID=14987 Type=TCP receiver >> >> Process:: ID=15 PID=14988 Type=TCP receiver >> >> Process:: ID=16 PID=14989 Type=TCP receiver >> >> Process:: ID=17 PID=14991 Type=Timer handler >> >> Process:: ID=18 PID=14992 Type=TCP main >> >> root at srv01:/etc/opensips# netstat -tlnp | grep opensips >> >> root at srv01:/etc/opensips# >> >> Thanks, >> >> Xaled >> >> *From:*Bogdan-Andrei Iancu [mailto:bogdan at opensips.org] >> *Sent:* Monday, February 26, 2018 4:57 PM >> *To:* xaled ; 'OpenSIPS >> users mailling list' >> >> *Subject:* Re: [OpenSIPS-Users] Wrong number of children >> >> If you do "netstat -tlnp | grep opensips", what do you get ? >> >> The TCP layer automatically enable itself if a TCP-based >> protocol is used from the OpenSIPS config (as listener), like >> SIP TCP/TLS/WS/WSS , HEP or BIN. >> >> Regards, >> >> >> >> Bogdan-Andrei Iancu >> >> >> >> OpenSIPS Founder and Developer >> >> http://www.opensips-solutions.com >> >> OpenSIPS Summit 2018 >> >> http://www.opensips.org/events/Summit-2018Amsterdam >> >> On 02/26/2018 05:53 PM, xaled wrote: >> >> Hi Bogdan, >> >> nope, non of them. >> >> Greetings, >> >> Xaled >> >> *From:*Bogdan-Andrei Iancu [mailto:bogdan at opensips.org] >> *Sent:* Monday, February 26, 2018 4:41 PM >> *To:* OpenSIPS users mailling list >> >> ; xaled >> >> *Subject:* Re: [OpenSIPS-Users] Wrong number of children >> >> Hi Xaled, >> >> Do you use HEP for tracing or BIN for replication ? >> >> Regards, >> >> >> >> >> Bogdan-Andrei Iancu >> >> >> >> OpenSIPS Founder and Developer >> >> http://www.opensips-solutions.com >> >> OpenSIPS Summit 2018 >> >> http://www.opensips.org/events/Summit-2018Amsterdam >> >> On 02/26/2018 01:36 PM, xaled wrote: >> >> Hi Callum, >> >> that helped thanks. >> >> I have 8 TCP receivers and one TCP main process that >> I did not expected as I only listen on udp. >> >> Will have to understand the TCP spawn logic. >> >> Greetings, >> >> Xaled >> > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Tue Feb 27 12:54:23 2018 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Tue, 27 Feb 2018 19:54:23 +0200 Subject: [OpenSIPS-Users] OpenSIPS crashing on 200 OK In-Reply-To: References: <530E1611-2A4D-424F-938D-2E409E1A78A2@genesys.com> <7cf6bb3d-967a-b8e2-d4b3-8b42c8dbf0b8@opensips.org> <755F831E-90E4-4105-80C5-42C7A2E2050A@genesys.com> <6D9EBB18-914E-49ED-988F-7D582007B88E@genesys.com> <085c6c6b-75ca-0557-aa5e-0fcb9e3eb07f@opensips.org> <4FD028FE-6607-496C-8B3A-EF4293FB6155@genesys.com> Message-ID: <8eeebbf4-9c57-dd6a-ebf1-bbe6dadb0bdb@opensips.org> Hi Ben, I just committedthe fix for this - could you update from GIT and give it a new try ? Thanks and regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Summit 2018 http://www.opensips.org/events/Summit-2018Amsterdam On 02/27/2018 06:19 PM, Bogdan-Andrei Iancu wrote: > Thanks Benfor the support in investigating this. The bug was > identified and confirm, let me work on a fix right now. I will shortly > ping you for testing ;) > > Regards, > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > OpenSIPS Summit 2018 > http://www.opensips.org/events/Summit-2018Amsterdam > On 02/27/2018 12:12 AM, Bogdan-Andrei Iancu wrote: >> OK, to be more precise, the crash is around an "If" statement where >> you have in the condition an REGEXP test (=~) with a left operand >> translated into "downstream" (maybe a $DLG_dir ??). >> Does it sound familiar to you :) ? >> >> Regards, >> Bogdan-Andrei Iancu >> >> OpenSIPS Founder and Developer >> http://www.opensips-solutions.com >> OpenSIPS Summit 2018 >> http://www.opensips.org/events/Summit-2018Amsterdam >> On 02/26/2018 11:32 PM, Ben Newlin wrote: >>> >>> Yes, that is from that exact file, line 435. It’s just a return >>> statement. I can send you the file off-list if you’d like. >>> >>> I do use the $DLG_dir in my script in a few places, but the only one >>> this crash would have hit is in some logging. >>> >>> Ben Newlin >>> >>> -------------- next part -------------- An HTML attachment was scrubbed... URL: From Ben.Newlin at genesys.com Tue Feb 27 14:20:34 2018 From: Ben.Newlin at genesys.com (Ben Newlin) Date: Tue, 27 Feb 2018 19:20:34 +0000 Subject: [OpenSIPS-Users] OpenSIPS crashing on 200 OK In-Reply-To: <8eeebbf4-9c57-dd6a-ebf1-bbe6dadb0bdb@opensips.org> References: <530E1611-2A4D-424F-938D-2E409E1A78A2@genesys.com> <7cf6bb3d-967a-b8e2-d4b3-8b42c8dbf0b8@opensips.org> <755F831E-90E4-4105-80C5-42C7A2E2050A@genesys.com> <6D9EBB18-914E-49ED-988F-7D582007B88E@genesys.com> <085c6c6b-75ca-0557-aa5e-0fcb9e3eb07f@opensips.org> <4FD028FE-6607-496C-8B3A-EF4293FB6155@genesys.com> <8eeebbf4-9c57-dd6a-ebf1-bbe6dadb0bdb@opensips.org> Message-ID: <279057EB-BEC2-4552-B0C9-EDD244263E32@genesys.com> Bogdan, I just tested it and that crash is no longer occurring. However I am now seeing a different crash, but it seems to still be related to regexp. Feb 27 19:07:17 [369] ERROR:core:subst_parser: string too short Feb 27 19:07:17 [369] ERROR:core:tr_eval_re: Can't compile regexp Feb 27 19:07:17 [369] WARNING:core:do_assign: no value in right expression at /opensips/control.cfg:1952 Feb 27 19:07:17 [361] INFO:core:handle_sigs: child process 369 exited by a signal 11 Feb 27 19:07:17 [361] INFO:core:handle_sigs: core was generated Feb 27 19:07:17 [361] INFO:core:handle_sigs: terminating due to SIGCHLD The do_assign error makes it seem like there may be something wrong in my script, but I can’t see what it is. And I’m not sure what the “string too short” error is either or why the regexp could not be compiled. It looks valid to me. Backtrace: https://pastebin.com/Fk2BRnsg Ben Newlin From: Bogdan-Andrei Iancu Date: Tuesday, February 27, 2018 at 12:55 PM To: OpenSIPS users mailling list , Ben Newlin Subject: Re: [OpenSIPS-Users] OpenSIPS crashing on 200 OK Hi Ben, I just committed the fix for this - could you update from GIT and give it a new try ? Thanks and regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Summit 2018 http://www.opensips.org/events/Summit-2018Amsterdam On 02/27/2018 06:19 PM, Bogdan-Andrei Iancu wrote: Thanks Ben for the support in investigating this. The bug was identified and confirm, let me work on a fix right now. I will shortly ping you for testing ;) Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Summit 2018 http://www.opensips.org/events/Summit-2018Amsterdam On 02/27/2018 12:12 AM, Bogdan-Andrei Iancu wrote: OK, to be more precise, the crash is around an "If" statement where you have in the condition an REGEXP test (=~) with a left operand translated into "downstream" (maybe a $DLG_dir ??). Does it sound familiar to you :) ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Summit 2018 http://www.opensips.org/events/Summit-2018Amsterdam On 02/26/2018 11:32 PM, Ben Newlin wrote: Yes, that is from that exact file, line 435. It’s just a return statement. I can send you the file off-list if you’d like. I do use the $DLG_dir in my script in a few places, but the only one this crash would have hit is in some logging. Ben Newlin -------------- next part -------------- An HTML attachment was scrubbed... URL: From Ben.Newlin at genesys.com Tue Feb 27 14:47:42 2018 From: Ben.Newlin at genesys.com (Ben Newlin) Date: Tue, 27 Feb 2018 19:47:42 +0000 Subject: [OpenSIPS-Users] OpenSIPS crashing on 200 OK In-Reply-To: <279057EB-BEC2-4552-B0C9-EDD244263E32@genesys.com> References: <530E1611-2A4D-424F-938D-2E409E1A78A2@genesys.com> <7cf6bb3d-967a-b8e2-d4b3-8b42c8dbf0b8@opensips.org> <755F831E-90E4-4105-80C5-42C7A2E2050A@genesys.com> <6D9EBB18-914E-49ED-988F-7D582007B88E@genesys.com> <085c6c6b-75ca-0557-aa5e-0fcb9e3eb07f@opensips.org> <4FD028FE-6607-496C-8B3A-EF4293FB6155@genesys.com> <8eeebbf4-9c57-dd6a-ebf1-bbe6dadb0bdb@opensips.org> <279057EB-BEC2-4552-B0C9-EDD244263E32@genesys.com> Message-ID: <478773D5-3955-4A6A-BA6F-15467E39CC29@genesys.com> Bogdan, I found this previous thread that has this error: http://lists.opensips.org/pipermail/users/2016-October/035635.html The fix suggested there worked. When I removed the quotes around the regexp used in re.subst the crash did not occur. Ben Newlin From: Ben Newlin Date: Tuesday, February 27, 2018 at 2:20 PM To: Bogdan-Andrei Iancu , OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] OpenSIPS crashing on 200 OK Bogdan, I just tested it and that crash is no longer occurring. However I am now seeing a different crash, but it seems to still be related to regexp. Feb 27 19:07:17 [369] ERROR:core:subst_parser: string too short Feb 27 19:07:17 [369] ERROR:core:tr_eval_re: Can't compile regexp Feb 27 19:07:17 [369] WARNING:core:do_assign: no value in right expression at /opensips/control.cfg:1952 Feb 27 19:07:17 [361] INFO:core:handle_sigs: child process 369 exited by a signal 11 Feb 27 19:07:17 [361] INFO:core:handle_sigs: core was generated Feb 27 19:07:17 [361] INFO:core:handle_sigs: terminating due to SIGCHLD The do_assign error makes it seem like there may be something wrong in my script, but I can’t see what it is. And I’m not sure what the “string too short” error is either or why the regexp could not be compiled. It looks valid to me. Backtrace: https://pastebin.com/Fk2BRnsg Ben Newlin From: Bogdan-Andrei Iancu Date: Tuesday, February 27, 2018 at 12:55 PM To: OpenSIPS users mailling list , Ben Newlin Subject: Re: [OpenSIPS-Users] OpenSIPS crashing on 200 OK Hi Ben, I just committed the fix for this - could you update from GIT and give it a new try ? Thanks and regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Summit 2018 http://www.opensips.org/events/Summit-2018Amsterdam On 02/27/2018 06:19 PM, Bogdan-Andrei Iancu wrote: Thanks Ben for the support in investigating this. The bug was identified and confirm, let me work on a fix right now. I will shortly ping you for testing ;) Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Summit 2018 http://www.opensips.org/events/Summit-2018Amsterdam On 02/27/2018 12:12 AM, Bogdan-Andrei Iancu wrote: OK, to be more precise, the crash is around an "If" statement where you have in the condition an REGEXP test (=~) with a left operand translated into "downstream" (maybe a $DLG_dir ??). Does it sound familiar to you :) ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Summit 2018 http://www.opensips.org/events/Summit-2018Amsterdam On 02/26/2018 11:32 PM, Ben Newlin wrote: Yes, that is from that exact file, line 435. It’s just a return statement. I can send you the file off-list if you’d like. I do use the $DLG_dir in my script in a few places, but the only one this crash would have hit is in some logging. Ben Newlin -------------- next part -------------- An HTML attachment was scrubbed... URL: From xaled at web.de Tue Feb 27 14:50:09 2018 From: xaled at web.de (xaled) Date: Tue, 27 Feb 2018 20:50:09 +0100 Subject: [OpenSIPS-Users] Wrong number of children References: <3f4b7059-74f6-e2ca-0697-123c3ccac637@opensips.org> Message-ID: An HTML attachment was scrubbed... URL: From pasandev at ymail.com Tue Feb 27 21:01:30 2018 From: pasandev at ymail.com (Pasan Meemaduma) Date: Wed, 28 Feb 2018 02:01:30 +0000 (UTC) Subject: [OpenSIPS-Users] dialog replication In-Reply-To: <48b7be0c-d5d9-b6f3-a81b-42d768116f07@opensips.org> References: <311827884.3546251.1519627085908.ref@mail.yahoo.com> <311827884.3546251.1519627085908@mail.yahoo.com> <704df948-6873-5fde-fe25-193f1cf2a420@opensips.org> <2103041839.6602135.1519697741901@mail.yahoo.com> <48b7be0c-d5d9-b6f3-a81b-42d768116f07@opensips.org> Message-ID: <107101422.7400213.1519783290651@mail.yahoo.com> Hi Vlad, That explains the issue then, my sip listener ip is an anycast one, and for some weired reason replies for INVITE that leaves node2 receive by node1 hence it can't create the dialog. I checked with node2 being down and recovered and calls init via node1 always received final reply hence dialog replication works. So it means my anycast configuration is broken right ? Thanks for you time to look in to it. On Tuesday, 27 February 2018, 21:42, Vlad Patrascu wrote: Hi Pasan, I don't see anything in the logs or your cluster configuration which could indicate that the dialogs don't replicate. Are you sure that the problem is consistently reproducible and that the INVITE for the call in question receives a final reply? Dialogs are replicated when they get to the confirmed state, and judging from the logs, this doesn't appear to be happening for that call. Regards, Vlad Patrascu OpenSIPS Developer http://www.opensips-solutions.com On 27.02.2018 04:15, Pasan Meemaduma via Users wrote: Hi Vlad, I have sent you the full debug logs as