From m.shirazi at gmail.com Mon Nov 4 03:42:03 2019 From: m.shirazi at gmail.com (Mehdi Shirazi) Date: Mon, 4 Nov 2019 12:12:03 +0330 Subject: [OpenSIPS-Users] opensip 3.1 rtpengine_play_media Message-ID: Hi I am trying to use rtpengine_play_media in opensips 3.1 If I use : if (is_method("INVITE") && !has_totag()) rtpengine_play_media("file=/path/to/ringback_tone_file.wav"); I receive error about no call ID. If I add rtpengine_manage(); before rtpengine_play_media functions it seems(in logs) rtpengine side is ok but still cannot hear wav file. I should also send 183/sdp myself or it is automatic? Is it possible to provide a working sample of playing a file for all incoming calls? Regards M.Shirazi -------------- next part -------------- An HTML attachment was scrubbed... URL: From razvan at opensips.org Mon Nov 4 04:27:01 2019 From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=) Date: Mon, 4 Nov 2019 11:27:01 +0200 Subject: [OpenSIPS-Users] opensip 3.1 rtpengine_play_media In-Reply-To: References: Message-ID: Hi, Mehdi! You should create the RTPengine session before playing back any media, using either rtpengine_manage() or rtpengine_offer(). Also, most likely the phone does not play the media because no one generates the 183 - it is not done automatically, it should be sent by you to the phone. Best regards, Razvan On 11/4/19 10:42 AM, Mehdi Shirazi wrote: > Hi > I am trying to use rtpengine_play_media in opensips 3.1 > If I use : > if (is_method("INVITE") && !has_totag()) >     rtpengine_play_media("file=/path/to/ringback_tone_file.wav"); > > I receive error about no call ID. If I add > rtpengine_manage();before rtpengine_play_media functions it > > seems(in logs) rtpengine side is ok but still cannot hear wav file. > > I should also send 183/sdp myself or it is automatic? > > Is it possible to provide a working sample of playing a file for all incoming calls? > > Regards > > M.Shirazi > > > _______________________________________________ > 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 From liviu at opensips.org Mon Nov 4 05:04:51 2019 From: liviu at opensips.org (Liviu Chircu) Date: Mon, 4 Nov 2019 12:04:51 +0200 Subject: [OpenSIPS-Users] Example of configuration "Full Sharing" Topology with NoSQL In-Reply-To: <19500210-1d34-06d5-be30-bdd5310a292f@opensips.org> References: <19500210-1d34-06d5-be30-bdd5310a292f@opensips.org> Message-ID: <0d50495e-adaa-5157-5f67-0ca6b380b652@opensips.org> As promised, a tutorial on the "full sharing" setups is now available on the opensips.org wiki [1], showing you how to configure both an active/backup setup and a NoSQL-based one. Enjoy, [1]: https://www.opensips.org/Documentation/Tutorials-Distributed-User-Location-Full-Sharing Liviu Chircu OpenSIPS Developer http://www.opensips-solutions.com On 26.10.2019 14:12, Liviu Chircu wrote: > Given that the configuration differences are so small, I have kept > postponing writing > the tutorial.  But I intend to finally complete it after next week's > AstriCon or > hopefully even during it. From gmaruzz at gmail.com Mon Nov 4 05:17:10 2019 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Mon, 4 Nov 2019 11:17:10 +0100 Subject: [OpenSIPS-Users] Example of configuration "Full Sharing" Topology with NoSQL In-Reply-To: <0d50495e-adaa-5157-5f67-0ca6b380b652@opensips.org> References: <19500210-1d34-06d5-be30-bdd5310a292f@opensips.org> <0d50495e-adaa-5157-5f67-0ca6b380b652@opensips.org> Message-ID: Hello Liviu, a question about the beautiful tutorial: in HA (active-passive) would not be better to have both servers marked as seed (in db), so if any of two goes down, when it comes back online will feed from the other (that in mean time was promoted to active)? If only one server is seed, and that server goes down, when it comes up, will wait 5 seconds, and then "think" to be ok, without feeding from the other... Correct? -giovanni On Mon, Nov 4, 2019 at 11:07 AM Liviu Chircu wrote: > As promised, a tutorial on the "full sharing" setups is now available on > the opensips.org wiki [1], > showing you how to configure both an active/backup setup and a > NoSQL-based one. > > Enjoy, > > [1]: > > https://www.opensips.org/Documentation/Tutorials-Distributed-User-Location-Full-Sharing > > Liviu Chircu > OpenSIPS Developer > http://www.opensips-solutions.com > > On 26.10.2019 14:12, Liviu Chircu wrote: > > Given that the configuration differences are so small, I have kept > > postponing writing > > the tutorial. But I intend to finally complete it after next week's > > AstriCon or > > hopefully even during it. > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -- Sincerely, Giovanni Maruzzelli OpenTelecom.IT cell: +39 347 266 56 18 -------------- next part -------------- An HTML attachment was scrubbed... URL: From social at bohboh.info Mon Nov 4 08:35:10 2019 From: social at bohboh.info (Social Boh) Date: Mon, 4 Nov 2019 08:35:10 -0500 Subject: [OpenSIPS-Users] Example of configuration "Full Sharing" Topology with NoSQL In-Reply-To: <0d50495e-adaa-5157-5f67-0ca6b380b652@opensips.org> References: <19500210-1d34-06d5-be30-bdd5310a292f@opensips.org> <0d50495e-adaa-5157-5f67-0ca6b380b652@opensips.org> Message-ID: <9e25e4eb-d391-e466-6f15-2863c46c8b3e@bohboh.info> thank you very much. Any doubt, i'll write here --- I'm SoCIaL, MayBe El 04/11/2019 a las 05:04, Liviu Chircu escribió: > As promised, a tutorial on the "full sharing" setups is now available > on the opensips.org wiki [1], > showing you how to configure both an active/backup setup and a > NoSQL-based one. > > Enjoy, > > [1]: > https://www.opensips.org/Documentation/Tutorials-Distributed-User-Location-Full-Sharing > > Liviu Chircu > OpenSIPS Developer > http://www.opensips-solutions.com > > On 26.10.2019 14:12, Liviu Chircu wrote: >> Given that the configuration differences are so small, I have kept >> postponing writing >> the tutorial.  But I intend to finally complete it after next week's >> AstriCon or >> hopefully even during it. > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > From liviu at opensips.org Mon Nov 4 08:56:13 2019 From: liviu at opensips.org (Liviu Chircu) Date: Mon, 4 Nov 2019 15:56:13 +0200 Subject: [OpenSIPS-Users] Example of configuration "Full Sharing" Topology with NoSQL In-Reply-To: References: <19500210-1d34-06d5-be30-bdd5310a292f@opensips.org> <0d50495e-adaa-5157-5f67-0ca6b380b652@opensips.org> Message-ID: <8aa0fdf9-cec8-2779-d323-a21510029092@opensips.org> Hi Giovanni! You are absolutely right, nowadays there are very little differences between flagging a node with either "seed" or "NULL", thanks to seed_fallback_interval [1], which makes both of them perform a sync anyway on startup.  The only difference is that a non-seed node will perform a sync _for sure_, no matter how long it waits for other nodes.  This can be a useful feature if your network is particularly poor, with links between nodes which fail often.  In this case, using "NULL" is actually better, as it protects you when restarting your isolated backup nodes -- eventually, they will fetch the data! TL;DR: unless you have very poor network links between your nodes, you can flag your nodes with either "1 x seed, (N-1) x NULL" or "N x seed".  As long as you set a seed_fallback_interval, both will be good. Cheers, [1]: https://opensips.org/html/docs/modules/2.4.x/clusterer.html#param_seed_fallback_interval Liviu Chircu OpenSIPS Developer http://www.opensips-solutions.com On 04.11.2019 12:17, Giovanni Maruzzelli wrote: > Hello Liviu, > > a question about the beautiful tutorial: in HA (active-passive) would > not be better to have both servers marked as seed (in db), so if any > of two goes down, when it comes back online will feed from the other > (that in mean time was promoted to active)? > > If only one server is seed, and that server goes down, when it comes > up, will wait 5 seconds, and then "think" to be ok, without feeding > from the other... Correct? > > -giovanni > > On Mon, Nov 4, 2019 at 11:07 AM Liviu Chircu > wrote: > > As promised, a tutorial on the "full sharing" setups is now > available on > the opensips.org wiki [1], > showing you how to configure both an active/backup setup and a > NoSQL-based one. > > Enjoy, > > [1]: > https://www.opensips.org/Documentation/Tutorials-Distributed-User-Location-Full-Sharing > > Liviu Chircu > OpenSIPS Developer > http://www.opensips-solutions.com > > On 26.10.2019 14:12, Liviu Chircu wrote: > > Given that the configuration differences are so small, I have kept > > postponing writing > > the tutorial.  But I intend to finally complete it after next > week's > > AstriCon or > > hopefully even during it. > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > -- > Sincerely, > > Giovanni Maruzzelli > OpenTelecom.IT > cell: +39 347 266 56 18 > > > _______________________________________________ > 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 darpan.gabani1093 at gmail.com Mon Nov 4 09:25:39 2019 From: darpan.gabani1093 at gmail.com (Darpan Patel) Date: Mon, 4 Nov 2019 19:55:39 +0530 Subject: [OpenSIPS-Users] $DLG_dir getting NULL in opensips 3.0. Message-ID: Hello , I have used *$DLG_dir* for sequential request in* LOCAL_ROUTE* and also did* loose route* but i am getting $DLG_dir * null .(opensips 3.0 version) .* ---------------------------------------------------------------------------------------------------------- *USAGE:* route[WITHINDIALOG] { if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { # validate the sequential request against dialog if ( $DLG_status!=NULL && !validate_dialog() ) { xlog("In-Dialog $rm from $si (callid=$ci) is not valid according to dialog\n"); } } } local_route { xlog("====>> [LOCAL_ROUTE] <<<=="); xlog("L_ERROR", "--- [LOCAL_ROUTE] METHOD: [$rm] DIRECTION: [*$DLG_dir*] ---"); } ----------------------------------------------------------------------------------------------------------- *OUTPUT :* [LOCAL_ROUTE] METHOD: [BYE] DIRECTION: [*null*] please , look into it . and thanks in advance -------------- next part -------------- An HTML attachment was scrubbed... URL: From m.shirazi at gmail.com Mon Nov 4 09:27:38 2019 From: m.shirazi at gmail.com (Mehdi Shirazi) Date: Mon, 4 Nov 2019 17:57:38 +0330 Subject: [OpenSIPS-Users] opensip 3.1 rtpengine_play_media Message-ID: Hi I used : $var(Session_owner) = $rb[1]; append_to_reply("Content-Type: application/sdp\r\nv=0\r\n$var(Session_owner)\r\ns=call\r\nc=IN IP4 10.105.80.3\r\nt=0 0\r\nm=audio 61896 RTP 0 8 3 101\r\na=rtpmap:0 gsm/8000\r\na=rtpmap:8 pcma/8000\r\na=rtpmap:3 gsm/8000\r\na=rtpmap:101 telephone-event/8000\r\na=fmtp:101 0-16\r\na=ptime:20\r\na=sendrecv\r\n"); t_reply_with_body(183, "Session Progress", "$var(body)"); for making a 183 reply but still play media do not works. Any comments? Regards M.Shirazi -------------- next part -------------- An HTML attachment was scrubbed... URL: From gmaruzz at gmail.com Mon Nov 4 14:10:13 2019 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Mon, 4 Nov 2019 20:10:13 +0100 Subject: [OpenSIPS-Users] opensip 3.1 rtpengine_play_media In-Reply-To: References: Message-ID: On Mon, Nov 4, 2019 at 3:29 PM Mehdi Shirazi wrote: > Hi > I used : > $var(Session_owner) = $rb[1]; > append_to_reply("Content-Type: > application/sdp\r\nv=0\r\n$var(Session_owner)\r\ns=call\r\nc=IN IP4 > 10.105.80.3\r\nt=0 0\r\nm=audio 61896 RTP 0 8 3 101\r\na=rtpmap:0 > gsm/8000\r\na=rtpmap:8 pcma/8000\r\na=rtpmap:3 gsm/8000\r\na=rtpmap:101 > telephone-event/8000\r\na=fmtp:101 0-16\r\na=ptime:20\r\na=sendrecv\r\n"); > t_reply_with_body(183, "Session Progress", > "$var(body)"); > for making a 183 reply but still play media do not works. > Best would be to see the full SIP trace... Maybe you can pastebin that trace somewhere and put the link here? -giovanni > Any comments? > Regards > M.Shirazi > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -- Sincerely, Giovanni Maruzzelli OpenTelecom.IT cell: +39 347 266 56 18 -------------- next part -------------- An HTML attachment was scrubbed... URL: From peter at thesimworks.net Tue Nov 5 05:34:46 2019 From: peter at thesimworks.net (Peter Pulham) Date: Tue, 5 Nov 2019 10:34:46 +0000 Subject: [OpenSIPS-Users] force send socket for load balancer ping Message-ID: Hi, I am looking into migrating from dispatcher to load balancer module. Currently I use the socket column in the dispatcher table to specify the socket to use for a particular destination. However, I dont see a similar option in load balancer module. While I could attempt to use force_send_socket in the route, I dont see a way to force OPTIONs ping out of a specific socket for load balancer module. Does anyone have any recommendations on how to deal with this? Many thanks Virus-free. www.avast.com <#DAB4FAD8-2DD7-40BB-A1B8-4E2AA1F9FDF2> -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Tue Nov 5 07:38:48 2019 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Tue, 5 Nov 2019 14:38:48 +0200 Subject: [OpenSIPS-Users] OpenSIPS Summit 2020 - Call For Papers Message-ID: <871ef334-d831-8918-fdb4-74e1f3ae958d@opensips.org> Call For Papers OpenSIPS Summit 2020 May 5th-8th, 2020 Amsterdam, The Netherlands *The Call for Papers is now open!* The OpenSIPS Summit is the meeting place for the OpenSIPS community, for experts, developers and users from all over the world, looking to learn and gain knowledge. The OpenSIPS Summit is a melting pot for discussion on new technology, for sharing experiences, for brainstorming on new trends, for building bridges in the Open-Source VoIP & RTC ecosystem. Submit a paper for OpenSIPS Summit and become a speaker for our experts, brilliant developers and fearless users. Even if we welcome any papers covering the VoIP & RTC area, the major topics of interest for our attendees are (but not limited to): * Implementations with OpenSIPS - scenarios, experience, challenges, solutions * Third-party OSS Integration around OpenSIPS - solutions * Open Source projects in the VoIP & RTC area - news, updates, presentation * Research or development done with OpenSIPS * News, concepts and updates from the VoIP & RTC ecosystem - future directions or current challenges We challenge the speakers to submit paper that bring value to the event, value to be shared with our audience. Our speaker will enjoy free admission to the event, covering lunches and evening events. Submit your paper now * * *Radisson Blu** **Rusland 17, 1012CK Amsterdam, The Netherlands* Meet us again at our familiar Venue, with the usual space and comfort! ** Interested in becoming a sponsor too? Please contact our team or email us! -- Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From Sunil.More at novanet.net Tue Nov 5 08:05:11 2019 From: Sunil.More at novanet.net (Sunil More) Date: Tue, 5 Nov 2019 18:35:11 +0530 Subject: [OpenSIPS-Users] Unable to connect to tls Message-ID: Hi All, I am having trouble to make connection using tls. I am using a polycom sound point 550 phone. I observe the below error. ERROR:proto_tls:tls_print_errstack: TLS errstack: error:1417A0C1:SSL routines:tls_post_process_client_hello:no shared cipher using cipher list as below: modparam("tls_mgm", "ciphers_list", "[dom1]ECDHE-ECDSA-AES128-GCM-SHA256,ECDHE-ECDSA-AES256-GCM-SHA384") Opensips Details: version: opensips 3.0.1 (x86_64/linux) flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, HP_MALLOC, DBG_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, sigio_rt, select. git revision: dac56fa49 main.c compiled on 06:28:01 Oct 24 2019 with gcc 7 openssl version : OpenSSL 1.1.1 11 Sep 2018 OS : ubuntu 18.04 LTS Regards, Sunil More -------------- next part -------------- An HTML attachment was scrubbed... URL: From callum.guy at x-on.co.uk Tue Nov 5 11:08:18 2019 From: callum.guy at x-on.co.uk (Callum Guy) Date: Tue, 5 Nov 2019 16:08:18 +0000 Subject: [OpenSIPS-Users] Unable to connect to tls In-Reply-To: References: Message-ID: The answer is in your email - the device and server don't share any ciphers. You'll have to find an acceptable cipher which meets your security requirements and which your phone supports. I'd suggest you either find a manual for the phone or run a trace to look into the handshake or similar If you are just testing and want to get it working try something basic like: modparam("tls_mgm", "ciphers_list", "[dom1]ALL:!aNULL:!eNULL:!MD5:!RC4:@STRENGTH") On Tue, 5 Nov 2019 at 13:06, Sunil More wrote: > Hi All, > > I am having trouble to make connection using tls. I am using a polycom > sound point 550 phone. > I observe the below error. > > ERROR:proto_tls:tls_print_errstack: TLS errstack: error:1417A0C1:SSL > routines:tls_post_process_client_hello:no shared cipher > > > using cipher list as below: > > modparam("tls_mgm", "ciphers_list", > "[dom1]ECDHE-ECDSA-AES128-GCM-SHA256,ECDHE-ECDSA-AES256-GCM-SHA384") > > > Opensips Details: > > version: opensips 3.0.1 (x86_64/linux) > > flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, > Q_MALLOC, F_MALLOC, HP_MALLOC, DBG_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, sigio_rt, select. > > git revision: dac56fa49 > > main.c compiled on 06:28:01 Oct 24 2019 with gcc 7 > > > openssl version : OpenSSL 1.1.1 11 Sep 2018 > > > OS : ubuntu 18.04 LTS > > > > Regards, > > Sunil More > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -- *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 ahmedmunir007 at gmail.com Tue Nov 5 14:35:41 2019 From: ahmedmunir007 at gmail.com (Ahmed Chohan) Date: Tue, 5 Nov 2019 14:35:41 -0500 Subject: [OpenSIPS-Users] OpenSIPs 2.2.3 SegFault Message-ID: Hi, I'm currently using OpenSIPs v 2.2.3 in our environment for over a year after migrated from version 1.6. As per on 1.6, we were experiencing Segfault and advised to move on 2.2.3 (update to date version during that time) to resolve this issue but it didn't help at all. After going through the forums, it stated root cause is memory allocation for pkmem as last week, I've upgraded from 16 M to 32 MB in config but still experiencing the same issue. See below message below from logs; Nov 4 17:49:50 qorclvsiproxy05 kernel: opensips[7800] general protection ip:4f6112 sp:7fffd35bc6b0 error:0 in opensips (deleted)[400000+1ed000 See below extracted coredump. Please advise to find the root cause and to resolve this issue. Core was generated by `/sbin/opensips -P /var/run/opensips/opensips.pid -m 4096 -M 32 -u opensips -g o'. Program terminated with signal 11, Segmentation fault. #0 fm_status (qm=) at mem/f_malloc.c:709 709 size+=f->size,f=f->u.nxt_free,i++,j++){ } (gdb) bt full #0 fm_status (qm=) at mem/f_malloc.c:709 f = 0x7f7994902d8000 i = 140018 j = h = unused = 0 size = 35881011432156432 __FUNCTION__ = "fm_status" #1 0x000000000043a32a in shm_status (show_status=) at mem/shm_mem.h:611 No locals. #2 cleanup (show_status=) at main.c:339 __FUNCTION__ = "cleanup" #3 0x000000000043ad2c in handle_sigs () at main.c:520 chld = chld_status = 139 overall_status = 139 i = do_exit = 1 __FUNCTION__ = "handle_sigs" #4 0x000000000043ea12 in main_loop (argc=, argv=) at main.c:720 startup_done = chd_rank = 0 #5 main (argc=, argv=) at main.c:1265 cfg_stream = c = r = tmp = 0x7fffd35be90d "" tmp_len = port = 0 proto = 5848501 protos_no = options = 0x59b568 "f:cCm:M:b:l:n:N:rRvdDFETSVhw:t:u:g:P:G:W:o:" ret = -1 seed = 666110575 rfd = __FUNCTION__ = "main" -- Regards, Ahmed Munir Chohan -------------- next part -------------- An HTML attachment was scrubbed... URL: From jwilkie at usipcom.com Tue Nov 5 20:55:30 2019 From: jwilkie at usipcom.com (Jeff Wilkie) Date: Tue, 5 Nov 2019 20:55:30 -0500 Subject: [OpenSIPS-Users] Opensips 2.4.x and CP 8 Message-ID: Attempting to get CP8 and 2.4.x talking to each other. I have modules in the config enabled loadmodule "httpd.so" modparam("httpd", "ip", "127.0.0.1") modparam("httpd", "port", 8888) loadmodule "mi_json.so" modparam("mi_json", "mi_json_root", "json") When attempting to execute any MI_json command from CP we're seeing this from the opensips logs. ERROR:mi_json:mi_json_answer_to_connection: unexpected method [POST] Where is this error being generated? How do we correct this? Thanks Jeff -------------- next part -------------- An HTML attachment was scrubbed... URL: From jwilkie at usipcom.com Tue Nov 5 22:23:34 2019 From: jwilkie at usipcom.com (Jeff Wilkie) Date: Tue, 5 Nov 2019 22:23:34 -0500 Subject: [OpenSIPS-Users] Recommended Radius client/server for AAA on 2.4.x under Debian 9 Message-ID: Attempting to find current docs since radiusclient-ng is referenced in several old docs but is no longer available. Currently, what is the recommending radius packages to use for accounting purposes on opensips for CDRs and ACC? Thanks Jeff -------------- next part -------------- An HTML attachment was scrubbed... URL: From qasimakhan at gmail.com Tue Nov 5 22:48:31 2019 From: qasimakhan at gmail.com (qasimakhan at gmail.com) Date: Wed, 6 Nov 2019 08:48:31 +0500 Subject: [OpenSIPS-Users] Recommended Radius client/server for AAA on 2.4.x under Debian 9 In-Reply-To: References: Message-ID: You can use latest version of freeradius it has both client and server. Regards, Qasim On Wed, 6 Nov 2019 at 8:27 AM, Jeff Wilkie wrote: > Attempting to find current docs since radiusclient-ng is referenced in > several old docs but is no longer available. Currently, what is the > recommending radius packages to use for accounting purposes on opensips for > CDRs and ACC? > > Thanks > Jeff > _______________________________________________ > 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 qasimakhan at gmail.com Tue Nov 5 22:51:46 2019 From: qasimakhan at gmail.com (qasimakhan at gmail.com) Date: Wed, 6 Nov 2019 08:51:46 +0500 Subject: [OpenSIPS-Users] Opensips 2.4.x and CP 8 In-Reply-To: References: Message-ID: Hi, Here is your error: ERROR:mi_json:mi_json_answer_to_connection: unexpected method [POST] If you look up in documentation it says on first line: JSON support via HTTP GET for Management Interface Regards Qasim On Wed, 6 Nov 2019 at 6:58 AM, Jeff Wilkie wrote: > Attempting to get CP8 and 2.4.x talking to each other. I have modules in > the config enabled > > loadmodule "httpd.so" > modparam("httpd", "ip", "127.0.0.1") > modparam("httpd", "port", 8888) > loadmodule "mi_json.so" > modparam("mi_json", "mi_json_root", "json") > > When attempting to execute any MI_json command from CP we're seeing this > from the opensips logs. > > ERROR:mi_json:mi_json_answer_to_connection: unexpected method [POST] > > Where is this error being generated? How do we correct this? > > Thanks > Jeff > > > _______________________________________________ > 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 darpan.gabani1093 at gmail.com Wed Nov 6 01:44:27 2019 From: darpan.gabani1093 at gmail.com (Darpan Patel) Date: Wed, 6 Nov 2019 12:14:27 +0530 Subject: [OpenSIPS-Users] $DLG_dir getting NULL in opensips 3.0. In-Reply-To: References: Message-ID: Any solution? On Mon, 4 Nov 2019, 7:55 pm Darpan Patel, wrote: > Hello , I have used *$DLG_dir* for sequential request in* LOCAL_ROUTE* > and also did* loose route* but i am getting $DLG_dir > * null .(opensips 3.0 version) .* > > > ---------------------------------------------------------------------------------------------------------- > *USAGE:* > route[WITHINDIALOG] { > if (has_totag()) { > # sequential request withing a dialog should > # take the path determined by record-routing > if (loose_route()) { > # validate the sequential request against dialog > if ( $DLG_status!=NULL && !validate_dialog() ) { > xlog("In-Dialog $rm from $si (callid=$ci) is not valid > according to dialog\n"); > } > } > } > local_route { > xlog("====>> [LOCAL_ROUTE] <<<=="); > > xlog("L_ERROR", "--- [LOCAL_ROUTE] METHOD: [$rm] DIRECTION: [ > *$DLG_dir*] ---"); > } > > ----------------------------------------------------------------------------------------------------------- > *OUTPUT :* > > [LOCAL_ROUTE] METHOD: [BYE] DIRECTION: [*null*] > > please , look into it . and thanks in advance > > > > > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From razvan at opensips.org Wed Nov 6 04:26:58 2019 From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=) Date: Wed, 6 Nov 2019 11:26:58 +0200 Subject: [OpenSIPS-Users] force send socket for load balancer ping In-Reply-To: References: Message-ID: <1e6912cb-40fe-f6a4-3bca-48912922d0f9@opensips.org> Hi, Peter! Unfortunately this feature is not available for now for the load_balancer module. There's actually a feature request already for doing this :)[1]. For now, what you can do is to "catch" that OPTIONS message in the `local_route` and force the send socket over there. [1] https://github.com/OpenSIPS/opensips/issues/1079 Best regards, Răzvan On 11/5/19 12:34 PM, Peter Pulham wrote: > Hi, > > I am looking into migrating from dispatcher to load balancer module. > > Currently I use the socket column in the dispatcher table to specify the > socket to use for a particular destination. > > However, I dont see a similar option in load balancer module. > > While I could attempt to use force_send_socket in the route, I dont see > a way to force OPTIONs ping out of a specific socket for load balancer > module. > > Does anyone have any recommendations on how to deal with this? > > Many thanks > > > Virus-free. www.avast.com > > > > <#DAB4FAD8-2DD7-40BB-A1B8-4E2AA1F9FDF2> > > _______________________________________________ > 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 From razvan at opensips.org Wed Nov 6 04:32:45 2019 From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=) Date: Wed, 6 Nov 2019 11:32:45 +0200 Subject: [OpenSIPS-Users] OpenSIPs 2.2.3 SegFault In-Reply-To: References: Message-ID: Hi, Ahmed! OpenSIPS 2.2 is quite an old version and is no longer supported for a while. If you have the possibility, I'd suggest you to migrate to 2.4, which is a stable LTS - if the issue occurs there, then we are able to investigate it and track down the problem. If you can't do that, you're best chance is to upgrade to the latest 2.2 release, which is 2.2.7. If that still crashes, there's not that much you can do, as 2.2 is out of support now, and you will need to migrate to 2.4 anyway. Best regards, Răzvan On 11/5/19 9:35 PM, Ahmed Chohan wrote: > Hi, > > I'm currently using OpenSIPs v 2.2.3 in our environment for over a year > after migrated from version 1.6. As per on 1.6, we were experiencing > Segfault and advised to move on 2.2.3 (update to date version during > that time) to resolve this issue but it didn't help at all. > > After going through the forums, it stated root cause is memory > allocation for pkmem as last week, I've upgraded from 16 M to 32 MB  in > config but still experiencing the same issue. See below message below > from logs; > > Nov  4 17:49:50 qorclvsiproxy05 kernel: opensips[7800] general > protection ip:4f6112 sp:7fffd35bc6b0 error:0 in opensips > (deleted)[400000+1ed000 > > See below extracted coredump. Please advise to find the root cause and > to resolve this issue. > > Core was generated by `/sbin/opensips -P /var/run/opensips/opensips.pid > -m 4096 -M 32 -u opensips -g o'. > Program terminated with signal 11, Segmentation fault. > #0  fm_status (qm=) at mem/f_malloc.c:709 > 709 > size+=f->size,f=f->u.nxt_free,i++,j++){ } > (gdb) bt full > #0  fm_status (qm=) at mem/f_malloc.c:709 >         f = 0x7f7994902d8000 >         i = 140018 >         j = >         h = >         unused = 0 >         size = 35881011432156432 >         __FUNCTION__ = "fm_status" > #1  0x000000000043a32a in shm_status (show_status=) > at mem/shm_mem.h:611 > No locals. > #2  cleanup (show_status=) at main.c:339 >         __FUNCTION__ = "cleanup" > #3  0x000000000043ad2c in handle_sigs () at main.c:520 >         chld = >         chld_status = 139 >         overall_status = 139 >         i = >         do_exit = 1 >         __FUNCTION__ = "handle_sigs" > #4  0x000000000043ea12 in main_loop (argc=, > argv=) at main.c:720 >         startup_done = >         chd_rank = 0 > #5  main (argc=, argv=) at > main.c:1265 >         cfg_stream = >         c = >         r = >         tmp = 0x7fffd35be90d "" >         tmp_len = >         port = 0 >         proto = 5848501 >         protos_no = >         options = 0x59b568 "f:cCm:M:b:l:n:N:rRvdDFETSVhw:t:u:g:P:G:W:o:" >         ret = -1 >         seed = 666110575 >         rfd = >         __FUNCTION__ = "main" > > > > -- > > Regards, > > Ahmed Munir Chohan > > > _______________________________________________ > 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 From razvan at opensips.org Wed Nov 6 04:33:27 2019 From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=) Date: Wed, 6 Nov 2019 11:33:27 +0200 Subject: [OpenSIPS-Users] Recommended Radius client/server for AAA on 2.4.x under Debian 9 In-Reply-To: References: Message-ID: radcli is the library we're using in our latest packages. Best regards, Răzvan On 11/6/19 5:48 AM, qasimakhan at gmail.com wrote: > You can use latest version of freeradius it has both client and server. > > Regards, > Qasim > > On Wed, 6 Nov 2019 at 8:27 AM, Jeff Wilkie > wrote: > > Attempting to find current docs since radiusclient-ng is referenced > in several old docs but is no longer available.  