requested. Also clusterer_list mi command gives following output when dialog replication stopped working. node1>>opensipsctl fifo clusterer_list Cluster:: 1         Node:: 2 DB_ID=5 URL=bin:10.3.1.137:5566 Enabled=1 Link_state=Up      Next_hop=2 Description=Node 2  node2>> opensipsctl fifo clusterer_list Cluster:: 1         Node:: 1 DB_ID=4 URL=bin:10.3.1.136:5566 Enabled=1 Link_state=Up      Next_hop=1 Description=Node 1 1 On Monday, 26 February 2018, 18:02, Vlad Patrascu wrote: Hi, Can you send the full logs for both nodes from the time that node1 restarts onwards? Also, what is the output of 'clusterer_list' mi command on the instances? Regards, Vlad Patrascu OpenSIPS Developer http://www.opensips-solutions.com On 26.02.2018 08:38, Pasan Meemaduma via Users wrote: Hi Guys, Its me again :). I'm using dialog replication in opensips 2.3.3 and it appears it doesn't work after a node goes down and come back online. The recovered node doesn't seems to receiving dialog info via binary interface. I have node1 and node2 with dialog replication on. everything works fine and if I shutdown node1 and bring it back online after a while node2 doesn't send new call dialog info via binary interface. I have attach the debug trace from both nodes, if you need anything else let me know. I'm also using an anycast ip as the service ip. on node2 for new call Feb 26 17:28:35 voip2-sip23b /usr/sbin/opensips[4703]: DBG:dialog:build_new_dlg: new dialog 0x7f188afd3bf8(c=ktQ0Pkdwz50qGYjED6Brpw..,f=sip:XXXXXXXXX at somedomain;transport=UDP,t=sip:+YYYYYYYYYYY at somedomain;transport=UDP,ft=2b161508) on hash 2317 Feb 26 17:28:35 voip2-sip23b /usr/sbin/opensips[4703]: DBG:dialog:init_leg_info: route_set , contact sip:XXXXXXXXX at 192.168.27.11:56419;transport=UDP, cseq 2 and bind_addr udp:10.3.3.1:5060 Feb 26 17:28:35 voip2-sip23b /usr/sbin/opensips[4703]: DBG:dialog:dlg_add_leg_info: set leg 0 for 0x7f188afd3bf8: tag=<2b161508> rcseq=<0> Feb 26 17:28:35 voip2-sip23b /usr/sbin/opensips[4703]: DBG:dialog:link_dlg: ref dlg 0x7f188afd3bf8 with 3 -> 3 in h_entry 0x7f188afc3828 - 2317 Feb 26 17:28:35 voip2-sip23b /usr/sbin/opensips[4703]: DBG:dialog:new_dlg_val: inserting =<45ad4c76-1abe-11e8-9410-831894b67d0c> Feb 26 17:28:35 voip2-sip23b /usr/sbin/opensips[4703]: DBG:dialog:dlg_onreq: t hash_index = 47425, t label = 2069082013 Feb 26 17:28:35 voip2-sip23b /usr/sbin/opensips[4703]: DBG:dialog:dlg_update_contact: Updated dialog 0x7f188afd3bf8 contact to Feb 26 17:28:35 voip2-sip23b /usr/sbin/opensips[4703]: DBG:dialog:unref_dlg: unref dlg 0x7f188afd3bf8 with 1 -> 2 in entry 0x7f188afc3828 Feb 26 17:28:52 voip2-sip23b /usr/sbin/opensips[4702]: DBG:dialog:ref_dlg: ref dlg 0x7f188afd3bf8 with 1 -> 3 Feb 26 17:28:52 voip2-sip23b /usr/sbin/opensips[4702]: DBG:dialog:ref_dlg: ref dlg 0x7f188afd3bf8 with 1 -> 4 Feb 26 17:28:52 voip2-sip23b /usr/sbin/opensips[4702]: DBG:dialog:unref_dlg: unref dlg 0x7f188afd3bf8 with 1 -> 3 in entry 0x7f188afc3828 Feb 26 17:28:56 voip2-sip23b /usr/sbin/opensips[4711]: DBG:dialog:unref_dlg: unref dlg 0x7f188afd3bf8 with 1 -> 2 in entry 0x7f188afc3828 Feb 26 17:29:05 voip2-sip23b /usr/sbin/opensips[4711]: DBG:dialog:next_state_dlg: dialog 0x7f188afd3bf8 changed from state 1 to state 5, due event 4 Feb 26 17:29:05 voip2-sip23b /usr/sbin/opensips[4711]: DBG:dialog:dlg_onreply: dialog 0x7f188afd3bf8 failed (negative reply) Feb 26 17:29:05 voip2-sip23b /usr/sbin/opensips[4711]: DBG:dialog:unref_dlg: unref dlg 0x7f188afd3bf8 with 1 -> 1 in entry 0x7f188afc3828 Feb 26 17:29:11 voip2-sip23b /usr/sbin/opensips[4711]: DBG:dialog:next_state_dlg: dialog 0x7f188afd3bf8 changed from state 5 to state 5, due event 1 Feb 26 17:29:11 voip2-sip23b /usr/sbin/opensips[4711]: DBG:dialog:unref_dlg: unref dlg 0x7f188afd3bf8 with 1 -> 0 in entry 0x7f188afc3828 Feb 26 17:29:11 voip2-sip23b /usr/sbin/opensips[4711]: DBG:dialog:unref_dlg: ref <=0 for dialog 0x7f188afd3bf8 Feb 26 17:29:11 voip2-sip23b /usr/sbin/opensips[4711]: DBG:dialog:destroy_dlg: destroying dialog 0x7f188afd3bf8 Feb 26 17:29:11 voip2-sip23b /usr/sbin/opensips[4711]: DBG:dialog:destroy_dlg: dlg expired or not in list - dlg 0x7f188afd3bf8 [2317:677053409] with clid 'ktQ0Pkdwz50qGYjED6Brpw..' and tags '2b161508' 'NULL' on node1 Feb 26 17:28:41 voip2-sip23a /usr/sbin/opensips[15820]: DBG:dialog:dlg_onroute: route param is 'd09.1e30b582' (len=12) Feb 26 17:28:41 voip2-sip23a /usr/sbin/opensips[15820]: DBG:dialog:lookup_dlg: no dialog id=677053409 found on entry 2317 Feb 26 17:28:41 voip2-sip23a /usr/sbin/opensips[15820]: DBG:dialog:dlg_onroute: unable to find dialog for BYE with route param 'd09.1e30b582' Feb 26 17:28:41 voip2-sip23a /usr/sbin/opensips[15822]: DBG:dialog:dlg_onroute: route param is 'd09.1e30b582' (len=12) Feb 26 17:28:41 voip2-sip23a /usr/sbin/opensips[15822]: DBG:dialog:lookup_dlg: no dialog id=677053409 found on entry 2317 Feb 26 17:28:41 voip2-sip23a /usr/sbin/opensips[15822]: DBG:dialog:dlg_onroute: unable to find dialog for BYE with route param 'd09.1e30b582' Feb 26 17:28:41 voip2-sip23a /usr/sbin/opensips[15822]: DBG:dialog:dlg_onroute: route param is 'd09.1e30b582' (len=12) Feb 26 17:28:41 voip2-sip23a /usr/sbin/opensips[15822]: DBG:dialog:lookup_dlg: no dialog id=677053409 found on entry 2317 Feb 26 17:28:41 voip2-sip23a /usr/sbin/opensips[15822]: DBG:dialog:dlg_onroute: unable to find dialog for BYE with route param 'd09.1e30b582' Feb 26 17:28:42 voip2-sip23a /usr/sbin/opensips[15820]: DBG:dialog:dlg_onroute: route param is 'd09.1e30b582' (len=12) Feb 26 17:28:42 voip2-sip23a /usr/sbin/opensips[15820]: DBG:dialog:lookup_dlg: no dialog id=677053409 found on entry 2317 Feb 26 17:28:42 voip2-sip23a /usr/sbin/opensips[15820]: DBG:dialog:dlg_onroute: unable to find dialog for BYE with route param 'd09.1e30b582' Feb 26 17:28:42 voip2-sip23a /usr/sbin/opensips[15820]: DBG:dialog:dlg_onroute: route param is 'd09.1e30b582' (len=12) Feb 26 17:28:42 voip2-sip23a /usr/sbin/opensips[15820]: DBG:dialog:lookup_dlg: no dialog id=677053409 found on entry 2317 If I purposely take down node1 again and try to make a call I can see that it indicates the failure to replicate dialog info on the cluster as below, Feb 26 17:27:01 voip2-sip23b /usr/sbin/opensips[4701]: ERROR:dialog:replicate_dialog_updated: All destinations in cluster: 1 are down or probing Feb 26 17:27:01 voip2-sip23b /usr/sbin/opensips[4701]: ERROR:dialog:replicate_dialog_updated: Failed to replicate updated dialog _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From pasandev at ymail.com Tue Feb 27 21:46:46 2018 From: pasandev at ymail.com (Pasan Meemaduma) Date: Wed, 28 Feb 2018 02:46:46 +0000 (UTC) Subject: [OpenSIPS-Users] dialog replication In-Reply-To: <107101422.7400213.1519783290651@mail.yahoo.com> References: <311827884.3546251.1519627085908.ref@mail.yahoo.com> <311827884.3546251.1519627085908@mail.yahoo.com> <704df948-6873-5fde-fe25-193f1cf2a420@opensips.org> <2103041839.6602135.1519697741901@mail.yahoo.com> <48b7be0c-d5d9-b6f3-a81b-42d768116f07@opensips.org> <107101422.7400213.1519783290651@mail.yahoo.com> Message-ID: <767753245.7406696.1519786006823@mail.yahoo.com> Hi Vlad, On a second thought its how anycast should work, packets could get to the closest node so basically with 2.3.3 we can't replicate transactions across mulitple opensips servers ? dialog replication  would only work for an active/passive setup doesn't it ? On Wednesday, 28 February 2018, 7:36, Pasan Meemaduma via Users wrote: Hi Vlad, That explains the issue then, my sip listener ip is an anycast one, and for some weired reason replies for INVITE that leaves node2 receive by node1 hence it can't create the dialog. I checked with node2 being down and recovered and calls init via node1 always received final reply hence dialog replication works. So it means my anycast configuration is broken right ? Thanks for you time to look in to it. On Tuesday, 27 February 2018, 21:42, Vlad Patrascu wrote: Hi Pasan, I don't see anything in the logs or your cluster configuration which could indicate that the dialogs don't replicate. Are you sure that the problem is consistently reproducible and that the INVITE for the call in question receives a final reply? Dialogs are replicated when they get to the confirmed state, and judging from the logs, this doesn't appear to be happening for that call. Regards, Vlad Patrascu OpenSIPS Developer http://www.opensips-solutions.com On 27.02.2018 04:15, Pasan Meemaduma via Users wrote: Hi Vlad, I have sent you the full debug logs as requested. Also clusterer_list mi command gives following output when dialog replication stopped working. node1>>opensipsctl fifo clusterer_list Cluster:: 1         Node:: 2 DB_ID=5 URL=bin:10.3.1.137:5566 Enabled=1 Link_state=Up      Next_hop=2 Description=Node 2  node2>> opensipsctl fifo clusterer_list Cluster:: 1         Node:: 1 DB_ID=4 URL=bin:10.3.1.136:5566 Enabled=1 Link_state=Up      Next_hop=1 Description=Node 1 1 On Monday, 26 February 2018, 18:02, Vlad Patrascu wrote: Hi, Can you send the full logs for both nodes from the time that node1 restarts onwards? Also, what is the output of 'clusterer_list' mi command on the instances? Regards, Vlad Patrascu OpenSIPS Developer http://www.opensips-solutions.com On 26.02.2018 08:38, Pasan Meemaduma via Users wrote: Hi Guys, Its me again :). I'm using dialog replication in opensips 2.3.3 and it appears it doesn't work after a node goes down and come back online. The recovered node doesn't seems to receiving dialog info via binary interface. I have node1 and node2 with dialog replication on. everything works fine and if I shutdown node1 and bring it back online after a while node2 doesn't send new call dialog info via binary interface. I have attach the debug trace from both nodes, if you need anything else let me know. I'm also using an anycast ip as the service ip. on node2 for new call Feb 26 17:28:35 voip2-sip23b /usr/sbin/opensips[4703]: DBG:dialog:build_new_dlg: new dialog 0x7f188afd3bf8(c=ktQ0Pkdwz50qGYjED6Brpw..,f=sip:XXXXXXXXX at somedomain;transport=UDP,t=sip:+YYYYYYYYYYY at somedomain;transport=UDP,ft=2b161508) on hash 2317 Feb 26 17:28:35 voip2-sip23b /usr/sbin/opensips[4703]: DBG:dialog:init_leg_info: route_set , contact sip:XXXXXXXXX at 192.168.27.11:56419;transport=UDP, cseq 2 and bind_addr udp:10.3.3.1:5060 Feb 26 17:28:35 voip2-sip23b /usr/sbin/opensips[4703]: DBG:dialog:dlg_add_leg_info: set leg 0 for 0x7f188afd3bf8: tag=<2b161508> rcseq=<0> Feb 26 17:28:35 voip2-sip23b /usr/sbin/opensips[4703]: DBG:dialog:link_dlg: ref dlg 0x7f188afd3bf8 with 3 -> 3 in h_entry 0x7f188afc3828 - 2317 Feb 26 17:28:35 voip2-sip23b /usr/sbin/opensips[4703]: DBG:dialog:new_dlg_val: inserting =<45ad4c76-1abe-11e8-9410-831894b67d0c> Feb 26 17:28:35 voip2-sip23b /usr/sbin/opensips[4703]: DBG:dialog:dlg_onreq: t hash_index = 47425, t label = 2069082013 Feb 26 17:28:35 voip2-sip23b /usr/sbin/opensips[4703]: DBG:dialog:dlg_update_contact: Updated dialog 0x7f188afd3bf8 contact to Feb 26 17:28:35 voip2-sip23b /usr/sbin/opensips[4703]: DBG:dialog:unref_dlg: unref dlg 0x7f188afd3bf8 with 1 -> 2 in entry 0x7f188afc3828 Feb 26 17:28:52 voip2-sip23b /usr/sbin/opensips[4702]: DBG:dialog:ref_dlg: ref dlg 0x7f188afd3bf8 with 1 -> 3 Feb 26 17:28:52 voip2-sip23b /usr/sbin/opensips[4702]: DBG:dialog:ref_dlg: ref dlg 0x7f188afd3bf8 with 1 -> 4 Feb 26 17:28:52 voip2-sip23b /usr/sbin/opensips[4702]: DBG:dialog:unref_dlg: unref dlg 0x7f188afd3bf8 with 1 -> 3 in entry 0x7f188afc3828 Feb 26 17:28:56 voip2-sip23b /usr/sbin/opensips[4711]: DBG:dialog:unref_dlg: unref dlg 0x7f188afd3bf8 with 1 -> 2 in entry 0x7f188afc3828 Feb 26 17:29:05 voip2-sip23b /usr/sbin/opensips[4711]: DBG:dialog:next_state_dlg: dialog 0x7f188afd3bf8 changed from state 1 to state 5, due event 4 Feb 26 17:29:05 voip2-sip23b /usr/sbin/opensips[4711]: DBG:dialog:dlg_onreply: dialog 0x7f188afd3bf8 failed (negative reply) Feb 26 17:29:05 voip2-sip23b /usr/sbin/opensips[4711]: DBG:dialog:unref_dlg: unref dlg 0x7f188afd3bf8 with 1 -> 1 in entry 0x7f188afc3828 Feb 26 17:29:11 voip2-sip23b /usr/sbin/opensips[4711]: DBG:dialog:next_state_dlg: dialog 0x7f188afd3bf8 changed from state 5 to state 5, due event 1 Feb 26 17:29:11 voip2-sip23b /usr/sbin/opensips[4711]: DBG:dialog:unref_dlg: unref dlg 0x7f188afd3bf8 with 1 -> 0 in entry 0x7f188afc3828 Feb 26 17:29:11 voip2-sip23b /usr/sbin/opensips[4711]: DBG:dialog:unref_dlg: ref <=0 for dialog 0x7f188afd3bf8 Feb 26 17:29:11 voip2-sip23b /usr/sbin/opensips[4711]: DBG:dialog:destroy_dlg: destroying dialog 0x7f188afd3bf8 Feb 26 17:29:11 voip2-sip23b /usr/sbin/opensips[4711]: DBG:dialog:destroy_dlg: dlg expired or not in list - dlg 0x7f188afd3bf8 [2317:677053409] with clid 'ktQ0Pkdwz50qGYjED6Brpw..' and tags '2b161508' 'NULL' on node1 Feb 26 17:28:41 voip2-sip23a /usr/sbin/opensips[15820]: DBG:dialog:dlg_onroute: route param is 'd09.1e30b582' (len=12) Feb 26 17:28:41 voip2-sip23a /usr/sbin/opensips[15820]: DBG:dialog:lookup_dlg: no dialog id=677053409 found on entry 2317 Feb 26 17:28:41 voip2-sip23a /usr/sbin/opensips[15820]: DBG:dialog:dlg_onroute: unable to find dialog for BYE with route param 'd09.1e30b582' Feb 26 17:28:41 voip2-sip23a /usr/sbin/opensips[15822]: DBG:dialog:dlg_onroute: route param is 'd09.1e30b582' (len=12) Feb 26 17:28:41 voip2-sip23a /usr/sbin/opensips[15822]: DBG:dialog:lookup_dlg: no dialog id=677053409 found on entry 2317 Feb 26 17:28:41 voip2-sip23a /usr/sbin/opensips[15822]: DBG:dialog:dlg_onroute: unable to find dialog for BYE with route param 'd09.1e30b582' Feb 26 17:28:41 voip2-sip23a /usr/sbin/opensips[15822]: DBG:dialog:dlg_onroute: route param is 'd09.1e30b582' (len=12) Feb 26 17:28:41 voip2-sip23a /usr/sbin/opensips[15822]: DBG:dialog:lookup_dlg: no dialog id=677053409 found on entry 2317 Feb 26 17:28:41 voip2-sip23a /usr/sbin/opensips[15822]: DBG:dialog:dlg_onroute: unable to find dialog for BYE with route param 'd09.1e30b582' Feb 26 17:28:42 voip2-sip23a /usr/sbin/opensips[15820]: DBG:dialog:dlg_onroute: route param is 'd09.1e30b582' (len=12) Feb 26 17:28:42 voip2-sip23a /usr/sbin/opensips[15820]: DBG:dialog:lookup_dlg: no dialog id=677053409 found on entry 2317 Feb 26 17:28:42 voip2-sip23a /usr/sbin/opensips[15820]: DBG:dialog:dlg_onroute: unable to find dialog for BYE with route param 'd09.1e30b582' Feb 26 17:28:42 voip2-sip23a /usr/sbin/opensips[15820]: DBG:dialog:dlg_onroute: route param is 'd09.1e30b582' (len=12) Feb 26 17:28:42 voip2-sip23a /usr/sbin/opensips[15820]: DBG:dialog:lookup_dlg: no dialog id=677053409 found on entry 2317 If I purposely take down node1 again and try to make a call I can see that it indicates the failure to replicate dialog info on the cluster as below, Feb 26 17:27:01 voip2-sip23b /usr/sbin/opensips[4701]: ERROR:dialog:replicate_dialog_updated: All destinations in cluster: 1 are down or probing Feb 26 17:27:01 voip2-sip23b /usr/sbin/opensips[4701]: ERROR:dialog:replicate_dialog_updated: Failed to replicate updated dialog _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From pasandev at ymail.com Tue Feb 27 21:46:46 2018 From: pasandev at ymail.com (Pasan Meemaduma) Date: Wed, 28 Feb 2018 02:46:46 +0000 (UTC) Subject: [OpenSIPS-Users] dialog replication In-Reply-To: <107101422.7400213.1519783290651@mail.yahoo.com> References: <311827884.3546251.1519627085908.ref@mail.yahoo.com> <311827884.3546251.1519627085908@mail.yahoo.com> <704df948-6873-5fde-fe25-193f1cf2a420@opensips.org> <2103041839.6602135.1519697741901@mail.yahoo.com> <48b7be0c-d5d9-b6f3-a81b-42d768116f07@opensips.org> <107101422.7400213.1519783290651@mail.yahoo.com> Message-ID: <767753245.7406696.1519786006823@mail.yahoo.com> Hi Vlad, On a second thought its how anycast should work, packets could get to the closest node so basically with 2.3.3 we can't replicate transactions across mulitple opensips servers ? dialog replication  would only work for an active/passive setup doesn't it ? On Wednesday, 28 February 2018, 7:36, Pasan Meemaduma via Users wrote: Hi Vlad, That explains the issue then, my sip listener ip is an anycast one, and for some weired reason replies for INVITE that leaves node2 receive by node1 hence it can't create the dialog. I checked with node2 being down and recovered and calls init via node1 always received final reply hence dialog replication works. So it means my anycast configuration is broken right ? Thanks for you time to look in to it. On Tuesday, 27 February 2018, 21:42, Vlad Patrascu wrote: Hi Pasan, I don't see anything in the logs or your cluster configuration which could indicate that the dialogs don't replicate. Are you sure that the problem is consistently reproducible and that the INVITE for the call in question receives a final reply? Dialogs are replicated when they get to the confirmed state, and judging from the logs, this doesn't appear to be happening for that call. Regards, Vlad Patrascu OpenSIPS Developer http://www.opensips-solutions.com On 27.02.2018 04:15, Pasan Meemaduma via Users wrote: Hi Vlad, I have sent you the full debug logs as requested. Also clusterer_list mi command gives following output when dialog replication stopped working. node1>>opensipsctl fifo clusterer_list Cluster:: 1         Node:: 2 DB_ID=5 URL=bin:10.3.1.137:5566 Enabled=1 Link_state=Up      Next_hop=2 Description=Node 2  node2>> opensipsctl fifo clusterer_list Cluster:: 1         Node:: 1 DB_ID=4 URL=bin:10.3.1.136:5566 Enabled=1 Link_state=Up      Next_hop=1 Description=Node 1 1 On Monday, 26 February 2018, 18:02, Vlad Patrascu wrote: Hi, Can you send the full logs for both nodes from the time that node1 restarts onwards? Also, what is the output of 'clusterer_list' mi command on the instances? Regards, Vlad Patrascu OpenSIPS Developer http://www.opensips-solutions.com On 26.02.2018 08:38, Pasan Meemaduma via Users wrote: Hi Guys, Its me again :). I'm using dialog replication in opensips 2.3.3 and it appears it doesn't work after a node goes down and come back online. The recovered node doesn't seems to receiving dialog info via binary interface. I have node1 and node2 with dialog replication on. everything works fine and if I shutdown node1 and bring it back online after a while node2 doesn't send new call dialog info via binary interface. I have attach the debug trace from both nodes, if you need anything else let me know. I'm also using an anycast ip as the service ip. on node2 for new call Feb 26 17:28:35 voip2-sip23b /usr/sbin/opensips[4703]: DBG:dialog:build_new_dlg: new dialog 0x7f188afd3bf8(c=ktQ0Pkdwz50qGYjED6Brpw..,f=sip:XXXXXXXXX at somedomain;transport=UDP,t=sip:+YYYYYYYYYYY at somedomain;transport=UDP,ft=2b161508) on hash 2317 Feb 26 17:28:35 voip2-sip23b /usr/sbin/opensips[4703]: DBG:dialog:init_leg_info: route_set , contact sip:XXXXXXXXX at 192.168.27.11:56419;transport=UDP, cseq 2 and bind_addr udp:10.3.3.1:5060 Feb 26 17:28:35 voip2-sip23b /usr/sbin/opensips[4703]: DBG:dialog:dlg_add_leg_info: set leg 0 for 0x7f188afd3bf8: tag=<2b161508> rcseq=<0> Feb 26 17:28:35 voip2-sip23b /usr/sbin/opensips[4703]: DBG:dialog:link_dlg: ref dlg 0x7f188afd3bf8 with 3 -> 3 in h_entry 0x7f188afc3828 - 2317 Feb 26 17:28:35 voip2-sip23b /usr/sbin/opensips[4703]: DBG:dialog:new_dlg_val: inserting =<45ad4c76-1abe-11e8-9410-831894b67d0c> Feb 26 17:28:35 voip2-sip23b /usr/sbin/opensips[4703]: DBG:dialog:dlg_onreq: t hash_index = 47425, t label = 2069082013 Feb 26 17:28:35 voip2-sip23b /usr/sbin/opensips[4703]: DBG:dialog:dlg_update_contact: Updated dialog 0x7f188afd3bf8 contact to Feb 26 17:28:35 voip2-sip23b /usr/sbin/opensips[4703]: DBG:dialog:unref_dlg: unref dlg 0x7f188afd3bf8 with 1 -> 2 in entry 0x7f188afc3828 Feb 26 17:28:52 voip2-sip23b /usr/sbin/opensips[4702]: DBG:dialog:ref_dlg: ref dlg 0x7f188afd3bf8 with 1 -> 3 Feb 26 17:28:52 voip2-sip23b /usr/sbin/opensips[4702]: DBG:dialog:ref_dlg: ref dlg 0x7f188afd3bf8 with 1 -> 4 Feb 26 17:28:52 voip2-sip23b /usr/sbin/opensips[4702]: DBG:dialog:unref_dlg: unref dlg 0x7f188afd3bf8 with 1 -> 3 in entry 0x7f188afc3828 Feb 26 17:28:56 voip2-sip23b /usr/sbin/opensips[4711]: DBG:dialog:unref_dlg: unref dlg 0x7f188afd3bf8 with 1 -> 2 in entry 0x7f188afc3828 Feb 26 17:29:05 voip2-sip23b /usr/sbin/opensips[4711]: DBG:dialog:next_state_dlg: dialog 0x7f188afd3bf8 changed from state 1 to state 5, due event 4 Feb 26 17:29:05 voip2-sip23b /usr/sbin/opensips[4711]: DBG:dialog:dlg_onreply: dialog 0x7f188afd3bf8 failed (negative reply) Feb 26 17:29:05 voip2-sip23b /usr/sbin/opensips[4711]: DBG:dialog:unref_dlg: unref dlg 0x7f188afd3bf8 with 1 -> 1 in entry 0x7f188afc3828 Feb 26 17:29:11 voip2-sip23b /usr/sbin/opensips[4711]: DBG:dialog:next_state_dlg: dialog 0x7f188afd3bf8 changed from state 5 to state 5, due event 1 Feb 26 17:29:11 voip2-sip23b /usr/sbin/opensips[4711]: DBG:dialog:unref_dlg: unref dlg 0x7f188afd3bf8 with 1 -> 0 in entry 0x7f188afc3828 Feb 26 17:29:11 voip2-sip23b /usr/sbin/opensips[4711]: DBG:dialog:unref_dlg: ref <=0 for dialog 0x7f188afd3bf8 Feb 26 17:29:11 voip2-sip23b /usr/sbin/opensips[4711]: DBG:dialog:destroy_dlg: destroying dialog 0x7f188afd3bf8 Feb 26 17:29:11 voip2-sip23b /usr/sbin/opensips[4711]: DBG:dialog:destroy_dlg: dlg expired or not in list - dlg 0x7f188afd3bf8 [2317:677053409] with clid 'ktQ0Pkdwz50qGYjED6Brpw..' and tags '2b161508' 'NULL' on node1 Feb 26 17:28:41 voip2-sip23a /usr/sbin/opensips[15820]: DBG:dialog:dlg_onroute: route param is 'd09.1e30b582' (len=12) Feb 26 17:28:41 voip2-sip23a /usr/sbin/opensips[15820]: DBG:dialog:lookup_dlg: no dialog id=677053409 found on entry 2317 Feb 26 17:28:41 voip2-sip23a /usr/sbin/opensips[15820]: DBG:dialog:dlg_onroute: unable to find dialog for BYE with route param 'd09.1e30b582' Feb 26 17:28:41 voip2-sip23a /usr/sbin/opensips[15822]: DBG:dialog:dlg_onroute: route param is 'd09.1e30b582' (len=12) Feb 26 17:28:41 voip2-sip23a /usr/sbin/opensips[15822]: DBG:dialog:lookup_dlg: no dialog id=677053409 found on entry 2317 Feb 26 17:28:41 voip2-sip23a /usr/sbin/opensips[15822]: DBG:dialog:dlg_onroute: unable to find dialog for BYE with route param 'd09.1e30b582' Feb 26 17:28:41 voip2-sip23a /usr/sbin/opensips[15822]: DBG:dialog:dlg_onroute: route param is 'd09.1e30b582' (len=12) Feb 26 17:28:41 voip2-sip23a /usr/sbin/opensips[15822]: DBG:dialog:lookup_dlg: no dialog id=677053409 found on entry 2317 Feb 26 17:28:41 voip2-sip23a /usr/sbin/opensips[15822]: DBG:dialog:dlg_onroute: unable to find dialog for BYE with route param 'd09.1e30b582' Feb 26 17:28:42 voip2-sip23a /usr/sbin/opensips[15820]: DBG:dialog:dlg_onroute: route param is 'd09.1e30b582' (len=12) Feb 26 17:28:42 voip2-sip23a /usr/sbin/opensips[15820]: DBG:dialog:lookup_dlg: no dialog id=677053409 found on entry 2317 Feb 26 17:28:42 voip2-sip23a /usr/sbin/opensips[15820]: DBG:dialog:dlg_onroute: unable to find dialog for BYE with route param 'd09.1e30b582' Feb 26 17:28:42 voip2-sip23a /usr/sbin/opensips[15820]: DBG:dialog:dlg_onroute: route param is 'd09.1e30b582' (len=12) Feb 26 17:28:42 voip2-sip23a /usr/sbin/opensips[15820]: DBG:dialog:lookup_dlg: no dialog id=677053409 found on entry 2317 If I purposely take down node1 again and try to make a call I can see that it indicates the failure to replicate dialog info on the cluster as below, Feb 26 17:27:01 voip2-sip23b /usr/sbin/opensips[4701]: ERROR:dialog:replicate_dialog_updated: All destinations in cluster: 1 are down or probing Feb 26 17:27:01 voip2-sip23b /usr/sbin/opensips[4701]: ERROR:dialog:replicate_dialog_updated: Failed to replicate updated dialog _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From Ben.Newlin at genesys.com Tue Feb 27 22:47:06 2018 From: Ben.Newlin at genesys.com (Ben Newlin) Date: Wed, 28 Feb 2018 03:47:06 +0000 Subject: [OpenSIPS-Users] OpenSIPS crashing on 200 OK In-Reply-To: <478773D5-3955-4A6A-BA6F-15467E39CC29@genesys.com> References: <530E1611-2A4D-424F-938D-2E409E1A78A2@genesys.com> <7cf6bb3d-967a-b8e2-d4b3-8b42c8dbf0b8@opensips.org> <755F831E-90E4-4105-80C5-42C7A2E2050A@genesys.com> <6D9EBB18-914E-49ED-988F-7D582007B88E@genesys.com> <085c6c6b-75ca-0557-aa5e-0fcb9e3eb07f@opensips.org> <4FD028FE-6607-496C-8B3A-EF4293FB6155@genesys.com> <8eeebbf4-9c57-dd6a-ebf1-bbe6dadb0bdb@opensips.org> <279057EB-BEC2-4552-B0C9-EDD244263E32@genesys.com> <478773D5-3955-4A6A-BA6F-15467E39CC29@genesys.com> Message-ID: Unfortunately, I have had another crash during testing. I was not able to capture the core, but I did get this log: Feb 28 03:35:22 [359] CRITICAL:core:qm_debug_frag: qm_*: prev. fragm. tail overwritten(c0c0c0c0c0c0c020, abcdefedabcdefed)[0x7fa6cd6c46b8:0x7fa6cd6c46f0] (del_uri_param, checks.c:242)! Feb 28 03:35:23 [353] INFO:core:handle_sigs: child process 359 exited by a signal 6 I am making adjustments to be able to collect the core if the crash occurs again. Thanks, Ben Newlin From: Ben Newlin Date: Tuesday, February 27, 2018 at 2:47 PM To: Bogdan-Andrei Iancu , OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] OpenSIPS crashing on 200 OK Bogdan, I found this previous thread that has this error: http://lists.opensips.org/pipermail/users/2016-October/035635.html The fix suggested there worked. When I removed the quotes around the regexp