Currently, what is > the recommending radius packages to use for accounting purposes on > opensips for CDRs and ACC? > > Thanks > Jeff > _______________________________________________ > 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 From callum.guy at x-on.co.uk Wed Nov 6 10:10:22 2019 From: callum.guy at x-on.co.uk (Callum Guy) Date: Wed, 6 Nov 2019 15:10:22 +0000 Subject: [OpenSIPS-Users] 3.0.1 Full sharing registrar - mysql write back not working Message-ID: Hi All, I've been testing a new cluster based registrar config and noticed that the location records weren't being written back to the database. The issue appears resolved when I switch usrloc.restart_persistency from "sync-from-cluster" to "load-from-sql". This is acceptable however it seems more correct to allow clusterer to sync the data for me. modparam("usrloc", "location_cluster", 25) modparam("usrloc", "cluster_mode", "full-sharing") modparam("usrloc", "pinging_mode", "ownership") modparam("usrloc", "restart_persistency", "*sync-from-cluster*") modparam("usrloc", "sql_write_mode", "write-back") modparam("usrloc", "timer_interval", 60) modparam("usrloc", "skip_replicated_db_ops", 1) modparam("usrloc", "max_contact_delete", 25) modparam("usrloc", "hash_size", 16) modparam("usrloc", "nat_bflag", "NATTED_CONTACT") modparam("usrloc", "use_domain", 1) modparam("usrloc", "db_url", "mysql://user:abc at 192.168.151.20/opensips") I have attempted several variations including removing skip_replicated_db_ops and changing sql_write_mode however nothing resolved the issue. The contacts were visible within the cluster (mi ul_dump) but I couldn't get them to sync back to the database at all. My target implementation is to have a pair of instances, sharing a VIP (keepalived), such that when a node is terminated the clusterer module has preshared all the data and registrations and dialogs do not drop. I have configuration for the dialog and clusterer module which I would be happy to share if required. I would be very interested to hear if I am doing something wrong or misunderstanding how it should work! Many thanks, Callum -- *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 liviu at opensips.org Wed Nov 6 11:00:03 2019 From: liviu at opensips.org (Liviu Chircu) Date: Wed, 6 Nov 2019 18:00:03 +0200 Subject: [OpenSIPS-Users] 3.0.1 Full sharing registrar - mysql write back not working In-Reply-To: References: Message-ID: <0c4eaf73-29d9-79ed-d991-4ef8a055ca4e@opensips.org> Hi Callum, In short: * "sync-from-cluster" will deny all DB writes.  It is useful in case you don't   mind the risk of losing all registrations in case the active node is offline   and you restart the backup for some reason.  On the positive side, you will   push a lot less queries to the disk. * "load-from-sql" will enable all DB writes.  By using it, you gain better   durability for the registrations, just like in the vanilla OpenSIPS usrloc.   This is just a restart persistency related setting -- the cluster nodes will   still continue to replicate data at runtime and you will still be able to   invoke the "ul_cluster_sync" MI command. * "skip_replicated_db_ops" is some hack from version 2.2, and we haven't taken   the time to remove it from the code.  It seems to be still functional, so as   long as you enable it, the DB writes will be denied. Regards, Liviu Chircu OpenSIPS Developer http://www.opensips-solutions.com On 06.11.2019 17:10, Callum Guy wrote: > Hi All, > > I've been testing a new cluster based registrar config and noticed > that the location records weren't being written back to the database. > > The issue appears resolved when I switch usrloc.restart_persistency > from "sync-from-cluster" to "load-from-sql". This is acceptable > however it seems more correct to allow clusterer to sync the data for me. > > modparam("usrloc", "location_cluster", 25) > modparam("usrloc", "cluster_mode", "full-sharing") > modparam("usrloc", "pinging_mode", "ownership") > modparam("usrloc", "restart_persistency", "*sync-from-cluster*") > modparam("usrloc", "sql_write_mode", "write-back") > modparam("usrloc", "timer_interval", 60) > modparam("usrloc", "skip_replicated_db_ops", 1) > modparam("usrloc", "max_contact_delete", 25) > modparam("usrloc", "hash_size", 16) > modparam("usrloc", "nat_bflag", "NATTED_CONTACT") > modparam("usrloc", "use_domain", 1) > modparam("usrloc", "db_url", "mysql://user:abc at 192.168.151.20/opensips > ") > > I have attempted several variations including > removing skip_replicated_db_ops and changing sql_write_mode however > nothing resolved the issue. The contacts were visible within the > cluster (mi ul_dump) but I couldn't get them to sync back to the > database at all. > > My target implementation is to have a pair of instances, sharing a VIP > (keepalived), such that when a node is terminated the clusterer module > has preshared all the data and registrations and dialogs do not drop. > I have configuration for the dialog and clusterer module which I would > be happy to share if required. I would be very interested to hear if I > am doing something wrong or misunderstanding how it should work! > > Many thanks, > > Callum > > > *^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 callum.guy at x-on.co.uk Wed Nov 6 11:52:12 2019 From: callum.guy at x-on.co.uk (Callum Guy) Date: Wed, 6 Nov 2019 16:52:12 +0000 Subject: [OpenSIPS-Users] 3.0.1 Full sharing registrar - mysql write back not working In-Reply-To: <0c4eaf73-29d9-79ed-d991-4ef8a055ca4e@opensips.org> References: <0c4eaf73-29d9-79ed-d991-4ef8a055ca4e@opensips.org> Message-ID: Thanks Liviu, that helps a lot. My expectation was that the sync-from-cluster option would favour a sync operation from a local active peer via the binary interface on first boot, that sounded like the ideal behaviour as it should be fully up to date; especially in db write back mode. For our production operations it isn't common for the server to be taken down unexpectedly so in reality it won't make too much difference to use the database as the primary data source on boot. I can confirm that the load-from-sql behaviour is replicating the contacts via clusterer and provides us with the resilience we need, in addition the ownership pinging_mode is working exactly as it should - thanks for these neat features! Should I remove skip_replicated_db_ops from our configuration, we're about to go live and if it is not a useful setting we would want to take this opportunity to purge it! To clarify, my interpretation of the docs for this option is that it would prevent duplication of the DB writes for new/updated registrations which we would not want. In our two node clusters we'd ideally just have the second as a hot backup which captures all the dialog and user data but doesn't do anything with it until it gains the sharing tag - including DB ops. If we can remove the skip_replicated_db_ops option and keep this behaviour please let me know, otherwise we'll leave it in place :) On Wed, 6 Nov 2019 at 16:01, Liviu Chircu wrote: > Hi Callum, > > In short: > > * "sync-from-cluster" will deny all DB writes. It is useful in case you > don't > mind the risk of losing all registrations in case the active node is > offline > and you restart the backup for some reason. On the positive side, you > will > push a lot less queries to the disk. > > * "load-from-sql" will enable all DB writes. By using it, you gain better > durability for the registrations, just like in the vanilla OpenSIPS > usrloc. > This is just a restart persistency related setting -- the cluster nodes > will > still continue to replicate data at runtime and you will still be able to > invoke the "ul_cluster_sync" MI command. > > * "skip_replicated_db_ops" is some hack from version 2.2, and we haven't > taken > the time to remove it from the code. It seems to be still functional, > so as > long as you enable it, the DB writes will be denied. > > Regards, > > Liviu Chircu > OpenSIPS Developerhttp://www.opensips-solutions.com > > On 06.11.2019 17:10, Callum Guy wrote: > > Hi All, > > I've been testing a new cluster based registrar config and noticed that > the location records weren't being written back to the database. > > The issue appears resolved when I switch usrloc.restart_persistency from > "sync-from-cluster" to "load-from-sql". This is acceptable however it seems > more correct to allow clusterer to sync the data for me. > > modparam("usrloc", "location_cluster", 25) > modparam("usrloc", "cluster_mode", "full-sharing") > modparam("usrloc", "pinging_mode", "ownership") > modparam("usrloc", "restart_persistency", "*sync-from-cluster*") > modparam("usrloc", "sql_write_mode", "write-back") > modparam("usrloc", "timer_interval", 60) > modparam("usrloc", "skip_replicated_db_ops", 1) > modparam("usrloc", "max_contact_delete", 25) > modparam("usrloc", "hash_size", 16) > modparam("usrloc", "nat_bflag", "NATTED_CONTACT") > modparam("usrloc", "use_domain", 1) > modparam("usrloc", "db_url", "mysql://user:abc at 192.168.151.20/opensips") > > I have attempted several variations including > removing skip_replicated_db_ops and changing sql_write_mode however nothing > resolved the issue. The contacts were visible within the cluster (mi > ul_dump) but I couldn't get them to sync back to the database at all. > > My target implementation is to have a pair of instances, sharing a VIP > (keepalived), such that when a node is terminated the clusterer module has > preshared all the data and registrations and dialogs do not drop. I have > configuration for the dialog and clusterer module which I would be happy to > share if required. I would be very interested to hear if I am doing > something wrong or misunderstanding how it should work! > > Many thanks, > > Callum > > > *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 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 > -- *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 liviu at opensips.org Wed Nov 6 12:39:03 2019 From: liviu at opensips.org (Liviu Chircu) Date: Wed, 6 Nov 2019 19:39:03 +0200 Subject: [OpenSIPS-Users] 3.0.1 Full sharing registrar - mysql write back not working In-Reply-To: References: <0c4eaf73-29d9-79ed-d991-4ef8a055ca4e@opensips.org> Message-ID: <201e625a-01b1-67d0-f7b5-3bfe0f69dde7@opensips.org> The answer is: it depends.  If each of your servers has its own, local "location" table for persistency reasons, then you must remove "skip_replicated_db_ops".  This is our recommended way of ensuring persistency:  no more shared DB shenanigans, keep the tables private. However, if your architecture forces each of your servers to share the same "location" table, then "skip_replicated_db_ops" may help, hopefully if it's still 100% functional. Cheers, Liviu Chircu OpenSIPS Developer http://www.opensips-solutions.com On 06.11.2019 18:52, Callum Guy wrote: > Thanks Liviu, that helps a lot. My expectation was that > the sync-from-cluster option would favour a sync operation from a > local active peer via the binary interface on first boot, that sounded > like the ideal behaviour as it should be fully up to date; especially > in db write back mode. For our production operations it isn't common > for the server to be taken down unexpectedly so in reality it won't > make too much difference to use the database as the primary data > source on boot. > > I can confirm that the load-from-sql behaviour is replicating the > contacts via clusterer and provides us with the resilience we need, in > addition the ownership pinging_mode is working exactly as it should - > thanks for these neat features! > > Should I remove skip_replicated_db_ops from our configuration, we're > about to go live and if it is not a useful setting we would want to > take this opportunity to purge it! To clarify, my interpretation of > the docs for this option is that it would prevent duplication of the > DB writes for new/updated registrations which we would not want. In > our two node clusters we'd ideally just have the second as a hot > backup which captures all the dialog and user data but doesn't do > anything with it until it gains the sharing tag - including DB ops. If > we can remove the skip_replicated_db_ops option and keep this > behaviour please let me know, otherwise we'll leave it in place :) > > > On Wed, 6 Nov 2019 at 16:01, Liviu Chircu > wrote: > > Hi Callum, > > In short: > > * "sync-from-cluster" will deny all DB writes.  It is useful in > case you don't >   mind the risk of losing all registrations in case the active > node is offline >   and you restart the backup for some reason.  On the positive > side, you will >   push a lot less queries to the disk. > > * "load-from-sql" will enable all DB writes.  By using it, you > gain better >   durability for the registrations, just like in the vanilla > OpenSIPS usrloc. >   This is just a restart persistency related setting -- the > cluster nodes will >   still continue to replicate data at runtime and you will still > be able to >   invoke the "ul_cluster_sync" MI command. > > * "skip_replicated_db_ops" is some hack from version 2.2, and we > haven't taken >   the time to remove it from the code.  It seems to be still > functional, so as >   long as you enable it, the DB writes will be denied. > > Regards, > > Liviu Chircu > OpenSIPS Developer > http://www.opensips-solutions.com > > On 06.11.2019 17:10, Callum Guy wrote: >> Hi All, >> >> I've been testing a new cluster based registrar config and >> noticed that the location records weren't being written back to >> the database. >> >> The issue appears resolved when I switch >> usrloc.restart_persistency from "sync-from-cluster" to >> "load-from-sql". This is acceptable however it seems more correct >> to allow clusterer to sync the data for me. >> >> modparam("usrloc", "location_cluster", 25) >> modparam("usrloc", "cluster_mode", "full-sharing") >> modparam("usrloc", "pinging_mode", "ownership") >> modparam("usrloc", "restart_persistency", "*sync-from-cluster*") >> modparam("usrloc", "sql_write_mode", "write-back") >> modparam("usrloc", "timer_interval", 60) >> modparam("usrloc", "skip_replicated_db_ops", 1) >> modparam("usrloc", "max_contact_delete", 25) >> modparam("usrloc", "hash_size", 16) >> modparam("usrloc", "nat_bflag", "NATTED_CONTACT") >> modparam("usrloc", "use_domain", 1) >> modparam("usrloc", "db_url", >> "mysql://user:abc at 192.168.151.20/opensips >> ") >> >> I have attempted several variations including >> removing skip_replicated_db_ops and changing sql_write_mode >> however nothing resolved the issue. The contacts were visible >> within the cluster (mi ul_dump) but I couldn't get them to sync >> back to the database at all. >> >> My target implementation is to have a pair of instances, sharing >> a VIP (keepalived), such that when a node is terminated the >> clusterer module has preshared all the data and registrations and >> dialogs do not drop. I have configuration for the dialog and >> clusterer module which I would be happy to share if required. I >> would be very interested to hear if I am doing something wrong or >> misunderstanding how it should work! >> >> Many thanks, >> >> Callum >> >> >> *^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 > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > *^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 jwilkie at usipcom.com Wed Nov 6 13:02:05 2019 From: jwilkie at usipcom.com (Jeff Wilkie) Date: Wed, 6 Nov 2019 13:02:05 -0500 Subject: [OpenSIPS-Users] Opensips 2.4.x and CP 8 In-Reply-To: References: Message-ID: > > Thanks for the response. That document does not explain why I'm seeing a > POST error using the CP8. If I directly http to opensips using requests > for json it works properly using a GET. Why would the the CP generate a > POST using JSON? How do I turn that off? Any reload command in the CP > generate this ERROR in the opensips logs. > -------------- next part -------------- An HTML attachment was scrubbed... URL: From jwilkie at usipcom.com Wed Nov 6 13:16:06 2019 From: jwilkie at usipcom.com (Jeff Wilkie) Date: Wed, 6 Nov 2019 13:16:06 -0500 Subject: [OpenSIPS-Users] Recommended Radius client/server for AAA on 2.4.x under Debian 9 In-Reply-To: References: Message-ID: Any build and config docs to support radcli installation with opensips 2.4 and 3.x by chance? Thanks Jeff On Wed, Nov 6, 2019 at 4:56 AM Răzvan Crainea wrote: > radcli is the library we're using in our latest packages. > > Best regards, > Răzvan > > On 11/6/19 5:48 AM, qasimakhan at gmail.com wrote: > > You can use latest version of freeradius it has both client and server. > > > > Regards, > > Qasim > > > > On Wed, 6 Nov 2019 at 8:27 AM, Jeff Wilkie > > wrote: > > > > Attempting to find current docs since radiusclient-ng is referenced > > in several old docs but is no longer available. Currently, what is > > the recommending radius packages to use for accounting purposes on > > opensips for CDRs and ACC? > > > > Thanks > > Jeff > > _______________________________________________ > > 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 > > _______________________________________________ > 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 Nov 6 14:12:46 2019 From: Ben.Newlin at genesys.com (Ben Newlin) Date: Wed, 6 Nov 2019 19:12:46 +0000 Subject: [OpenSIPS-Users] Opensips 2.4.x and CP 8 In-Reply-To: References: Message-ID: <80ECE515-1618-4671-A834-FA084CD204B8@genesys.com> Jeff, I didn’t see this in your original post but have you verified you are using CP 8.2.4? Ben Newlin From: Users on behalf of Jeff Wilkie Reply-To: OpenSIPS users mailling list Date: Wednesday, November 6, 2019 at 1:03 PM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] Opensips 2.4.x and CP 8 Thanks for the response. That document does not explain why I'm seeing a POST error using the CP8. If I directly http to opensips using requests for json it works properly using a GET. Why would the the CP generate a POST using JSON? How do I turn that off? Any reload command in the CP generate this ERROR in the opensips logs. -------------- next part -------------- An HTML attachment was scrubbed... URL: From jwilkie at usipcom.com Wed Nov 6 15:31:25 2019 From: jwilkie at usipcom.com (Jeff Wilkie) Date: Wed, 6 Nov 2019 15:31:25 -0500 Subject: [OpenSIPS-Users] Opensips 2.4.x and CP 8 In-Reply-To: <80ECE515-1618-4671-A834-FA084CD204B8@genesys.com> References: <80ECE515-1618-4671-A834-FA084CD204B8@genesys.com> Message-ID: Last comments in Changelog are the following dates 2019-07-22 Bogdan Iancu I think I downloaded 8.3.0 by mistake! oooof. Let me reinstall with 8.2.4 git Thanks for the catch. -------------- next part -------------- An HTML attachment was scrubbed... URL: From callum.guy at x-on.co.uk Wed Nov 6 18:13:15 2019 From: callum.guy at x-on.co.uk (Callum Guy) Date: Wed, 6 Nov 2019 23:13:15 +0000 Subject: [OpenSIPS-Users] 3.0.1 Full sharing registrar - mysql write back not working In-Reply-To: <201e625a-01b1-67d0-f7b5-3bfe0f69dde7@opensips.org> References: <0c4eaf73-29d9-79ed-d991-4ef8a055ca4e@opensips.org> <201e625a-01b1-67d0-f7b5-3bfe0f69dde7@opensips.org> Message-ID: Ok, in this case we are running a central database per geographic location with two local instances per site, each site serves a unique domain. The goal is to have it setup such that clients will have registrations in two sites and are served by a single server per site, a VIP owner. The second instance serves as a hot standby and would ideally be primed with dialog and location awareness in the event that the first instance fails. When the failed node is restored we'd like it to fall back cleanly with as little impact on the users as possible. skip_replicated_db_ops sounds like it should prevent both servers writing back to the shared table which could be problematic and a waste of electricity. If that is the expected behaviour then it sounds helpful so I'll use it until I know better. I'll update you if I find anything worrying in the source (or production!) :) On Wed, 6 Nov 2019, 17:40 Liviu Chircu, wrote: > The answer is: it depends. If each of your servers has its own, local > "location" table for > persistency reasons, then you must remove "skip_replicated_db_ops". This > is our > recommended way of ensuring persistency: no more shared DB shenanigans, > keep the tables private. > > However, if your architecture forces each of your servers to share the > same "location" > table, then "skip_replicated_db_ops" may help, hopefully if it's still > 100% functional. > > Cheers, > > Liviu Chircu > OpenSIPS Developerhttp://www.opensips-solutions.com > > On 06.11.2019 18:52, Callum Guy wrote: > > Thanks Liviu, that helps a lot. My expectation was that > the sync-from-cluster option would favour a sync operation from a local > active peer via the binary interface on first boot, that sounded like the > ideal behaviour as it should be fully up to date; especially in db write > back mode. For our production operations it isn't common for the server to > be taken down unexpectedly so in reality it won't make too much difference > to use the database as the primary data source on boot. > > I can confirm that the load-from-sql behaviour is replicating the contacts > via clusterer and provides us with the resilience we need, in addition the > ownership pinging_mode is working exactly as it should - thanks for these > neat features! > > Should I remove skip_replicated_db_ops from our configuration, we're about > to go live and if it is not a useful setting we would want to take this > opportunity to purge it! To clarify, my interpretation of the docs for this > option is that it would prevent duplication of the DB writes for > new/updated registrations which we would not want. In our two node clusters > we'd ideally just have the second as a hot backup which captures all the > dialog and user data but doesn't do anything with it until it gains the > sharing tag - including DB ops. If we can remove the skip_replicated_db_ops > option and keep this behaviour please let me know, otherwise we'll leave it > in place :) > > > On Wed, 6 Nov 2019 at 16:01, Liviu Chircu wrote: > >> Hi Callum, >> >> In short: >> >> * "sync-from-cluster" will deny all DB writes. It is useful in case you >> don't >> mind the risk of losing all registrations in case the active node is >> offline >> and you restart the backup for some reason. On the positive side, you >> will >> push a lot less queries to the disk. >> >> * "load-from-sql" will enable all DB writes. By using it, you gain better >> durability for the registrations, just like in the vanilla OpenSIPS >> usrloc. >> This is just a restart persistency related setting -- the cluster nodes >> will >> still continue to replicate data at runtime and you will still be able >> to >> invoke the "ul_cluster_sync" MI command. >> >> * "skip_replicated_db_ops" is some hack from version 2.2, and we haven't >> taken >> the time to remove it from the code. It seems to be still functional, >> so as >> long as you enable it, the DB writes will be denied. >> >> Regards, >> >> Liviu Chircu >> OpenSIPS Developerhttp://www.opensips-solutions.com >> >> On 06.11.2019 17:10, Callum Guy wrote: >> >> Hi All, >> >> I've been testing a new cluster based registrar config and noticed that >> the location records weren't being written back to the database. >> >> The issue appears resolved when I switch usrloc.restart_persistency from >> "sync-from-cluster" to "load-from-sql". This is acceptable however it seems >> more correct to allow clusterer to sync the data for me. >> >> modparam("usrloc", "location_cluster", 25) >> modparam("usrloc", "cluster_mode", "full-sharing") >> modparam("usrloc", "pinging_mode", "ownership") >> modparam("usrloc", "restart_persistency", "*sync-from-cluster*") >> modparam("usrloc", "sql_write_mode", "write-back") >> modparam("usrloc", "timer_interval", 60) >> modparam("usrloc", "skip_replicated_db_ops", 1) >> modparam("usrloc", "max_contact_delete", 25) >> modparam("usrloc", "hash_size", 16) >> modparam("usrloc", "nat_bflag", "NATTED_CONTACT") >> modparam("usrloc", "use_domain", 1) >> modparam("usrloc", "db_url", "mysql://user:abc at 192.168.151.20/opensips") >> >> I have attempted several variations including >> removing skip_replicated_db_ops and changing sql_write_mode however nothing >> resolved the issue. The contacts were visible within the cluster (mi >> ul_dump) but I couldn't get them to sync back to the database at all. >> >> My target implementation is to have a pair of instances, sharing a VIP >> (keepalived), such that when a node is terminated the clusterer module has >> preshared all the data and registrations and dialogs do not drop. I have >> configuration for the dialog and clusterer module which I would be happy to >> share if required. I would be very interested to hear if I am doing >> something wrong or misunderstanding how it should work! >> >> Many thanks, >> >> Callum >> >> >> *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 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 >> > > > *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 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 > -- *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 suharik71 at gmail.com Thu Nov 7 03:19:19 2019 From: suharik71 at gmail.com (=?UTF-8?B?0JDQvdGC0L7QvSDQldGA0YjQvtCy?=) Date: Thu, 7 Nov 2019 11:19:19 +0300 Subject: [OpenSIPS-Users] segfault in dialog.so Message-ID: Hello friends, I already wrote about this problem and there was no answer. http://lists.opensips.org/pipermail/users/2019-October/041771.html I still want to process call profiles in event_route, but opensips falls into error at the same time. opensips -V version: opensips 3.0.1 (x86_64/linux) flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, HP_MALLOC, DBG_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, sigio_rt, select. git revision: 3d2bd5318 main.c compiled on 02:53:35 Nov 7 2019 with gcc 4.8.5 my event_route event_route[E_DLG_STATE_CHANGED] { xlog("L_INFO", "[$param(callid)] - new state dialog changed $param(new_state) old state $param(old_state) \n"); cache_raw_query("redis:group3", "HGET callid $param(callid)","$avp(reknum)"); if ($avp(reknum) != NULL) { xlog("L_INFO", "[$param(callid)] - get reknum $avp(reknum) \n"); if ( get_dialogs_by_profile("reknumber", $avp(reknum), $avp(dlg_jsons), $avp(callcount)) ) { xlog("L_INFO", "[$param(callid)] - advertising number $avp(reknum) has $avp(callcount) other calls \n"); } else { xlog("L_INFO", "[$param(callid)] - this profile does not have active dialogs \n"); $avp(callcount) = 0; } switch($param(new_state)) { case 4: xlog("L_INFO", "[$param(callid)] - write in REDIS \"callcount $avp(reknum) $avp(callcount) \n"); if ($avp(reknum) != "" && $avp(callcount) != "") { cache_raw_query("redis:group3","HSET callcount $avp(reknum) $avp(callcount)"); cache_raw_query("redis:group3","EXPIRE callconut 360"); xlog("L_INFO", "[$param(callid)] - call established. Now on number $avp(reknum) $avp(callcount) calls \n"); } break; case 5: xlog("L_INFO", "[$param(callid)] - write in REDIS \"callcount $avp(reknum) $avp(callcount) \n"); if ($avp(reknum) != "" && $avp(callcount) != "") { cache_raw_query("redis:group3","HSET callcount $avp(reknum) $avp(callcount)"); cache_raw_query("redis:group3","EXPIRE callcount 360"); xlog("L_INFO", "[$param(callid)] - call end. Now on number $avp(reknum) $avp(callcount) calls \n"); cache_raw_query("redis:group3","HDEL callid $param(callid)"); } break; } } else { xlog("L_INFO", "[$param(callid)] - reknum vareable is NULL"); exit; } } when we try to get a profile of an already dead dialog opensips falls. please tell me is it possible to implement what I want or is it worth looking for another solution? -------------- next part -------------- An HTML attachment was scrubbed... URL: From podguiko at mail.ru Thu Nov 7 13:15:10 2019 From: podguiko at mail.ru (=?UTF-8?B?T2xlZyBQb2RndXlrbw==?