used in re.subst the crash did not occur. Ben Newlin From: Ben Newlin Date: Tuesday, February 27, 2018 at 2:20 PM To: Bogdan-Andrei Iancu , OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] OpenSIPS crashing on 200 OK Bogdan, I just tested it and that crash is no longer occurring. However I am now seeing a different crash, but it seems to still be related to regexp. Feb 27 19:07:17 [369] ERROR:core:subst_parser: string too short Feb 27 19:07:17 [369] ERROR:core:tr_eval_re: Can't compile regexp Feb 27 19:07:17 [369] WARNING:core:do_assign: no value in right expression at /opensips/control.cfg:1952 Feb 27 19:07:17 [361] INFO:core:handle_sigs: child process 369 exited by a signal 11 Feb 27 19:07:17 [361] INFO:core:handle_sigs: core was generated Feb 27 19:07:17 [361] INFO:core:handle_sigs: terminating due to SIGCHLD The do_assign error makes it seem like there may be something wrong in my script, but I can’t see what it is. And I’m not sure what the “string too short” error is either or why the regexp could not be compiled. It looks valid to me. Backtrace: https://pastebin.com/Fk2BRnsg Ben Newlin From: Bogdan-Andrei Iancu Date: Tuesday, February 27, 2018 at 12:55 PM To: OpenSIPS users mailling list , Ben Newlin Subject: Re: [OpenSIPS-Users] OpenSIPS crashing on 200 OK Hi Ben, I just committed the fix for this - could you update from GIT and give it a new try ? Thanks and regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Summit 2018 http://www.opensips.org/events/Summit-2018Amsterdam On 02/27/2018 06:19 PM, Bogdan-Andrei Iancu wrote: Thanks Ben for the support in investigating this. The bug was identified and confirm, let me work on a fix right now. I will shortly ping you for testing ;) Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Summit 2018 http://www.opensips.org/events/Summit-2018Amsterdam On 02/27/2018 12:12 AM, Bogdan-Andrei Iancu wrote: OK, to be more precise, the crash is around an "If" statement where you have in the condition an REGEXP test (=~) with a left operand translated into "downstream" (maybe a $DLG_dir ??). Does it sound familiar to you :) ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Summit 2018 http://www.opensips.org/events/Summit-2018Amsterdam On 02/26/2018 11:32 PM, Ben Newlin wrote: Yes, that is from that exact file, line 435. It’s just a return statement. I can send you the file off-list if you’d like. I do use the $DLG_dir in my script in a few places, but the only one this crash would have hit is in some logging. Ben Newlin -------------- next part -------------- An HTML attachment was scrubbed... URL: From Ben.Newlin at genesys.com Tue Feb 27 23:19:13 2018 From: Ben.Newlin at genesys.com (Ben Newlin) Date: Wed, 28 Feb 2018 04:19:13 +0000 Subject: [OpenSIPS-Users] OpenSIPS crashing on 200 OK In-Reply-To: References: <530E1611-2A4D-424F-938D-2E409E1A78A2@genesys.com> <7cf6bb3d-967a-b8e2-d4b3-8b42c8dbf0b8@opensips.org> <755F831E-90E4-4105-80C5-42C7A2E2050A@genesys.com> <6D9EBB18-914E-49ED-988F-7D582007B88E@genesys.com> <085c6c6b-75ca-0557-aa5e-0fcb9e3eb07f@opensips.org> <4FD028FE-6607-496C-8B3A-EF4293FB6155@genesys.com> <8eeebbf4-9c57-dd6a-ebf1-bbe6dadb0bdb@opensips.org> <279057EB-BEC2-4552-B0C9-EDD244263E32@genesys.com> <478773D5-3955-4A6A-BA6F-15467E39CC29@genesys.com> Message-ID: <99C36C64-1230-4967-8DD9-2CBF51768742@genesys.com> I was able to reproduce and capture a core dump of the crash. Backtrace is available here: https://pastebin.com/WtBB6cKm Thanks, Ben Newlin From: Ben Newlin Date: Tuesday, February 27, 2018 at 10:47 PM To: Bogdan-Andrei Iancu , OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] OpenSIPS crashing on 200 OK Unfortunately, I have had another crash during testing. I was not able to capture the core, but I did get this log: Feb 28 03:35:22 [359] CRITICAL:core:qm_debug_frag: qm_*: prev. fragm. tail overwritten(c0c0c0c0c0c0c020, abcdefedabcdefed)[0x7fa6cd6c46b8:0x7fa6cd6c46f0] (del_uri_param, checks.c:242)! Feb 28 03:35:23 [353] INFO:core:handle_sigs: child process 359 exited by a signal 6 I am making adjustments to be able to collect the core if the crash occurs again. Thanks, Ben Newlin From: Ben Newlin Date: Tuesday, February 27, 2018 at 2:47 PM To: Bogdan-Andrei Iancu , OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] OpenSIPS crashing on 200 OK Bogdan, I found this previous thread that has this error: http://lists.opensips.org/pipermail/users/2016-October/035635.html The fix suggested there worked. When I removed the quotes around the regexp used in re.subst the crash did not occur. Ben Newlin From: Ben Newlin Date: Tuesday, February 27, 2018 at 2:20 PM To: Bogdan-Andrei Iancu , OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] OpenSIPS crashing on 200 OK Bogdan, I just tested it and that crash is no longer occurring. However I am now seeing a different crash, but it seems to still be related to regexp. Feb 27 19:07:17 [369] ERROR:core:subst_parser: string too short Feb 27 19:07:17 [369] ERROR:core:tr_eval_re: Can't compile regexp Feb 27 19:07:17 [369] WARNING:core:do_assign: no value in right expression at /opensips/control.cfg:1952 Feb 27 19:07:17 [361] INFO:core:handle_sigs: child process 369 exited by a signal 11 Feb 27 19:07:17 [361] INFO:core:handle_sigs: core was generated Feb 27 19:07:17 [361] INFO:core:handle_sigs: terminating due to SIGCHLD The do_assign error makes it seem like there may be something wrong in my script, but I can’t see what it is. And I’m not sure what the “string too short” error is either or why the regexp could not be compiled. It looks valid to me. Backtrace: https://pastebin.com/Fk2BRnsg Ben Newlin From: Bogdan-Andrei Iancu Date: Tuesday, February 27, 2018 at 12:55 PM To: OpenSIPS users mailling list , Ben Newlin Subject: Re: [OpenSIPS-Users] OpenSIPS crashing on 200 OK Hi Ben, I just committed the fix for this - could you update from GIT and give it a new try ? Thanks and regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Summit 2018 http://www.opensips.org/events/Summit-2018Amsterdam On 02/27/2018 06:19 PM, Bogdan-Andrei Iancu wrote: Thanks Ben for the support in investigating this. The bug was identified and confirm, let me work on a fix right now. I will shortly ping you for testing ;) Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Summit 2018 http://www.opensips.org/events/Summit-2018Amsterdam On 02/27/2018 12:12 AM, Bogdan-Andrei Iancu wrote: OK, to be more precise, the crash is around an "If" statement where you have in the condition an REGEXP test (=~) with a left operand translated into "downstream" (maybe a $DLG_dir ??). Does it sound familiar to you :) ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Summit 2018 http://www.opensips.org/events/Summit-2018Amsterdam On 02/26/2018 11:32 PM, Ben Newlin wrote: Yes, that is from that exact file, line 435. It’s just a return statement. I can send you the file off-list if you’d like. I do use the $DLG_dir in my script in a few places, but the only one this crash would have hit is in some logging. Ben Newlin -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Wed Feb 28 05:37:17 2018 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Wed, 28 Feb 2018 12:37:17 +0200 Subject: [OpenSIPS-Users] OpenSIPS crashing on 200 OK In-Reply-To: <478773D5-3955-4A6A-BA6F-15467E39CC29@genesys.com> References: <530E1611-2A4D-424F-938D-2E409E1A78A2@genesys.com> <7cf6bb3d-967a-b8e2-d4b3-8b42c8dbf0b8@opensips.org> <755F831E-90E4-4105-80C5-42C7A2E2050A@genesys.com> <6D9EBB18-914E-49ED-988F-7D582007B88E@genesys.com> <085c6c6b-75ca-0557-aa5e-0fcb9e3eb07f@opensips.org> <4FD028FE-6607-496C-8B3A-EF4293FB6155@genesys.com> <8eeebbf4-9c57-dd6a-ebf1-bbe6dadb0bdb@opensips.org> <279057EB-BEC2-4552-B0C9-EDD244263E32@genesys.com> <478773D5-3955-4A6A-BA6F-15467E39CC29@genesys.com> Message-ID: <689fb7e7-6bc3-2f0a-21e5-ed3b69d28af8@opensips.org> Hi Ben Thanks for more testing :). I managed to identify the bug for this crash - even if the regexp is bogus, opensips should not crash. See https://github.com/OpenSIPS/opensips/commit/37eab9a8c2dd19a17f2617057ca2014e9ffee263 Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Summit 2018 http://www.opensips.org/events/Summit-2018Amsterdam On 02/27/2018 09:47 PM, Ben Newlin wrote: > > Bogdan, > > I found this previous thread that has this error: > http://lists.opensips.org/pipermail/users/2016-October/035635.html > > The fix suggested there worked. When I removed the quotes around the > regexp used in re.subst the crash did not occur. > > Ben Newlin > > *From: *Ben Newlin > *Date: *Tuesday, February 27, 2018 at 2:20 PM > *To: *Bogdan-Andrei Iancu , OpenSIPS users > mailling list > *Subject: *Re: [OpenSIPS-Users] OpenSIPS crashing on 200 OK > > Bogdan, > > I just tested it and that crash is no longer occurring. However I am > now seeing a different crash, but it seems to still be related to regexp. > > Feb 27 19:07:17 [369] ERROR:core:subst_parser: string too short > > Feb 27 19:07:17 [369] ERROR:core:tr_eval_re: Can't compile regexp > > Feb 27 19:07:17 [369] WARNING:core:do_assign: no value in right > expression at /opensips/control.cfg:1952 > > Feb 27 19:07:17 [361] INFO:core:handle_sigs: child process 369 exited > by a signal 11 > > Feb 27 19:07:17 [361] INFO:core:handle_sigs: core was generated > > Feb 27 19:07:17 [361] INFO:core:handle_sigs: terminating due to SIGCHLD > > The do_assign error makes it seem like there may be something wrong in > my script, but I can’t see what it is. And I’m not sure what the > “string too short” error is either or why the regexp could not be > compiled. It looks valid to me. > > Backtrace: https://pastebin.com/Fk2BRnsg > > Ben Newlin > > *From: *Bogdan-Andrei Iancu > *Date: *Tuesday, February 27, 2018 at 12:55 PM > *To: *OpenSIPS users mailling list , Ben > Newlin > *Subject: *Re: [OpenSIPS-Users] OpenSIPS crashing on 200 OK > > Hi Ben, > > I just committed the fix for this - could you update from GIT and give > it a new try ? > > Thanks and regards, > > > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > OpenSIPS Summit 2018 > http://www.opensips.org/events/Summit-2018Amsterdam > > On 02/27/2018 06:19 PM, Bogdan-Andrei Iancu wrote: > > Thanks Ben for the support in investigating this. The bug was > identified and confirm, let me work on a fix right now. I will > shortly ping you for testing ;) > > Regards, > > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > > http://www.opensips-solutions.com > > OpenSIPS Summit 2018 > > http://www.opensips.org/events/Summit-2018Amsterdam > > On 02/27/2018 12:12 AM, Bogdan-Andrei Iancu wrote: > > OK, to be more precise, the crash is around an "If" statement > where you have in the condition an REGEXP test (=~) with a > left operand translated into "downstream" (maybe a $DLG_dir ??). > Does it sound familiar to you :) ? > > Regards, > > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > > http://www.opensips-solutions.com > > OpenSIPS Summit 2018 > > http://www.opensips.org/events/Summit-2018Amsterdam > > On 02/26/2018 11:32 PM, Ben Newlin wrote: > > Yes, that is from that exact file, line 435. It’s just a > return statement. I can send you the file off-list if > you’d like. > > I do use the $DLG_dir in my script in a few places, but > the only one this crash would have hit is in some logging. > > Ben Newlin > > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Wed Feb 28 06:27:32 2018 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Wed, 28 Feb 2018 13:27:32 +0200 Subject: [OpenSIPS-Users] OpenSIPS crashing on 200 OK In-Reply-To: <99C36C64-1230-4967-8DD9-2CBF51768742@genesys.com> References: <530E1611-2A4D-424F-938D-2E409E1A78A2@genesys.com> <7cf6bb3d-967a-b8e2-d4b3-8b42c8dbf0b8@opensips.org> <755F831E-90E4-4105-80C5-42C7A2E2050A@genesys.com> <6D9EBB18-914E-49ED-988F-7D582007B88E@genesys.com> <085c6c6b-75ca-0557-aa5e-0fcb9e3eb07f@opensips.org> <4FD028FE-6607-496C-8B3A-EF4293FB6155@genesys.com> <8eeebbf4-9c57-dd6a-ebf1-bbe6dadb0bdb@opensips.org> <279057EB-BEC2-4552-B0C9-EDD244263E32@genesys.com> <478773D5-3955-4A6A-BA6F-15467E39CC29@genesys.com> <99C36C64-1230-4967-8DD9-2CBF51768742@genesys.com> Message-ID: <138a08a6-8021-1d6d-3d7d-444fe136a61a@opensips.org> Hi Ben, This new crash was also fixed - see https://github.com/OpenSIPS/opensips/commit/34f81b223fb466ecddf648f72f615ab24d4a86dc . Do you have any more crashes on the pipe to report :D ? I'm getting warm :) Thanks and regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Summit 2018 http://www.opensips.org/events/Summit-2018Amsterdam On 02/28/2018 06:19 