=) Date: Thu, 07 Nov 2019 21:15:10 +0300 Subject: [OpenSIPS-Users] =?utf-8?q?Opensips_generate_two_Content-lenth_he?= =?utf-8?q?aders?= Message-ID: <1573150510.509115405@f514.i.mail.ru> I’m using opensips as proxy. I got INVITE from one side, do some logics ,remove ISUP body and send INVITE to destination via dispatcher module. And I see that outgoing INVITE has two content-lenth headers. Remote side after received such INVITE sends 400  Bad Contenth-lenth header.       Incoming INVITE:   INVITE sip:9222992040 at 10.66.107.169;transport=sctp;user=phone SIP/2.0 Via: SIP/2.0/SCTP 10.111.18.3:7010;branch=z9hG4bKiz5cvz9x5v5zi4bg84g4vx4zs;X-DispMsg=1408 Route: Call-ID: i44czbcvax4gabcdst4iavdbaisji94s at 10.18.5.64 From: "9269918424";tag=i4z5545v-CC-1027-TRC-613805-OFC-14 To: "9222992040" CSeq: 1 INVITE P-Access-Network-Info: GEN-ACCESS;"area-number=+79262000601" Max-Forwards: 70 Contact: Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,INFO,PRACK,NOTIFY,MESSAGE,REFER,UPDATE P-Asserted-Identity: History-Info: ;index=1 History-Info: ;index=1.1 P-Early-Media: supported Supported: 100rel,timer,histinfo Min-SE: 90 Session-Expires: 1800;refresher=uac Content-Length: 477 Content-Type: multipart/mixed;boundary=ssboundary --ssboundary Content-Length: 205 Content-Type: application/sdp v=0 o=xyz 1154444372 1154444373 IN IP4 10.111.18.3 s=SipCall c=IN IP4 10.249.66.133 t=0 0 m=audio 28644 RTP/AVP 8 116 a=rtpmap:8 PCMA/8000 a=rtpmap:116 telephone-event/8000 a=ptime:20 --ssboundary Content-Length: 100 Content-Type: application/isup;version=itu-t92+   )" )HB()EdRx1Z )EdRxa4}à§=?bñ£9 1À4À?ÀoÀuÀ --ssboundary--     Outgoing INVITE   INVITE sip:9222992040 at 10.66.107.169;transport=sctp;user=phone SIP/2.0 Record-Route: Record-Route: Via: SIP/2.0/UDP 192.168.9.84:5060;branch=z9hG4bKfa6a.b4f80a51.0 Via: SIP/2.0/SCTP 10.111.18.3:7010;branch=z9hG4bKiz5cvz9x5v5zi4bg84g4vx4zs;X-DispMsg=1408 Call-ID: i44czbcvax4gabcdst4iavdbaisji94s at 10.18.5.64 From: "9269918424";tag=i4z5545v-CC-1027-TRC-613805-OFC-14 To: "9222992040" CSeq: 1 INVITE P-Access-Network-Info: GEN-ACCESS;"area-number=+79262000601" Max-Forwards: 69 Contact: Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,INFO,PRACK,NOTIFY,MESSAGE,REFER,UPDATE P-Asserted-Identity: History-Info: ;index=1 History-Info: ;index=1.1 P-Early-Media: supported Supported: 100rel,timer,histinfo Min-SE: 90 Session-Expires: 1800;refresher=uac Content-Length: 205 Content-Length: 205 Content-Type: application/sdp v=0 o=xyz 1154444372 1154444373 IN IP4 10.111.18.3 s=SipCall c=IN IP4 10.249.66.133 t=0 0 m=audio 28644 RTP/AVP 8 116 a=rtpmap:8 PCMA/8000 a=rtpmap:116 telephone-event/8000 a=ptime:20           -- Oleg Podguyko -------------- next part -------------- An HTML attachment was scrubbed... URL: From alain.bieuzent at free.fr Fri Nov 8 03:13:43 2019 From: alain.bieuzent at free.fr (Alain Bieuzent) Date: Fri, 08 Nov 2019 09:13:43 +0100 Subject: [OpenSIPS-Users] Opensips generate two Content-lenth headers In-Reply-To: <1573150510.509115405@f514.i.mail.ru> References: <1573150510.509115405@f514.i.mail.ru> Message-ID: <56ECC46C-F074-4609-AE6B-79F3657F3D10@free.fr> Hi Oleg, Is it normal your outgoing INVITE have twice « Content-Length: 205 » ? Regards De : Users au nom de Oleg Podguyko via Users Répondre à : Oleg Podguyko , OpenSIPS users mailling list Date : jeudi 7 novembre 2019 à 19:16 À : Objet : [OpenSIPS-Users] Opensips generate two Content-lenth headers I’m using opensips as proxy. I got INVITE from one side, do some logics ,remove ISUP body and send INVITE to destination via dispatcher module. And I see that outgoing INVITE has two content-lenth headers. Remote side after received such INVITE sends 400 Bad Contenth-lenth header. Incoming INVITE: INVITE sip:9222992040 at 10.66.107.169;transport=sctp;user=phone SIP/2.0 Via: SIP/2.0/SCTP 10.111.18.3:7010;branch=z9hG4bKiz5cvz9x5v5zi4bg84g4vx4zs;X-DispMsg=1408 Route: Call-ID: i44czbcvax4gabcdst4iavdbaisji94s at 10.18.5.64 From: "9269918424";tag=i4z5545v-CC-1027-TRC-613805-OFC-14 To: "9222992040" CSeq: 1 INVITE P-Access-Network-Info: GEN-ACCESS;"area-number=+79262000601" Max-Forwards: 70 Contact: Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,INFO,PRACK,NOTIFY,MESSAGE,REFER,UPDATE P-Asserted-Identity: History-Info: ;index=1 History-Info: ;index=1.1 P-Early-Media: supported Supported: 100rel,timer,histinfo Min-SE: 90 Session-Expires: 1800;refresher=uac Content-Length: 477 Content-Type: multipart/mixed;boundary=ssboundary --ssboundary Content-Length: 205 Content-Type: application/sdp v=0 o=xyz 1154444372 1154444373 IN IP4 10.111.18.3 s=SipCall c=IN IP4 10.249.66.133 t=0 0 m=audio 28644 RTP/AVP 8 116 a=rtpmap:8 PCMA/8000 a=rtpmap:116 telephone-event/8000 a=ptime:20 --ssboundary Content-Length: 100 Content-Type: application/isup;version=itu-t92+ )" )HB()EdRx1Z )EdRxa4}à§=?bñ£9 1À4À?ÀoÀuÀ --ssboundary-- Outgoing INVITE INVITE sip:9222992040 at 10.66.107.169;transport=sctp;user=phone SIP/2.0 Record-Route: Record-Route: Via: SIP/2.0/UDP 192.168.9.84:5060;branch=z9hG4bKfa6a.b4f80a51.0 Via: SIP/2.0/SCTP 10.111.18.3:7010;branch=z9hG4bKiz5cvz9x5v5zi4bg84g4vx4zs;X-DispMsg=1408 Call-ID: i44czbcvax4gabcdst4iavdbaisji94s at 10.18.5.64 From: "9269918424";tag=i4z5545v-CC-1027-TRC-613805-OFC-14 To: "9222992040" CSeq: 1 INVITE P-Access-Network-Info: GEN-ACCESS;"area-number=+79262000601" Max-Forwards: 69 Contact: Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,INFO,PRACK,NOTIFY,MESSAGE,REFER,UPDATE P-Asserted-Identity: History-Info: ;index=1 History-Info: ;index=1.1 P-Early-Media: supported Supported: 100rel,timer,histinfo Min-SE: 90 Session-Expires: 1800;refresher=uac Content-Length: 205 Content-Length: 205 Content-Type: application/sdp v=0 o=xyz 1154444372 1154444373 IN IP4 10.111.18.3 s=SipCall c=IN IP4 10.249.66.133 t=0 0 m=audio 28644 RTP/AVP 8 116 a=rtpmap:8 PCMA/8000 a=rtpmap:116 telephone-event/8000 a=ptime:20 -- Oleg Podguyko _______________________________________________ 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 farmorg at gmail.com Fri Nov 8 04:23:54 2019 From: farmorg at gmail.com (Mark Farmer) Date: Fri, 8 Nov 2019 09:23:54 +0000 Subject: [OpenSIPS-Users] 3.x LTS Message-ID: Good morning I have some upcoming projects for which I will be using OpenSIPS and I would really like to use features from 3.x but I need to stick to LTS releases. Is there any kind of timeline for the next LTS release please? Best regards Mark. -------------- next part -------------- An HTML attachment was scrubbed... URL: From vladp at opensips.org Fri Nov 8 04:35:04 2019 From: vladp at opensips.org (Vlad Patrascu) Date: Fri, 8 Nov 2019 11:35:04 +0200 Subject: [OpenSIPS-Users] segfault in dialog.so In-Reply-To: References: Message-ID: <675ac805-9407-af68-9fd6-d6d99123d404@opensips.org> Hello, Can you open a ticket on Github and also extract a full backtrace of the crash? Regards, Vlad Patrascu OpenSIPS Developer http://www.opensips-solutions.com On 11/7/19 10:19 AM, Антон Ершов wrote: > Hello friends, > I already wrote about this problem and there was no answer. > http://lists.opensips.org/pipermail/users/2019-October/041771.html > I still want to process call profiles in event_route, but opensips > falls into error at the same time. > > opensips -V > version: opensips 3.0.1 (x86_64/linux) > flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, > Q_MALLOC, F_MALLOC, HP_MALLOC, DBG_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, sigio_rt, select. > git revision: 3d2bd5318 > main.c compiled on 02:53:35 Nov  7 2019 with gcc 4.8.5 > > my event_route > > event_route[E_DLG_STATE_CHANGED] { >     xlog("L_INFO", "[$param(callid)] - new state dialog changed > $param(new_state) old state $param(old_state) \n"); > >     cache_raw_query("redis:group3", "HGET callid > $param(callid)","$avp(reknum)"); >     if ($avp(reknum) != NULL) { >         xlog("L_INFO", "[$param(callid)] - get reknum $avp(reknum) \n"); > > if ( get_dialogs_by_profile("reknumber", $avp(reknum), > $avp(dlg_jsons), $avp(callcount)) ) { >    xlog("L_INFO", "[$param(callid)] - advertising number  $avp(reknum) > has $avp(callcount) other calls \n"); > } else { >    xlog("L_INFO", "[$param(callid)] - this profile does not have > active dialogs \n"); >    $avp(callcount) = 0; > } > > switch($param(new_state)) { >    case 4: > xlog("L_INFO", "[$param(callid)] - write in REDIS \"callcount > $avp(reknum) $avp(callcount) \n"); > if ($avp(reknum) != "" && $avp(callcount) != "") { >    cache_raw_query("redis:group3","HSET callcount $avp(reknum) > $avp(callcount)"); >    cache_raw_query("redis:group3","EXPIRE callconut 360"); >    xlog("L_INFO", "[$param(callid)] - call established. Now on number > $avp(reknum) $avp(callcount) calls \n"); > } > break; >    case 5: > xlog("L_INFO", "[$param(callid)] - write in REDIS \"callcount > $avp(reknum) $avp(callcount) \n"); > if ($avp(reknum) != "" && $avp(callcount) != "") { >    cache_raw_query("redis:group3","HSET callcount $avp(reknum) > $avp(callcount)"); >    cache_raw_query("redis:group3","EXPIRE callcount 360"); >    xlog("L_INFO", "[$param(callid)] - call end. Now on number > $avp(reknum) $avp(callcount) calls \n"); >    cache_raw_query("redis:group3","HDEL callid $param(callid)"); > } > break; > } >     } else { >         xlog("L_INFO", "[$param(callid)] - reknum vareable is NULL"); >         exit; >     } > } > > when we try to get a profile of an already dead dialog opensips falls. > please tell me is it possible to implement what I want or is it worth > looking for another solution? > > > _______________________________________________ > 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 gmaruzz at gmail.com Fri Nov 8 04:43:45 2019 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Fri, 8 Nov 2019 10:43:45 +0100 Subject: [OpenSIPS-Users] 3.x LTS In-Reply-To: References: Message-ID: 3.0 IS LTS (if I understood correctly) On Fri, Nov 8, 2019 at 10:26 AM Mark Farmer wrote: > Good morning > > I have some upcoming projects for which I will be using OpenSIPS and I > would really like to use features from 3.x but I need to stick to LTS > releases. > > Is there any kind of timeline for the next LTS release please? > > Best regards > Mark. > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -- Sincerely, Giovanni Maruzzelli OpenTelecom.IT cell: +39 347 266 56 18 -------------- next part -------------- An HTML attachment was scrubbed... URL: From farmorg at gmail.com Fri Nov 8 05:03:02 2019 From: farmorg at gmail.com (Mark Farmer) Date: Fri, 8 Nov 2019 10:03:02 +0000 Subject: [OpenSIPS-Users] 3.x LTS In-Reply-To: References: Message-ID: According to the versions page only 2.4.6 is currently LTS if I understood correctly :) https://www.opensips.org/About/AvailableVersions On Fri, 8 Nov 2019 at 09:46, Giovanni Maruzzelli wrote: > 3.0 IS LTS (if I understood correctly) > > > > On Fri, Nov 8, 2019 at 10:26 AM Mark Farmer wrote: > >> Good morning >> >> I have some upcoming projects for which I will be using OpenSIPS and I >> would really like to use features from 3.x but I need to stick to LTS >> releases. >> >> Is there any kind of timeline for the next LTS release please? >> >> Best regards >> Mark. >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> > > > -- > Sincerely, > > Giovanni Maruzzelli > OpenTelecom.IT > cell: +39 347 266 56 18 > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -- Mark Farmer farmorg at gmail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From podguiko at mail.ru Fri Nov 8 05:41:15 2019 From: podguiko at mail.ru (=?UTF-8?B?T2xlZyBQb2RndXlrbw==?=) Date: Fri, 08 Nov 2019 13:41:15 +0300 Subject: [OpenSIPS-Users] =?utf-8?q?Users_Digest=2C_Vol_136=2C_Issue_11?= In-Reply-To: References: Message-ID: <1573209675.99458447@f436.i.mail.ru> In my opinion it is not correct. So the remote party sends me 400 Bad Content-lenth in response to my outgoing INVITE. That's why I'm asking this question. How to make this header only one. And why does opensips do this?   >Пятница, 8 ноября 2019, 11:14 +03:00 от users-request at lists.opensips.org: >  >Send Users mailing list submissions to >users at lists.opensips.org > >To subscribe or unsubscribe via the World Wide Web, visit >http://lists.opensips.org/cgi-bin/mailman/listinfo/users >or, via email, send a message with subject or body 'help' to >users-request at lists.opensips.org > >You can reach the person managing the list at >users-owner at lists.opensips.org > >When replying, please edit your Subject line so it is more specific >than "Re: Contents of Users digest..." > > >Today's Topics: > >   1. Opensips generate two Content-lenth headers (Oleg Podguyko) >   2. Re: Opensips generate two Content-lenth headers (Alain Bieuzent) > > >---------------------------------------------------------------------- > >Message: 1 >Date: Thu, 07 Nov 2019 21:15:10 +0300 >From: Oleg Podguyko < podguiko at mail.ru > >To: users at lists.opensips.org >Subject: [OpenSIPS-Users] Opensips generate two Content-lenth headers >Message-ID: < 1573150510.509115405 at f514.i.mail.ru > >Content-Type: text/plain; charset="utf-8" > > >I’m using opensips as proxy. I got INVITE from one side, do some logics ,remove ISUP body and send INVITE to destination via dispatcher module. >And I see that outgoing INVITE has two content-lenth headers. Remote side after received such INVITE sends 400  >Bad Contenth-lenth header. >  >  >  >Incoming INVITE: >  >INVITE sip:9222992040 at 10.66.107.169;transport=sctp;user=phone SIP/2.0 >Via: SIP/2.0/SCTP 10.111.18.3:7010;branch=z9hG4bKiz5cvz9x5v5zi4bg84g4vx4zs;X-DispMsg=1408 >Route: >Call-ID: i44czbcvax4gabcdst4iavdbaisji94s at 10.18.5.64 >From: "9269918424";tag=i4z5545v-CC-1027-TRC-613805-OFC-14 >To: "9222992040" >CSeq: 1 INVITE >P-Access-Network-Info: GEN-ACCESS;"area-number= +79262000601 " >Max-Forwards: 70 >Contact: >Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,INFO,PRACK,NOTIFY,MESSAGE,REFER,UPDATE >P-Asserted-Identity: >History-Info: ;index=1 >History-Info: ;index=1.1 >P-Early-Media: supported >Supported: 100rel,timer,histinfo >Min-SE: 90 >Session-Expires: 1800;refresher=uac >Content-Length: 477 >Content-Type: multipart/mixed;boundary=ssboundary >--ssboundary >Content-Length: 205 >Content-Type: application/sdp >v=0 >o=xyz 1154444372 1154444373 IN IP4 10.111.18.3 >s=SipCall >c=IN IP4 10.249.66.133 >t=0 0 >m=audio 28644 RTP/AVP 8 116 >a=rtpmap:8 PCMA/8000 >a=rtpmap:116 telephone-event/8000 >a=ptime:20 >--ssboundary >Content-Length: 100 >Content-Type: application/isup;version=itu-t92+ >  >)" >)HB()EdRx1Z )EdRxa4}à§=?bñ£9 >1À4À?ÀoÀuÀ >--ssboundary-- >  >  >Outgoing INVITE >  >INVITE sip:9222992040 at 10.66.107.169;transport=sctp;user=phone SIP/2.0 >Record-Route: >Record-Route: >Via: SIP/2.0/UDP 192.168.9.84:5060;branch=z9hG4bKfa6a.b4f80a51.0 >Via: SIP/2.0/SCTP 10.111.18.3:7010;branch=z9hG4bKiz5cvz9x5v5zi4bg84g4vx4zs;X-DispMsg=1408 >Call-ID: i44czbcvax4gabcdst4iavdbaisji94s at 10.18.5.64 >From: "9269918424";tag=i4z5545v-CC-1027-TRC-613805-OFC-14 >To: "9222992040" >CSeq: 1 INVITE >P-Access-Network-Info: GEN-ACCESS;"area-number= +79262000601 " >Max-Forwards: 69 >Contact: >Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,INFO,PRACK,NOTIFY,MESSAGE,REFER,UPDATE >P-Asserted-Identity: >History-Info: ;index=1 >History-Info: ;index=1.1 >P-Early-Media: supported >Supported: 100rel,timer,histinfo >Min-SE: 90 >Session-Expires: 1800;refresher=uac >Content-Length: 205 >Content-Length: 205 >Content-Type: application/sdp >v=0 >o=xyz 1154444372 1154444373 IN IP4 10.111.18.3 >s=SipCall >c=IN IP4 10.249.66.133 >t=0 0 >m=audio 28644 RTP/AVP 8 116 >a=rtpmap:8 PCMA/8000 >a=rtpmap:116 telephone-event/8000 >a=ptime:20 >  >  >  >  >  >-- >Oleg Podguyko >-------------- next part -------------- >An HTML attachment was scrubbed... >URL: < http://lists.opensips.org/pipermail/users/attachments/20191107/99140cb6/attachment-0001.html > > >------------------------------ > >Message: 2 >Date: Fri, 08 Nov 2019 09:13:43 +0100 >From: Alain Bieuzent < alain.bieuzent at free.fr > >To: Oleg Podguyko < podguiko at mail.ru >, OpenSIPS users mailling list >< users at lists.opensips.org > >Subject: Re: [OpenSIPS-Users] Opensips generate two Content-lenth >headers >Message-ID: < 56ECC46C-F074-4609-AE6B-79F3657F3D10 at free.fr > >Content-Type: text/plain; charset="utf-8" > >Hi Oleg, > >  > >Is it normal your outgoing INVITE have twice « Content-Length: 205 » ? > >  > >Regards > >  > >De : Users < users-bounces at lists.opensips.org > au nom de Oleg Podguyko via Users < users at lists.opensips.org > >Répondre à : Oleg Podguyko < podguiko at mail.ru >, OpenSIPS users mailling list < users at lists.opensips.org > >Date : jeudi 7 novembre 2019 à 19:16 >À : < users at lists.opensips.org > >Objet : [OpenSIPS-Users] Opensips generate two Content-lenth headers > >  > >I’m using opensips as proxy. I got INVITE from one side, do some logics ,remove ISUP body and send INVITE to destination via dispatcher module. > >And I see that outgoing INVITE has two content-lenth headers. Remote side after received such INVITE sends 400 > >Bad Contenth-lenth header. > >  > >  > >  > >Incoming INVITE: > >  > >INVITE sip:9222992040 at 10.66.107.169;transport=sctp;user=phone SIP/2.0 >Via: SIP/2.0/SCTP 10.111.18.3:7010;branch=z9hG4bKiz5cvz9x5v5zi4bg84g4vx4zs;X-DispMsg=1408 >Route: >Call-ID: i44czbcvax4gabcdst4iavdbaisji94s at 10.18.5.64 >From: "9269918424";tag=i4z5545v-CC-1027-TRC-613805-OFC-14 >To: "9222992040" >CSeq: 1 INVITE >P-Access-Network-Info: GEN-ACCESS;"area-number= +79262000601 " >Max-Forwards: 70 >Contact: >Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,INFO,PRACK,NOTIFY,MESSAGE,REFER,UPDATE >P-Asserted-Identity: >History-Info: ;index=1 >History-Info: ;index=1.1 >P-Early-Media: supported >Supported: 100rel,timer,histinfo >Min-SE: 90 >Session-Expires: 1800;refresher=uac >Content-Length: 477 >Content-Type: multipart/mixed;boundary=ssboundary > >--ssboundary >Content-Length: 205 >Content-Type: application/sdp > >v=0 >o=xyz 1154444372 1154444373 IN IP4 10.111.18.3 >s=SipCall >c=IN IP4 10.249.66.133 >t=0 0 >m=audio 28644 RTP/AVP 8 116 >a=rtpmap:8 PCMA/8000 >a=rtpmap:116 telephone-event/8000 >a=ptime:20 > >--ssboundary >Content-Length: 100 >Content-Type: application/isup;version=itu-t92+ > >  > >)" >)HB()EdRx1Z >)EdRxa4}à§=?bñ£9 >1À4À?ÀoÀuÀ >--ssboundary-- > >  > >  > >Outgoing INVITE > >  > >INVITE sip:9222992040 at 10.66.107.169;transport=sctp;user=phone SIP/2.0 >Record-Route: >Record-Route: >Via: SIP/2.0/UDP 192.168.9.84:5060;branch=z9hG4bKfa6a.b4f80a51.0 >Via: SIP/2.0/SCTP 10.111.18.3:7010;branch=z9hG4bKiz5cvz9x5v5zi4bg84g4vx4zs;X-DispMsg=1408 >Call-ID: i44czbcvax4gabcdst4iavdbaisji94s at 10.18.5.64 >From: "9269918424";tag=i4z5545v-CC-1027-TRC-613805-OFC-14 >To: "9222992040" >CSeq: 1 INVITE >P-Access-Network-Info: GEN-ACCESS;"area-number= +79262000601 " >Max-Forwards: 69 >Contact: >Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,INFO,PRACK,NOTIFY,MESSAGE,REFER,UPDATE >P-Asserted-Identity: >History-Info: ;index=1 >History-Info: ;index=1.1 >P-Early-Media: supported >Supported: 100rel,timer,histinfo >Min-SE: 90 >Session-Expires: 1800;refresher=uac >Content-Length: 205 >Content-Length: 205 >Content-Type: application/sdp >v=0 >o=xyz 1154444372 1154444373 IN IP4 10.111.18.3 >s=SipCall >c=IN IP4 10.249.66.133 >t=0 0 >m=audio 28644 RTP/AVP 8 116 >a=rtpmap:8 PCMA/8000 >a=rtpmap:116 telephone-event/8000 >a=ptime:20 >  > >  > >  > >  > >  > >-- >Oleg Podguyko > >_______________________________________________ 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: < http://lists.opensips.org/pipermail/users/attachments/20191108/c6dfac88/attachment.html > > >------------------------------ > >Subject: Digest Footer > >_______________________________________________ >Users mailing list >Users at lists.opensips.org >http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > >------------------------------ > >End of Users Digest, Vol 136, Issue 11 >**************************************     -- Олег Подгуйко   -------------- next part -------------- An HTML attachment was scrubbed... URL: From podguiko at mail.ru Fri Nov 8 05:42:03 2019 From: podguiko at mail.ru (=?UTF-8?B?T2xlZyBQb2RndXlrbw==?=) Date: Fri, 08 Nov 2019 13:42:03 +0300 Subject: [OpenSIPS-Users] =?utf-8?q?Opensips_generate_two_Content-lenth_he?= =?utf-8?q?aders?= In-Reply-To: <56ECC46C-F074-4609-AE6B-79F3657F3D10@free.fr> References: <1573150510.509115405@f514.i.mail.ru> <56ECC46C-F074-4609-AE6B-79F3657F3D10@free.fr> Message-ID: <1573209723.605950531@f436.i.mail.ru> In my opinion it is not correct. So the remote party sends me 400 Bad Content-lenth in response to my outgoing INVITE. That's why I'm asking this question. How to make this header only one. And why does opensips do this?   >Пятница, 8 ноября 2019, 11:13 +03:00 от Alain Bieuzent : >  >Hi Oleg, >  >Is it normal your outgoing INVITE have twice «  Content-Length: 205 » ? >  >Regards >  >De : Users < users-bounces at lists.opensips.org > au nom de Oleg Podguyko via Users < users at lists.opensips.org > >Répondre à : Oleg Podguyko < podguiko at mail.ru >, OpenSIPS users mailling list < users at lists.opensips.org > >Date : jeudi 7 novembre 2019 à 19:16 >À : < users at lists.opensips.org > >Objet : [OpenSIPS-Users] Opensips generate two Content-lenth headers >  >I’m using opensips as proxy. I got INVITE from one side, do some logics ,remove ISUP body and send INVITE to destination via dispatcher module. >And I see that outgoing INVITE has two content-lenth headers. Remote side after received such INVITE sends 400  >Bad Contenth-lenth header. >  >  >  >Incoming INVITE: >  >INVITE sip:9222992040 at 10.66.107.169;transport=sctp;user=phone SIP/2.0 >Via: SIP/2.0/SCTP 10.111.18.3:7010;branch=z9hG4bKiz5cvz9x5v5zi4bg84g4vx4zs;X-DispMsg=1408 >Route: >Call-ID: i44czbcvax4gabcdst4iavdbaisji94s at 10.18.5.64 >From: "9269918424";tag=i4z5545v-CC-1027-TRC-613805-OFC-14 >To: "9222992040" >CSeq: 1 INVITE >P-Access-Network-Info: GEN-ACCESS;"area-number= +79262000601 " >Max-Forwards: 70 >Contact: >Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,INFO,PRACK,NOTIFY,MESSAGE,REFER,UPDATE >P-Asserted-Identity: >History-Info: ;index=1 >History-Info: ;index=1.1 >P-Early-Media: supported >Supported: 100rel,timer,histinfo >Min-SE: 90 >Session-Expires: 1800;refresher=uac >Content-Length: 477 >Content-Type: multipart/mixed;boundary=ssboundary >--ssboundary >Content-Length: 205 >Content-Type: application/sdp >v=0 >o=xyz 1154444372 1154444373 IN IP4 10.111.18.3 >s=SipCall >c=IN IP4 10.249.66.133 >t=0 0 >m=audio 28644 RTP/AVP 8 116 >a=rtpmap:8 PCMA/8000 >a=rtpmap:116 telephone-event/8000 >a=ptime:20 >--ssboundary >Content-Length: 100 >Content-Type: application/isup;version=itu-t92+ >  >)" >)HB()EdRx1Z >)EdRxa4}à§=?bñ£9 >1À4À?ÀoÀuÀ >--ssboundary-- >  >  >Outgoing INVITE >  >INVITE sip:9222992040 at 10.66.107.169;transport=sctp;user=phone SIP/2.0 >Record-Route: >Record-Route: >Via: SIP/2.0/UDP 192.168.9.84:5060;branch=z9hG4bKfa6a.b4f80a51.0 >Via: SIP/2.0/SCTP 10.111.18.3:7010;branch=z9hG4bKiz5cvz9x5v5zi4bg84g4vx4zs;X-DispMsg=1408 >Call-ID: i44czbcvax4gabcdst4iavdbaisji94s at 10.18.5.64 >From: "9269918424";tag=i4z5545v-CC-1027-TRC-613805-OFC-14 >To: "9222992040" >CSeq: 1 INVITE >P-Access-Network-Info: GEN-ACCESS;"area-number= +79262000601 " >Max-Forwards: 69 >Contact: >Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,INFO,PRACK,NOTIFY,MESSAGE,REFER,UPDATE >P-Asserted-Identity: >History-Info: ;index=1 >History-Info: ;index=1.1 >P-Early-Media: supported >Supported: 100rel,timer,histinfo >Min-SE: 90 >Session-Expires: 1800;refresher=uac >Content-Length: 205 >Content-Length: 205 >Content-Type: application/sdp >v=0 >o=xyz 1154444372 1154444373 IN IP4 10.111.18.3 >s=SipCall >c=IN IP4 10.249.66.133 >t=0 0 >m=audio 28644 RTP/AVP 8 116 >a=rtpmap:8 PCMA/8000 >a=rtpmap:116 telephone-event/8000 >a=ptime:20 >  >  >  >  >  >-- >Oleg Podguyko >_______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users     -- Олег Подгуйко   -------------- next part -------------- An HTML attachment was scrubbed... URL: From kurgan-rus at inbox.ru Fri Nov 8 05:44:57 2019 From: kurgan-rus at inbox.ru (=?UTF-8?B?QWxleGV5IEthemFudHNldg==?=) Date: Fri, 08 Nov 2019 13:44:57 +0300 Subject: [OpenSIPS-Users] =?utf-8?q?Opensips_generate_two_Content-lenth_he?= =?utf-8?q?aders?= In-Reply-To: <56ECC46C-F074-4609-AE6B-79F3657F3D10@free.fr> References: <1573150510.509115405@f514.i.mail.ru> <56ECC46C-F074-4609-AE6B-79F3657F3D10@free.fr> Message-ID: <1573209897.95814737@f126.i.mail.ru> Hi guys,   as we see, there are also 2 Content-length headers in the incoming to OpenSIPS INVITE.   I haven’t found anything regarding the number of reoccurrences of this header in the SIP message [1].   What about measuring the length with Wireshark, to determine the position of this header with the right size, and then removing the odd  one with OpenSIPS?   [1]  https://tools.ietf.org/html/rfc3261#section-20.14   ----------------------------------------------- BR, Alexey http://alexeyka.zantsev.com/   -------------- next part -------------- An HTML attachment was scrubbed... URL: From podguiko at mail.ru Fri Nov 8 05:52:20 2019 From: podguiko at mail.ru (=?UTF-8?B?T2xlZyBQb2RndXlrbw==?=) Date: Fri, 08 Nov 2019 13:52:20 +0300 Subject: [OpenSIPS-Users] =?utf-8?q?Opensips_generate_two_Content-lenth_he?= =?utf-8?q?aders?= In-Reply-To: References: Message-ID: <1573210340.498849117@f520.i.mail.ru> In my opinion it is not correct. So the remote party sends me 400 Bad Content-lenth in response to my outgoing INVITE. That's why I'm asking this question. How to make this header only one. And why does opensips do this?   >Пятница, 8 ноября 2019, 11:14 +03:00 от users-request at lists.opensips.org: >  >Send Users mailing list submissions to >users at lists.opensips.org > >To subscribe or unsubscribe via the World Wide Web, visit >http://lists.opensips.org/cgi-bin/mailman/listinfo/users >or, via email, send a message with subject or body 'help' to >users-request at lists.opensips.org > >You can reach the person managing the list at >users-owner at lists.opensips.org > >When replying, please edit your Subject line so it is more specific >than "Re: Contents of Users digest..." > > >Today's Topics: > >   1. Opensips generate two Content-lenth headers (Oleg Podguyko) >   2. Re: Opensips generate two Content-lenth headers (Alain Bieuzent) > > >---------------------------------------------------------------------- > >Message: 1 >Date: Thu, 07 Nov 2019 21:15:10 +0300 >From: Oleg Podguyko < podguiko at mail.ru > >To: users at lists.opensips.org >Subject: [OpenSIPS-Users] Opensips generate two Content-lenth headers >Message-ID: < 1573150510.509115405 at f514.i.mail.ru > >Content-Type: text/plain; charset="utf-8" > > >I’m using opensips as proxy. I got INVITE from one side, do some logics ,remove ISUP body and send INVITE to destination via dispatcher module. >And I see that outgoing INVITE has two content-lenth headers. Remote side after received such INVITE sends 400  >Bad Contenth-lenth header. >  >  >  >Incoming INVITE: >  >INVITE sip:9222992040 at 10.66.107.169;transport=sctp;user=phone SIP/2.0 >Via: SIP/2.0/SCTP 10.111.18.3:7010;branch=z9hG4bKiz5cvz9x5v5zi4bg84g4vx4zs;X-DispMsg=1408 >Route: >Call-ID: i44czbcvax4gabcdst4iavdbaisji94s at 10.18.5.64 >From: "9269918424";tag=i4z5545v-CC-1027-TRC-613805-OFC-14 >To: "9222992040" >CSeq: 1 INVITE >P-Access-Network-Info: GEN-ACCESS;"area-number= +79262000601 " >Max-Forwards: 70 >Contact: >Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,INFO,PRACK,NOTIFY,MESSAGE,REFER,UPDATE >P-Asserted-Identity: >History-Info: ;index=1 >History-Info: ;index=1.1 >P-Early-Media: supported >Supported: 100rel,timer,histinfo >Min-SE: 90 >Session-Expires: 1800;refresher=uac >Content-Length: 477 >Content-Type: multipart/mixed;boundary=ssboundary >--ssboundary >Content-Length: 205 >Content-Type: application/sdp >v=0 >o=xyz 1154444372 1154444373 IN IP4 10.111.18.3 >s=SipCall >c=IN IP4 10.249.66.133 >t=0 0 >m=audio 28644 RTP/AVP 8 116 >a=rtpmap:8 PCMA/8000 >a=rtpmap:116 telephone-event/8000 >a=ptime:20 >--ssboundary >Content-Length: 100 >Content-Type: application/isup;version=itu-t92+ >  >)" >)HB()EdRx1Z )EdRxa4}à§=?bñ£9 >1À4À?