AM, Ben Newlin wrote: > > I was able to reproduce and capture a core dump of the crash. > Backtrace is available here: https://pastebin.com/WtBB6cKm > > Thanks, > > Ben Newlin > > *From: *Ben Newlin > *Date: *Tuesday, February 27, 2018 at 10:47 PM > *To: *Bogdan-Andrei Iancu , OpenSIPS users > mailling list > *Subject: *Re: [OpenSIPS-Users] OpenSIPS crashing on 200 OK > > Unfortunately, I have had another crash during testing. I was not able > to capture the core, but I did get this log: > > Feb 28 03:35:22 [359] CRITICAL:core:qm_debug_frag: qm_*: prev. fragm. > tail overwritten(c0c0c0c0c0c0c020, > abcdefedabcdefed)[0x7fa6cd6c46b8:0x7fa6cd6c46f0] (del_uri_param, > checks.c:242)! > > Feb 28 03:35:23 [353] INFO:core:handle_sigs: child process 359 exited > by a signal 6 > > I am making adjustments to be able to collect the core if the crash > occurs again. > > Thanks, > > Ben Newlin > > *From: *Ben Newlin > *Date: *Tuesday, February 27, 2018 at 2:47 PM > *To: *Bogdan-Andrei Iancu , OpenSIPS users > mailling list > *Subject: *Re: [OpenSIPS-Users] OpenSIPS crashing on 200 OK > > Bogdan, > > I found this previous thread that has this error: > http://lists.opensips.org/pipermail/users/2016-October/035635.html > > The fix suggested there worked. When I removed the quotes around the > regexp used in re.subst the crash did not occur. > > Ben Newlin > > *From: *Ben Newlin > *Date: *Tuesday, February 27, 2018 at 2:20 PM > *To: *Bogdan-Andrei Iancu , OpenSIPS users > mailling list > *Subject: *Re: [OpenSIPS-Users] OpenSIPS crashing on 200 OK > > Bogdan, > > I just tested it and that crash is no longer occurring. However I am > now seeing a different crash, but it seems to still be related to regexp. > > Feb 27 19:07:17 [369] ERROR:core:subst_parser: string too short > > Feb 27 19:07:17 [369] ERROR:core:tr_eval_re: Can't compile regexp > > Feb 27 19:07:17 [369] WARNING:core:do_assign: no value in right > expression at /opensips/control.cfg:1952 > > Feb 27 19:07:17 [361] INFO:core:handle_sigs: child process 369 exited > by a signal 11 > > Feb 27 19:07:17 [361] INFO:core:handle_sigs: core was generated > > Feb 27 19:07:17 [361] INFO:core:handle_sigs: terminating due to SIGCHLD > > The do_assign error makes it seem like there may be something wrong in > my script, but I can’t see what it is. And I’m not sure what the > “string too short” error is either or why the regexp could not be > compiled. It looks valid to me. > > Backtrace: https://pastebin.com/Fk2BRnsg > > Ben Newlin > > *From: *Bogdan-Andrei Iancu > *Date: *Tuesday, February 27, 2018 at 12:55 PM > *To: *OpenSIPS users mailling list , Ben > Newlin > *Subject: *Re: [OpenSIPS-Users] OpenSIPS crashing on 200 OK > > Hi Ben, > > I just committed the fix for this - could you update from GIT and give > it a new try ? > > Thanks and regards, > > > > > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > OpenSIPS Summit 2018 > http://www.opensips.org/events/Summit-2018Amsterdam > > On 02/27/2018 06:19 PM, Bogdan-Andrei Iancu wrote: > > Thanks Ben for the support in investigating this. The bug was > identified and confirm, let me work on a fix right now. I will > shortly ping you for testing ;) > > Regards, > > > > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > > http://www.opensips-solutions.com > > OpenSIPS Summit 2018 > > http://www.opensips.org/events/Summit-2018Amsterdam > > On 02/27/2018 12:12 AM, Bogdan-Andrei Iancu wrote: > > OK, to be more precise, the crash is around an "If" statement > where you have in the condition an REGEXP test (=~) with a > left operand translated into "downstream" (maybe a $DLG_dir ??). > Does it sound familiar to you :) ? > > Regards, > > > > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > > http://www.opensips-solutions.com > > OpenSIPS Summit 2018 > > http://www.opensips.org/events/Summit-2018Amsterdam > > On 02/26/2018 11:32 PM, Ben Newlin wrote: > > Yes, that is from that exact file, line 435. It’s just a > return statement. I can send you the file off-list if > you’d like. > > I do use the $DLG_dir in my script in a few places, but > the only one this crash would have hit is in some logging. > > Ben Newlin > > > > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From xaled at web.de Wed Feb 28 05:36:07 2018 From: xaled at web.de (xaled) Date: Wed, 28 Feb 2018 11:36:07 +0100 Subject: [OpenSIPS-Users] Wrong number of children In-Reply-To: References: <027501d3aca1$66551340$32ff39c0$@web.de> <010001d3aef6$02f70f90$08e52eb0$@web.de> <4e50d0e9-d0c8-5317-4f21-433cbc5cde13@opensips.org> <01ff01d3af19$fda10db0$f8e32910$@web.de> <15812914-24ca-8dfa-9371-732ce42c91bd@opensips.org> <026201d3af1e$a7be42a0$f73ac7e0$@web.de> <02dd01d3af2d$b3380b20$19a82160$@web.de> <00a601d3afe8$3c0a0740$b41e15c0$@web.de> <3f4b7059-74f6-e2ca-0697-123c3ccac637@opensips.org> Message-ID: <016101d3b07f$f1ae3f20$d50abd60$@web.de> Hi Bodgan, thanks, now there are no unwanted TCP children ;) Number of SIP receiver UDP children matches the children param value. ivr01:/usr/local/etc/opensips # /usr/local/sbin/opensipsctl fifo ps Process:: ID=0 PID=12872 Type=attendant Process:: ID=1 PID=12873 Type=MI FIFO Process:: ID=2 PID=12874 Type=time_keeper Process:: ID=3 PID=12875 Type=timer Process:: ID=4 PID=12876 Type=SIP receiver udp: x.x.x.x:5060 Process:: ID=5 PID=12878 Type=SIP receiver udp: x.x.x.x:5060 Process:: ID=6 PID=12879 Type=SIP receiver udp: x.x.x.x:5060 Process:: ID=7 PID=12880 Type=SIP receiver udp: x.x.x.x:5060 Process:: ID=8 PID=12881 Type=Timer handler Xaled From: Bogdan-Andrei Iancu [mailto:bogdan at opensips.org] Sent: Tuesday, February 27, 2018 6:39 PM To: OpenSIPS users mailling list ; xaled Subject: Re: [OpenSIPS-Users] Wrong number of children Ok, thanks to Razvan, we have an fix for this: https://github.com/OpenSIPS/opensips/commit/b022ad5a7c1c24aa9cf3ddf22d5f50510807c9f4 Please update from GIT and give it a try. Many Thanks, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Summit 2018 http://www.opensips.org/events/Summit-2018Amsterdam On 02/27/2018 06:38 PM, Bogdan-Andrei Iancu wrote: Thank you Xaled, Based on the output you provided I managed to identify the issue. There is a tricky internal test that actually forces the TCP layer to start more or less all the time. Let us have this fixed !! I will update you Best regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Summit 2018 http://www.opensips.org/events/Summit-2018Amsterdam On 02/27/2018 06:30 PM, xaled wrote: Hi Bogdan, It’s a different server, but the same tcp issue: 2018-02-27T17:19:29.939207+01:00 dus-ivr01 opensips: DBG:core:main: Loaded 1 transport protocols 2018-02-27T17:19:29.939411+01:00 dus-ivr01 opensips: DBG:core:socket2str: 2018-02-27T17:19:29.939630+01:00 dus-ivr01 opensips: INFO:core:tcp_init: enabling TCP for proto 5 2018-02-27T17:19:29.939842+01:00 dus-ivr01 opensips: DBG:core:ipc_register_handler: IPC type 0 [TCP reporting] registered with handler 0x5f9fee 2018-02-27T17:19:29.940120+01:00 dus-ivr01 opensips: DBG:core:create_status_pipe: pipe created ? rc = 0, errno = Success 2018-02-27T17:19:29.940326+01:00 dus-ivr01 opensips: DBG:core:daemonize: waiting for status code from children 2018-02-27T17:19:29.940526+01:00 dus-ivr01 opensips: DBG:core:count_init_children: 15 children are going to be inited 2018-02-27T17:19:29.940724+01:00 dus-ivr01 /usr/local/sbin/opensips[13076]: DBG:core:set_core_dump: core dump limits set to 18 446744073709551615 2018-02-27T17:19:29.940951+01:00 dus-ivr01 /usr/local/sbin/opensips[13076]: NOTICE:core:main: version: opensips 2.3.3 (x86_64/ linux) Do you need full logs, or is this enough? Thanks, Xale From: Bogdan-Andrei Iancu [mailto:bogdan at opensips.org] Sent: Monday, February 26, 2018 10:38 PM To: xaled Subject: Re: [OpenSIPS-Users] Wrong number of children Please apply the below patch, recompile and send again the startup logs: diff --git a/net/net_tcp.c b/net/net_tcp.c index 2dbea49..2d63940 100644 --- a/net/net_tcp.c +++ b/net/net_tcp.c @@ -1624,7 +1624,7 @@ int tcp_init(void) /* first we do auto-detection to see if there are any TCP based * protocols loaded */ for ( i=PROTO_FIRST ; i ; 'OpenSIPS users mailling list' Subject: Re: [OpenSIPS-Users] Wrong number of children Hmm.... that is strange. Could you send me (off list) your cfg to give it a try - I see you use the 2.3.3 version , right ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Summit 2018 http://www.opensips.org/events/Summit-2018Amsterdam On 02/26/2018 06:27 PM, xaled wrote: Nothing root at dus-appsrv01:/etc/opensips# opensipsctl fifo ps Process:: ID=0 PID=14966 Type=attendant Process:: ID=1 PID=14967 Type=MI FIFO Process:: ID=2 PID=14968 Type=time_keeper Process:: ID=3 PID=14969 Type=timer Process:: ID=4 PID=14970 Type=SIP receiver udp:1.2.3.4:5060 Process:: ID=5 PID=14975 Type=SIP receiver udp:1.2.3.4:5060 Process:: ID=6 PID=14976 Type=SIP receiver udp:1.2.3.4:5060 Process:: ID=7 PID=14977 Type=SIP receiver udp:1.2.3.4:5060 Process:: ID=8 PID=14978 Type=SIP receiver udp:1.2.3.4:5060 Process:: ID=9 PID=14979 Type=TCP receiver Process:: ID=10 PID=14982 Type=TCP receiver Process:: ID=11 PID=14984 Type=TCP receiver Process:: ID=12 PID=14985 Type=TCP receiver Process:: ID=13 PID=14986 Type=TCP receiver Process:: ID=14 PID=14987 Type=TCP receiver Process:: ID=15 PID=14988 Type=TCP receiver Process:: ID=16 PID=14989 Type=TCP receiver Process:: ID=17 PID=14991 Type=Timer handler Process:: ID=18 PID=14992 Type=TCP main root at srv01:/etc/opensips# netstat -tlnp | grep opensips root at srv01:/etc/opensips# Thanks, Xaled From: Bogdan-Andrei Iancu [mailto:bogdan at opensips.org] Sent: Monday, February 26, 2018 4:57 PM To: xaled ; 'OpenSIPS users mailling list' Subject: Re: [OpenSIPS-Users] Wrong number of children If you do "netstat -tlnp | grep opensips", what do you get ? The TCP layer automatically enable itself if a TCP-based protocol is used from the OpenSIPS config (as listener), like SIP TCP/TLS/WS/WSS , HEP or BIN. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Summit 2018 http://www.opensips.org/events/Summit-2018Amsterdam On 02/26/2018 05:53 PM, xaled wrote: Hi Bogdan, nope, non of them. Greetings, Xaled From: Bogdan-Andrei Iancu [mailto:bogdan at opensips.org] Sent: Monday, February 26, 2018 4:41 PM To: OpenSIPS users mailling list ; xaled Subject: Re: [OpenSIPS-Users] Wrong number of children Hi Xaled, Do you use HEP for tracing or BIN for replication ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Summit 2018 http://www.opensips.org/events/Summit-2018Amsterdam On 02/26/2018 01:36 PM, xaled wrote: Hi Callum, that helped thanks. I have 8 TCP receivers and one TCP main process that I did not expected as I only listen on udp. Will have to understand the TCP spawn logic. Greetings, Xaled _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From vladp at opensips.org Wed Feb 28 06:28:50 2018 From: vladp at opensips.org (Vlad Patrascu) Date: Wed, 28 Feb 2018 13:28:50 +0200 Subject: [OpenSIPS-Users] dialog replication In-Reply-To: <767753245.7406696.1519786006823@mail.yahoo.com> References: <311827884.3546251.1519627085908.ref@mail.yahoo.com> <311827884.3546251.1519627085908@mail.yahoo.com> <704df948-6873-5fde-fe25-193f1cf2a420@opensips.org> <2103041839.6602135.1519697741901@mail.yahoo.com> <48b7be0c-d5d9-b6f3-a81b-42d768116f07@opensips.org> <107101422.7400213.1519783290651@mail.yahoo.com> <767753245.7406696.1519786006823@mail.yahoo.com> Message-ID: <8811a93e-37c1-42ae-b197-7ccef7db42ff@opensips.org> Hi, Yes, you are correct, this is currently a limitation. But it could also work in an "active/active" sort of setup if you have 2 movable IPs and each dialog is tied to one of the IPs. Full anycast support though is coming up in OpenSIPS 2.4. Regards, Vlad Patrascu OpenSIPS Developer http://www.opensips-solutions.com On 28.02.2018 04:46, Pasan Meemaduma via Users wrote: > Hi Vlad, > > On a second thought its how anycast should work, packets could get to > the closest node so basically with 2.3.3 we can't replicate > transactions across mulitple opensips servers ? dialog replication  > would only work for an active/passive setup doesn't it ? > > > > On Wednesday, 28 February 2018, 7:36, Pasan Meemaduma via Users > wrote: > > > Hi Vlad, > > That explains the issue then, my sip listener ip is an anycast one, > and for some weired reason replies for INVITE that leaves node2 > receive by node1 hence it can't create the dialog. I checked with > node2 being down and recovered and calls init via node1 always > received final reply hence dialog replication