ÀoÀuÀ >--ssboundary-- >  >  >Outgoing INVITE >  >INVITE sip:9222992040 at 10.66.107.169;transport=sctp;user=phone SIP/2.0 >Record-Route: >Record-Route: >Via: SIP/2.0/UDP 192.168.9.84:5060;branch=z9hG4bKfa6a.b4f80a51.0 >Via: SIP/2.0/SCTP 10.111.18.3:7010;branch=z9hG4bKiz5cvz9x5v5zi4bg84g4vx4zs;X-DispMsg=1408 >Call-ID: i44czbcvax4gabcdst4iavdbaisji94s at 10.18.5.64 >From: "9269918424";tag=i4z5545v-CC-1027-TRC-613805-OFC-14 >To: "9222992040" >CSeq: 1 INVITE >P-Access-Network-Info: GEN-ACCESS;"area-number= +79262000601 " >Max-Forwards: 69 >Contact: >Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,INFO,PRACK,NOTIFY,MESSAGE,REFER,UPDATE >P-Asserted-Identity: >History-Info: ;index=1 >History-Info: ;index=1.1 >P-Early-Media: supported >Supported: 100rel,timer,histinfo >Min-SE: 90 >Session-Expires: 1800;refresher=uac >Content-Length: 205 >Content-Length: 205 >Content-Type: application/sdp >v=0 >o=xyz 1154444372 1154444373 IN IP4 10.111.18.3 >s=SipCall >c=IN IP4 10.249.66.133 >t=0 0 >m=audio 28644 RTP/AVP 8 116 >a=rtpmap:8 PCMA/8000 >a=rtpmap:116 telephone-event/8000 >a=ptime:20 >  >  >  >  >  >-- >Oleg Podguyko >-------------- next part -------------- >An HTML attachment was scrubbed... >URL: < http://lists.opensips.org/pipermail/users/attachments/20191107/99140cb6/attachment-0001.html > > >------------------------------ > >Message: 2 >Date: Fri, 08 Nov 2019 09:13:43 +0100 >From: Alain Bieuzent < alain.bieuzent at free.fr > >To: Oleg Podguyko < podguiko at mail.ru >, OpenSIPS users mailling list >< users at lists.opensips.org > >Subject: Re: [OpenSIPS-Users] Opensips generate two Content-lenth >headers >Message-ID: < 56ECC46C-F074-4609-AE6B-79F3657F3D10 at free.fr > >Content-Type: text/plain; charset="utf-8" > >Hi Oleg, > >  > >Is it normal your outgoing INVITE have twice « Content-Length: 205 » ? > >  > >Regards > >  > >De : Users < users-bounces at lists.opensips.org > au nom de Oleg Podguyko via Users < users at lists.opensips.org > >Répondre à : Oleg Podguyko < podguiko at mail.ru >, OpenSIPS users mailling list < users at lists.opensips.org > >Date : jeudi 7 novembre 2019 à 19:16 >À : < users at lists.opensips.org > >Objet : [OpenSIPS-Users] Opensips generate two Content-lenth headers > >  > >I’m using opensips as proxy. I got INVITE from one side, do some logics ,remove ISUP body and send INVITE to destination via dispatcher module. > >And I see that outgoing INVITE has two content-lenth headers. Remote side after received such INVITE sends 400 > >Bad Contenth-lenth header. > >  > >  > >  > >Incoming INVITE: > >  > >INVITE sip:9222992040 at 10.66.107.169;transport=sctp;user=phone SIP/2.0 >Via: SIP/2.0/SCTP 10.111.18.3:7010;branch=z9hG4bKiz5cvz9x5v5zi4bg84g4vx4zs;X-DispMsg=1408 >Route: >Call-ID: i44czbcvax4gabcdst4iavdbaisji94s at 10.18.5.64 >From: "9269918424";tag=i4z5545v-CC-1027-TRC-613805-OFC-14 >To: "9222992040" >CSeq: 1 INVITE >P-Access-Network-Info: GEN-ACCESS;"area-number= +79262000601 " >Max-Forwards: 70 >Contact: >Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,INFO,PRACK,NOTIFY,MESSAGE,REFER,UPDATE >P-Asserted-Identity: >History-Info: ;index=1 >History-Info: ;index=1.1 >P-Early-Media: supported >Supported: 100rel,timer,histinfo >Min-SE: 90 >Session-Expires: 1800;refresher=uac >Content-Length: 477 >Content-Type: multipart/mixed;boundary=ssboundary > >--ssboundary >Content-Length: 205 >Content-Type: application/sdp > >v=0 >o=xyz 1154444372 1154444373 IN IP4 10.111.18.3 >s=SipCall >c=IN IP4 10.249.66.133 >t=0 0 >m=audio 28644 RTP/AVP 8 116 >a=rtpmap:8 PCMA/8000 >a=rtpmap:116 telephone-event/8000 >a=ptime:20 > >--ssboundary >Content-Length: 100 >Content-Type: application/isup;version=itu-t92+ > >  > >)" >)HB()EdRx1Z >)EdRxa4}à§=?bñ£9 >1À4À?ÀoÀuÀ >--ssboundary-- > >  > >  > >Outgoing INVITE > >  > >INVITE sip:9222992040 at 10.66.107.169;transport=sctp;user=phone SIP/2.0 >Record-Route: >Record-Route: >Via: SIP/2.0/UDP 192.168.9.84:5060;branch=z9hG4bKfa6a.b4f80a51.0 >Via: SIP/2.0/SCTP 10.111.18.3:7010;branch=z9hG4bKiz5cvz9x5v5zi4bg84g4vx4zs;X-DispMsg=1408 >Call-ID: i44czbcvax4gabcdst4iavdbaisji94s at 10.18.5.64 >From: "9269918424";tag=i4z5545v-CC-1027-TRC-613805-OFC-14 >To: "9222992040" >CSeq: 1 INVITE >P-Access-Network-Info: GEN-ACCESS;"area-number= +79262000601 " >Max-Forwards: 69 >Contact: >Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,INFO,PRACK,NOTIFY,MESSAGE,REFER,UPDATE >P-Asserted-Identity: >History-Info: ;index=1 >History-Info: ;index=1.1 >P-Early-Media: supported >Supported: 100rel,timer,histinfo >Min-SE: 90 >Session-Expires: 1800;refresher=uac >Content-Length: 205 >Content-Length: 205 >Content-Type: application/sdp >v=0 >o=xyz 1154444372 1154444373 IN IP4 10.111.18.3 >s=SipCall >c=IN IP4 10.249.66.133 >t=0 0 >m=audio 28644 RTP/AVP 8 116 >a=rtpmap:8 PCMA/8000 >a=rtpmap:116 telephone-event/8000 >a=ptime:20 >  > >  > >  > >  > >  > >-- >Oleg Podguyko > >_______________________________________________ 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: < http://lists.opensips.org/pipermail/users/attachments/20191108/c6dfac88/attachment.html > > >------------------------------ > >Subject: Digest Footer > >_______________________________________________ >Users mailing list >Users at lists.opensips.org >http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > >------------------------------ > >End of Users Digest, Vol 136, Issue 11 >**************************************     -- Олег Подгуйко   -------------- next part -------------- An HTML attachment was scrubbed... URL: From farmorg at gmail.com Fri Nov 8 06:59:49 2019 From: farmorg at gmail.com (Mark Farmer) Date: Fri, 8 Nov 2019 11:59:49 +0000 Subject: [OpenSIPS-Users] Call to local user Message-ID: Hi all I'm suddenly having a problem calling from one registered user to another. Even though both users are locally registered, the INVITE is trying to go to the public IP of OpenSIPS (itself) instead of the IP:rport of the other phone. OpenSIPS is behind NAT as are the phones and I have fix_nated_register() & fix_nated_contact() which seem to be working. Also the contact in the location table is correct. Fairly sure I'm going to end up kicking myself over this one but I just can't see the problem! All suggestions gratefully received. Thanks Mark. -------------- next part -------------- An HTML attachment was scrubbed... URL: From vladp at opensips.org Fri Nov 8 07:00:50 2019 From: vladp at opensips.org (Vlad Patrascu) Date: Fri, 8 Nov 2019 14:00:50 +0200 Subject: [OpenSIPS-Users] 3.x LTS In-Reply-To: References: Message-ID: <38d9b999-1216-5df8-2ca9-ccc5623026d5@opensips.org> Hi Mark, Indeed 3.0 is not LTS, the next LTS version will be 3.1. Regards, Vlad Patrascu OpenSIPS Developer http://www.opensips-solutions.com On 11/8/19 12:03 PM, Mark Farmer wrote: > According to the versions page only 2.4.6 is currently LTS if I > understood correctly :) > > https://www.opensips.org/About/AvailableVersions > > > On Fri, 8 Nov 2019 at 09:46, Giovanni Maruzzelli > wrote: > > 3.0 IS LTS (if I understood correctly) > > > > On Fri, Nov 8, 2019 at 10:26 AM Mark Farmer > wrote: > > Good morning > > I have some upcoming projects for which I will be using > OpenSIPS and I would really like to use features from 3.x but > I need to stick to LTS releases. > > Is there any kind of timeline for the next LTS release please? > > Best regards > Mark. > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > -- > Sincerely, > > Giovanni Maruzzelli > OpenTelecom.IT > cell: +39 347 266 56 18 > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > -- > Mark Farmer > farmorg at gmail.com > > _______________________________________________ > 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 gmaruzz at gmail.com Fri Nov 8 08:33:01 2019 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Fri, 8 Nov 2019 14:33:01 +0100 Subject: [OpenSIPS-Users] 3.x LTS In-Reply-To: References: Message-ID: I see, Mark, you right! -giovanni On Fri, Nov 8, 2019 at 11:05 AM Mark Farmer wrote: > According to the versions page only 2.4.6 is currently LTS if I > understood correctly :) > > https://www.opensips.org/About/AvailableVersions > > > On Fri, 8 Nov 2019 at 09:46, Giovanni Maruzzelli > wrote: > >> 3.0 IS LTS (if I understood correctly) >> >> >> >> On Fri, Nov 8, 2019 at 10:26 AM Mark Farmer wrote: >> >>> Good morning >>> >>> I have some upcoming projects for which I will be using OpenSIPS and I >>> would really like to use features from 3.x but I need to stick to LTS >>> releases. >>> >>> Is there any kind of timeline for the next LTS release please? >>> >>> Best regards >>> Mark. >>> >>> _______________________________________________ >>> Users mailing list >>> Users at lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >> >> >> -- >> Sincerely, >> >> Giovanni Maruzzelli >> OpenTelecom.IT >> cell: +39 347 266 56 18 >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> > > > -- > Mark Farmer > farmorg at gmail.com > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -- Sincerely, Giovanni Maruzzelli OpenTelecom.IT cell: +39 347 266 56 18 -------------- next part -------------- An HTML attachment was scrubbed... URL: From farmorg at gmail.com Fri Nov 8 08:40:07 2019 From: farmorg at gmail.com (Mark Farmer) Date: Fri, 8 Nov 2019 13:40:07 +0000 Subject: [OpenSIPS-Users] 3.x LTS In-Reply-To: References: Message-ID: :) Is there an estimated timeline for 3.1? On Fri, 8 Nov 2019 at 13:36, Giovanni Maruzzelli wrote: > I see, Mark, you right! > > -giovanni > > > > On Fri, Nov 8, 2019 at 11:05 AM Mark Farmer wrote: > >> According to the versions page only 2.4.6 is currently LTS if I >> understood correctly :) >> >> https://www.opensips.org/About/AvailableVersions >> >> >> On Fri, 8 Nov 2019 at 09:46, Giovanni Maruzzelli >> wrote: >> >>> 3.0 IS LTS (if I understood correctly) >>> >>> >>> >>> On Fri, Nov 8, 2019 at 10:26 AM Mark Farmer wrote: >>> >>>> Good morning >>>> >>>> I have some upcoming projects for which I will be using OpenSIPS and I >>>> would really like to use features from 3.x but I need to stick to LTS >>>> releases. >>>> >>>> Is there any kind of timeline for the next LTS release please? >>>> >>>> Best regards >>>> Mark. >>>> >>>> _______________________________________________ >>>> Users mailing list >>>> Users at lists.opensips.org >>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>> >>> >>> >>> -- >>> Sincerely, >>> >>> Giovanni Maruzzelli >>> OpenTelecom.IT >>> cell: +39 347 266 56 18 >>> >>> _______________________________________________ >>> Users mailing list >>> Users at lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >> >> >> -- >> Mark Farmer >> farmorg at gmail.com >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> > > > -- > Sincerely, > > Giovanni Maruzzelli > OpenTelecom.IT > cell: +39 347 266 56 18 > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -- Mark Farmer farmorg at gmail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From farmorg at gmail.com Fri Nov 8 08:42:56 2019 From: farmorg at gmail.com (Mark Farmer) Date: Fri, 8 Nov 2019 13:42:56 +0000 Subject: [OpenSIPS-Users] Call to local user In-Reply-To: References: Message-ID: I'm running with an advertised IP in mhomed mode. It's as though OpenSIPS doesn't realise that it is responsible for the domain (IP) but I've got the domain module loaded and the advertised IP in the table. Very odd. On Fri, 8 Nov 2019 at 11:59, Mark Farmer wrote: > Hi all > > I'm suddenly having a problem calling from one registered user to another. > Even though both users are locally registered, the INVITE is trying to go > to the public IP of OpenSIPS (itself) instead of the IP:rport of the other > phone. > > OpenSIPS is behind NAT as are the phones and I have fix_nated_register() > & fix_nated_contact() which seem to be working. Also the contact in the > location table is correct. > > Fairly sure I'm going to end up kicking myself over this one but I just > can't see the problem! > > All suggestions gratefully received. > > Thanks > Mark. > > -- Mark Farmer farmorg at gmail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From stalker_rulezzz at mail.ru Fri Nov 8 09:01:19 2019 From: stalker_rulezzz at mail.ru (=?UTF-8?B?0JDQvdC00YDQtdC5INCd0LXQstCw0LbQvdC+?=) Date: Fri, 08 Nov 2019 17:01:19 +0300 Subject: [OpenSIPS-Users] =?utf-8?q?Cisco_SIP_phone_7940_do_not_put_parame?= =?utf-8?q?ter_=22did=22_in_R-URI_of_ACK_request?= Message-ID: <1573221679.974456026@f157.i.mail.ru> Hello guys! Sorry for desturbing you, but maybe someone know, how i can fix this problem on Cisco SIP phone 7940: 1) Cisco SIP phone received 200 ok with Contact like: Contact: 2) Cisco SIP phone send ACK with R-URI like: Request-Line: ACK sip:1.1.1.1:5060 SIP/2.0 Parameter "did" lost in R-URI, and ACK can not pass topology_hiding_match() properly. Maybe anybody know how this problem can fixed from phone side? Many thanks! --  Andrei -------------- next part -------------- An HTML attachment was scrubbed... URL: From johan at democon.be Fri Nov 8 10:43:02 2019 From: johan at democon.be (johan) Date: Fri, 8 Nov 2019 16:43:02 +0100 Subject: [OpenSIPS-Users] help requested on debugging. Message-ID: Hi, can somebody explain to me how to debug a module from source code ? Is there somewhere a document describing this ? Br, From razvan at opensips.org Fri Nov 8 11:53:53 2019 From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=) Date: Fri, 8 Nov 2019 18:53:53 +0200 Subject: [OpenSIPS-Users] help requested on debugging. In-Reply-To: References: Message-ID: <050504bd-86b3-a2f1-74ca-288f78839a58@opensips.org> Hi, Johan! Unfortunately there's no specific document for describing how to debug OpenSIPS. We use the gdb tool to debug, which provides plethora of means to track down an issue: set breakpoints, trace variables, etc. I admit it sometimes become a bit hard core, but that depends on what you want to achieve. But unfortunately we've never documented the steps to do. Best regards, Răzvan On 11/8/19 5:43 PM, johan wrote: > Hi, > > can somebody explain to me how to debug a module from source code ? > > Is there somewhere a document describing this ? > > > Br, > > > _______________________________________________ > 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 From razvan at opensips.org Fri Nov 8 11:55:08 2019 From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=) Date: Fri, 8 Nov 2019 18:55:08 +0200 Subject: [OpenSIPS-Users] Recommended Radius client/server for AAA on 2.4.x under Debian 9 In-Reply-To: References: Message-ID: <2fa577b0-47a3-de6a-97c4-a5a52f43122c@opensips.org> Unfortunately I don't have such config file, but radcli seems quite similar to radiusclient-ng. So just change the paths of the config file and it should work. Best regards, Răzvan On 11/6/19 8:16 PM, Jeff Wilkie wrote: > Any build and config docs to support radcli installation with opensips > 2.4 and 3.x by chance? > > Thanks > Jeff > > On Wed, Nov 6, 2019 at 4:56 AM Răzvan Crainea > wrote: > > radcli is the library we're using in our latest packages. > > Best regards, > Răzvan > > On 11/6/19 5:48 AM, qasimakhan at gmail.com > wrote: > > You can use latest version of freeradius it has both client and > server. > > > > Regards, > > Qasim > > > > On Wed, 6 Nov 2019 at 8:27 AM, Jeff Wilkie > > >> wrote: > > > >     Attempting to find current docs since radiusclient-ng is > referenced > >     in several old docs but is no longer available.  Currently, > what is > >     the recommending radius packages to use for accounting > purposes on > >     opensips for CDRs and ACC? > > > >     Thanks > >     Jeff > >     _______________________________________________ > >     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 > > _______________________________________________ > 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 From johan at democon.be Fri Nov 8 11:58:55 2019 From: johan at democon.be (johan) Date: Fri, 8 Nov 2019 17:58:55 +0100 Subject: [OpenSIPS-Users] help requested on debugging. In-Reply-To: <050504bd-86b3-a2f1-74ca-288f78839a58@opensips.org> References: <050504bd-86b3-a2f1-74ca-288f78839a58@opensips.org> Message-ID: Okay, I do assume that in a default make menuconfig build, the debug info is present ? On 08.11.19 17:53, Răzvan Crainea wrote: > Hi, Johan! > > Unfortunately there's no specific document for describing how to debug > OpenSIPS. We use the gdb tool to debug, which provides plethora of > means to track down an issue: set breakpoints, trace variables, etc. I > admit it sometimes become a bit hard core, but that depends on what > you want to achieve. But unfortunately we've never documented the > steps to do. > > Best regards, > Răzvan > > On 11/8/19 5:43 PM, johan wrote: >> Hi, >> >> can somebody explain to me how to debug a module from source code ? >> >> Is there somewhere a document describing this ? >> >> >> Br, >> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users > From bogdan at opensips.org Fri Nov 8 12:04:55 2019 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Fri, 8 Nov 2019 19:04:55 +0200 Subject: [OpenSIPS-Users] $DLG_dir getting NULL in opensips 3.0. In-Reply-To: References: Message-ID: <5afd6cbc-ab4b-36e2-b03e-05b4bc338021@opensips.org> Please open a bug report on the GITHUB tracker. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com OpenSIPS Bootcamp Pre-Registration https://opensips.org/training/OpenSIPS_Bootcamp/ On 11/6/19 8:44 AM, Darpan Patel wrote: > Any solution? > > > On Mon, 4 Nov 2019, 7:55 pm Darpan Patel, > wrote: > > Hello , I have used *$DLG_dir*  for sequential request > in*LOCAL_ROUTE* and also did*loose route* but i am getting > $DLG_dir *null .(opensips 3.0 version) . > * > > ---------------------------------------------------------------------------------------------------------- > *USAGE:* > route[WITHINDIALOG] { >     if (has_totag()) { > # sequential request withing a dialog should > # take the path determined by record-routing >         if (loose_route()) { > # validate the sequential request against dialog >             if ( $DLG_status!=NULL && !validate_dialog() ) { >                 xlog("In-Dialog $rm from $si (callid=$ci) is not > valid according to dialog\n"); >                } >             } > } > local_route { >     xlog("====>> [LOCAL_ROUTE] <<<=="); > >     xlog("L_ERROR", "--- [LOCAL_ROUTE] METHOD: [$rm] DIRECTION: > [*$DLG_dir*] ---"); >     } > ----------------------------------------------------------------------------------------------------------- > *OUTPUT :* > > [LOCAL_ROUTE] METHOD: [BYE] DIRECTION: [*null*] > > please , look into it . and thanks in advance > > > > > > > > _______________________________________________ > 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 Nov 8 12:11:45 2019 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Fri, 8 Nov 2019 19:11:45 +0200 Subject: [OpenSIPS-Users] Cisco SIP phone 7940 do not put parameter "did" in R-URI of ACK request In-Reply-To: <1573221679.974456026@f157.i.mail.ru> References: <1573221679.974456026@f157.i.mail.ru> Message-ID: Hi Andrei, Do you use TH without dialog support right ? IF so, try with dialog support (just create the dialog before doing TH) - this will shorten the "did" param A LOT and maybe Cisco will not discard it. Also, do you obfuscate the Call-ID too ? if not, with dialog support, you may try to use dlg_match_mode [1] with 1 (DID_FALLBACK) to try a SIP matching if the DID param is matching [1] https://opensips.org/html/docs/modules/2.4.x/dialog.html#param_dlg_match_mode Regards Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com OpenSIPS Bootcamp Pre-Registration https://opensips.org/training/OpenSIPS_Bootcamp/ On 11/8/19 4:01 PM, Андрей Неважно via Users wrote: > > Hello guys! Sorry for desturbing you, but maybe someone know, how i > can fix this problem on Cisco SIP phone 7940: > 1) Cisco SIP phone received 200 ok with Contact like: > Contact: > > > 2) Cisco SIP phone send ACK with R-URI like: > Request-Line: ACK sip:1.1.1.1:5060 SIP/2.0 > > Parameter "did" lost in R-URI, and ACK can not pass > topology_hiding_match() properly. > Maybe anybody know how this problem can fixed from phone side? > Many thanks! > > -- > Andrei > > _______________________________________________ > 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 Nov 8 12:13:47 2019 From: callum.guy at x-on.co.uk (Callum Guy) Date: Fri, 8 Nov 2019 17:13:47 +0000 Subject: [OpenSIPS-Users] usrloc contact_id Message-ID: New installation running on OpenSIPs 3.0.1. Quick question, has anything changed with regards to the contact_id field of the location tables? The platform I am using implements the database on MySQL NDBCLUSTER however the autoincrement values are going wild - I'm trying to work out if its my database or if we've moved away from auto-inc? I only ask as the newest ID's are not necessarily the largest! We've only got three subscribers and look: [image: image.png] Bonkers! That table has been truncated and recreated just before these first registrations came in! Here is our table definition in case that's useful: CREATE TABLE `location_th` ( `contact_id` BIGINT(10) UNSIGNED NOT NULL AUTO_INCREMENT, `username` CHAR(64) NOT NULL DEFAULT '', `domain` CHAR(64) DEFAULT NULL, `contact` CHAR(255) NOT NULL DEFAULT '', `received` CHAR(255) DEFAULT NULL, `path` CHAR(255) DEFAULT NULL, `expires` INT(10) UNSIGNED NOT NULL, `q` FLOAT(10,2) NOT NULL DEFAULT '1.00', `callid` CHAR(255) NOT NULL DEFAULT 'Default-Call-ID', `cseq` INT(11) NOT NULL DEFAULT '13', `last_modified` DATETIME NOT NULL DEFAULT '1900-01-01 00:00:01', `flags` INT(11) NOT NULL DEFAULT '0', `cflags` CHAR(255) DEFAULT NULL, `user_agent` CHAR(255) NOT NULL DEFAULT '', `socket` CHAR(64) DEFAULT NULL, `methods` INT(11) DEFAULT NULL, `sip_instance` CHAR(255) DEFAULT NULL, `kv_store` VARCHAR(1000) DEFAULT NULL, `attr` CHAR(255) DEFAULT NULL, PRIMARY KEY (`contact_id`) ) ENGINE=NDBCLUSTER AUTO_INCREMENT=1 DEFAULT CHARSET=latin1; -- *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: -------------- next part -------------- A non-text attachment was scrubbed... Name: image.png Type: image/png Size: 20417 bytes Desc: not available URL: From volga629 at networklab.ca Sun Nov 10 20:34:03 2019 From: volga629 at networklab.ca (volga629 at networklab.ca) Date: Sun, 10 Nov 2019 21:34:03 -0400 Subject: [OpenSIPS-Users] switch statment Message-ID: <1573436043.9670.7@skillsearch.ca> Hello Everyone, Using latest version 3.0.1 can't match switch statement based on $rU. Here are example Debug output Nov 11 02:16:45 dev1-fr /usr/sbin/opensips[30243]: [MESSAGE] Sending to outside provider ~> [3845637810] Nov 11 02:16:45 dev1-fr /usr/sbin/opensips[30243]: SMS_ROUTE: Got 3845637810 looking for correct sms gateway Nov 11 02:16:45 dev1-fr /usr/sbin/opensips[30243]: DBG:core:do_action: switch: running default statement Nov 11 02:16:45 dev1-fr /usr/sbin/opensips[30243]: Unknown destination number Any help thank you volga629 -------------- next part -------------- An HTML attachment was scrubbed... URL: From razvan at opensips.org Mon Nov 11 02:57:42 2019 From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=) Date: Mon, 11 Nov 2019 09:57:42 +0200 Subject: [OpenSIPS-Users] help requested on debugging. In-Reply-To: References: <050504bd-86b3-a2f1-74ca-288f78839a58@opensips.org> Message-ID: Yes, you are correct, you will get the debugging info with the default menuconfig build. On 11/8/19 6:58 PM, johan wrote: > Okay, > > I do assume that in a default make menuconfig build, the debug info is > present ? > > On 08.11.19 17:53, Răzvan Crainea wrote: >> Hi, Johan! >> >> Unfortunately there's no specific document for describing how to debug >> OpenSIPS. We use the gdb tool to debug, which provides plethora of >> means to track down an issue: set breakpoints, trace variables, etc. I >> admit it sometimes become a bit hard core, but that depends on what >> you want to achieve. But unfortunately we've never documented the >> steps to do. >> >> Best regards, >> Răzvan >> >> On 11/8/19 5:43 PM, johan wrote: >>> Hi, >>> >>> can somebody explain to me how to debug a module from source code ? >>> >>> Is there somewhere a document describing this ? >>> >>> >>> Br, >>> >>> >>> _______________________________________________ >>> 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 From razvan at opensips.org Mon Nov 11 03:00:53 2019 From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=) Date: Mon, 11 Nov 2019 10:00:53 +0200 Subject: [OpenSIPS-Users] switch statment In-Reply-To: <1573436043.9670.7@skillsearch.ca> References: <1573436043.9670.7@skillsearch.ca> Message-ID: Hi, Volga! Switch statements don't work with regular expressions, only with string and int comparison. If you need to do regex matching, you need to use multiple ifs. Best regards, Răzvan On 11/11/19 3:34 AM, volga629 via Users wrote: > Hello Everyone, > Using latest version 3.0.1 can't match switch statement based on $rU. > > Here are example > > https://paste.fedoraproject.org/paste/bFBAnYJYly-5PRSmNPKVtg > > > > Debug output > > Nov 11 02:16:45 dev1-fr /usr/sbin/opensips[30243]: [MESSAGE] Sending to > outside provider ~> [3845637810] > Nov 11 02:16:45 dev1-fr /usr/sbin/opensips[30243]: SMS_ROUTE: Got >  3845637810 looking for correct sms gateway > Nov 11 02:16:45 dev1-fr /usr/sbin/opensips[30243]: DBG:core:do_action: > switch: running default statement > Nov 11 02:16:45 dev1-fr /usr/sbin/opensips[30243]: Unknown destination > number > > > Any help thank  you volga629 > > > ** > ** > > _______________________________________________ > 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 From callum.guy at x-on.co.uk Mon Nov 11 04:26:40 2019 From: callum.guy at x-on.co.uk (Callum Guy) Date: Mon, 11 Nov 2019 09:26:40 +0000 Subject: [OpenSIPS-Users] usrloc contact_id In-Reply-To: References: Message-ID: Hi All, I just wanted to feedback on this after a run through of the usrloc source (ucontact.c:518). It appears that unless we are running in sql-only cluster mode (we are using full-sharing) the contact_id generation is handled by the module rather than using the database autoincrement. This makes sense now that I think about it, especially given that we are using write-back so the contacts will share an identifier within the cluster before being persisted. On that basis I guess we would be sensible to remove the auto_increment option from the table definition to reduce the ambiguity. Callum On Fri, 8 Nov 2019 at 17:13, Callum Guy wrote: > New installation running on OpenSIPs 3.0.1. > > Quick question, has anything changed with regards to the contact_id field > of the location tables? The platform I am using implements the database on > MySQL NDBCLUSTER however the autoincrement values are going wild - I'm > trying to work out if its my database or if we've moved away from auto-inc? > I only ask as the newest ID's are not necessarily the largest! > > We've only got three subscribers and look: > > [image: image.png] > > Bonkers! That table has been truncated and recreated just before these > first registrations came in! > > Here is our table definition in case that's useful: > > CREATE TABLE `location_th` ( > `contact_id` BIGINT(10) UNSIGNED NOT NULL AUTO_INCREMENT, > `username` CHAR(64) NOT NULL DEFAULT '', > `domain` CHAR(64) DEFAULT NULL, > `contact` CHAR(255) NOT NULL DEFAULT '', > `received` CHAR(255) DEFAULT NULL, > `path` CHAR(255) DEFAULT NULL, > `expires` INT(10) UNSIGNED NOT NULL, > `q` FLOAT(10,2) NOT NULL DEFAULT '1.00', > `callid` CHAR(255) NOT NULL DEFAULT 'Default-Call-ID', > `cseq` INT(11) NOT NULL DEFAULT '13', > `last_modified` DATETIME NOT NULL DEFAULT '1900-01-01 00:00:01', > `flags` INT(11) NOT NULL DEFAULT '0', > `cflags` CHAR(255) DEFAULT NULL, > `user_agent` CHAR(255) NOT NULL DEFAULT '', > `socket` CHAR(64) DEFAULT NULL, > `methods` INT(11) DEFAULT NULL, > `sip_instance` CHAR(255) DEFAULT NULL, > `kv_store` VARCHAR(1000) DEFAULT NULL, > `attr` CHAR(255) DEFAULT NULL, > PRIMARY KEY (`contact_id`) > ) ENGINE=NDBCLUSTER AUTO_INCREMENT=1 DEFAULT CHARSET=latin1; > -- *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: -------------- next part -------------- A non-text attachment was scrubbed... Name: image.png Type: image/png Size: 20417 bytes Desc: not available URL: From johan at democon.be Mon Nov 11 04:55:37 2019 From: johan at democon.be (Johan De Clercq) Date: Mon, 11 Nov 2019 09:55:37 +0000 Subject: [OpenSIPS-Users] help requested on debugging. In-Reply-To: References: <050504bd-86b3-a2f1-74ca-288f78839a58@opensips.org> , Message-ID: Is there a way to limit tcp workers? Whatever I do, I end up with 8 instances ... Outlook voor iOS downloaden ________________________________ Van: Users namens Răzvan Crainea Verzonden: maandag, november 11, 2019 9:01 AM Aan: users at lists.opensips.org Onderwerp: Re: [OpenSIPS-Users] help requested on debugging. Yes, you