works. So it means my > anycast configuration is broken right ? Thanks for you time to look in > to it. > > > On Tuesday, 27 February 2018, 21:42, Vlad Patrascu > wrote: > > > Hi Pasan, > I don't see anything in the logs or your cluster configuration which > could indicate that the dialogs don't replicate. Are you sure that the > problem is consistently reproducible and that the INVITE for the call > in question receives a final reply? Dialogs are replicated when they > get to the confirmed state, and judging from the logs, this doesn't > appear to be happening for that call. > Regards, > Vlad Patrascu > OpenSIPS Developer > http://www.opensips-solutions.com > On 27.02.2018 04:15, Pasan Meemaduma via Users wrote: >> Hi Vlad, >> >> I have sent you the full debug logs as requested. Also clusterer_list >> mi command gives following output when dialog replication stopped >> working. >> >> node1>>opensipsctl fifo clusterer_list >> Cluster:: 1 >>         Node:: 2 DB_ID=5 URL=bin:10.3.1.137:5566 Enabled=1 >> Link_state=Up Next_hop=2 Description=Node 2 >> >> >>  node2>> opensipsctl fifo clusterer_list >> Cluster:: 1 >>         Node:: 1 DB_ID=4 URL=bin:10.3.1.136:5566 Enabled=1 >> Link_state=Up Next_hop=1 Description=Node 1 >> 1 >> >> >> >> On Monday, 26 February 2018, 18:02, Vlad Patrascu >> wrote: >> >> >> Hi, >> Can you send the full logs for both nodes from the time that node1 >> restarts onwards? Also, what is the output of 'clusterer_list' mi >> command on the instances? >> Regards, >> Vlad Patrascu >> OpenSIPS Developer >> http://www.opensips-solutions.com >> On 26.02.2018 08:38, Pasan Meemaduma via Users wrote: >>> Hi Guys, >>> >>> Its me again :). I'm using dialog replication in opensips 2.3.3 and >>> it appears it doesn't work after a node goes down and come back >>> online. The recovered node >>> doesn't seems to receiving dialog info via binary interface. >>> >>> I have node1 and node2 with dialog replication on. everything works >>> fine and if I shutdown node1 and bring it back online after a while >>> node2 doesn't send new call dialog info via binary interface. >>> >>> I have attach the debug trace from both nodes, if you need anything >>> else let me know. I'm also using an anycast ip as the service ip. >>> >>> >>> on node2 for new call >>> >>> Feb 26 17:28:35 voip2-sip23b /usr/sbin/opensips[4703]: >>> DBG:dialog:build_new_dlg: new dialog 0x7f188afd3bf8 >>> (c=ktQ0Pkdwz50qGYjED6Brpw..,f=sip:XXXXXXXXX at somedomain;transport=UDP,t=sip:+YYYYYYYYYYY at somedomain;transport=UDP,ft=2b161508) >>> on hash 2317 >>> Feb 26 17:28:35 voip2-sip23b /usr/sbin/opensips[4703]: >>> DBG:dialog:init_leg_info: route_set , contact >>> sip:XXXXXXXXX at 192.168.27.11:56419;transport=UDP >>> , cseq 2 and >>> bind_addr udp:10.3.3.1:5060 >>> Feb 26 17:28:35 voip2-sip23b /usr/sbin/opensips[4703]: >>> DBG:dialog:dlg_add_leg_info: set leg 0 for 0x7f188afd3bf8: >>> tag=<2b161508> rcseq=<0> >>> Feb 26 17:28:35 voip2-sip23b /usr/sbin/opensips[4703]: >>> DBG:dialog:link_dlg: ref dlg 0x7f188afd3bf8 with 3 -> 3 in h_entry >>> 0x7f188afc3828 - 2317 >>> Feb 26 17:28:35 voip2-sip23b /usr/sbin/opensips[4703]: >>> DBG:dialog:new_dlg_val: inserting >>> =<45ad4c76-1abe-11e8-9410-831894b67d0c> >>> Feb 26 17:28:35 voip2-sip23b /usr/sbin/opensips[4703]: >>> DBG:dialog:dlg_onreq: t hash_index = 47425, t label = 2069082013 >>> Feb 26 17:28:35 voip2-sip23b /usr/sbin/opensips[4703]: >>> DBG:dialog:dlg_update_contact: Updated dialog 0x7f188afd3bf8 contact >>> to >>> >>> Feb 26 17:28:35 voip2-sip23b /usr/sbin/opensips[4703]: >>> DBG:dialog:unref_dlg: unref dlg 0x7f188afd3bf8 with 1 -> 2 in entry >>> 0x7f188afc3828 >>> Feb 26 17:28:52 voip2-sip23b /usr/sbin/opensips[4702]: >>> DBG:dialog:ref_dlg: ref dlg 0x7f188afd3bf8 with 1 -> 3 >>> Feb 26 17:28:52 voip2-sip23b /usr/sbin/opensips[4702]: >>> DBG:dialog:ref_dlg: ref dlg 0x7f188afd3bf8 with 1 -> 4 >>> Feb 26 17:28:52 voip2-sip23b /usr/sbin/opensips[4702]: >>> DBG:dialog:unref_dlg: unref dlg 0x7f188afd3bf8 with 1 -> 3 in entry >>> 0x7f188afc3828 >>> Feb 26 17:28:56 voip2-sip23b /usr/sbin/opensips[4711]: >>> DBG:dialog:unref_dlg: unref dlg 0x7f188afd3bf8 with 1 -> 2 in entry >>> 0x7f188afc3828 >>> Feb 26 17:29:05 voip2-sip23b /usr/sbin/opensips[4711]: >>> DBG:dialog:next_state_dlg: dialog 0x7f188afd3bf8 changed from state >>> 1 to state 5, due event 4 >>> Feb 26 17:29:05 voip2-sip23b /usr/sbin/opensips[4711]: >>> DBG:dialog:dlg_onreply: dialog 0x7f188afd3bf8 failed (negative reply) >>> Feb 26 17:29:05 voip2-sip23b /usr/sbin/opensips[4711]: >>> DBG:dialog:unref_dlg: unref dlg 0x7f188afd3bf8 with 1 -> 1 in entry >>> 0x7f188afc3828 >>> Feb 26 17:29:11 voip2-sip23b /usr/sbin/opensips[4711]: >>> DBG:dialog:next_state_dlg: dialog 0x7f188afd3bf8 changed from state >>> 5 to state 5, due event 1 >>> Feb 26 17:29:11 voip2-sip23b /usr/sbin/opensips[4711]: >>> DBG:dialog:unref_dlg: unref dlg 0x7f188afd3bf8 with 1 -> 0 in entry >>> 0x7f188afc3828 >>> Feb 26 17:29:11 voip2-sip23b /usr/sbin/opensips[4711]: >>> DBG:dialog:unref_dlg: ref <=0 for dialog 0x7f188afd3bf8 >>> Feb 26 17:29:11 voip2-sip23b /usr/sbin/opensips[4711]: >>> DBG:dialog:destroy_dlg: destroying dialog 0x7f188afd3bf8 >>> Feb 26 17:29:11 voip2-sip23b /usr/sbin/opensips[4711]: >>> DBG:dialog:destroy_dlg: dlg expired or not in list - dlg >>> 0x7f188afd3bf8 [2317:677053409] with clid 'ktQ0Pkdwz50qGYjED6Brpw..' >>> and tags '2b161508' 'NULL' >>> >>> >>> on node1 >>> >>> Feb 26 17:28:41 voip2-sip23a /usr/sbin/opensips[15820]: >>> DBG:dialog:dlg_onroute: route param is 'd09.1e30b582' (len=12) >>> Feb 26 17:28:41 voip2-sip23a /usr/sbin/opensips[15820]: >>> DBG:dialog:lookup_dlg: no dialog id=677053409 found on entry 2317 >>> Feb 26 17:28:41 voip2-sip23a /usr/sbin/opensips[15820]: >>> DBG:dialog:dlg_onroute: unable to find dialog for BYE with route >>> param 'd09.1e30b582' >>> Feb 26 17:28:41 voip2-sip23a /usr/sbin/opensips[15822]: >>> DBG:dialog:dlg_onroute: route param is 'd09.1e30b582' (len=12) >>> Feb 26 17:28:41 voip2-sip23a /usr/sbin/opensips[15822]: >>> DBG:dialog:lookup_dlg: no dialog id=677053409 found on entry 2317 >>> Feb 26 17:28:41 voip2-sip23a /usr/sbin/opensips[15822]: >>> DBG:dialog:dlg_onroute: unable to find dialog for BYE with route >>> param 'd09.1e30b582' >>> Feb 26 17:28:41 voip2-sip23a /usr/sbin/opensips[15822]: >>> DBG:dialog:dlg_onroute: route param is 'd09.1e30b582' (len=12) >>> Feb 26 17:28:41 voip2-sip23a /usr/sbin/opensips[15822]: >>> DBG:dialog:lookup_dlg: no dialog id=677053409 found on entry 2317 >>> Feb 26 17:28:41 voip2-sip23a /usr/sbin/opensips[15822]: >>> DBG:dialog:dlg_onroute: unable to find dialog for BYE with route >>> param 'd09.1e30b582' >>> Feb 26 17:28:42 voip2-sip23a /usr/sbin/opensips[15820]: >>> DBG:dialog:dlg_onroute: route param is 'd09.1e30b582' (len=12) >>> Feb 26 17:28:42 voip2-sip23a /usr/sbin/opensips[15820]: >>> DBG:dialog:lookup_dlg: no dialog id=677053409 found on entry 2317 >>> Feb 26 17:28:42 voip2-sip23a /usr/sbin/opensips[15820]: >>> DBG:dialog:dlg_onroute: unable to find dialog for BYE with route >>> param 'd09.1e30b582' >>> Feb 26 17:28:42 voip2-sip23a /usr/sbin/opensips[15820]: >>> DBG:dialog:dlg_onroute: route param is 'd09.1e30b582' (len=12) >>> Feb 26 17:28:42 voip2-sip23a /usr/sbin/opensips[15820]: >>> DBG:dialog:lookup_dlg: no dialog id=677053409 found on entry 2317 >>> >>> >>> If I purposely take down node1 again and try to make a call I can >>> see that it indicates the failure to replicate dialog info on the >>> cluster as below, >>> >>> Feb 26 17:27:01 voip2-sip23b /usr/sbin/opensips[4701]: >>> ERROR:dialog:replicate_dialog_updated: All destinations in cluster: >>> 1 are down or probing >>> Feb 26 17:27:01 voip2-sip23b /usr/sbin/opensips[4701]: >>> ERROR:dialog:replicate_dialog_updated: Failed to replicate updated >>> dialog >>> >>> >>> >>> >>> _______________________________________________ >>> Users mailing list >>> Users at lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> >> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Wed Feb 28 06:30:19 2018 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Wed, 28 Feb 2018 13:30:19 +0200 Subject: [OpenSIPS-Users] Wrong number of children In-Reply-To: <016101d3b07f$f1ae3f20$d50abd60$@web.de> References: <027501d3aca1$66551340$32ff39c0$@web.de> <010001d3aef6$02f70f90$08e52eb0$@web.de> <4e50d0e9-d0c8-5317-4f21-433cbc5cde13@opensips.org> <01ff01d3af19$fda10db0$f8e32910$@web.de> <15812914-24ca-8dfa-9371-732ce42c91bd@opensips.org> <026201d3af1e$a7be42a0$f73ac7e0$@web.de> <02dd01d3af2d$b3380b20$19a82160$@web.de> <00a601d3afe8$3c0a0740$b41e15c0$@web.de> <3f4b7059-74f6-e2ca-0697-123c3ccac637@opensips.org> <016101d3b07f$f1ae3f20$d50abd60$@web.de> Message-ID: <95ab58eb-49d3-3d62-ce7d-01b51b61af1e@opensips.org> Super ! I like this "unwanted children" :P Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Summit 2018 http://www.opensips.org/events/Summit-2018Amsterdam On 02/28/2018 12:36 PM, xaled wrote: > > Hi Bodgan, > > thanks, now there are no unwanted TCP children ;) > > Number of SIP receiver UDP children matches the children param value. > > ivr01:/usr/local/etc/opensips # /usr/local/sbin/opensipsctl fifo ps > > Process:: ID=0 PID=12872 Type=attendant > > Process:: ID=1 PID=12873 Type=MI FIFO > > Process:: ID=2 PID=12874 Type=time_keeper > > Process:: ID=3 PID=12875 Type=timer > > Process:: ID=4 PID=12876 Type=SIP receiver udp: x.x.x.x:5060 > > Process:: ID=5 PID=12878 Type=SIP receiver udp: x.x.x.x:5060 > > Process:: ID=6 PID=12879 Type=SIP receiver udp: x.x.x.x:5060 > > Process:: ID=7 PID=12880 Type=SIP receiver udp: x.x.x.x:5060 > > Process:: ID=8 PID=12881 Type=Timer handler > > Xaled > > *From:*Bogdan-Andrei Iancu [mailto:bogdan at opensips.org] > *Sent:* Tuesday, February 27, 2018 6:39 PM > *To:* OpenSIPS users mailling list ; xaled > > *Subject:* Re: [OpenSIPS-Users] Wrong number of children > > Ok, thanks to Razvan, we have an fix for this: > https://github.com/OpenSIPS/opensips/commit/b022ad5a7c1c24aa9cf3ddf22d5f50510807c9f4 > > Please update from GIT and give it a try. > > Many Thanks, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > OpenSIPS Summit 2018 > http://www.opensips.org/events/Summit-2018Amsterdam > > On 02/27/2018 06:38 PM, Bogdan-Andrei Iancu wrote: > > Thank you Xaled, > > Based on the output you provided I managed to identify the issue. > There is a tricky internal test that actually forces the TCP layer > to start more or less all the time. > Let us have this fixed !! I will update you > > Best regards, > > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > > http://www.opensips-solutions.com > > OpenSIPS Summit 2018 > > http://www.opensips.org/events/Summit-2018Amsterdam > > On 02/27/2018 06:30 PM, xaled wrote: > > Hi Bogdan, > > It’s a different server, but the same tcp issue: > > 2018-02-27T17:19:29.939207+01:00 dus-ivr01 opensips: > DBG:core:main: Loaded 1 transport protocols > > 2018-02-27T17:19:29.939411+01:00 dus-ivr01 opensips: > DBG:core:socket2str: > > 2018-02-27T17:19:29.939630+01:00 dus-ivr01 opensips: > INFO:core:tcp_init: enabling TCP for proto 5 > > 2018-02-27T17:19:29.939842+01:00 dus-ivr01 opensips: > DBG:core:ipc_register_handler: IPC type 0 [TCP reporting] > registered with > > handler 0x5f9fee > > 2018-02-27T17:19:29.940120+01:00 dus-ivr01 opensips: > DBG:core:create_status_pipe: pipe created ? rc = 0, errno = > Success > > 2018-02-27T17:19:29.940326+01:00 dus-ivr01 opensips: > DBG:core:daemonize: waiting for status code from children > > 2018-02-27T17:19:29.940526+01:00 dus-ivr01 opensips: > DBG:core:count_init_children: 15 children are going to be inited > > 2018-02-27T17:19:29.940724+01:00 dus-ivr01 > /usr/local/sbin/opensips[13076]: DBG:core:set_core_dump: core > dump limits set to 18 > > 446744073709551615 > > 2018-02-27T17:19:29.940951+01:00 dus-ivr01 > /usr/local/sbin/opensips[13076]: NOTICE:core:main: version: > opensips 2.3.3 (x86_64/ > > linux) > > Do you need full logs, or is this enough? > > Thanks, > > Xale > > *From:*Bogdan-Andrei Iancu [mailto:bogdan at opensips.org] > *Sent:* Monday, February 26, 2018 10:38 PM > *To:* xaled > *Subject:* Re: [OpenSIPS-Users] Wrong number of children > > Please apply the below patch, recompile and send again the > startup logs: > > diff --git a/net/net_tcp.c b/net/net_tcp.c > index 2dbea49..2d63940 100644 > --- a/net/net_tcp.c > +++ b/net/net_tcp.c > @@ -1624,7 +1624,7 @@ int tcp_init(void) > /* first we do auto-detection to see if there are any > TCP based > * protocols loaded */ > for ( i=PROTO_FIRST ; i - if (is_tcp_based_proto(i)) {tcp_disabled=0;break;} > + if (is_tcp_based_proto(i)) {LM_INFO("enabling > TCP for proto %d\n",i);tcp_disabled=0;break;} > > if (tcp_disabled) > return 0; > > > Regards, > > > Bogdan-Andrei Iancu > > > > OpenSIPS Founder and Developer > > http://www.opensips-solutions.com > > OpenSIPS Summit 2018 > > http://www.opensips.org/events/Summit-2018Amsterdam > > On 02/26/2018 08:14 PM, xaled wrote: > > Hi Bogdan, > > would a full debug log will be sufficient? > > The config is spread across quite a few files. > > Thanks, > > Xaled > > *From:*Bogdan-Andrei Iancu [mailto:bogdan at opensips.org] > *Sent:* Monday, February 26, 2018 5:25 PM > *To:* xaled ; > 'OpenSIPS users mailling list' > > *Subject:* Re: [OpenSIPS-Users] Wrong number of children > > Hmm.... that is strange. > > Could you send me (off list) your cfg to give it a try - I > see you use the 2.3.3 version , right ? > > Regards, > > > > Bogdan-Andrei Iancu > > > > OpenSIPS Founder and Developer > > http://www.opensips-solutions.com > > OpenSIPS Summit 2018 > > http://www.opensips.org/events/Summit-2018Amsterdam > > On 02/26/2018 06:27 PM, xaled wrote: > > Nothing > > root at dus-appsrv01:/etc/opensips# opensipsctl fifo ps > > Process:: ID=0 PID=14966 Type=attendant > > Process:: ID=1 PID=14967 Type=MI FIFO > > Process:: ID=2 PID=14968 Type=time_keeper > > Process:: ID=3 PID=14969 Type=timer > > Process:: ID=4 PID=14970 Type=SIP receiver > udp:1.2.3.4:5060 > > Process:: ID=5 PID=14975 Type=SIP receiver > udp:1.2.3.4:5060 > > Process:: ID=6 PID=14976 Type=SIP receiver > udp:1.2.3.4:5060 > > Process:: ID=7 PID=14977 Type=SIP receiver > udp:1.2.3.4:5060 > > Process:: ID=8 PID=14978 Type=SIP receiver > udp:1.2.3.4:5060 > > Process:: ID=9 PID=14979 Type=TCP receiver > > Process:: ID=10 PID=14982 Type=TCP receiver > > Process:: ID=11 PID=14984 Type=TCP receiver > > Process:: ID=12 PID=14985 Type=TCP receiver > > Process:: ID=13 PID=14986 Type=TCP receiver > > Process:: ID=14 PID=14987 Type=TCP receiver > > Process:: ID=15 PID=14988 Type=TCP receiver > > Process:: ID=16 PID=14989 Type=TCP receiver > > Process:: ID=17 PID=14991 Type=Timer handler > > Process:: ID=18 PID=14992 Type=TCP main > > root at srv01:/etc/opensips# netstat -tlnp | grep opensips > > root at srv01:/etc/opensips# > > Thanks, > > Xaled > > *From:*Bogdan-Andrei Iancu [mailto:bogdan at opensips.org] > *Sent:* Monday, February 26, 2018 4:57 PM > *To:* xaled ; > 'OpenSIPS users mailling list' > > > *Subject:* Re: [OpenSIPS-Users] Wrong number of children > > If you do "netstat -tlnp | grep opensips", what do you > get ? > > The TCP layer automatically enable itself if a > TCP-based protocol is used from the OpenSIPS config > (as listener), like SIP TCP/TLS/WS/WSS , HEP or BIN. > > Regards, > > > > > Bogdan-Andrei Iancu > > > > OpenSIPS Founder and Developer > > http://www.opensips-solutions.com > > OpenSIPS Summit 2018 > > http://www.opensips.org/events/Summit-2018Amsterdam > > On 02/26/2018 05:53 PM, xaled wrote: > > Hi Bogdan, > > nope, non of them. > > Greetings, > > Xaled > > *From:*Bogdan-Andrei Iancu > [mailto:bogdan at opensips.org] > *Sent:* Monday, February 26, 2018 4:41 PM > *To:* OpenSIPS users mailling list > > ; xaled > > *Subject:* Re: [OpenSIPS-Users] Wrong number of > children > > Hi Xaled, > > Do you use HEP for tracing or BIN for replication ? > > Regards, > > > > > > Bogdan-Andrei Iancu > > > > OpenSIPS Founder and Developer > > http://www.opensips-solutions.com > > OpenSIPS Summit 2018 > > http://www.opensips.org/events/Summit-2018Amsterdam > > On 02/26/2018 01:36 PM, xaled wrote: > > Hi Callum, > > that helped thanks. > > I have 8 TCP receivers and one TCP main > process that I did not expected as I only > listen on udp. > > Will have to understand the TCP spawn logic. > > Greetings, > > Xaled > > > > > > _______________________________________________ > > Users mailing list > > Users at lists.opensips.org > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Wed Feb 28 07:27:15 2018 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Wed, 28 Feb 2018 14:27:15 +0200 Subject: [OpenSIPS-Users] textops module replace functions don't work In-Reply-To: References: Message-ID: <5ef04a27-6cfd-0f70-e341-b691bb782c42@opensips.org> Hi Yuval, First, please subscribe to the list in order to be able to properly post messages here. Secondly, on your issue - before the replace function do you perform any other changes over the body ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Summit 2018 http://www.opensips.org/events/Summit-2018Amsterdam On 02/26/2018 06:45 PM, Dinari, Yuval wrote: > Hi, > I tried all of them, not just those two but also replace_all(), > replace_body_all(), replace_body_atonce(), subst(), and subst_body(). > Thanks > > On Mon, Feb 26, 2018 at 5:54 PM, Bogdan-Andrei Iancu > > wrote: > > Hi, > > Have you used replace() or replace_body() ? > > Regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > OpenSIPS Summit 2018 > http://www.opensips.org/events/Summit-2018Amsterdam > > > On 02/26/2018 12:31 PM, Dinari, Yuval via Users wrote: >> Hi, >> I tried using textops module functions to replace a part of the >> message body, but the body remains unchanged. A search() with the >> regex to be replaced succeeds. >> I tried all replacement functions (replace*, subst, subst_body). >> I see that the body is unchanged by printing $rb, and also >> ngreping the SDP that opensips sends out to rtpengine via >> rtpengine_offer(). >> opensips version - 2.2.5 >> Thanks >> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From Ben.Newlin at genesys.com Wed Feb 28 17:13:06 2018 From: Ben.Newlin at genesys.com (Ben Newlin) Date: Wed, 28 Feb 2018 22:13:06 +0000 Subject: [OpenSIPS-Users] OpenSIPS crashing on 200 OK In-Reply-To: <138a08a6-8021-1d6d-3d7d-444fe136a61a@opensips.org> References: <530E1611-2A4D-424F-938D-2E409E1A78A2@genesys.com> <7cf6bb3d-967a-b8e2-d4b3-8b42c8dbf0b8@opensips.org> <755F831E-90E4-4105-80C5-42C7A2E2050A@genesys.com> <6D9EBB18-914E-49ED-988F-7D582007B88E@genesys.com> <085c6c6b-75ca-0557-aa5e-0fcb9e3eb07f@opensips.org> <4FD028FE-6607-496C-8B3A-EF4293FB6155@genesys.com> <8eeebbf4-9c57-dd6a-ebf1-bbe6dadb0bdb@opensips.org> <279057EB-BEC2-4552-B0C9-EDD244263E32@genesys.com> <478773D5-3955-4A6A-BA6F-15467E39CC29@genesys.com> <99C36C64-1230-4967-8DD9-2CBF51768742@genesys.com> <138a08a6-8021-1d6d-3d7d-444fe136a61a@opensips.org> Message-ID: Bogdan, You’re more than warm! Thanks for the quick turnaround on these fixes! I’ve tested and verified each of them. I’ve not run into any more issues so far. Ben Newlin From: Bogdan-Andrei Iancu Date: Wednesday, February 28, 2018 at 6:27 AM To: Ben Newlin , OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] OpenSIPS crashing on 200 OK Hi Ben, This new crash was also fixed - see https://github.com/OpenSIPS/opensips/commit/34f81b223fb466ecddf648f72f615ab24d4a86dc . Do you have any more crashes on the pipe to report :D ? I'm getting warm :) Thanks and regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Summit 2018 http://www.opensips.org/events/Summit-2018Amsterdam On 02/28/2018 06:19 AM, Ben Newlin wrote: I was able to reproduce and capture a core dump of the crash. Backtrace is available here: https://pastebin.com/WtBB6cKm Thanks, Ben Newlin From: Ben Newlin Date: Tuesday, February 27, 2018 at 10:47 PM To: Bogdan-Andrei Iancu , OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] OpenSIPS crashing on 200 OK Unfortunately, I have had another crash during testing. I was not able to capture the core, but I did get this log: Feb 28 03:35:22 [359] CRITICAL:core:qm_debug_frag: qm_*: prev. fragm. tail overwritten(c0c0c0c0c0c0c020, abcdefedabcdefed)[0x7fa6cd6c46b8:0x7fa6cd6c46f0] (del_uri_param, checks.c:242)! Feb 28 03:35:23 [353] INFO:core:handle_sigs: child process 359 exited by a signal 6 I am making adjustments to be able to collect the core if the crash occurs again. Thanks, Ben Newlin From: Ben Newlin Date: Tuesday, February 27, 2018 at 2:47 PM To: Bogdan-Andrei Iancu , OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] OpenSIPS crashing on 200 OK Bogdan, I found this previous thread that has this error: http://lists.opensips.org/pipermail/users/2016-October/035635.html The fix suggested there worked. When I removed the quotes around the regexp used in re.subst the crash did not occur. Ben Newlin From: Ben Newlin Date: Tuesday, February 27, 2018 at 2:20 PM To: Bogdan-Andrei Iancu , OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] OpenSIPS crashing on 200 OK Bogdan, I just tested it and that crash is no longer occurring. However I am now seeing a different crash, but it seems to still be related to regexp. Feb 27 19:07:17 [369] ERROR:core:subst_parser: string too short Feb 27 19:07:17 [369] ERROR:core:tr_eval_re: Can't compile regexp Feb 27 19:07:17 [369] WARNING:core:do_assign: no value in right expression at /opensips/control.cfg:1952 Feb 27 19:07:17 [361] INFO:core:handle_sigs: child process 369 exited by a signal 11 Feb 27 19:07:17 [361] INFO:core:handle_sigs: core was generated Feb 27 19:07:17 [361] INFO:core:handle_sigs: terminating due to SIGCHLD The do_assign error makes it seem like there may be something wrong in my script, but I can’t see what it is. And I’m not sure what the “string too short” error is either or why the regexp could not be compiled. It looks valid to me. Backtrace: https://pastebin.com/Fk2BRnsg Ben Newlin From: Bogdan-Andrei Iancu Date: Tuesday, February 27, 2018 at 12:55 PM To: OpenSIPS users mailling list , Ben Newlin Subject: Re: [OpenSIPS-Users] OpenSIPS crashing on 200 OK Hi Ben, I just committed the fix for this - could you update from GIT and give it a new try ? Thanks and regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Summit 2018 http://www.opensips.org/events/Summit-2018Amsterdam On 02/27/2018 06:19 PM, Bogdan-Andrei Iancu wrote: Thanks Ben for the support in investigating this. The bug was identified and confirm, let me work on a fix right now. I will shortly ping you for testing ;) Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Summit 2018 http://www.opensips.org/events/Summit-2018Amsterdam On 02/27/2018 12:12 AM, Bogdan-Andrei Iancu wrote: OK, to be more precise, the crash is around an "If" statement where you have in the condition an REGEXP test (=~) with a left operand translated into "downstream" (maybe a $DLG_dir ??). Does it sound familiar to you :) ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com OpenSIPS Summit 2018 http://www.opensips.org/events/Summit-2018Amsterdam On 02/26/2018 11:32 PM, Ben Newlin wrote: Yes, that is from that exact file, line 435. It’s just a return statement. I can send you the file off-list if you’d like. I do use the $DLG_dir in my script in a few places, but the only one this crash would have hit is in some logging. Ben Newlin -------------- next part -------------- An HTML attachment was scrubbed... URL: From pat at voxtelesys.com Wed Feb 28 22:01:18 2018 From: pat at voxtelesys.com (Pat Burke) Date: Wed, 28 Feb 2018 21:01:18 -0600 Subject: [OpenSIPS-Users] RTP Proxy Port not updating on T38 reinvite Message-ID: We have a situation where a fax comes in, all of the negotiation looks good (ports are translated like they are supposed to be), and the interface with RTP Proxy looks correct (at least to me).  But what happens is that after the Re-INVITE for T38, the RTP Proxy continues to sent the RTP to the customer's old port.  In the example below, RTP Proxy sends the RTP to the customer on port 7186 AFTER the Re-INVITE with the new port. SIP trace is located at https://www.4shared.com/file/wIVrkOtnei/SIP.html >From the SIP responses, it appears that the code is performing correctly.  Not certain what to try next.   Also, the RTP Proxy server was rebooted. Any suggestions? Thanks in advance, Pat Configuration:     OpenSIPS 2.2.3 on Ubuntu 16.04         RTP Proxy 2.2.alpha.20160822 on Ubuntu 16.04 *************************************************** *** OpenSIPS re-Invite code. if (has_totag()) { if (is_method("BYE")) { if ($avp(call_end_seconds) == NULL) { $avp(bye_src_ip) := $si; $avp(bye_time) := $Ts; get_timestamp($avp(call_end_seconds),$avp(call_end_useconds)); $json(end_recording_details) := "{}"; $json(end_recording_details/endsec) = $avp(call_end_seconds); $json(end_recording_details/endusec) = $avp(call_end_useconds); xlog("L_INFO", "$dlg_val(rU) SCRIPT:RTPPROXY:DEBUG: End call recording callid=$ci ; ft= $ft data = $json(end_recording_details) \n"); #async(avp_db_query("UPDATE recordings SET end_recording_details = '$json(end_recording_details)' WHERE call_id = '$ci'", ""), resume_totag); avp_db_query("UPDATE recordings SET end_recording_details = '$json(end_recording_details)' WHERE call_id = '$ci'", ""); } else { xlog("L_INFO", "$dlg_val(rU) SCRIPT:BILLING:DEBUG: Extra BYE received.