are correct, you will get the debugging info with the default menuconfig build. On 11/8/19 6:58 PM, johan wrote: > Okay, > > I do assume that in a default make menuconfig build, the debug info is > present ? > > On 08.11.19 17:53, Răzvan Crainea wrote: >> Hi, Johan! >> >> Unfortunately there's no specific document for describing how to debug >> OpenSIPS. We use the gdb tool to debug, which provides plethora of >> means to track down an issue: set breakpoints, trace variables, etc. I >> admit it sometimes become a bit hard core, but that depends on what >> you want to achieve. But unfortunately we've never documented the >> steps to do. >> >> Best regards, >> Răzvan >> >> On 11/8/19 5:43 PM, johan wrote: >>> Hi, >>> >>> can somebody explain to me how to debug a module from source code ? >>> >>> Is there somewhere a document describing this ? >>> >>> >>> Br, >>> >>> >>> _______________________________________________ >>> 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 _______________________________________________ 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 david.villasmil.work at gmail.com Mon Nov 11 05:13:24 2019 From: david.villasmil.work at gmail.com (David Villasmil) Date: Mon, 11 Nov 2019 10:13:24 +0000 Subject: [OpenSIPS-Users] help requested on debugging. In-Reply-To: References: <050504bd-86b3-a2f1-74ca-288f78839a58@opensips.org> Message-ID: https://www.opensips.org/Documentation/Script-CoreParameters-3-0#udp_workers https://www.opensips.org/Documentation/Script-CoreParameters-3-0#tcp_workers On Mon, 11 Nov 2019 at 09:55, Johan De Clercq wrote: > Is there a way to limit tcp workers? Whatever I do, I end up with 8 > instances ... > > Outlook voor iOS downloaden > > ------------------------------ > *Van:* Users namens Răzvan Crainea < > razvan at opensips.org> > *Verzonden:* maandag, november 11, 2019 9:01 AM > *Aan:* users at lists.opensips.org > *Onderwerp:* Re: [OpenSIPS-Users] help requested on debugging. > > Yes, you are correct, you will get the debugging info with the default > menuconfig build. > > On 11/8/19 6:58 PM, johan wrote: > > Okay, > > > > I do assume that in a default make menuconfig build, the debug info is > > present ? > > > > On 08.11.19 17:53, Răzvan Crainea wrote: > >> Hi, Johan! > >> > >> Unfortunately there's no specific document for describing how to debug > >> OpenSIPS. We use the gdb tool to debug, which provides plethora of > >> means to track down an issue: set breakpoints, trace variables, etc. I > >> admit it sometimes become a bit hard core, but that depends on what > >> you want to achieve. But unfortunately we've never documented the > >> steps to do. > >> > >> Best regards, > >> Răzvan > >> > >> On 11/8/19 5:43 PM, johan wrote: > >>> Hi, > >>> > >>> can somebody explain to me how to debug a module from source code ? > >>> > >>> Is there somewhere a document describing this ? > >>> > >>> > >>> Br, > >>> > >>> > >>> _______________________________________________ > >>> 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 > > _______________________________________________ > 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 > -- Regards, David Villasmil email: david.villasmil.work at gmail.com phone: +34669448337 -------------- next part -------------- An HTML attachment was scrubbed... URL: From johan at democon.be Mon Nov 11 05:52:30 2019 From: johan at democon.be (Johan De Clercq) Date: Mon, 11 Nov 2019 10:52:30 +0000 Subject: [OpenSIPS-Users] help requested on debugging. In-Reply-To: References: <050504bd-86b3-a2f1-74ca-288f78839a58@opensips.org> , Message-ID: Nope, I tried that. I still have eight opensips processes. Outlook voor iOS downloaden ________________________________ Van: Users namens David Villasmil Verzonden: maandag, november 11, 2019 11:17 AM Aan: OpenSIPS users mailling list Onderwerp: Re: [OpenSIPS-Users] help requested on debugging. https://www.opensips.org/Documentation/Script-CoreParameters-3-0#udp_workers https://www.opensips.org/Documentation/Script-CoreParameters-3-0#tcp_workers On Mon, 11 Nov 2019 at 09:55, Johan De Clercq > wrote: Is there a way to limit tcp workers? Whatever I do, I end up with 8 instances ... Outlook voor iOS downloaden ________________________________ Van: Users > namens Răzvan Crainea > Verzonden: maandag, november 11, 2019 9:01 AM Aan: users at lists.opensips.org Onderwerp: Re: [OpenSIPS-Users] help requested on debugging. Yes, you are correct, you will get the debugging info with the default menuconfig build. On 11/8/19 6:58 PM, johan wrote: > Okay, > > I do assume that in a default make menuconfig build, the debug info is > present ? > > On 08.11.19 17:53, Răzvan Crainea wrote: >> Hi, Johan! >> >> Unfortunately there's no specific document for describing how to debug >> OpenSIPS. We use the gdb tool to debug, which provides plethora of >> means to track down an issue: set breakpoints, trace variables, etc. I >> admit it sometimes become a bit hard core, but that depends on what >> you want to achieve. But unfortunately we've never documented the >> steps to do. >> >> Best regards, >> Răzvan >> >> On 11/8/19 5:43 PM, johan wrote: >>> Hi, >>> >>> can somebody explain to me how to debug a module from source code ? >>> >>> Is there somewhere a document describing this ? >>> >>> >>> Br, >>> >>> >>> _______________________________________________ >>> 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 _______________________________________________ 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 -- Regards, David Villasmil email: david.villasmil.work at gmail.com phone: +34669448337 -------------- next part -------------- An HTML attachment was scrubbed... URL: From razvan at opensips.org Mon Nov 11 06:09:10 2019 From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=) Date: Mon, 11 Nov 2019 13:09:10 +0200 Subject: [OpenSIPS-Users] help requested on debugging. In-Reply-To: References: <050504bd-86b3-a2f1-74ca-288f78839a58@opensips.org> Message-ID: <92b1c65f-ff7a-b2a4-72d3-86bddadec1ba@opensips.org> David is right, in order to limit the TCP workers you need to set the tcp_workers parameter. Did you try to set it to a higher value, to make sure that param is properly read? On 11/11/19 12:52 PM, Johan De Clercq wrote: > Nope, I tried that. I still have eight opensips processes. > > Outlook voor iOS downloaden > ------------------------------------------------------------------------ > *Van:* Users namens David Villasmil > > *Verzonden:* maandag, november 11, 2019 11:17 AM > *Aan:* OpenSIPS users mailling list > *Onderwerp:* Re: [OpenSIPS-Users] help requested on debugging. > https://www.opensips.org/Documentation/Script-CoreParameters-3-0#udp_workers > > https://www.opensips.org/Documentation/Script-CoreParameters-3-0#tcp_workers > > On Mon, 11 Nov 2019 at 09:55, Johan De Clercq > wrote: > > Is there a way to limit tcp workers? Whatever I do, I end up with 8 > instances ... > > Outlook voor iOS downloaden > ------------------------------------------------------------------------ > *Van:* Users > namens Răzvan Crainea > > > *Verzonden:* maandag, november 11, 2019 9:01 AM > *Aan:* users at lists.opensips.org > *Onderwerp:* Re: [OpenSIPS-Users] help requested on debugging. > Yes, you are correct, you will get the debugging info with the default > menuconfig build. > > On 11/8/19 6:58 PM, johan wrote: > > Okay, > > > > I do assume that in a default make menuconfig build, the debug > info is > > present ? > > > > On 08.11.19 17:53, Răzvan Crainea wrote: > >> Hi, Johan! > >> > >> Unfortunately there's no specific document for describing how to > debug > >> OpenSIPS. We use the gdb tool to debug, which provides plethora of > >> means to track down an issue: set breakpoints, trace variables, > etc. I > >> admit it sometimes become a bit hard core, but that depends on what > >> you want to achieve. But unfortunately we've never documented the > >> steps to do. > >> > >> Best regards, > >> Răzvan > >> > >> On 11/8/19 5:43 PM, johan wrote: > >>> Hi, > >>> > >>> can somebody explain to me how to debug a module from source code ? > >>> > >>> Is there somewhere a document describing this ? > >>> > >>> > >>> Br, > >>> > >>> > >>> _______________________________________________ > >>> 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 > > _______________________________________________ > 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 > > -- > Regards, > > David Villasmil > email: david.villasmil.work at gmail.com > > phone: +34669448337 > > _______________________________________________ > 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 From jon at hul.me.uk Mon Nov 11 06:44:13 2019 From: jon at hul.me.uk (Jonathan Hulme) Date: Mon, 11 Nov 2019 11:44:13 +0000 Subject: [OpenSIPS-Users] help requested on debugging. In-Reply-To: References: <050504bd-86b3-a2f1-74ca-288f78839a58@opensips.org> Message-ID: <10726e32-7118-eb76-af57-1e1745ccac9f@hul.me.uk> Do you definitely have 8 tcp processes, checked using either: opensips-cli -x mi ps For version 3.x, or opensipsctl fifo ps For version 2.x or are you checking ps x or top on the os? Regards Jonathan On 11/11/2019 10:52, Johan De Clercq wrote: > Nope, I tried that. I still have eight opensips processes. > > Outlook voor iOS downloaden > ------------------------------------------------------------------------ > *Van:* Users namens David Villasmil > > *Verzonden:* maandag, november 11, 2019 11:17 AM > *Aan:* OpenSIPS users mailling list > *Onderwerp:* Re: [OpenSIPS-Users] help requested on debugging. > https://www.opensips.org/Documentation/Script-CoreParameters-3-0#udp_workers > > https://www.opensips.org/Documentation/Script-CoreParameters-3-0#tcp_workers > > On Mon, 11 Nov 2019 at 09:55, Johan De Clercq > wrote: > > Is there a way to limit tcp workers? Whatever I do, I end up with > 8 instances ... > > Outlook voor iOS downloaden > ------------------------------------------------------------------------ > *Van:* Users > namens Răzvan Crainea > > > *Verzonden:* maandag, november 11, 2019 9:01 AM > *Aan:* users at lists.opensips.org > *Onderwerp:* Re: [OpenSIPS-Users] help requested on debugging. > Yes, you are correct, you will get the debugging info with the > default > menuconfig build. > > On 11/8/19 6:58 PM, johan wrote: > > Okay, > > > > I do assume that in a default make menuconfig build, the debug > info is > > present ? > > > > On 08.11.19 17:53, Răzvan Crainea wrote: > >> Hi, Johan! > >> > >> Unfortunately there's no specific document for describing how > to debug > >> OpenSIPS. We use the gdb tool to debug, which provides plethora of > >> means to track down an issue: set breakpoints, trace variables, > etc. I > >> admit it sometimes become a bit hard core, but that depends on > what > >> you want to achieve. But unfortunately we've never documented the > >> steps to do. > >> > >> Best regards, > >> Răzvan > >> > >> On 11/8/19 5:43 PM, johan wrote: > >>> Hi, > >>> > >>> can somebody explain to me how to debug a module from source > code ? > >>> > >>> Is there somewhere a document describing this ? > >>> > >>> > >>> Br, > >>> > >>> > >>> _______________________________________________ > >>> 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 > > _______________________________________________ > 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 > > -- > Regards, > > David Villasmil > email: david.villasmil.work at gmail.com > > phone: +34669448337 > > _______________________________________________ > 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 farmorg at gmail.com Mon Nov 11 10:47:58 2019 From: farmorg at gmail.com (Mark Farmer) Date: Mon, 11 Nov 2019 15:47:58 +0000 Subject: [OpenSIPS-Users] opensips-cli/Domain Issues Message-ID: Hi all I'm still trying to figure out why my OpenSIPS 3.0.1 is not handling calls between 2 registered users.The problem seems to be that the server does not realise that it is responsible for the domain so I tried to use opensips-cli to list the domains that the server is responsible for but I'm getting an issue there too: opensips-cli -d -x mi domain_dump DEBUG: using config file /etc/opensips-cli.cfg DEBUG: Loaded module 'database' DEBUG: Loaded module 'diagnose' DEBUG: Loaded module 'instance' DEBUG: Loaded module 'mi' DEBUG: sent command ':opensips_fifo_reply_12340:{"jsonrpc": "2.0", "id": "16973", "method": "which", "params": []}' DEBUG: Loaded module 'tls' DEBUG: Loaded module 'trace' DEBUG: Loaded module 'trap' DEBUG: Loaded module 'user' DEBUG: running in non-interactive mode '['mi', 'domain_dump']' DEBUG: running command 'domain_dump' '[]' DEBUG: named parameters are used DEBUG: running command 'domain_dump' '{}' DEBUG: sent command ':opensips_fifo_reply_32518:{"jsonrpc": "2.0", "id": "24947", "method": "domain_dump", "params": {}}' ERROR: command 'domain_dump' returned: 500: command not activated I have the domain module loaded: loadmodule "domain.so" modparam("domain", "db_mode", 0) modparam("domain", "db_url", "mysql://opensips:mypassword at localhost/opensips3") My opensips-cli.cfg [default] log_level: DEBUG prompt_name: opensips-cli - treg1.tx prompt_intro: Welcome to OpenSIPS Command Line Interface! prompt_emptyline_repeat_cmd: False history_file: ~/.opensips-cli.history history_file_size: 1000 output_type: pretty-print communication_type: fifo fifo_file: /tmp/opensips_fifo url: http://127.0.0.1:8888/mi database_url=mysql://root:rootpasswd at localhost database_name=opensips3 database_path=/usr/local/share/opensips/ domain=XXX.XXX.XXX.XXX plain_text_passwords=false According to the docs, it should work: https://opensips.org/html/docs/modules/3.0.x/domain.html#mi_domain_dump -------------- next part -------------- An HTML attachment was scrubbed... URL: From razvan at opensips.org Mon Nov 11 10:55:28 2019 From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=) Date: Mon, 11 Nov 2019 17:55:28 +0200 Subject: [OpenSIPS-Users] opensips-cli/Domain Issues In-Reply-To: References: Message-ID: <4c981e51-0603-ba7d-3c35-54cfd0beb221@opensips.org> Hi, Mark! If you're using db_mode = 0, OpenSIPS doesn't have any cache - it will go to the database every time. Therefore OpenSIPS returns 500, because it doesn't have any cache - the domains that he's using are in the database. So you should check you DB, and if the domains are in there, then your issue is somewhere else. Best regards, Razvan On 11/11/19 5:47 PM, Mark Farmer wrote: > Hi all > > I'm still trying to figure out why my OpenSIPS 3.0.1 is not handling > calls between 2 registered users.The problem seems to be that the server > does not realise that it is responsible for the domain so I tried to use > opensips-cli to list the domains that the server is responsible for but > I'm getting an issue there too: > > opensips-cli -d -x mi domain_dump > DEBUG: using config file /etc/opensips-cli.cfg > DEBUG: Loaded module 'database' > DEBUG: Loaded module 'diagnose' > DEBUG: Loaded module 'instance' > DEBUG: Loaded module 'mi' > DEBUG: sent command ':opensips_fifo_reply_12340:{"jsonrpc": "2.0", "id": > "16973", "method": "which", "params": []}' > DEBUG: Loaded module 'tls' > DEBUG: Loaded module 'trace' > DEBUG: Loaded module 'trap' > DEBUG: Loaded module 'user' > DEBUG: running in non-interactive mode '['mi', 'domain_dump']' > DEBUG: running command 'domain_dump' '[]' > DEBUG: named parameters are used > DEBUG: running command 'domain_dump' '{}' > DEBUG: sent command ':opensips_fifo_reply_32518:{"jsonrpc": "2.0", "id": > "24947", "method": "domain_dump", "params": {}}' > ERROR: command 'domain_dump' returned: 500: command not activated > > I have the domain module loaded: > > loadmodule "domain.so" > modparam("domain", "db_mode", 0) > modparam("domain", "db_url", >         "mysql://opensips:mypassword at localhost/opensips3") > > My opensips-cli.cfg > > [default] > log_level: DEBUG > prompt_name: opensips-cli - treg1.tx > prompt_intro: Welcome to OpenSIPS Command Line Interface! > prompt_emptyline_repeat_cmd: False > history_file: ~/.opensips-cli.history > history_file_size: 1000 > output_type: pretty-print > communication_type: fifo > fifo_file: /tmp/opensips_fifo > url: http://127.0.0.1:8888/mi > database_url=mysql://root:rootpasswd at localhost > database_name=opensips3 > database_path=/usr/local/share/opensips/ > domain=XXX.XXX.XXX.XXX > plain_text_passwords=false > > According to the docs, it should work: > https://opensips.org/html/docs/modules/3.0.x/domain.html#mi_domain_dump > > > _______________________________________________ > 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 From farmorg at gmail.com Mon Nov 11 11:04:47 2019 From: farmorg at gmail.com (Mark Farmer) Date: Mon, 11 Nov 2019 16:04:47 +0000 Subject: [OpenSIPS-Users] opensips-cli/Domain Issues In-Reply-To: <4c981e51-0603-ba7d-3c35-54cfd0beb221@opensips.org> References: <4c981e51-0603-ba7d-3c35-54cfd0beb221@opensips.org> Message-ID: Ah, now I understand. Thank you. On Mon, 11 Nov 2019 at 15:57, Răzvan Crainea wrote: > Hi, Mark! > > If you're using db_mode = 0, OpenSIPS doesn't have any cache - it will > go to the database every time. Therefore OpenSIPS returns 500, because > it doesn't have any cache - the domains that he's using are in the > database. So you should check you DB, and if the domains are in there, > then your issue is somewhere else. > > Best regards, > Razvan > > On 11/11/19 5:47 PM, Mark Farmer wrote: > > Hi all > > > > I'm still trying to figure out why my OpenSIPS 3.0.1 is not handling > > calls between 2 registered users.The problem seems to be that the server > > does not realise that it is responsible for the domain so I tried to use > > opensips-cli to list the domains that the server is responsible for but > > I'm getting an issue there too: > > > > opensips-cli -d -x mi domain_dump > > DEBUG: using config file /etc/opensips-cli.cfg > > DEBUG: Loaded module 'database' > > DEBUG: Loaded module 'diagnose' > > DEBUG: Loaded module 'instance' > > DEBUG: Loaded module 'mi' > > DEBUG: sent command ':opensips_fifo_reply_12340:{"jsonrpc": "2.0", "id": > > "16973", "method": "which", "params": []}' > > DEBUG: Loaded module 'tls' > > DEBUG: Loaded module 'trace' > > DEBUG: Loaded module 'trap' > > DEBUG: Loaded module 'user' > > DEBUG: running in non-interactive mode '['mi', 'domain_dump']' > > DEBUG: running command 'domain_dump' '[]' > > DEBUG: named parameters are used > > DEBUG: running command 'domain_dump' '{}' > > DEBUG: sent command ':opensips_fifo_reply_32518:{"jsonrpc": "2.0", "id": > > "24947", "method": "domain_dump", "params": {}}' > > ERROR: command 'domain_dump' returned: 500: command not activated > > > > I have the domain module loaded: > > > > loadmodule "domain.so" > > modparam("domain", "db_mode", 0) > > modparam("domain", "db_url", > > "mysql://opensips:mypassword at localhost/opensips3") > > > > My opensips-cli.cfg > > > > [default] > > log_level: DEBUG > > prompt_name: opensips-cli - treg1.tx > > prompt_intro: Welcome to OpenSIPS Command Line Interface! > > prompt_emptyline_repeat_cmd: False > > history_file: ~/.opensips-cli.history > > history_file_size: 1000 > > output_type: pretty-print > > communication_type: fifo > > fifo_file: /tmp/opensips_fifo > > url: http://127.0.0.1:8888/mi > > database_url=mysql://root:rootpasswd at localhost > > database_name=opensips3 > > database_path=/usr/local/share/opensips/ > > domain=XXX.XXX.XXX.XXX > > plain_text_passwords=false > > > > According to the docs, it should work: > > https://opensips.org/html/docs/modules/3.0.x/domain.html#mi_domain_dump > > > > > > _______________________________________________ > > 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 > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -- Mark Farmer farmorg at gmail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From ahmedmunir007 at gmail.com Mon Nov 11 11:35:42 2019 From: ahmedmunir007 at gmail.com (Ahmed Chohan) Date: Mon, 11 Nov 2019 11:35:42 -0500 Subject: [OpenSIPS-Users] OpenSIPs 2.2.3 SegFault In-Reply-To: References: Message-ID: Thanks Razvan. First I would like to know few things; 1- which is the stable release for v2.2? As I see there are two versions available 2.2.7 & 2.2.8. 2- Is routing configuration for v2.2.3 works with v2.2.7 or v2.2.8? As I was going through the release notes and the source codes for v2.4, v2.2.7 and v2.2.8. For now I may need to stick with v2.2 due to routing configuration in v2.4 as alot of changes need to be made (in term of routing script) and may need sometime for deployment and testing. Please advise. Date: Wed, 6 Nov 2019 11:32:45 +0200 > From: Răzvan Crainea > To: users at lists.opensips.org > Subject: Re: [OpenSIPS-Users] OpenSIPs 2.2.3 SegFault > Message-ID: > Content-Type: text/plain; charset=utf-8; format=flowed > > Hi, Ahmed! > > OpenSIPS 2.2 is quite an old version and is no longer supported for a > while. If you have the possibility, I'd suggest you to migrate to 2.4, > which is a stable LTS - if the issue occurs there, then we are able to > investigate it and track down the problem. > If you can't do that, you're best chance is to upgrade to the latest 2.2 > release, which is 2.2.7. If that still crashes, there's not that much > you can do, as 2.2 is out of support now, and you will need to migrate > to 2.4 anyway. > > Best regards, > Răzvan > > On 11/5/19 9:35 PM, Ahmed Chohan wrote: > > Hi, > > > > I'm currently using OpenSIPs v 2.2.3 in our environment for over a year > > after migrated from version 1.6. As per on 1.6, we were experiencing > > Segfault and advised to move on 2.2.3 (update to date version during > > that time) to resolve this issue but it didn't help at all. > > > > After going through the forums, it stated root cause is memory > > allocation for pkmem as last week, I've upgraded from 16 M to 32 MB in > > config but still experiencing the same issue. See below message below > > from logs; > > > > Nov 4 17:49:50 qorclvsiproxy05 kernel: opensips[7800] general > > protection ip:4f6112 sp:7fffd35bc6b0 error:0 in opensips > > (deleted)[400000+1ed000 > > > > See below extracted coredump. Please advise to find the root cause and > > to resolve this issue. > > > > Core was generated by `/sbin/opensips -P /var/run/opensips/opensips.pid > > -m 4096 -M 32 -u opensips -g o'. > > Program terminated with signal 11, Segmentation fault. > > #0 fm_status (qm=) at mem/f_malloc.c:709 > > 709 > > size+=f->size,f=f->u.nxt_free,i++,j++){ } > > (gdb) bt full > > #0 fm_status (qm=) at mem/f_malloc.c:709 > > f = 0x7f7994902d8000 > > i = 140018 > > j = > > h = > > unused = 0 > > size = 35881011432156432 > > __FUNCTION__ = "fm_status" > > #1 0x000000000043a32a in shm_status (show_status=) > > at mem/shm_mem.h:611 > > No locals. > > #2 cleanup (show_status=) at main.c:339 > > __FUNCTION__ = "cleanup" > > #3 0x000000000043ad2c in handle_sigs () at main.c:520 > > chld = > > chld_status = 139 > > overall_status = 139 > > i = > > do_exit = 1 > > __FUNCTION__ = "handle_sigs" > > #4 0x000000000043ea12 in main_loop (argc=, > > argv=) at main.c:720 > > startup_done = > > chd_rank = 0 > > #5 main (argc=, argv=) at > > main.c:1265 > > cfg_stream = > > c = > > r = > > tmp = 0x7fffd35be90d "" > > tmp_len = > > port = 0 > > proto = 5848501 > > protos_no = > > options = 0x59b568 "f:cCm:M:b:l:n:N:rRvdDFETSVhw:t:u:g:P:G:W:o:" > > ret = -1 > > seed = 666110575 > > rfd = > > __FUNCTION__ = "main" > > > > > > > > -- > > > > Regards, > > > > Ahmed Munir Chohan > > > > > > ------------------------------ > > End of Users Digest, Vol 136, Issue 6 > ************************************* > -- Regards, Ahmed Munir Chohan -------------- next part -------------- An HTML attachment was scrubbed... URL: From razvan at opensips.org Mon Nov 11 11:52:18 2019 From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=) Date: Mon, 11 Nov 2019 18:52:18 +0200 Subject: [OpenSIPS-Users] OpenSIPs 2.2.3 SegFault In-Reply-To: References: Message-ID: <34bef069-ce6a-91b2-6a65-f1e4a4b186a7@opensips.org> Hi, Ahmed! The higher the number, the stabler it is :). So you should go with 2.2.8. In terms of scripting, if you're using the same version (2.2 in this case), we are trying as hard as possible to prevent any semantics changes between minor versions (2.2.3 vs 2.2.8 in your case). So 99% the script should work without any touches. I would, however, run a few tests before putting it blindly in production, just to make sure you're not hitting a but that we haven't thought of. Best regards, Răzvan On 11/11/19 6:35 PM, Ahmed Chohan wrote: > Thanks Razvan. > > First I would like to know few things; > > 1- which is the stable release for v2.2? As I see there are two versions > available 2.2.7 & 2.2.8. > 2- Is routing configuration for v2.2.3 works with v2.2.7 or v2.2.8? > > As I was going through the release notes and the source codes for v2.4, > v2.2.7 and v2.2.8. For now I may need to stick with v2.2 due to routing > configuration in v2.4 as alot of changes need to be made (in term of > routing script) and may need sometime for deployment and testing. > > Please advise. > > > Date: Wed, 6 Nov 2019 11:32:45 +0200 > From: Răzvan Crainea > > To: users at lists.opensips.org > Subject: Re: [OpenSIPS-Users] OpenSIPs 2.2.3 SegFault > Message-ID: > > Content-Type: text/plain; charset=utf-8; format=flowed > > Hi, Ahmed! > > OpenSIPS 2.2 is quite an old version and is no longer supported for a > while. If you have the possibility, I'd suggest you to migrate to 2.4, > which is a stable LTS - if the issue occurs there, then we are able to > investigate it and track down the problem. > If you can't do that, you're best chance is to upgrade to the latest > 2.2 > release, which is 2.2.7. If that still crashes, there's not that much > you can do, as 2.2 is out of support now, and you will need to migrate > to 2.4 anyway. > > Best regards, > Răzvan > > On 11/5/19 9:35 PM, Ahmed Chohan wrote: > > Hi, > > > > I'm currently using OpenSIPs v 2.2.3 in our environment for over > a year > > after migrated from version 1.6. As per on 1.6, we were experiencing > > Segfault and advised to move on 2.2.3 (update to date version during > > that time) to resolve this issue but it didn't help at all. > > > > After going through the forums, it stated root cause is memory > > allocation for pkmem as last week, I've upgraded from 16 M to 32 > MB  in > > config but still experiencing the same issue. See below message > below > > from logs; > > > > Nov  4 17:49:50 qorclvsiproxy05 kernel: opensips[7800] general > > protection ip:4f6112 sp:7fffd35bc6b0 error:0 in opensips > > (deleted)[400000+1ed000 > > > > See below extracted coredump. Please advise to find the root > cause and > > to resolve this issue. > > > > Core was generated by `/sbin/opensips -P > /var/run/opensips/opensips.pid > > -m 4096 -M 32 -u opensips -g o'. > > Program terminated with signal 11, Segmentation fault. > > #0  fm_status (qm=) at mem/f_malloc.c:709 > > 709 > > size+=f->size,f=f->u.nxt_free,i++,j++){ } > > (gdb) bt full > > #0  fm_status (qm=) at mem/f_malloc.c:709 > >          f = 0x7f7994902d8000 > >          i = 140018 > >          j = > >          h = > >          unused = 0 > >          size = 35881011432156432 > >          __FUNCTION__ = "fm_status" > > #1  0x000000000043a32a in shm_status (show_status= optimized out>) > > at mem/shm_mem.h:611 > > No locals. > > #2  cleanup (show_status=) at main.c:339 > >          __FUNCTION__ = "cleanup" > > #3  0x000000000043ad2c in handle_sigs () at main.c:520 > >          chld = > >          chld_status = 139 > >          overall_status = 139 > >          i = > >          do_exit = 1 > >          __FUNCTION__ = "handle_sigs" > > #4  0x000000000043ea12 in main_loop (argc=, > > argv=) at main.c:720 > >          startup_done = > >          chd_rank = 0 > > #5  main (argc=, argv=) at > > main.c:1265 > >          cfg_stream = > >          c = > >          r = > >          tmp = 0x7fffd35be90d "" > >          tmp_len = > >          port = 0 > >          proto = 5848501 > >          protos_no = > >          options = 0x59b568 > "f:cCm:M:b:l:n:N:rRvdDFETSVhw:t:u:g:P:G:W:o:" > >          ret = -1 > >          seed = 666110575 > >          rfd = > >          __FUNCTION__ = "main" > > > > > > > > -- > > > > Regards, > > > > Ahmed Munir Chohan > > > > > > ------------------------------ > > End of Users Digest, Vol 136, Issue 6 > ************************************* > > > > -- > Regards, > > Ahmed Munir Chohan > > > _______________________________________________ > 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 From farmorg at gmail.com Mon Nov 11 12:01:55 2019 From: farmorg at gmail.com (Mark Farmer) Date: Mon, 11 Nov 2019 17:01:55 +0000 Subject: [OpenSIPS-Users] Call to local user In-Reply-To: References: Message-ID: Finally it works! In case this helps someone else, I needed to do a lookup("location"); before calling route(relay); This rewrites the request URI to be the contact saved in the location table. Mark. On Fri, 8 Nov 2019 at 13:42, Mark Farmer wrote: > I'm running with an advertised IP in mhomed mode. It's as though OpenSIPS > doesn't realise that it is responsible for the domain (IP) but I've got the > domain module loaded and the advertised IP in the table. > > Very odd. > > > On Fri, 8 Nov 2019 at 11:59, Mark Farmer wrote: > >> Hi all >> >> I'm suddenly having a problem calling from one registered user to >> another. Even though both users are locally registered, the INVITE is >> trying to go to the public IP of OpenSIPS (itself) instead of the IP:rport >> of the other phone. >> >> OpenSIPS is behind NAT as are the phones and I have fix_nated_register() >> & fix_nated_contact() which seem to be working. Also the contact in the >> location table is correct. >> >> Fairly sure I'm going to end up kicking myself over this one but I just >> can't see the problem! >> >> All suggestions gratefully received. >> >> Thanks >> Mark. >> >> > > -- > Mark Farmer > farmorg at gmail.com > -- Mark Farmer farmorg at gmail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From volga629 at networklab.ca Tue Nov 12 21:14:09 2019 From: volga629 at networklab.ca (volga629 at networklab.ca) Date: Tue, 12 Nov 2019 22:14:09 -0400 Subject: [OpenSIPS-Users] switch statment In-Reply-To: References: <1573436043.9670.7@skillsearch.ca> Message-ID: <1573611249.9239.10@skillsearch.ca> Thank you fixed. volga629 On Mon, Nov 11, 2019 at 10:00, Răzvan Crainea wrote: > Hi, Volga! > > Switch statements don't work with regular expressions, only with > string and int comparison. If you need to do regex matching, you need > to use multiple ifs. > > Best regards, > Răzvan > > On 11/11/19 3:34 AM, volga629 via Users wrote: >> Hello Everyone, >> Using latest version 3.0.1 can't match switch statement based on $rU. >> >> Here are example >> >> >> >> >> >> Debug output >> >> Nov 11 02:16:45 dev1-fr /usr/sbin/opensips[30243]: [MESSAGE] Sending >> to outside provider ~> [3845637810] >> Nov 11 02:16:45 dev1-fr /usr/sbin/opensips[30243]: SMS_ROUTE: Got  >> 3845637810 looking for correct sms gateway >> Nov 11 02:16:45 dev1-fr /usr/sbin/opensips[30243]: >> DBG:core:do_action: switch: running default statement >> Nov 11 02:16:45 dev1-fr /usr/sbin/opensips[30243]: Unknown >> destination number >> >> >> Any help thank you volga629 >> >> >> ** >> ** >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> >> > > -- > Răzvan Crainea > OpenSIPS Core Developer > http://www.opensips-solutions.com > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: From volga629 at networklab.ca Tue Nov 12 21:17:05 2019 From: volga629 at networklab.ca (volga629 at networklab.ca) Date: Tue, 12 Nov 2019 22:17:05 -0400 Subject: [OpenSIPS-Users] replace From Message-ID: <1573611425.9239.11@skillsearch.ca> Hello Everyone, Which way will be correct to replace From with alias. Code # Check valid DID is present alias_db_find("dbaliases", $fu, $avp(from_alias), "r"); xlog("Found DID ~> $avp(from_alias)\r"); if($avp(from_alias)) { uac_replace_from( , $avp(from_alias)); } else { sl_send_reply(404, "Assigned did is required\n\r"); exit; } I see in log Nov 13 01:13:18 dev1-fr /usr/sbin/opensips[5217]: Found DID ~> sip:4384783197 at dev-sip.networklab.tld Nov 13 01:13:18 dev1-fr /usr/sbin/opensips[5217]: DBG:uac:w_replace_from: dsp=(nil) (len=0) , uri=0x7ffde612a6c0 (len=37) Nov 13 01:13:18 dev1-fr /usr/sbin/opensips[5217]: DBG:uac:replace_uri: uri to replace [sip:452390 at dev-sip.networklab.tld], replacement is [sip:4384783197 at dev-sip.networklab.tld], enclosed=0 Nov 13 01:13:18 dev1-fr /usr/sbin/opensips[5217]: DBG:uac:replace_uri: encode is= len=52 Nov 13 01:13:18 dev1-fr /usr/sbin/opensips[5217]: DBG:rr:add_rr_param: adding (;vsf=AAAAAAAGCgcOCHNVXEFtFwwGAx0MBFkBFx8bDhBFGA0GLnRsZA--) But original from still in place Nov 13 01:13:18 dev1-fr /usr/sbin/opensips[5217]: DBG:proto_smpp:send_submit_or_deliver_request: sending submit_sm Nov 13 01:13:18 dev1-fr /usr/sbin/opensips[5217]: DBG:proto_smpp:send_submit_or_deliver_request: *FROM: 452390* Nov 13 01:13:18 dev1-fr /usr/sbin/opensips[5217]: DBG:proto_smpp:send_submit_or_deliver_request: TO: destination number Nov 13 01:13:18 dev1-fr /usr/sbin/opensips[5217]: DBG:proto_smpp:send_submit_or_deliver_request: MESSAGE: Hello type = 65537 Any help thank you. volga629 -------------- next part -------------- An HTML attachment was scrubbed... URL: From johan at democon.be Wed Nov 13 02:57:27 2019 From: johan at democon.be (Johan De Clercq) Date: Wed, 13 Nov 2019 07:57:27 +0000 Subject: [OpenSIPS-Users] replace From In-Reply-To: <1573611425.9239.11@skillsearch.ca> References: <1573611425.9239.11@skillsearch.ca> Message-ID: You need to resubmit the message to opensips. It,s only then that you will see the modification of the from header. Same applies for to. Outlook voor iOS downloaden ________________________________ Van: Users namens volga629 via Users Verzonden: woensdag, november 13, 2019 3:19 AM Aan: Users Onderwerp: [OpenSIPS-Users] replace From Hello Everyone, Which way will be correct to replace From with alias. Code # Check valid DID is present alias_db_find("dbaliases", $fu, $avp(from_alias), "r"); xlog("Found DID ~> $avp(from_alias)\r"); if($avp(from_alias)) { uac_replace_from( , $avp(from_alias)); } else { sl_send_reply(404, "Assigned did is required\n\r"); exit; } I see in log Nov 13 01:13:18 dev1-fr /usr/sbin/opensips[5217]: Found DID ~> sip:4384783197 at dev-sip.networklab.tld Nov 13 01:13:18 dev1-fr /usr/sbin/opensips[5217]: DBG:uac:w_replace_from: dsp=(nil) (len=0) , uri=0x7ffde612a6c0 (len=37) Nov 13 01:13:18 dev1-fr /usr/sbin/opensips[5217]: DBG:uac:replace_uri: uri to replace [sip:452390 at dev-sip.networklab.tld], replacement is [sip:4384783197 at dev-sip.networklab.tld], enclosed=0 Nov 13 01:13:18 dev1-fr /usr/sbin/opensips[5217]: DBG:uac:replace_uri: encode is= len=52 Nov 13 01:13:18 dev1-fr /usr/sbin/opensips[5217]: DBG:rr:add_rr_param: adding (;vsf=AAAAAAAGCgcOCHNVXEFtFwwGAx0MBFkBFx8bDhBFGA0GLnRsZA--) But original from still in place Nov 13 01:13:18 dev1-fr /usr/sbin/opensips[5217]: DBG:proto_smpp:send_submit_or_deliver_request: sending submit_sm Nov 13 01:13:18 dev1-fr /usr/sbin/opensips[5217]: DBG:proto_smpp:send_submit_or_del iver_request: FROM: 452390 Nov 13 01:13:18 dev1-fr /usr/sbin/opensips[5217]: DBG:proto_smpp:send_submit_or_deliver_request: TO: destination number Nov 13 01:13:18 dev1-fr /usr/sbin/opensips[5217]: DBG:proto_smpp:send_submit_or_deliver_request: MESSAGE: Hello type = 65537 Any help thank you. volga629 -------------- next part -------------- An HTML attachment was scrubbed... URL: From razvan at opensips.org Wed Nov 13 03:47:50 2019 From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=) Date: Wed, 13 Nov 2019 10:47:50 +0200 Subject: [OpenSIPS-Users] replace From In-Reply-To: References: <1573611425.9239.11@skillsearch.ca> Message-ID: <6b846029-47f9-1f37-2dbe-d5f154a1379b@opensips.org> Johan is right, there's currently no way the SMPP module sees the changes in the script. But you can open a feature request for this, to add additional fields to the send_smpp_msg() command, to specify the from and to fields. Best regards, Răzvan On 11/13/19 9:57 AM, Johan De Clercq wrote: > You need to resubmit the message to opensips. It,s only then that you > will see the modification of the from header. Same applies for to. > > Outlook voor iOS downloaden > ------------------------------------------------------------------------ > *Van:* Users namens volga629 via > Users > *Verzonden:* woensdag, november 13, 2019 3:19 AM > *Aan:* Users > *Onderwerp:* [OpenSIPS-Users] replace From > Hello Everyone, > Which way will be correct to replace From with alias. > > Code > >         # Check valid DID is present >         alias_db_find("dbaliases", $fu, $avp(from_alias), "r"); >         xlog("Found DID ~> $avp(from_alias)\r"); >         if($avp(from_alias)) { >                 uac_replace_from( , $avp(from_alias)); >         } else { >                 sl_send_reply(404, "Assigned did is required\n\r"); >                 exit; >         } > > I see in log > > Nov 13 01:13:18 dev1-fr /usr/sbin/opensips[5217]: Found DID ~> > sip:4384783197 at dev-sip.networklab.tld > Nov 13 01:13:18 dev1-fr /usr/sbin/opensips[5217]: > DBG:uac:w_replace_from: dsp=(nil) (len=0) , uri=0x7ffde612a6c0 (len=37) > Nov 13 01:13:18 dev1-fr /usr/sbin/opensips[5217]: DBG:uac:replace_uri: > uri to replace [sip:452390 at dev-sip.networklab.tld], replacement is > [sip:4384783197 at dev-sip.networklab.tld], enclosed=0 > Nov 13 01:13:18 dev1-fr /usr/sbin/opensips[5217]: DBG:uac:replace_uri: > encode is= len=52 > Nov 13 01:13:18 dev1-fr /usr/sbin/opensips[5217]: DBG:rr:add_rr_param: > adding (;vsf=AAAAAAAGCgcOCHNVXEFtFwwGAx0MBFkBFx8bDhBFGA0GLnRsZA--) > > But original  from still in place > > Nov 13 01:13:18 dev1-fr /usr/sbin/opensips[5217]: > DBG:proto_smpp:send_submit_or_deliver_request: sending submit_sm > Nov 13 01:13:18 dev1-fr /usr/sbin/opensips[5217]: > DBG:proto_smpp:send_submit_or_del iver_request: *FROM: 452390* > Nov 13 01:13:18 dev1-fr /usr/sbin/opensips[5217]: > DBG:proto_smpp:send_submit_or_deliver_request: TO:  destination number > Nov 13 01:13:18 dev1-fr /usr/sbin/opensips[5217]: > DBG:proto_smpp:send_submit_or_deliver_request: MESSAGE: Hello type = 65537 > > Any help thank you. > > volga629 > > _______________________________________________ > 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 From farmorg at gmail.com Wed Nov 13 06:51:45 2019 From: farmorg at gmail.com (Mark Farmer) Date: Wed, 13 Nov 2019 11:51:45 +0000 Subject: [OpenSIPS-Users] fix_nated_sdp() not taking effect Message-ID: Hi everyone In my failure_route I'm routing to an Asterisk box for voicemail & I need to change the SDP c/o parameters to use the correct internal IP address but using fix_nated_sdp() is not taking effect. if (t_check_status("486|408|603")) { xlog("CUSTOM_LOG: User replied $T_reply_code - Routing to Asterisk Voicemail service."); prefix("VMR_"); rewritehostport("10.150.50.53:2404"); force_send_socket(udp:10.150.50.51); fix_nated_sdp(10,"10.150.50.51"); if (!t_relay()) { send_reply(500,"Internal Error"); } exit; } I get the CUSTOM_LOG entry so I know that the route is executing. Maybe I'm doing something wrong with the flags, I've tried: fix_nated_sdp(2,"10.150.50.51"); fix_nated_sdp(8,"10.150.50.51"); fix_nated_sdp(10,"10.150.50.51"); But when I examine the SDP in the resulting invite, the c/o parameters are never changed. I'm using rtpengine_offer/answer in the initial routing, could it be related to that? I'm using OpenSIPS 3.0.1 Best regards Mark. -------------- next part -------------- An HTML attachment was scrubbed... URL: From volga629 at networklab.ca Wed Nov 13 07:59:33 2019 From: volga629 at networklab.ca (volga629 at networklab.ca) Date: Wed, 13 Nov 2019 08:59:33 -0400 Subject: [OpenSIPS-Users] replace From In-Reply-To: <6b846029-47f9-1f37-2dbe-d5f154a1379b@opensips.org> References: <1573611425.9239.11@skillsearch.ca> <6b846029-47f9-1f37-2dbe-d5f154a1379b@opensips.org> Message-ID: <1573649973.9239.12@skillsearch.ca> Thank you for reply, You referring to ticket volga629 On Wed, Nov 13, 2019 at 10:47, Răzvan Crainea wrote: > Johan is right, there's currently no way the SMPP module sees the > changes in the script. > But you can open a feature request for this, to add additional fields > to the send_smpp_msg() command, to specify the from and to fields. > > Best regards, > Răzvan > > On 11/13/19 9:57 AM, Johan De Clercq wrote: >> You need to resubmit the message to opensips. It,s only then that >> you will see the modification of the from header. Same applies for >> to. >> >> Outlook voor iOS <> downloaden >> ------------------------------------------------------------------------ >> *Van:* Users > > namens volga629 via >> Users > >> *Verzonden:* woensdag, november 13, 2019 3:19 AM >> *Aan:* Users >> *Onderwerp:* [OpenSIPS-Users] replace From >> Hello Everyone, >> Which way will be correct to replace From with alias. >> >> Code >> >> # Check valid DID is present >> alias_db_find("dbaliases", $fu, $avp(from_alias), "r"); >> xlog("Found DID ~> $avp(from_alias)\r"); >> if($avp(from_alias)) { >> uac_replace_from( , $avp(from_alias)); >> } else { >> sl_send_reply(404, "Assigned did is required\n\r"); >> exit; >> } >> >> I see in log >> >> Nov 13 01:13:18 dev1-fr /usr/sbin/opensips[5217]: Found DID ~> >> sip:4384783197 at dev-sip.networklab.tld >> Nov 13 01:13:18 dev1-fr /usr/sbin/opensips[5217]: >> DBG:uac:w_replace_from: dsp=(nil) (len=0) , uri=0x7ffde612a6c0 >> (len=37) >> Nov 13 01:13:18 dev1-fr /usr/sbin/opensips[5217]: >> DBG:uac:replace_uri: uri to replace >> [sip:452390 at dev-sip.networklab.tld], replacement is >> [sip:4384783197 at dev-sip.networklab.tld], enclosed=0 >> Nov 13 01:13:18 dev1-fr /usr/sbin/opensips[5217]: >> DBG:uac:replace_uri: encode >> is= len=52 >> Nov 13 01:13:18 dev1-fr /usr/sbin/opensips[5217]: >> DBG:rr:add_rr_param: adding >> (;vsf=AAAAAAAGCgcOCHNVXEFtFwwGAx0MBFkBFx8bDhBFGA0GLnRsZA--) >> >> But original from still in place >> >> Nov 13 01:13:18 dev1-fr /usr/sbin/opensips[5217]: >> DBG:proto_smpp:send_submit_or_deliver_request: sending submit_sm >> Nov 13 01:13:18 dev1-fr /usr/sbin/opensips[5217]: >> DBG:proto_smpp:send_submit_or_del iver_request: *FROM: 452390* >> Nov 13 01:13:18 dev1-fr /usr/sbin/opensips[5217]: >> DBG:proto_smpp:send_submit_or_deliver_request: TO: destination >> number >> Nov 13 01:13:18 dev1-fr /usr/sbin/opensips[5217]: >> DBG:proto_smpp:send_submit_or_deliver_request: MESSAGE: Hello type >> = 65537 >> >> Any help thank you. >> >> volga629 >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> >> > > -- > Răzvan Crainea > OpenSIPS Core Developer > http://www.opensips-solutions.com > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: From Johan at democon.be Wed Nov 13 08:31:59 2019 From: Johan at democon.be (Johan De Clercq) Date: Wed, 13 Nov 2019 14:31:59 +0100 Subject: [OpenSIPS-Users] https://blog.opensips.org/2017/07/13/hunting-down-complex-opensips-bugs-in-production-environments/ Message-ID: Hi, this post is about struct history support. Is there any chance to share some example code of the case mentioned in the final part ? BR, -------------- next part -------------- An HTML attachment was scrubbed... URL: From liviu at opensips.org Wed Nov 13 08:42:19 2019 From: liviu at opensips.org (Liviu Chircu) Date: Wed, 13 Nov 2019 15:42:19 +0200 Subject: [OpenSIPS-Users] https://blog.opensips.org/2017/07/13/hunting-down-complex-opensips-bugs-in-production-environments/ In-Reply-To: References: Message-ID: <39f0d231-7685-945b-5cd6-3b78996a77f4@opensips.org> Hey Johan, The code you're mentioning actually made it into upstream OpenSIPS.  See the net/net_tcp.c and net/net_tcp_proc.c files for examples on how the "shl_init()", "sh_push()" and "sh_log()" API is being used to log each phase that a TCP connection is going through. And the cool part is that by commenting -DDBG_TCPCON in Makefile.conf, all this troubleshooting code suddenly disappears :) Cheers, Liviu Chircu OpenSIPS Developer http://www.opensips-solutions.com On 13.11.2019 15:31, Johan De Clercq wrote: > Hi, > > this post is about struct history support. > Is there any chance to share some example code of the case mentioned > in the final part ? > > BR, > > _______________________________________________ > 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 podguiko at mail.ru Wed Nov 13 08:47:34 2019 From: podguiko at mail.ru (=?UTF-8?B?T2xlZyBQb2RndXlrbw==?=) Date: Wed, 13 Nov 2019 16:47:34 +0300 Subject: [OpenSIPS-Users] =?utf-8?q?Opensips_generate_two_Content-lenth_he?= =?utf-8?q?aders?= In-Reply-To: References: Message-ID: <1573652854.747465734@f452.i.mail.ru> It looks like a bug of opensips. This problem can occur for those who use opensips as a bridge between SIP-I and SIP.  For example, you received a SIP-I message with body, you will want to delete the ISUP part in the processing process in order to send a regular SIP further. If the incoming body (sdp+isup) is described by a single Content-Length header then everything will be fine. And if the incoming body will have three Content-Length headers (the first Content-Length-shared of all body, the second on ISUP body and the third on SDP body)   Content-Length: 477 Content-Type: multipart/mixed;boundary=ssboundary --ssboundary Content-Length: 205 Content-Type: application/sdp v=0 o=xyz 1154444372 1154444373 IN IP4 10.111.18.3 s=SipCall c=IN IP4 10.249.66.133 t=0 0 m=audio 28644 RTP/AVP 8 116 a=rtpmap:8 PCMA/8000 a=rtpmap:116 telephone-event/8000 a=ptime:20 --ssboundary Content-Length: 100 Content-Type: application/isup;version=itu-t92+ )" )HB()EdRx1Z )EdRxa4}à§=?bñ£9 1À4À?ÀoÀuÀ --ssboundary--  and you delete ISUP body # delete a mime body from incoming INVITE     remove_body_part("application/isup"); your outgoing INVITE will contain two identical Content-Length headers and the remote side will drop your call 400. Content-Length: 205 Content-Length: 205 Content-Type: application/sdp v=0 o=xyz 1154444372 1154444373 IN IP4 10.111.18.3 s=SipCall c=IN IP4 10.249.66.133 t=0 0 m=audio 28644 RTP/AVP 8 116 a=rtpmap:8 PCMA/8000 a=rtpmap:116 telephone-event/8000 a=ptime:20   One Content-Length header is automatically generated by the "remove_body_part" function. And the second is taken from the incoming INVITE probably Then the most interesting. You cannot remove this unnecessary extra Content-Length header. remove_hf("Content-Length");  - doesn't work. I don't know why. Helped here is such a deception. When you get an incoming INVITE (let me remind you in it a body with three Content-Length) Before all manipulations do like this:    ## hack for removing Content-Length    replace_all("Content-Length", "X-"); And then your outgoing invite will have only one Content-Length and the remote party will not send you 400. In General, all will be well. The only nuance you will send is an X-header that is not needed by the remote side))   >Пятница, 8 ноября 2019, 11:14 +03:00 от users-request at lists.opensips.org: >  >Send Users mailing list submissions to >users at lists.opensips.org > >To subscribe or unsubscribe via the World Wide Web, visit >http://lists.opensips.org/cgi-bin/mailman/listinfo/users >or, via email, send a message with subject or body 'help' to >users-request at lists.opensips.org > >You can reach the person managing the list at >users-owner at lists.opensips.org > >When replying, please edit your Subject line so it is more specific >than "Re: Contents of Users digest..." > > >Today's Topics: > >   1. Opensips generate two Content-lenth headers (Oleg Podguyko) >   2. Re: Opensips generate two Content-lenth headers (Alain Bieuzent) > > >---------------------------------------------------------------------- > >Message: 1 >Date: Thu, 07 Nov 2019 21:15:10 +0300 >From: Oleg Podguyko < podguiko at mail.ru > >To: users at lists.opensips.org >Subject: [OpenSIPS-Users] Opensips generate two Content-lenth headers >Message-ID: < 1573150510.509115405 at f514.i.mail.ru > >Content-Type: text/plain; charset="utf-8" > > >I’m using opensips as proxy. I got INVITE from one side, do some logics ,remove ISUP body and send INVITE to destination via dispatcher module. >And I see that outgoing INVITE has two content-lenth headers. Remote side after received such INVITE sends 400  >Bad Contenth-lenth header. >  >  >  >Incoming INVITE: >  >INVITE sip:9222992040 at 10.66.107.169;transport=sctp;user=phone SIP/2.0 >Via: SIP/2.0/SCTP 10.111.18.3:7010;branch=z9hG4bKiz5cvz9x5v5zi4bg84g4vx4zs;X-DispMsg=1408 >Route: >Call-ID: i44czbcvax4gabcdst4iavdbaisji94s at 10.18.5.64 >From: "9269918424";tag=i4z5545v-CC-1027-TRC-613805-OFC-14 >To: "9222992040" >CSeq: 1 INVITE >P-Access-Network-Info: GEN-ACCESS;"area-number= +79262000601 " >Max-Forwards: 70 >Contact: >Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,INFO,PRACK,NOTIFY,MESSAGE,REFER,UPDATE >P-Asserted-Identity: >History-Info: ;index=1 >History-Info: ;index=1.1 >P-Early-Media: supported >Supported: 100rel,timer,histinfo >Min-SE: 90 >Session-Expires: 1800;refresher=uac >Content-Length: 477 >Content-Type: multipart/mixed;boundary=ssboundary >--ssboundary >Content-Length: 205 >Content-Type: application/sdp >v=0 >o=xyz 1154444372 1154444373 IN IP4 10.111.18.3 >s=SipCall >c=IN IP4 10.249.66.133 >t=0 0 >m=audio 28644 RTP/AVP 8 116 >a=rtpmap:8 PCMA/8000 >a=rtpmap:116 telephone-event/8000 >a=ptime:20 >--ssboundary >Content-Length: 100 >Content-Type: application/isup;version=itu-t92+ >  >)" >)HB()EdRx1Z )EdRxa4}à§=?bñ£9 >1À4À?ÀoÀuÀ >--ssboundary-- >  >  >Outgoing INVITE >  >INVITE sip:9222992040 at 10.66.107.169;transport=sctp;user=phone SIP/2.0 >Record-Route: >Record-Route: >Via: SIP/2.0/UDP 192.168.9.84:5060;branch=z9hG4bKfa6a.b4f80a51.0 >Via: SIP/2.0/SCTP 10.111.18.3:7010;branch=z9hG4bKiz5cvz9x5v5zi4bg84g4vx4zs;X-DispMsg=1408 >Call-ID: i44czbcvax4gabcdst4iavdbaisji94s at 10.18.5.64 >From: "9269918424";tag=i4z5545v-CC-1027-TRC-613805-OFC-14 >To: "9222992040" >CSeq: 1 INVITE >P-Access-Network-Info: GEN-ACCESS;"area-number= +79262000601 " >Max-Forwards: 69 >Contact: >Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,INFO,PRACK,NOTIFY,MESSAGE,REFER,UPDATE >P-Asserted-Identity: >History-Info: ;index=1 >History-Info: ;index=1.1 >P-Early-Media: supported >Supported: 100rel,timer,histinfo >Min-SE: 90 >Session-Expires: 1800;refresher=uac >Content-Length: 205 >Content-Length: 205 >Content-Type: application/sdp >v=0 >o=xyz 1154444372 1154444373 IN IP4 10.111.18.3 >s=SipCall >c=IN IP4 10.249.66.133 >t=0 0 >m=audio 28644 RTP/AVP 8 116 >a=rtpmap:8 PCMA/8000 >a=rtpmap:116 telephone-event/8000 >a=ptime:20 >  >  >  >  >  >-- >Oleg Podguyko >-------------- next part -------------- >An HTML attachment was scrubbed... >URL: < http://lists.opensips.org/pipermail/users/attachments/20191107/99140cb6/attachment-0001.html > > >------------------------------ > >Message: 2 >Date: Fri, 08 Nov 2019 09:13:43 +0100 >From: Alain Bieuzent < alain.bieuzent at free.fr > >To: Oleg Podguyko < podguiko at mail.ru >, OpenSIPS users mailling list >< users at lists.opensips.org > >Subject: Re: [OpenSIPS-Users] Opensips generate two Content-lenth >headers >Message-ID: < 56ECC46C-F074-4609-AE6B-79F3657F3D10 at free.fr > >Content-Type: text/plain; charset="utf-8" > >Hi Oleg, > >  > >Is it normal your outgoing INVITE have twice « Content-Length: 205 » ? > >  > >Regards > >  > >De : Users < users-bounces at lists.opensips.org > au nom de Oleg Podguyko via Users < users at lists.opensips.org > >Répondre à : Oleg Podguyko < podguiko at mail.ru >, OpenSIPS users mailling list < users at lists.opensips.org > >Date : jeudi 7 novembre 2019 à 19:16 >À : < users at lists.opensips.org > >Objet : [OpenSIPS-Users] Opensips generate two Content-lenth headers > >  > >I’m using opensips as proxy. I got INVITE from one side, do some logics ,remove ISUP body and send INVITE to destination via dispatcher module. > >And I see that outgoing INVITE has two content-lenth headers. Remote side after received such INVITE sends 400 > >Bad Contenth-lenth header. > >  > >  > >  > >Incoming INVITE: > >  > >INVITE sip:9222992040 at 10.66.107.169;transport=sctp;user=phone SIP/2.0 >Via: SIP/2.0/SCTP 10.111.18.3:7010;branch=z9hG4bKiz5cvz9x5v5zi4bg84g4vx4zs;X-DispMsg=1408 >Route: >Call-ID: i44czbcvax4gabcdst4iavdbaisji94s at 10.18.5.64 >From: "9269918424";tag=i4z5545v-CC-1027-TRC-613805-OFC-14 >To: "9222992040" >CSeq: 1 INVITE >P-Access-Network-Info: GEN-ACCESS;"area-number= +79262000601 " >Max-Forwards: 70 >Contact: >Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,INFO,PRACK,NOTIFY,MESSAGE,REFER,UPDATE >P-Asserted-Identity: >History-Info: ;index=1 >History-Info: ;index=1.1 >P-Early-Media: supported >Supported: 100rel,timer,histinfo >Min-SE: 90 >Session-Expires: 1800;refresher=uac >Content-Length: 477 >Content-Type: multipart/mixed;boundary=ssboundary > >--ssboundary >Content-Length: 205 >Content-Type: application/sdp > >v=0 >o=xyz 1154444372 1154444373 IN IP4 10.111.18.3 >s=SipCall >c=IN IP4 10.249.66.133 >t=0 0 >m=audio 28644 RTP/AVP 8 116 >a=rtpmap:8 PCMA/8000 >a=rtpmap:116 telephone-event/8000 >a=ptime:20 > >--ssboundary >Content-Length: 100 >Content-Type: application/isup;version=itu-t92+ > >  > >)" >)HB()EdRx1Z >)EdRxa4}à§=?bñ£9 >1À4À?ÀoÀuÀ >--ssboundary-- > >  > >  > >Outgoing INVITE > >  > >INVITE sip:9222992040 at 10.66.107.169;transport=sctp;user=phone SIP/2.0 >Record-Route: >Record-Route: >Via: SIP/2.0/UDP 192.168.9.84:5060;branch=z9hG4bKfa6a.b4f80a51.0 >Via: SIP/2.0/SCTP 10.111.18.3:7010;branch=z9hG4bKiz5cvz9x5v5zi4bg84g4vx4zs;X-DispMsg=1408 >Call-ID: i44czbcvax4gabcdst4iavdbaisji94s at 10.18.5.64 >From: "9269918424";tag=i4z5545v-CC-1027-TRC-613805-OFC-14 >To: "9222992040" >CSeq: 1 INVITE >P-Access-Network-Info: GEN-ACCESS;"area-number= +79262000601 " >Max-Forwards: 69 >Contact: >Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,INFO,PRACK,NOTIFY,MESSAGE,REFER,UPDATE >P-Asserted-Identity: >History-Info: ;index=1 >History-Info: ;index=1.1 >P-Early-Media: supported >Supported: 100rel,timer,histinfo >Min-SE: 90 >Session-Expires: 1800;refresher=uac >Content-Length: 205 >Content-Length: 205 >Content-Type: application/sdp >v=0 >o=xyz 1154444372 1154444373 IN IP4 10.111.18.3 >s=SipCall >c=IN IP4 10.249.66.133 >t=0 0 >m=audio 28644 RTP/AVP 8 116 >a=rtpmap:8 PCMA/8000 >a=rtpmap:116 telephone-event/8000 >a=ptime:20 >  > >  > >  > >  > >  > >-- >Oleg Podguyko > >_______________________________________________ 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: < http://lists.opensips.org/pipermail/users/attachments/20191108/c6dfac88/attachment.html > > >------------------------------ > >Subject: Digest Footer > >_______________________________________________ >Users mailing list >Users at lists.opensips.org >http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > >------------------------------ > >End of Users Digest, Vol 136, Issue 11 >**************************************     -- Oleg Podguiko   -------------- next part -------------- An HTML attachment was scrubbed... URL: From ulus_egemen at hotmail.com Wed Nov 13 12:25:32 2019 From: ulus_egemen at hotmail.com (egemen ulus) Date: Wed, 13 Nov 2019 17:25:32 +0000 Subject: [OpenSIPS-Users] 483 too many hops Message-ID: Hi, I've installed Opensips 2.4 on a GCP vm, when I try to register my softphone to server, I am getting "483 too many hops" error. I am aware that there is loop on the server side, it sends the packet itself. But could not fix it. I have private and public IP addresses defined on google cloud vm. Here is the my conf and logs: auto_aliases=no listen=udp:10.138.0.3:5060 # private ip address listen=tcp:10.138.0.3:5060 logs: opensips[2671]: Nov 12 18:56:53 [2727] DBG:core:parse_headers: via found, flags=22 opensips[2671]: Nov 12 18:56:53 [2727] DBG:core:parse_via_param: found param type 234, = <34.83.194.202>; state=6 opensips[2671]: Nov 12 18:56:53 [2727] DBG:core:parse_via_param: found param type 232, = ; state=16 opensips[2671]: Nov 12 18:56:53 [2727] DBG:core:parse_via: end of header reached, state=5 thank you for help from now! -------------- next part -------------- An HTML attachment was scrubbed... URL: From david.villasmil.work at gmail.com Wed Nov 13 13:40:19 2019 From: david.villasmil.work at gmail.com (David Villasmil) Date: Wed, 13 Nov 2019 18:40:19 +0000 Subject: [OpenSIPS-Users] 483 too many hops In-Reply-To: References: Message-ID: You need to set the alias to its addresses, and domains. This is done so that OpenSIPS/kamailio know the message is directed to it. Hope that helps On Wed, 13 Nov 2019 at 17:25, egemen ulus wrote: > Hi, > I've installed Opensips 2.4 on a GCP vm, when I try to register my > softphone to server, I am getting "483 too many hops" error. I am aware > that there is loop on the server side, it sends the packet itself. But > could not fix it. I have private and public IP addresses defined on google > cloud vm. > > Here is the my conf and logs: > > > *auto_aliases=no * > > *listen=udp:10.138.0.3:5060 # private ip address > * > *listen=tcp:10.138.0.3:5060 * > > logs: > > *opensips[2671]: Nov 12 18:56:53 [2727] DBG:core:parse_headers: via found, > flags=22 * > > *opensips[2671]: Nov 12 18:56:53 [2727] DBG:core:parse_via_param: found > param type 234, = <34.83.194.202>; state=6 * > > *opensips[2671]: Nov 12 18:56:53 [2727] DBG:core:parse_via_param: found > param type 232, = ; state=16 * > *opensips[2671]: Nov 12 18:56:53 [2727] DBG:core:parse_via: end of header > reached, state=5* > > *thank you for help from now!* > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -- Regards, David Villasmil email: david.villasmil.work at gmail.com phone: +34669448337 -------------- next part -------------- An HTML attachment was scrubbed... URL: From virendra at cloud-connect.in Thu Nov 14 00:07:01 2019 From: virendra at cloud-connect.in (Virendra Bhati) Date: Thu, 14 Nov 2019 10:37:01 +0530 Subject: [OpenSIPS-Users] Users Digest, Vol 135, Issue 41 In-Reply-To: References: Message-ID: Dear Team, Does anyone face such issue of "crash of OS at the time of MariDB connection"? -- Regards Virendra Bhati Lead- Architecture and Software Solutions On Wed, Oct 30, 2019 at 9:30 PM wrote: > Send Users mailing list submissions to > users at lists.opensips.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > or, via email, send a message with subject or body 'help' to > users-request at lists.opensips.org > > You can reach the person managing the list at > users-owner at lists.opensips.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of Users digest..." > > > Today's Topics: > > 1. Opensips crash while connecting mariadb (Virendra Bhati) > 2. Re: Example of configuration "Full Sharing" Topology with > NoSQL (Liviu Chircu) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Wed, 30 Oct 2019 13:26:44 +0530 > From: Virendra Bhati > To: users at lists.opensips.org > Subject: [OpenSIPS-Users] Opensips crash while connecting mariadb > Message-ID: > dTEeKNT7w9fDZMAqZmORTSJGOT5QK3RZAcLmpiWU2vHPL1A at mail.gmail.com> > Content-Type: text/plain; charset="utf-8" > > Dear Team, > > We are using opensips 3.0.1 with mariadb 10.4.8 . We are facing issue at > starting opensips. As opensips crash with below given message: > > Oct 15 03:27:17 [22849] DBG:core:find_mod_export: found in > module db_mysql [/usr/local/lib64/opensips/modules/] > Oct 15 03:27:17 [22849] DBG:core:db_bind_mod: using db bind api for > db_mysql > Oct 15 03:27:17 [22849] DBG:core:db_do_init: connection 0x7fba0e093510 not > found in pool > Got ERROR: "InnoDB: Unable to lock /var/lib/mysql/ibdata1 error: 11" errno: > 2000 > > I have attached logs and backtrace from core so please provide your > suggestions. Please note credentials given in db_url are working while > using from mysql client > > > -- > Regards > Virendra Bhati > Lead- Architecture and Software Solutions > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: < > http://lists.opensips.org/pipermail/users/attachments/20191030/dfb9da26/attachment-0001.html > > > -------------- next part -------------- > Oct 15 03:27:17 [22849] INFO:core:evi_publish_event: Registered event > > Oct 15 03:27:17 [22849] DBG:core:find_cmd_export_t: found in > module tm [/usr/local/lib64/opensips/modules/] > Oct 15 03:27:17 [22849] DBG:core:find_cmd_export_t: found in > module rr [/usr/local/lib64/opensips/modules/] > Oct 15 03:27:17 [22849] DBG:core:find_mod_export: found in > module db_mysql [/usr/local/lib64/opensips/modules/] > Oct 15 03:27:17 [22849] DBG:core:db_bind_mod: using db bind api for > db_mysql > Oct 15 03:27:17 [22849] DBG:core:db_do_init: connection 0x7fba0e093510 not > found in pool > Got ERROR: "InnoDB: Unable to lock /var/lib/mysql/ibdata1 error: 11" > errno: 2000 > Got ERROR: "InnoDB: Unable to lock /var/lib/mysql/ibdata1 error: 11" > errno: 2000 > Got ERROR: "InnoDB: Unable to lock /var/lib/mysql/ibdata1 error: 11" > errno: 2000 > Got ERROR: "InnoDB: Unable to lock /var/lib/mysql/ibdata1 error: 11" > errno: 2000 > Got ERROR: "InnoDB: Unable to lock /var/lib/mysql/ibdata1 error: 11" > errno: 2000 > Got ERROR: "InnoDB: Unable to lock /var/lib/mysql/ibdata1 error: 11" > errno: 2000 > Got ERROR: "InnoDB: Unable to lock /var/lib/mysql/ibdata1 error: 11" > errno: 2000 > > Got ERROR: "InnoDB: Operating system error number 11 in a file operation." > errno: 2000 > Got ERROR: "InnoDB: Error number 11 means 'Resource temporarily > unavailable'" errno: 2000 > Got ERROR: "InnoDB: Cannot open datafile '/var/lib/mysql/ibdata1'" errno: > 2000 > Got ERROR: "InnoDB: Could not open or create the system tablespace. If you > tried to add new data files to the system tablespace, and it failed here, > you should now edit innodb_data_file_path in my.cnf back to what it was, > and remove the new ibdata files InnoDB created in this failed attempt. > InnoDB only wrote those files full of zeros, but did not yet use them in > any way. But be careful: do not remove old data files which contain your > precious data!" errno: 2000 > Got ERROR: "InnoDB: Plugin initialization aborted with error Cannot open a > file" errno: 2000 > Got ERROR: "Plugin 'InnoDB' init function returned error." errno: 2000 > Got ERROR: "Plugin 'InnoDB' registration as a STORAGE ENGINE failed." > errno: 2000 > Got ERROR: "unknown: Can't lock aria control file > '/var/lib/mysql/aria_log_control' for exclusive use, error: 11. Will retry > for 30 seconds" errno: 2000 > > > > ^C^CGot ERROR: "unknown: Got error 'Could not get an exclusive lock; file > is probably in use by another process' when trying to use aria control file > '/var/lib/mysql/aria_log_control'" errno: 2000 > Got ERROR: "Plugin 'Aria' init function returned error." errno: 2000 > Got ERROR: "Plugin 'Aria' registration as a STORAGE ENGINE failed." errno: > 2000 > Got ERROR: "Unknown/unsupported storage engine: InnoDB" errno: 2000 > DBG:db_mysql:db_mysql_connect: opening connection: > mysql://xxxx:xxxx at localhost/cc_master > CRITICAL:core:sig_usr: segfault in attendant (starter) process! > DBG:core:restore_segv_handler: restoring SIGSEGV handler... > DBG:core:restore_segv_handler: successfully restored system SIGSEGV > handler > ^CSegmentation fault (core dumped) > > -------------- next part -------------- > (gdb) bt full > #0 intern_plugin_lock (lex=0x0, state_mask=14, rc=0x0) > at /usr/src/debug/MariaDB-10.3.18/src_0/sql/sql_plugin.cc:948 > pi = 0x0 > #1 plugin_thdvar_init (thd=0x28af578) at > /usr/src/debug/MariaDB-10.3.18/src_0/sql/sql_plugin.cc:3155 > old_table_plugin = 0x0 > old_tmp_table_plugin = 0x0 > old_enforced_table_plugin = 0x0 > #2 0x00007fc7d58722b1 in THD::init (this=this at entry=0x28af578) > at /usr/src/debug/MariaDB-10.3.18/src_0/sql/sql_class.cc:1177 > No locals. > #3 0x00007fc7d587310a in THD::THD (this=0x28af578, id=, > is_wsrep_applier=) > at /usr/src/debug/MariaDB-10.3.18/src_0/sql/sql_class.cc:798 > tmp = > #4 0x00007fc7d57f2bdc in create_embedded_thd > (client_flag=client_flag at entry=-2143837683) > at /usr/src/debug/MariaDB-10.3.18/src_0/libmysqld/lib_sql.cc:685 > thd = 0x7fff689c20b0 > #5 0x00007fc7d57fa1e4 in mysql_real_connect (mysql=0x7fc7da7190b0, > host=, user=, > passwd=, db=0x7fc7da717620 "cc_master", port=port at entry=0, > unix_socket=unix_socket at entry=0x0, > client_flag=2151129613, client_flag at entry=2147549184) > at /usr/src/debug/MariaDB-10.3.18/src_0/libmysqld/libmysqld.c:179 > name_buff = > "\200\314|\326\307\177\000\000\060!\234h\377\177\000\000 > !\234h\377\177\000\000\060!\234h\377\177\000\000\200\314|\326\307\177\000\000\001\000\000\000\000\000\000\000\340r\242\000\000\000\000\000p\251\215\000\000\000\000\000\340r\242\000\000\000\000\000\270\"\234h\377\177\000\000 > vq\332\307\177\000\000\362\344s\333\307\177\000\000\000\000\000\000\000\000\000\000\060\000\000\000\060\000\000\000\060\"\234h\377\177\000\000P!\234h\377\177", > '\000' , > "\224*\207\333\307\177\000\000\005+\207\333\307\177\000\000\300\061\253\333\307\177\000\000\200\363\252\333\307\177\000\000c\226u\333\307\177\000\000 > vq\332\307\177\000\000"... > #6 0x00007fc7d70e8a63 in db_mysql_connect (ptr=ptr at entry=0x7fc7da717668) > at my_con.c:105 > reconnect = 0 '\000' > ---Type to continue, or q to quit--- > __FUNCTION__ = "db_mysql_connect" > #7 0x00007fc7d70e94ff in db_mysql_new_connection (id=0x7fc7da717510) at > my_con.c:165 > ptr = 0x7fc7da717668 > __FUNCTION__ = "db_mysql_new_connection" > #8 0x00000000005ec4ba in db_do_init (url=, > new_connection=0x7fc7d70e9417 ) at db/db.c:338 > id = 0x7fc7da717510 > con = > res = 0x7fc7da717468 > con_size = > __FUNCTION__ = "db_do_init" > #9 0x00007fc7d01044cd in dlg_connect_db (db_url=db_url at entry=0x7fc7d0340610 > ) at dlg_db_handler.c:135 > __FUNCTION__ = "dlg_connect_db" > #10 0x00007fc7d0104511 in init_dlg_db (db_url=db_url at entry=0x7fc7d0340610 > , > dlg_hash_size=, db_update_period=60) at > dlg_db_handler.c:176 > __FUNCTION__ = "init_dlg_db" > #11 0x00007fc7d0100c27 in mod_init () at dialog.c:900 > n = > __FUNCTION__ = "mod_init" > #12 0x000000000050bfa7 in init_mod (m=0x7fc7da6face0, > skip_others=skip_others at entry=0) at sr_module.c:697 > dep = > __FUNCTION__ = "init_mod" > #13 0x000000000050c028 in init_mod (m=0x7fc7da6fb970, > skip_others=skip_others at entry=0) at sr_module.c:678 > dep = > __FUNCTION__ = "init_mod" > #14 0x000000000050c028 in init_mod (m=0x7fc7da6fbd28, > skip_others=skip_others at entry=0) at sr_module.c:678 > dep = > __FUNCTION__ = "init_mod" > ---Type to continue, or q to quit--- > #15 0x000000000050c028 in init_mod (m=0x7fc7da6fbeb8, > skip_others=skip_others at entry=0) at sr_module.c:678 > dep = > __FUNCTION__ = "init_mod" > #16 0x000000000050c028 in init_mod (m=0x7fc7da6fc0c0, > skip_others=skip_others at entry=0) at sr_module.c:678 > dep = > __FUNCTION__ = "init_mod" > #17 0x000000000050c028 in init_mod (m=0x7fc7da6fc340, > skip_others=skip_others at entry=0) at sr_module.c:678 > dep = > __FUNCTION__ = "init_mod" > #18 0x000000000050c028 in init_mod (m=0x7fc7da6fc4d8, > skip_others=skip_others at entry=0) at sr_module.c:678 > dep = > __FUNCTION__ = "init_mod" > #19 0x000000000050c028 in init_mod (m=0x7fc7da6fd488, > skip_others=skip_others at entry=0) at sr_module.c:678 > dep = > __FUNCTION__ = "init_mod" > #20 0x000000000050c028 in init_mod (m=0x7fc7da6fde20, > skip_others=skip_others at entry=0) at sr_module.c:678 > dep = > __FUNCTION__ = "init_mod" > #21 0x000000000050f45d in init_modules () at sr_module.c:759 > currentMod = 0x0 > ret = > __FUNCTION__ = "init_modules" > #22 0x0000000000420351 in main (argc=, argv= out>) at main.c:1421 > c = > r = > tmp = 0x1
> tmp_len = > port = > ---Type to continue, or q to quit--- > proto = > protos_no = > options = 0x67e9f8 > "f:cCm:M:b:l:n:N:rRvdDFEVhw:t:u:g:p:P:G:W:o:a:k:s:" > ret = -1 > seed = 961253727 > rfd = > __FUNCTION__ = "main" > > > ------------------------------ > > Message: 2 > Date: Wed, 30 Oct 2019 11:34:19 -0400 > From: Liviu Chircu > To: OpenSIPS users mailling list > Subject: Re: [OpenSIPS-Users] Example of configuration "Full Sharing" > Topology with NoSQL > Message-ID: <33579d6f-a9ef-6e4d-5ab9-535450c25714 at opensips.org> > Content-Type: text/plain; charset=utf-8; format=flowed > > Can you rather tell us more about the problem you are trying to solve, > rather than > inquiring about various (random) solutions? For example: > > * will your platform have 1 POP or multiple POPs? > * do you want high availability for the user location nodes? > > Regards, > > -- > Liviu Chircu > OpenSIPS Developer > http://www.opensips-solutions.com > > On 10/28/19 10:27 AM, Social Boh wrote: > > > > Hello, > > > > if i don't want use a SBC how can I known is a node of cluster is up > > or down? which method do you advise? > > > > Thank you > > > > > > ------------------------------ > > Subject: Digest Footer > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > ------------------------------ > > End of Users Digest, Vol 135, Issue 41 > ************************************** > -------------- next part -------------- An HTML attachment was scrubbed... URL: From ulus_egemen at hotmail.com Wed Nov 13 08:43:19 2019 From: ulus_egemen at hotmail.com (egemen ulus) Date: Wed, 13 Nov 2019 13:43:19 +0000 Subject: [OpenSIPS-Users] 483 Too Many Hops on GCP Message-ID: Hi, I've installed Opensips 2.4 on a GCP vm, when I try to register my softphone to server, I am getting "483 too many hops" error. I am aware that there is loop on the server side, it sends the packet itself. But could not fix it. I have private and public IP addresses defined on google cloud vm. Here is the my conf and logs: auto_aliases=no listen=udp:10.138.0.3:5060 # private ip address listen=tcp:10.138.0.3:5060 logs: opensips[2671]: Nov 12 18:56:53 [2727] DBG:core:parse_headers: via found, flags=22 opensips[2671]: Nov 12 18:56:53 [2727] DBG:core:parse_via_param: found param type 234, = <34.83.194.202>; state=6 opensips[2671]: Nov 12 18:56:53 [2727] DBG:core:parse_via_param: found param type 232, = ; state=16 opensips[2671]: Nov 12 18:56:53 [2727] DBG:core:parse_via: end of header reached, state=5 thank you for help from now! Egemen -------------- next part -------------- An HTML attachment was scrubbed... URL: From Ben.Newlin at genesys.com Thu Nov 14 15:44:00 2019 From: Ben.Newlin at genesys.com (Ben Newlin) Date: Thu, 14 Nov 2019 20:44:00 +0000 Subject: [OpenSIPS-Users] Rate Limit Module Implementation Message-ID: <59054909-5B9B-4D25-B067-7ACFF6E73D81@genesys.com> Hello, We are looking to implement some rate limiting using the module, but I have a few questions and wanted to see if anyone has run into the same issues or has working experience with the module to answer them. The module provides $rl_count and also a counter value in the output of the rl_list command, but both of these values only reflect the number of rl_check calls that were made in the time window. In the case of TAILDROP, RED, and SBT algorithms, the count is directly related to the limit and drop_rate. But for the NETWORK and FEEDBACK algorithms, the counter is essentially meaningless. What we are looking for is a way to track the values these modules are using to make drop decisions, so that we can set our limit value appropriately. But unless I am missing something, there is no way to access these values for either NETWORK or FEEDBACK. Is that correct? Ben Newlin -------------- next part -------------- An HTML attachment was scrubbed... URL: From david.villasmil.work at gmail.com Thu Nov 14 16:05:17 2019 From: david.villasmil.work at gmail.com (David Villasmil) Date: Thu, 14 Nov 2019 21:05:17 +0000 Subject: [OpenSIPS-Users] 483 Too Many Hops on GCP In-Reply-To: References: Message-ID: Didn't i respond on this already? You only have no aliases, you need to add an alias. i.e.: If the PUBLIC IP address is 1.2.3.4 you need to add an alias: alias=1.2.3.4 Otherwise, there is no way for the proxy to know the packet is directed to it. https://www.opensips.org/Documentation/Script-CoreParameters-3-0#toc5 Regards, David Villasmil email: david.villasmil.work at gmail.com phone: +34669448337 On Thu, Nov 14, 2019 at 8:27 PM egemen ulus wrote: > Hi, > I've installed Opensips 2.4 on a GCP vm, when I try to register my > softphone to server, I am getting "483 too many hops" error. I am aware > that there is loop on the server side, it sends the packet itself. But > could not fix it. I have private and public IP addresses defined on google > cloud vm. > > Here is the my conf and logs: > > > *auto_aliases=no * > > *listen=udp:10.138.0.3:5060 # private ip address > * > *listen=tcp:10.138.0.3:5060 * > > logs: > > *opensips[2671]: Nov 12 18:56:53 [2727] DBG:core:parse_headers: via found, > flags=22 * > > *opensips[2671]: Nov 12 18:56:53 [2727] DBG:core:parse_via_param: found > param type 234, = <34.83.194.202>; state=6 * > > *opensips[2671]: Nov 12 18:56:53 [2727] DBG:core:parse_via_param: found > param type 232, = ; state=16 * > *opensips[2671]: Nov 12 18:56:53 [2727] DBG:core:parse_via: end of header > reached, state=5* > > *thank you for help from now!* > *Egemen* > _______________________________________________ > 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 farmorg at gmail.com Fri Nov 15 09:06:42 2019 From: farmorg at gmail.com (Mark Farmer) Date: Fri, 15 Nov 2019 14:06:42 +0000 Subject: [OpenSIPS-Users] Dual NIC - How to handle media on failure? Message-ID: Good afternoon I've read the document on bridging interfaces [1], reviewed the redirection part of the residential script & I think I understand the mhomed/manual methods etc but I'm really struggling with this problem. My OpenSIPS server has a NAT'ed public IP and an internal IP on separate interfaces. Phones register via the public IP and I have an Asterisk server connected via the internal IP for voicemail etc: I'm using topology_hiding/match() for the initial call & SIP/RTP for a working call from one phone to another goes: PhoneA<-->NATedPublicIP<-->PhoneB If a call results in a 603 for example, the SIP is routed to Asterisk OK using the internal IP's but the media connection on the OpenSIPS end is still it's public IP so audio fails. How do I have OpenSIPS re-route the media so that in order to reach Asterisk it flows via the internal IP like this? My media server is rtpengine. Phone<-->NATedPublicIP<-->InternalIP<-->AsteriskInternal [1] https://blog.opensips.org/2018/09/04/sip-bridging-over-multiple-interfaces/ Many thanks! Mark -------------- next part -------------- An HTML attachment was scrubbed... URL: From volga629 at networklab.ca Fri Nov 15 09:31:33 2019 From: volga629 at networklab.ca (volga629 at networklab.ca) Date: Fri, 15 Nov 2019 10:31:33 -0400 Subject: [OpenSIPS-Users] dbalias Message-ID: <1573828293.9239.27@skillsearch.ca> Hello Everyone, I am trying use in multi domain environment dbalias table, but hit limitation where impossible insert more then one user per domain. opensips=# INSERT INTO dbaliases (alias_username, alias_domain, username, domain) VALUES ('4384783197', 'dev-sip.networklab.tld', '452392', 'dev-sip.networklab.tld'); ERROR: duplicate key value violates unique constraint "dbaliases_alias_idx" We use PgSQL 10 Any help thank you volga629 -------------- next part -------------- An HTML attachment was scrubbed... URL: From johan at democon.be Fri Nov 15 09:52:18 2019 From: johan at democon.be (johan) Date: Fri, 15 Nov 2019 15:52:18 +0100 Subject: [OpenSIPS-Users] dbalias In-Reply-To: <1573828293.9239.27@skillsearch.ca> References: <1573828293.9239.27@skillsearch.ca> Message-ID: <6e0d352c-013a-4da6-49f5-28aa04ea9b01@democon.be> That's odd : the primary key is on index and there are no foreign keys. Can you check if you have the same in your db ? mysql> describe dbaliases; +----------------+------------------+------+-----+---------+----------------+ | Field          | Type             | Null | Key | Default | Extra          | +----------------+------------------+------+-----+---------+----------------+ | id             | int(10) unsigned | NO   | PRI | NULL    | auto_increment | | alias_username | char(64)         | NO   | MUL | |                | | alias_domain   | char(64)         | NO   |     | |                | | username       | char(64)         | NO   | MUL | |                | | domain         | char(64)         | NO   |     | |                | +----------------+------------------+------+-----+---------+----------------+ 5 rows in set (0.01 sec) mysql> SELECT    TABLE_NAME,COLUMN_NAME,CONSTRAINT_NAME, REFERENCED_TABLE_NAME,REFERENCED_COLUMN_NAME FROM INFORMATION_SCHEMA.KEY_COLUMN_USAGE WHERE REFERENCED_TABLE_SCHEMA = 'opensips' AND   REFERENCED_TABLE_NAME = 'dbaliases'; Empty set (0.01 sec) mysql> On 15.11.19 15:31, volga629 via Users wrote: > Hello Everyone, > I am trying use in multi domain environment dbalias table, but hit > limitation where impossible  insert more then one user per domain. > > opensips=# INSERT INTO dbaliases (alias_username, alias_domain, > username, domain) VALUES ('4384783197', 'dev-sip.networklab.tld', > '452392', 'dev-sip.networklab.tld'); > ERROR:  duplicate key value violates unique constraint > "dbaliases_alias_idx" > > > We use PgSQL  10 > > Any help thank you > volga629 > > _______________________________________________ > 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 Fri Nov 15 09:58:08 2019 From: volga629 at networklab.ca (volga629 at networklab.ca) Date: Fri, 15 Nov 2019 10:58:08 -0400 Subject: [OpenSIPS-Users] dbalias In-Reply-To: <6e0d352c-013a-4da6-49f5-28aa04ea9b01@democon.be> References: <1573828293.9239.27@skillsearch.ca> <6e0d352c-013a-4da6-49f5-28aa04ea9b01@democon.be> Message-ID: <1573829888.9239.28@skillsearch.ca> This PgSQL table structure opensips=# \d dbaliases Table "public.dbaliases" Column | Type | Collation | Nullable | Default ----------------+-----------------------+-----------+----------+--------------------------------------- id | integer | | not null | nextval('dbaliases_id_seq'::regclass) alias_username | character varying(64) | | not null | ''::character varying alias_domain | character varying(64) | | not null | ''::character varying username | character varying(64) | | not null | ''::character varying domain | character varying(64) | | not null | ''::character varying Indexes: "dbaliases_pkey" PRIMARY KEY, btree (id) "dbaliases_alias_idx" UNIQUE CONSTRAINT, btree (alias_username, alias_domain) "dbaliases_target_idx" btree (username, domain) volga629 On Fri, Nov 15, 2019 at 15:52, johan wrote: > That's odd : the primary key is on index and there are no foreign > keys. > > Can you check if you have the same in your db ? > > > mysql> describe dbaliases; > > +----------------+------------------+------+-----+---------+----------------+ > | Field | Type | Null | Key | Default | Extra > | > > +----------------+------------------+------+-----+---------+----------------+ > | id | int(10) unsigned | NO | PRI | NULL | > auto_increment | > | alias_username | char(64) | NO | MUL | | > | > | alias_domain | char(64) | NO | | | > | > | username | char(64) | NO | MUL | | > | > | domain | char(64) | NO | | | > | > > +----------------+------------------+------+-----+---------+----------------+ > 5 rows in set (0.01 sec) > > mysql> SELECT TABLE_NAME,COLUMN_NAME,CONSTRAINT_NAME, > REFERENCED_TABLE_NAME,REFERENCED_COLUMN_NAME FROM > INFORMATION_SCHEMA.KEY_COLUMN_USAGE WHERE REFERENCED_TABLE_SCHEMA = > 'opensips' AND REFERENCED_TABLE_NAME = 'dbaliases'; > Empty set (0.01 sec) > > mysql> > > > On 15.11.19 15:31, volga629 via Users wrote: >> Hello Everyone, >> I am trying use in multi domain environment dbalias table, but hit >> limitation where impossible insert more then one user per domain. >> >> opensips=# INSERT INTO dbaliases (alias_username, alias_domain, >> username, domain) VALUES ('4384783197', 'dev-sip.networklab.tld', >> '452392', 'dev-sip.networklab.tld'); >> ERROR: duplicate key value violates unique constraint >> "dbaliases_alias_idx" >> >> >> We use PgSQL 10 >> >> Any help thank you >> volga629 >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> -------------- next part -------------- An HTML attachment was scrubbed... URL: From Ben.Newlin at genesys.com Fri Nov 15 17:07:25 2019 From: Ben.Newlin at genesys.com (Ben Newlin) Date: Fri, 15 Nov 2019 22:07:25 +0000 Subject: [OpenSIPS-Users] Rate Limit Module Implementation Message-ID: As a follow up to this, when I tried to set the limit for the FEEDBACK algorithm to 0 for testing, I received this error: invalid limit for FEEDBACK algorithm (must be between 0 and 100) Per the documentation this algorithm should be tracking CPU Load, but the range being 0 to 100 makes me wonder if it is actually reporting a percentage, similar to the load statistics exported by OpenSIPS core [1]. Can anyone using this module clarify what unit the limit for this algorithm is expected to be in? The ambiguity is one of the reasons I had wanted to track the values being obtained for a time before actually engaging limited, which is what prompted my original question. [1] - https://www.opensips.org/Documentation/Interface-CoreStatistics-2-4#toc14 Ben Newlin From: Users on behalf of Ben Newlin Reply-To: OpenSIPS users mailling list Date: Thursday, November 14, 2019 at 3:45 PM To: OpenSIPS users mailling list Subject: [OpenSIPS-Users] Rate Limit Module Implementation Hello, We are looking to implement some rate limiting using the module, but I have a few questions and wanted to see if anyone has run into the same issues or has working experience with the module to answer them. The module provides $rl_count and also a counter value in the output of the rl_list command, but both of these values only reflect the number of rl_check calls that were made in the time window. In the case of TAILDROP, RED, and SBT algorithms, the count is directly related to the limit and drop_rate. But for the NETWORK and FEEDBACK algorithms, the counter is essentially meaningless. What we are looking for is a way to track the values these modules are using to make drop decisions, so that we can set our limit value appropriately. But unless I am missing something, there is no way to access these values for either NETWORK or FEEDBACK. Is that correct? Ben Newlin -------------- next part -------------- An HTML attachment was scrubbed... URL: From diptesh.patel at ecosmob.com Sat Nov 16 11:47:18 2019 From: diptesh.patel at ecosmob.com (Dipteshkumar Patel) Date: Sat, 16 Nov 2019 22:17:18 +0530 Subject: [OpenSIPS-Users] How to limit parallel calls duration of prepaid customers? Message-ID: Hello Team, I want to use opensips as a pbx system. I have prepaid customers so how can i manage prepaid calls scheduling based on customers' balance(dialog timeout). As specially in case we have parallel calls of a user. I found call-control from ag-projects for that. Can you please suggest any other possible way to implement this feature? Thanks & Regards *Diptesh Patel* Software Developer Ecosmob Technologies Ltd, Ahmedabad Mo:*+919898962659* -- *Disclaimer* In addition to generic Disclaimer which you have agreed on our website, any views or opinions presented in this email are solely those of the originator and do not necessarily represent those of the Company or its sister concerns. Any liability (in negligence, contract or otherwise) arising from any third party taking any action, or refraining from taking any action on the basis of any of the information contained in this email is hereby excluded. *Confidentiality* This communication (including any attachment/s) is intended only for the use of the addressee(s) and contains information that is PRIVILEGED AND CONFIDENTIAL. Unauthorized reading, dissemination, distribution, or copying of this communication is prohibited. Please inform originator if you have received it in error. *Caution for viruses, malware etc.* This communication, including any attachments, may not be free of viruses, trojans, similar or new contaminants/malware, interceptions or interference, and may not be compatible with your systems. You shall carry out virus/malware scanning on your own before opening any attachment to this e-mail. The sender of this e-mail and Company including its sister concerns shall not be liable for any damage that may incur to you as a result of viruses, incompleteness of this message, a delay in receipt of this message or any other computer problems.  -------------- next part -------------- An HTML attachment was scrubbed... URL: From daniel.zanutti at gmail.com Sat Nov 16 12:59:50 2019 From: daniel.zanutti at gmail.com (Daniel Zanutti) Date: Sat, 16 Nov 2019 14:59:50 -0300 Subject: [OpenSIPS-Users] How to limit parallel calls duration of prepaid customers? In-Reply-To: References: Message-ID: Hi Diptesh We tried to implement a native prepaid system on Opensips but didn't found a way to do this natively, so we developed a custom prepaid mechanism to our solution. Our company (http://dazsoft.com) is focused on complete systems but we can negotiate this specific part if you want. Let me know. Regards On Sat, Nov 16, 2019 at 1:50 PM Dipteshkumar Patel < diptesh.patel at ecosmob.com> wrote: > Hello Team, > > I want to use opensips as a pbx system. I have prepaid customers so how > can i manage prepaid calls scheduling based on customers' balance(dialog > timeout). As specially in case we have parallel calls of a user. I found > call-control from ag-projects for that. > > Can you please suggest any other possible way to implement this feature? > > Thanks & Regards > *Diptesh Patel* > Software Developer > Ecosmob Technologies Ltd, > Ahmedabad > Mo:*+919898962659* > > *Disclaimer* > In addition to generic Disclaimer which you have agreed on our website, > any views or opinions presented in this email are solely those of the > originator and do not necessarily represent those of the Company or its > sister concerns. Any liability (in negligence, contract or otherwise) > arising from any third party taking any action, or refraining from taking > any action on the basis of any of the information contained in this email > is hereby excluded. > > *Confidentiality* > This communication (including any attachment/s) is intended only for the > use of the addressee(s) and contains information that is PRIVILEGED AND > CONFIDENTIAL. Unauthorized reading, dissemination, distribution, or copying > of this communication is prohibited. Please inform originator if you have > received it in error. > > *Caution for viruses, malware etc.