\n"); } } if (loose_route() || match_dialog()) { if ( $DLG_status==NULL ) { if (is_method("ACK") && t_check_trans() ) { t_relay(); exit; } xlog("L_WARN", "$rU SCRIPT:TRAFFIC:WARNING: - $rm is not in-dialog ( callid=$ci ; ruri=$ru ; route=$(hdr(Route)[*]) )\n"); exit; } else { if (is_method("INVITE")) { record_route(); if (stream_exists("image") && $dlg_val(t38) == "0") { xlog("L_WARN", "$rU SCRIPT:FAX:DEBUG: Received RE-INVITE containing FAX, but carrier doesn't support it - rejecting $ci \n"); send_reply("488","Not Acceptable"); exit; } xlog("L_INFO", "$dlg_val(rU) SCRIPT:RTPPROXY:DEBUG: - $dlg_val(proxy_media) - $dlg_val(rtpproxy_group) - $dlg_val(customer_record)\n"); # Setup proxy media if ($dlg_val(proxy_media) != NULL && $dlg_val(proxy_media) == "1") { if (nat_uac_test("8")) { rtpproxy_offer("co",,"$dlg_val(rtpproxy_group)", "$avp(rtpprpoxy_server)"); } else { rtpproxy_offer("cor",,"$dlg_val(rtpproxy_group)", "$avp(rtpprpoxy_server)"); } $avp(rtpprpoxy_server) := $(avp(rtpprpoxy_server){s.select,1,:}); if ($dlg_val(customer_record) == "1") { rtpproxy_start_recording("$dlg_val(rtpproxy_group)"); } xlog("L_INFO", "$dlg_val(rU) PHB - D - media = $(rb{sdp.line,c}) rtp = $avp(rtpprpoxy_server) \n"); } t_on_reply("media"); } t_relay(); exit; } } else { if ( is_method("ACK") ) { if ( t_check_trans() ) { t_relay(); exit; } else exit; } sl_send_reply("404","Not here"); exit; } *************************************************** *** RTP Ports in the call flow               Carrier                               OpenSIPS                              Customer Audio      61616 <<<<<<<<<<<<<<<<< 18986 <<<<<<<<<<                                     >>>>>>>>>>>>>> 16930 >>>>>>>>>>>>>> 7186    (intial INVITE / 200 OK)                                           Image                       >>>>>>>>>>>>>> 16930 >>>>>>>>>>>>>> 7192    (Re-INVITE / 200 OK - T38)                   61616 <<<<<<<<<<<<<<<<< 18986 <<<<<<<<<< *************************************************** *** RTP Proxy Command TRACE # U 216.147.191.162:41657 -> 216.147.191.232:9992 1422_78032 Uc0,18,101 1917630487_62491918 at 208.93.41.184 208.93.41.140 61616 gK0c00bce5;1 # U 216.147.191.232:9992 -> 216.147.191.162:41657 1422_78032 18986 216.147.191.232 # U 216.147.191.162:41657 -> 216.147.191.232:9992 1422_78033 Lc0,101 1917630487_62491918 at 208.93.41.184 216.147.191.171 7186 gK0c00bce5;1 5a142640;1 # U 216.147.191.232:9992 -> 216.147.191.162:41657 1422_78033 16930 216.147.191.232 # U 216.147.191.162:41657 -> 216.147.191.232:9992 1422_78034 U 1917630487_62491918 at 208.93.41.184 216.147.191.171 7192 5a142640;1 gK0c00bce5;1 # U 216.147.191.232:9992 -> 216.147.191.162:41657 1422_78034 16930 216.147.191.232 # U 216.147.191.162:41657 -> 216.147.191.232:9992 1422_78035 L 1917630487_62491918 at 208.93.41.184 208.93.41.140 61616 5a142640;1 gK0c00bce5;1 -------------- next part -------------- An HTML attachment was scrubbed... URL: From pasandev at ymail.com Wed Feb 28 20:21:01 2018 From: pasandev at ymail.com (Pasan Meemaduma) Date: Thu, 1 Mar 2018 01:21:01 +0000 (UTC) Subject: [OpenSIPS-Users] dialog replication In-Reply-To: <8811a93e-37c1-42ae-b197-7ccef7db42ff@opensips.org> References: <311827884.3546251.1519627085908.ref@mail.yahoo.com> <311827884.3546251.1519627085908@mail.yahoo.com> <704df948-6873-5fde-fe25-193f1cf2a420@opensips.org> <2103041839.6602135.1519697741901@mail.yahoo.com> <48b7be0c-d5d9-b6f3-a81b-42d768116f07@opensips.org> <107101422.7400213.1519783290651@mail.yahoo.com> <767753245.7406696.1519786006823@mail.yahoo.com> <8811a93e-37c1-42ae-b197-7ccef7db42ff@opensips.org> Message-ID: <1737101176.8117773.1519867261545@mail.yahoo.com> Hi Vlad, Thanks for your response. I think I'll have to wait until 2.4 transaction replication come in to action if I am to go with anycast ip. I didn't understand how it would solve with 2 moveable ips. how do i make sure dialog will tie to a particular ip ? On Wednesday, 28 February 2018, 17:04, Vlad Patrascu wrote: Hi, Yes, you are correct, this is currently a limitation. But it could also work in an "active/active" sort of setup if you have 2 movable IPs and each dialog is tied to one of the IPs. Full anycast support though is coming up in OpenSIPS 2.4. Regards, Vlad Patrascu OpenSIPS Developer http://www.opensips-solutions.com On 28.02.2018 04:46, Pasan Meemaduma via Users wrote: Hi Vlad, On a second thought its how anycast should work, packets could get to the closest node so basically with 2.3.3 we can't replicate transactions across mulitple opensips servers ? dialog replication  would only work for an active/passive setup doesn't it ? On Wednesday, 28 February 2018, 7:36, Pasan Meemaduma via Users wrote: Hi Vlad, That explains the issue then, my sip listener ip is an anycast one, and for some weired reason replies for INVITE that leaves node2 receive by node1 hence it can't create the dialog. I checked with node2 being down and recovered and calls init via node1 always received final reply hence dialog replication works. So it means my anycast configuration is broken right ? Thanks for you time to look in to it. On Tuesday, 27 February 2018, 21:42, Vlad Patrascu wrote: Hi Pasan, I don't see anything in the logs or your cluster configuration which could indicate that the dialogs don't replicate. Are you sure that the problem is consistently reproducible and that the INVITE for the call in question receives a final reply? Dialogs are replicated when they get to the confirmed state, and judging from the logs, this doesn't appear to be happening for that call. Regards, Vlad Patrascu OpenSIPS Developer http://www.opensips-solutions.com On 27.02.2018 04:15, Pasan Meemaduma via Users wrote: Hi Vlad, I have sent you the full debug logs as requested. Also clusterer_list mi command gives following output when dialog replication stopped working. node1>>opensipsctl fifo clusterer_list Cluster:: 1         Node:: 2 DB_ID=5 URL=bin:10.3.1.137:5566 Enabled=1 Link_state=Up      Next_hop=2 Description=Node 2  node2>> opensipsctl fifo clusterer_list Cluster:: 1         Node:: 1 DB_ID=4 URL=bin:10.3.1.136:5566 Enabled=1 Link_state=Up      Next_hop=1 Description=Node 1 1 On Monday, 26 February 2018, 18:02, Vlad Patrascu wrote: Hi, Can you send the full logs for both nodes from the time that node1 restarts onwards? Also, what is the output of 'clusterer_list' mi command on the instances? Regards, Vlad Patrascu OpenSIPS Developer http://www.opensips-solutions.com On 26.02.2018 08:38, Pasan Meemaduma via Users wrote: Hi Guys, Its me again :). I'm using dialog replication in opensips 2.3.3 and it appears it doesn't work after a node goes down and come back online. The recovered node doesn't seems to receiving dialog info via binary interface. I have node1 and node2 with dialog replication on. everything works fine and if I shutdown node1 and bring it back online after a while node2 doesn't send new call dialog info via binary interface. I have attach the debug trace from both nodes, if you need anything else let me know. I'm also using an anycast ip as the service ip. on node2 for new call Feb 26 17:28:35 voip2-sip23b /usr/sbin/opensips[4703]:DBG:dialog:build_new_dlg: new dialog 0x7f188afd3bf8(c=ktQ0Pkdwz50qGYjED6Brpw..,f=sip:XXXXXXXXX at somedomain;transport=UDP,t=sip:+YYYYYYYYYYY at somedomain;transport=UDP,ft=2b161508) on hash 2317 Feb 26 17:28:35 voip2-sip23b /usr/sbin/opensips[4703]:DBG:dialog:init_leg_info: route_set , contact sip:XXXXXXXXX at 192.168.27.11:56419;transport=UDP, cseq 2 and bind_addr udp:10.3.3.1:5060 Feb 26 17:28:35 voip2-sip23b /usr/sbin/opensips[4703]:DBG:dialog:dlg_add_leg_info: set leg 0 for 0x7f188afd3bf8: tag=<2b161508>rcseq=<0> Feb 26 17:28:35 voip2-sip23b /usr/sbin/opensips[4703]:DBG:dialog:link_dlg: ref dlg 0x7f188afd3bf8 with 3 -> 3 in h_entry 0x7f188afc3828 - 2317 Feb 26 17:28:35 voip2-sip23b /usr/sbin/opensips[4703]:DBG:dialog:new_dlg_val: inserting =<45ad4c76-1abe-11e8-9410-831894b67d0c> Feb 26 17:28:35 voip2-sip23b /usr/sbin/opensips[4703]:DBG:dialog:dlg_onreq: t hash_index = 47425, t label = 2069082013 Feb 26 17:28:35 voip2-sip23b /usr/sbin/opensips[4703]:DBG:dialog:dlg_update_contact: Updated dialog 0x7f188afd3bf8 contact to Feb 26 17:28:35 voip2-sip23b /usr/sbin/opensips[4703]:DBG:dialog:unref_dlg: unref dlg 0x7f188afd3bf8 with 1 -> 2 in entry 0x7f188afc3828 Feb 26 17:28:52 voip2-sip23b /usr/sbin/opensips[4702]:DBG:dialog:ref_dlg: ref dlg 0x7f188afd3bf8 with 1 -> 3 Feb 26 17:28:52 voip2-sip23b /usr/sbin/opensips[4702]:DBG:dialog:ref_dlg: ref dlg 0x7f188afd3bf8 with 1 -> 4 Feb 26 17:28:52 voip2-sip23b /usr/sbin/opensips[4702]:DBG:dialog:unref_dlg: unref dlg 0x7f188afd3bf8 with 1 -> 3 in entry 0x7f188afc3828 Feb 26 17:28:56 voip2-sip23b /usr/sbin/opensips[4711]:DBG:dialog:unref_dlg: unref dlg 0x7f188afd3bf8 with 1 -> 2 in entry 0x7f188afc3828 Feb 26 17:29:05 voip2-sip23b /usr/sbin/opensips[4711]:DBG:dialog:next_state_dlg: dialog 0x7f188afd3bf8 changed from state 1 to state 5, due event 4 Feb 26 17:29:05 voip2-sip23b /usr/sbin/opensips[4711]:DBG:dialog:dlg_onreply: dialog 0x7f188afd3bf8 failed (negative reply) Feb 26 17:29:05 voip2-sip23b /usr/sbin/opensips[4711]:DBG:dialog:unref_dlg: unref dlg 0x7f188afd3bf8 with 1 -> 1 in entry 0x7f188afc3828 Feb 26 17:29:11 voip2-sip23b /usr/sbin/opensips[4711]:DBG:dialog:next_state_dlg: dialog 0x7f188afd3bf8 changed from state 5 to state 5, due event 1 Feb 26 17:29:11 voip2-sip23b /usr/sbin/opensips[4711]:DBG:dialog:unref_dlg: unref dlg 0x7f188afd3bf8 with 1 -> 0 in entry 0x7f188afc3828 Feb 26 17:29:11 voip2-sip23b /usr/sbin/opensips[4711]:DBG:dialog:unref_dlg: ref <=0 for dialog 0x7f188afd3bf8 Feb 26 17:29:11 voip2-sip23b /usr/sbin/opensips[4711]:DBG:dialog:destroy_dlg: destroying dialog 0x7f188afd3bf8 Feb 26 17:29:11 voip2-sip23b /usr/sbin/opensips[4711]:DBG:dialog:destroy_dlg: dlg expired or not in list - dlg 0x7f188afd3bf8[2317:677053409] with clid 'ktQ0Pkdwz50qGYjED6Brpw..' and tags '2b161508' 'NULL' on node1 Feb 26 17:28:41 voip2-sip23a /usr/sbin/opensips[15820]:DBG:dialog:dlg_onroute: route param is 'd09.1e30b582' (len=12) Feb 26 17:28:41 voip2-sip23a /usr/sbin/opensips[15820]:DBG:dialog:lookup_dlg: no dialog id=677053409 found on entry 2317 Feb 26 17:28:41 voip2-sip23a /usr/sbin/opensips[15820]:DBG:dialog:dlg_onroute: unable to find dialog for BYE with route param 'd09.1e30b582' Feb 26 17:28:41 voip2-sip23a /usr/sbin/opensips[15822]:DBG:dialog:dlg_onroute: route param is 'd09.1e30b582' (len=12) Feb 26 17:28:41 voip2-sip23a /usr/sbin/opensips[15822]:DBG:dialog:lookup_dlg: no dialog id=677053409 found on entry 2317 Feb 26 17:28:41 voip2-sip23a /usr/sbin/opensips[15822]:DBG:dialog:dlg_onroute: unable to find dialog for BYE with route param 'd09.1e30b582' Feb 26 17:28:41 voip2-sip23a /usr/sbin/opensips[15822]:DBG:dialog:dlg_onroute: route param is 'd09.1e30b582' (len=12) Feb 26 17:28:41 voip2-sip23a /usr/sbin/opensips[15822]:DBG:dialog:lookup_dlg: no dialog id=677053409 found on entry 2317 Feb 26 17:28:41 voip2-sip23a /usr/sbin/opensips[15822]:DBG:dialog:dlg_onroute: unable to find dialog for BYE with route param 'd09.1e30b582' Feb 26 17:28:42 voip2-sip23a /usr/sbin/opensips[15820]:DBG:dialog:dlg_onroute: route param is 'd09.1e30b582' (len=12) Feb 26 17:28:42 voip2-sip23a /usr/sbin/opensips[15820]:DBG:dialog:lookup_dlg: no dialog id=677053409 found on entry 2317 Feb 26 17:28:42 voip2-sip23a /usr/sbin/opensips[15820]:DBG:dialog:dlg_onroute: unable to find dialog for BYE with route param 'd09.1e30b582' Feb 26 17:28:42 voip2-sip23a /usr/sbin/opensips[15820]:DBG:dialog:dlg_onroute: route param is 'd09.1e30b582' (len=12) Feb 26 17:28:42 voip2-sip23a /usr/sbin/opensips[15820]:DBG:dialog:lookup_dlg: no dialog id=677053409 found on entry 2317 If I purposely take down node1 again and try to make a call I can see that it indicates the failure to replicate dialog info on the cluster as below, Feb 26 17:27:01 voip2-sip23b /usr/sbin/opensips[4701]:ERROR:dialog:replicate_dialog_updated: All destinations in cluster: 1 are down or probing Feb 26 17:27:01 voip2-sip23b /usr/sbin/opensips[4701]:ERROR:dialog:replicate_dialog_updated: Failed to replicate updated dialog _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... 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