* > This communication, including any attachments, may not be free of viruses, > trojans, similar or new contaminants/malware, interceptions or > interference, and may not be compatible with your systems. You shall carry > out virus/malware scanning on your own before opening any attachment to > this e-mail. The sender of this e-mail and Company including its sister > concerns shall not be liable for any damage that may incur to you as a > result of viruses, incompleteness of this message, a delay in receipt of > this message or any other computer problems. > _______________________________________________ > 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 raobilal28 at yahoo.com Sat Nov 16 17:29:41 2019 From: raobilal28 at yahoo.com (Muhammad Bilal Rao) Date: Sat, 16 Nov 2019 22:29:41 +0000 (UTC) Subject: [OpenSIPS-Users] wait_for_event DO NOT TIMEOUT References: <876568390.832986.1573943381817.ref@mail.yahoo.com> Message-ID: <876568390.832986.1573943381817@mail.yahoo.com> Hi, I am using below function to wait for B-party to come online so that INVITE can be sent to him. async( wait_for_event("E_UL_AOR_INSERT","$avp(filter)", "15"),  fork_call); But the issue is that this event never gets timeout and triggers whenever the B-party comes online. Event should not be triggered after the specified time or when the call is cancelled/timeout. Please let me know that how this can be handled.  Thanks Regards,Bilal -------------- next part -------------- An HTML attachment was scrubbed... URL: From darpan.gabani1093 at gmail.com Sun Nov 17 09:06:21 2019 From: darpan.gabani1093 at gmail.com (Darpan Patel) Date: Sun, 17 Nov 2019 19:36:21 +0530 Subject: [OpenSIPS-Users] CDR is not generated in rabbitmq server while call detected as fraud Message-ID: I have subscribed events like E_ACC_CDR ,E_ACC_EVENT , E_ACC_MISSED_EVENT in startup_route to connect rabbitmq server for sending CDR related information of calls and also used event_routes E_FRD_WARNING and E_FRD_CRITICAL to detect fraud calls. while call detected as a fraud after hang up the same call's cdr is not generated in queue of rabbitmq server (ISSUE FOUND IN OPENSIPS 2.4 AND OPENSIPS 3.0) Thanks & Regards, Darpan Patel -------------- next part -------------- An HTML attachment was scrubbed... URL: From ulus_egemen at hotmail.com Mon Nov 18 00:36:38 2019 From: ulus_egemen at hotmail.com (egemen ulus) Date: Mon, 18 Nov 2019 05:36:38 +0000 Subject: [OpenSIPS-Users] Too Many Hops on GCP. Message-ID: Hi, I've installed Opensips 2.4 on a GCP vm, when I try to register my softphone to server, I am getting "483 too many hops" error. I am aware that there is loop on the server side, it sends the packet itself. But could not fix it. I have private and public IP addresses defined on google cloud vm. Here is the my conf and logs: auto_aliases=no listen=udp:10.138.0.3:5060 # private ip address listen=tcp:10.138.0.3:5060 logs: opensips[2671]: Nov 12 18:56:53 [2727] DBG:core:parse_headers: via found, flags=22 opensips[2671]: Nov 12 18:56:53 [2727] DBG:core:parse_via_param: found param type 234, = <34.83.194.202>; state=6 opensips[2671]: Nov 12 18:56:53 [2727] DBG:core:parse_via_param: found param type 232, = ; state=16 opensips[2671]: Nov 12 18:56:53 [2727] DBG:core:parse_via: end of header reached, state=5 thank you for help! Egemen -------------- next part -------------- An HTML attachment was scrubbed... URL: From john at voxtelesys.net Mon Nov 18 00:38:17 2019 From: john at voxtelesys.net (John Burke) Date: Mon, 18 Nov 2019 00:38:17 -0500 Subject: [OpenSIPS-Users] Too Many Hops on GCP. Message-ID: <3069669d235a571fd6eae01072a81360-1574055497@ops-icewarp.voxtelesys.net> I will be out of the office until 11/25. For immediate concerns please contact support at voxtelesys.com or 402-403-4435. From razvan at opensips.org Mon Nov 18 07:44:51 2019 From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=) Date: Mon, 18 Nov 2019 14:44:51 +0200 Subject: [OpenSIPS-Users] Too Many Hops on GCP. In-Reply-To: References: Message-ID: <4abd0344-58fd-7ccb-dc83-f9e01bb00821@opensips.org> Hi, Egemen! Is the REGISTER looping? Are you exiting `exit;` after `save()`? Best regards, Răzvan On 11/18/19 7:36 AM, egemen ulus wrote: > Hi, > I've installed Opensips 2.4 on a GCP vm, when I try to register my > softphone to server, I am getting "483 too many hops" error. I am aware > that there is loop on the server side, it sends the packet itself. But > could not fix it. I have private and public IP addresses defined on > google cloud vm. > > Here is the my conf and logs: > > /auto_aliases=no > / > /listen=udp:10.138.0.3:5060  # private ip address > / > /listen=tcp:10.138.0.3:5060/ > > logs: > /opensips[2671]: Nov 12 18:56:53 [2727] DBG:core:parse_headers: via > found, flags=22 > / > /opensips[2671]: Nov 12 18:56:53 [2727] DBG:core:parse_via_param: found > param type 234, = <34.83.194.202>; state=6 > / > /opensips[2671]: Nov 12 18:56:53 [2727] DBG:core:parse_via_param: found > param type 232, = ; state=16 > / > /opensips[2671]: Nov 12 18:56:53 [2727] DBG:core:parse_via: end of > header reached, state=5/ > / > / > /thank you for help!/ > /Egemen/ > > > _______________________________________________ > 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 From razvan at opensips.org Mon Nov 18 07:48:28 2019 From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=) Date: Mon, 18 Nov 2019 14:48:28 +0200 Subject: [OpenSIPS-Users] fix_nated_sdp() not taking effect In-Reply-To: References: Message-ID: <2fb96c37-2ed0-5ddd-eacc-9bd249c88774@opensips.org> Yes, the problem is definitely the fact that you are calling `rtpproxy_offer()` for the initial invite. Hence, when you run `fix_nated_sdp()`, you're trying to change the same IP once again - this is not possile in OpenSIPS. But I wonder why you need the `fix_nated_sdp()` if you are using RTPProxy. Can't you just use the `ip_address`[1] field to advertise the proper IP int he c= line. [1] https://opensips.org/html/docs/modules/3.0.x/rtpproxy.html#func_rtpproxy_offer Best regards, Răzvan On 11/13/19 1:51 PM, Mark Farmer wrote: > Hi everyone > > In my failure_route I'm routing to an Asterisk box for voicemail & I > need to change the SDP c/o parameters to use the correct internal IP > address but using fix_nated_sdp() is not taking effect. > > if (t_check_status("486|408|603")) { >                 xlog("CUSTOM_LOG: User replied $T_reply_code - Routing > to Asterisk Voicemail service."); >                 prefix("VMR_"); >                 rewritehostport("10.150.50.53:2404 > "); >                 force_send_socket(udp:10.150.50.51); >                 fix_nated_sdp(10,"10.150.50.51"); > >                 if (!t_relay()) { >                         send_reply(500,"Internal Error"); >                 } >                 exit; > } > > I get the CUSTOM_LOG entry so I know that the route is executing. > > Maybe I'm doing something wrong with the flags, I've tried: > fix_nated_sdp(2,"10.150.50.51"); > fix_nated_sdp(8,"10.150.50.51"); > fix_nated_sdp(10,"10.150.50.51"); > > But when I examine the SDP in the resulting invite, the c/o parameters > are never changed. > I'm using rtpengine_offer/answer in the initial routing, could it be > related to that? > > I'm using OpenSIPS 3.0.1 > > Best regards > Mark. > > > > _______________________________________________ > 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 From farmorg at gmail.com Tue Nov 19 07:26:23 2019 From: farmorg at gmail.com (Mark Farmer) Date: Tue, 19 Nov 2019 12:26:23 +0000 Subject: [OpenSIPS-Users] fix_nated_sdp() not taking effect In-Reply-To: <2fb96c37-2ed0-5ddd-eacc-9bd249c88774@opensips.org> References: <2fb96c37-2ed0-5ddd-eacc-9bd249c88774@opensips.org> Message-ID: Hi Răzvan My OpenSIPS/RTPProxy box has 2 interfaces, public(NAT) - for phones & internal - for Asterisk. The issue is that if a call from one registered user to another is rejected & goes to failure_route() then I send the call to an Asterisk box for voicemail which is connected via the internal interface. When the call is routed to Asterisk, I need the RTP to flow between RTPproxy & Asterisk on the internal interfaces so I need to have the SDP correct before it hits Asterisk. RTP to & from the phone needs to use the public interface. Initial media flow: phone<-->OpenSIPS/RTPproxy<-->phone Voicemail media flow: phone<-->OpenSIPS/RTPproxy<-->Asterisk What is the best way to achieve this? Many thanks! Mark. On Mon, 18 Nov 2019 at 12:50, Răzvan Crainea wrote: > Yes, the problem is definitely the fact that you are calling > `rtpproxy_offer()` for the initial invite. Hence, when you run > `fix_nated_sdp()`, you're trying to change the same IP once again - this > is not possile in OpenSIPS. > But I wonder why you need the `fix_nated_sdp()` if you are using > RTPProxy. Can't you just use the `ip_address`[1] field to advertise the > proper IP int he c= line. > > [1] > > https://opensips.org/html/docs/modules/3.0.x/rtpproxy.html#func_rtpproxy_offer > > Best regards, > Răzvan > > On 11/13/19 1:51 PM, Mark Farmer wrote: > > Hi everyone > > > > In my failure_route I'm routing to an Asterisk box for voicemail & I > > need to change the SDP c/o parameters to use the correct internal IP > > address but using fix_nated_sdp() is not taking effect. > > > > if (t_check_status("486|408|603")) { > > xlog("CUSTOM_LOG: User replied $T_reply_code - Routing > > to Asterisk Voicemail service."); > > prefix("VMR_"); > > rewritehostport("10.150.50.53:2404 > > "); > > force_send_socket(udp:10.150.50.51); > > fix_nated_sdp(10,"10.150.50.51"); > > > > if (!t_relay()) { > > send_reply(500,"Internal Error"); > > } > > exit; > > } > > > > I get the CUSTOM_LOG entry so I know that the route is executing. > > > > Maybe I'm doing something wrong with the flags, I've tried: > > fix_nated_sdp(2,"10.150.50.51"); > > fix_nated_sdp(8,"10.150.50.51"); > > fix_nated_sdp(10,"10.150.50.51"); > > > > But when I examine the SDP in the resulting invite, the c/o parameters > > are never changed. > > I'm using rtpengine_offer/answer in the initial routing, could it be > > related to that? > > > > I'm using OpenSIPS 3.0.1 > > > > Best regards > > Mark. > > > > > > > > _______________________________________________ > > 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 > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -- Mark Farmer farmorg at gmail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From kurgan-rus at inbox.ru Tue Nov 19 08:16:15 2019 From: kurgan-rus at inbox.ru (=?UTF-8?B?QWxleGV5IEthemFudHNldg==?=) Date: Tue, 19 Nov 2019 16:16:15 +0300 Subject: [OpenSIPS-Users] =?utf-8?q?fix=5Fnated=5Fsdp=28=29_not_taking_eff?= =?utf-8?q?ect?= In-Reply-To: References: <2fb96c37-2ed0-5ddd-eacc-9bd249c88774@opensips.org> Message-ID: <1574169375.518554473@f137.i.mail.ru> Hi Mark,   have you tried to achieve this with rtpengine_offer/rtpengine_answer functions?   It will rewrite the o= and c= SDP fields. ----------------------------------------------- BR, Alexey http://alexeyka.zantsev.com/   -------------- next part -------------- An HTML attachment was scrubbed... URL: From callum.guy at x-on.co.uk Tue Nov 19 08:18:25 2019 From: callum.guy at x-on.co.uk (Callum Guy) Date: Tue, 19 Nov 2019 13:18:25 +0000 Subject: [OpenSIPS-Users] fix_nated_sdp() not taking effect In-Reply-To: References: <2fb96c37-2ed0-5ddd-eacc-9bd249c88774@opensips.org> Message-ID: You might want to read up on bridge mode, it allows you to meet the finely control which interface is presented during the SDP rewrites. All of the information on the various use cases is available in the module docs, I've used both successfully including some pretty complex request routing. The move to offer/answer with interface specifications works great, you'll just have to fire off the offer with different params when in the failure route so it will override your initial public/public selection from the initial invite processing On Tue, 19 Nov 2019, 12:27 Mark Farmer, wrote: > Hi Răzvan > > My OpenSIPS/RTPProxy box has 2 interfaces, public(NAT) - for phones & > internal - for Asterisk. > The issue is that if a call from one registered user to another is > rejected & goes to failure_route() then I send the call to an Asterisk box > for voicemail which is connected via the internal interface. > > When the call is routed to Asterisk, I need the RTP to flow between > RTPproxy & Asterisk on the internal interfaces so I need to have the SDP > correct before it hits Asterisk. RTP to & from the phone needs to use the > public interface. > > Initial media flow: > phone<-->OpenSIPS/RTPproxy<-->phone > > Voicemail media flow: > phone<-->OpenSIPS/RTPproxy<-->Asterisk > > What is the best way to achieve this? > > Many thanks! > Mark. > > > On Mon, 18 Nov 2019 at 12:50, Răzvan Crainea wrote: > >> Yes, the problem is definitely the fact that you are calling >> `rtpproxy_offer()` for the initial invite. Hence, when you run >> `fix_nated_sdp()`, you're trying to change the same IP once again - this >> is not possile in OpenSIPS. >> But I wonder why you need the `fix_nated_sdp()` if you are using >> RTPProxy. Can't you just use the `ip_address`[1] field to advertise the >> proper IP int he c= line. >> >> [1] >> >> https://opensips.org/html/docs/modules/3.0.x/rtpproxy.html#func_rtpproxy_offer >> >> Best regards, >> Răzvan >> >> On 11/13/19 1:51 PM, Mark Farmer wrote: >> > Hi everyone >> > >> > In my failure_route I'm routing to an Asterisk box for voicemail & I >> > need to change the SDP c/o parameters to use the correct internal IP >> > address but using fix_nated_sdp() is not taking effect. >> > >> > if (t_check_status("486|408|603")) { >> > xlog("CUSTOM_LOG: User replied $T_reply_code - Routing >> > to Asterisk Voicemail service."); >> > prefix("VMR_"); >> > rewritehostport("10.150.50.53:2404 >> > "); >> > force_send_socket(udp:10.150.50.51); >> > fix_nated_sdp(10,"10.150.50.51"); >> > >> > if (!t_relay()) { >> > send_reply(500,"Internal Error"); >> > } >> > exit; >> > } >> > >> > I get the CUSTOM_LOG entry so I know that the route is executing. >> > >> > Maybe I'm doing something wrong with the flags, I've tried: >> > fix_nated_sdp(2,"10.150.50.51"); >> > fix_nated_sdp(8,"10.150.50.51"); >> > fix_nated_sdp(10,"10.150.50.51"); >> > >> > But when I examine the SDP in the resulting invite, the c/o parameters >> > are never changed. >> > I'm using rtpengine_offer/answer in the initial routing, could it be >> > related to that? >> > >> > I'm using OpenSIPS 3.0.1 >> > >> > Best regards >> > Mark. >> > >> > >> > >> > _______________________________________________ >> > 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 >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> > > > -- > Mark Farmer > farmorg at gmail.com > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -- *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 farmorg at gmail.com Tue Nov 19 08:35:33 2019 From: farmorg at gmail.com (Mark Farmer) Date: Tue, 19 Nov 2019 13:35:33 +0000 Subject: [OpenSIPS-Users] fix_nated_sdp() not taking effect In-Reply-To: <1574169375.518554473@f137.i.mail.ru> References: <2fb96c37-2ed0-5ddd-eacc-9bd249c88774@opensips.org> <1574169375.518554473@f137.i.mail.ru> Message-ID: Yes. The issue is that the media path needs to change for the voicemail leg only. On Tue, 19 Nov 2019 at 13:19, Alexey Kazantsev via Users < users at lists.opensips.org> wrote: > Hi Mark, > > have you tried to achieve this with rtpengine_offer/rtpengine_answer > functions? > > It will rewrite the o= and c= SDP fields. > ----------------------------------------------- > BR, Alexey > http://alexeyka.zantsev.com/ > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -- Mark Farmer farmorg at gmail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From farmorg at gmail.com Tue Nov 19 08:53:15 2019 From: farmorg at gmail.com (Mark Farmer) Date: Tue, 19 Nov 2019 13:53:15 +0000 Subject: [OpenSIPS-Users] fix_nated_sdp() not taking effect In-Reply-To: References: <2fb96c37-2ed0-5ddd-eacc-9bd249c88774@opensips.org> Message-ID: I have been using bridged mode all along. I've flipped over to RTPproxy for now, this is my interface configuration, 10.147 being the NATed subnet & 10.150 being the internal subnet: -l 10.150.50.51/10.147.52.52 -A 10.150.50.51/XXX.XXX.XXX.XXX <-- PUBLIC IP I think I understand offer/answer params but calling rtpproxy_offer() in both the initial routing and again in the failure route breaks the SDP which I believe is expected. If I don't call rtpproxy_offer for the initial INVITE then the SDP is broken for that leg. Clearly I'm missing something somewhere... On Tue, 19 Nov 2019 at 13:28, Callum Guy wrote: > You might want to read up on bridge mode, it allows you to meet the finely > control which interface is presented during the SDP rewrites. > > All of the information on the various use cases is available in the module > docs, I've used both successfully including some pretty complex request > routing. > > The move to offer/answer with interface specifications works great, you'll > just have to fire off the offer with different params when in the failure > route so it will override your initial public/public selection from the > initial invite processing > > On Tue, 19 Nov 2019, 12:27 Mark Farmer, wrote: > >> Hi Răzvan >> >> My OpenSIPS/RTPProxy box has 2 interfaces, public(NAT) - for phones & >> internal - for Asterisk. >> The issue is that if a call from one registered user to another is >> rejected & goes to failure_route() then I send the call to an Asterisk box >> for voicemail which is connected via the internal interface. >> >> When the call is routed to Asterisk, I need the RTP to flow between >> RTPproxy & Asterisk on the internal interfaces so I need to have the SDP >> correct before it hits Asterisk. RTP to & from the phone needs to use the >> public interface. >> >> Initial media flow: >> phone<-->OpenSIPS/RTPproxy<-->phone >> >> Voicemail media flow: >> phone<-->OpenSIPS/RTPproxy<-->Asterisk >> >> What is the best way to achieve this? >> >> Many thanks! >> Mark. >> >> >> On Mon, 18 Nov 2019 at 12:50, Răzvan Crainea wrote: >> >>> Yes, the problem is definitely the fact that you are calling >>> `rtpproxy_offer()` for the initial invite. Hence, when you run >>> `fix_nated_sdp()`, you're trying to change the same IP once again - this >>> is not possile in OpenSIPS. >>> But I wonder why you need the `fix_nated_sdp()` if you are using >>> RTPProxy. Can't you just use the `ip_address`[1] field to advertise the >>> proper IP int he c= line. >>> >>> [1] >>> >>> https://opensips.org/html/docs/modules/3.0.x/rtpproxy.html#func_rtpproxy_offer >>> >>> Best regards, >>> Răzvan >>> >>> On 11/13/19 1:51 PM, Mark Farmer wrote: >>> > Hi everyone >>> > >>> > In my failure_route I'm routing to an Asterisk box for voicemail & I >>> > need to change the SDP c/o parameters to use the correct internal IP >>> > address but using fix_nated_sdp() is not taking effect. >>> > >>> > if (t_check_status("486|408|603")) { >>> > xlog("CUSTOM_LOG: User replied $T_reply_code - >>> Routing >>> > to Asterisk Voicemail service."); >>> > prefix("VMR_"); >>> > rewritehostport("10.150.50.53:2404 >>> > "); >>> > force_send_socket(udp:10.150.50.51); >>> > fix_nated_sdp(10,"10.150.50.51"); >>> > >>> > if (!t_relay()) { >>> > send_reply(500,"Internal Error"); >>> > } >>> > exit; >>> > } >>> > >>> > I get the CUSTOM_LOG entry so I know that the route is executing. >>> > >>> > Maybe I'm doing something wrong with the flags, I've tried: >>> > fix_nated_sdp(2,"10.150.50.51"); >>> > fix_nated_sdp(8,"10.150.50.51"); >>> > fix_nated_sdp(10,"10.150.50.51"); >>> > >>> > But when I examine the SDP in the resulting invite, the c/o parameters >>> > are never changed. >>> > I'm using rtpengine_offer/answer in the initial routing, could it be >>> > related to that? >>> > >>> > I'm using OpenSIPS 3.0.1 >>> > >>> > Best regards >>> > Mark. >>> > >>> > >>> > >>> > _______________________________________________ >>> > 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 >>> >>> _______________________________________________ >>> Users mailing list >>> Users at lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >> >> >> -- >> Mark Farmer >> farmorg at gmail.com >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> > > > *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 > -- Mark Farmer farmorg at gmail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From callum.guy at x-on.co.uk Wed Nov 20 04:55:23 2019 From: callum.guy at x-on.co.uk (Callum Guy) Date: Wed, 20 Nov 2019 09:55:23 +0000 Subject: [OpenSIPS-Users] fix_nated_sdp() not taking effect In-Reply-To: References: <2fb96c37-2ed0-5ddd-eacc-9bd249c88774@opensips.org> Message-ID: It should be acceptable to repeat rtpproxy_offer in the failure route - perhaps there is an argument that you'd need to disengage the first session before starting a new one but I don't recall doing that myself. Just to clarify you are using the direction options to manage the bridge interfaces used for each scenario? i.e. rtpproxy_offer("rfie"), rtpproxy_offer("rfei"), rtpproxy_offer("rfee") etc? On Tue, 19 Nov 2019 at 13:54, Mark Farmer wrote: > I have been using bridged mode all along. > I've flipped over to RTPproxy for now, this is my interface configuration, > 10.147 being the NATed subnet & 10.150 being the internal subnet: > > -l 10.150.50.51/10.147.52.52 -A 10.150.50.51/XXX.XXX.XXX.XXX <-- PUBLIC IP > > I think I understand offer/answer params but calling rtpproxy_offer() in > both the initial routing and again in the failure route breaks the SDP > which I believe is expected. > If I don't call rtpproxy_offer for the initial INVITE then the SDP is > broken for that leg. > > Clearly I'm missing something somewhere... > > > > On Tue, 19 Nov 2019 at 13:28, Callum Guy wrote: > >> You might want to read up on bridge mode, it allows you to meet the >> finely control which interface is presented during the SDP rewrites. >> >> All of the information on the various use cases is available in the >> module docs, I've used both successfully including some pretty complex >> request routing. >> >> The move to offer/answer with interface specifications works great, >> you'll just have to fire off the offer with different params when in the >> failure route so it will override your initial public/public selection from >> the initial invite processing >> >> On Tue, 19 Nov 2019, 12:27 Mark Farmer, wrote: >> >>> Hi Răzvan >>> >>> My OpenSIPS/RTPProxy box has 2 interfaces, public(NAT) - for phones & >>> internal - for Asterisk. >>> The issue is that if a call from one registered user to another is >>> rejected & goes to failure_route() then I send the call to an Asterisk box >>> for voicemail which is connected via the internal interface. >>> >>> When the call is routed to Asterisk, I need the RTP to flow between >>> RTPproxy & Asterisk on the internal interfaces so I need to have the SDP >>> correct before it hits Asterisk. RTP to & from the phone needs to use the >>> public interface. >>> >>> Initial media flow: >>> phone<-->OpenSIPS/RTPproxy<-->phone >>> >>> Voicemail media flow: >>> phone<-->OpenSIPS/RTPproxy<-->Asterisk >>> >>> What is the best way to achieve this? >>> >>> Many thanks! >>> Mark. >>> >>> >>> On Mon, 18 Nov 2019 at 12:50, Răzvan Crainea >>> wrote: >>> >>>> Yes, the problem is definitely the fact that you are calling >>>> `rtpproxy_offer()` for the initial invite. Hence, when you run >>>> `fix_nated_sdp()`, you're trying to change the same IP once again - >>>> this >>>> is not possile in OpenSIPS. >>>> But I wonder why you need the `fix_nated_sdp()` if you are using >>>> RTPProxy. Can't you just use the `ip_address`[1] field to advertise the >>>> proper IP int he c= line. >>>> >>>> [1] >>>> >>>> https://opensips.org/html/docs/modules/3.0.x/rtpproxy.html#func_rtpproxy_offer >>>> >>>> Best regards, >>>> Răzvan >>>> >>>> On 11/13/19 1:51 PM, Mark Farmer wrote: >>>> > Hi everyone >>>> > >>>> > In my failure_route I'm routing to an Asterisk box for voicemail & I >>>> > need to change the SDP c/o parameters to use the correct internal IP >>>> > address but using fix_nated_sdp() is not taking effect. >>>> > >>>> > if (t_check_status("486|408|603")) { >>>> > xlog("CUSTOM_LOG: User replied $T_reply_code - >>>> Routing >>>> > to Asterisk Voicemail service."); >>>> > prefix("VMR_"); >>>> > rewritehostport("10.150.50.53:2404 >>>> > "); >>>> > force_send_socket(udp:10.150.50.51); >>>> > fix_nated_sdp(10,"10.150.50.51"); >>>> > >>>> > if (!t_relay()) { >>>> > send_reply(500,"Internal Error"); >>>> > } >>>> > exit; >>>> > } >>>> > >>>> > I get the CUSTOM_LOG entry so I know that the route is executing. >>>> > >>>> > Maybe I'm doing something wrong with the flags, I've tried: >>>> > fix_nated_sdp(2,"10.150.50.51"); >>>> > fix_nated_sdp(8,"10.150.50.51"); >>>> > fix_nated_sdp(10,"10.150.50.51"); >>>> > >>>> > But when I examine the SDP in the resulting invite, the c/o >>>> parameters >>>> > are never changed. >>>> > I'm using rtpengine_offer/answer in the initial routing, could it be >>>> > related to that? >>>> > >>>> > I'm using OpenSIPS 3.0.1 >>>> > >>>> > Best regards >>>> > Mark. >>>> > >>>> > >>>> > >>>> > _______________________________________________ >>>> > 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 >>>> >>>> _______________________________________________ >>>> Users mailing list >>>> Users at lists.opensips.org >>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>> >>> >>> >>> -- >>> Mark Farmer >>> farmorg at gmail.com >>> _______________________________________________ >>> Users mailing list >>> Users at lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >> >> >> *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 >> > > > -- > Mark Farmer > farmorg at gmail.com > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -- *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 a.martin at alphalink.fr Wed Nov 20 06:28:08 2019 From: a.martin at alphalink.fr (Adrien Martin) Date: Wed, 20 Nov 2019 12:28:08 +0100 Subject: [OpenSIPS-Users] APT and YUM repositories down In-Reply-To: <3ec8a6e2-cf5b-dba1-7380-c5097a10aaf6@opensips.org> References: <3aa7678b-31b3-ac61-3792-1f00edca78eb@alphalink.fr> <3ec8a6e2-cf5b-dba1-7380-c5097a10aaf6@opensips.org> Message-ID: Hello, It seems there is another server outage with apt and yum repositories. Regards, Adrien Martin Le 30/09/2019 à 15:34, Bogdan-Andrei Iancu a écrit : > And now back online, thanks to Nick Altmann !! (some server outage) > > Regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer >   https://www.opensips-solutions.com > OpenSIPS Summit 2019 >   https://www.opensips.org/events/Summit-2019Amsterdam/ > > On 9/30/19 11:06 AM, Callum Guy wrote: >> Hopefully just in prep for 3.0.1! >> >> On Mon, 30 Sep 2019 at 08:42, Adrien Martin > wrote: >> >>     Hello, >> >> >>     Both apt.opensips.org and >>     yum.opensips.org seem down (IPv4 and IPv6). >> >> >>     Regards, >> >>     Adrien Martin >> >>     _______________________________________________ >>     Users mailing list >>     Users at lists.opensips.org >>     http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> >> >> *^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 > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > From bogdan at opensips.org Wed Nov 20 06:53:52 2019 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Wed, 20 Nov 2019 13:53:52 +0200 Subject: [OpenSIPS-Users] APT and YUM repositories down In-Reply-To: References: <3aa7678b-31b3-ac61-3792-1f00edca78eb@alphalink.fr> <3ec8a6e2-cf5b-dba1-7380-c5097a10aaf6@opensips.org> Message-ID: <7d5246e0-a7c5-7724-51e0-c40959bfe5e1@opensips.org> Hi Adrien, Thanks for the heads up, I just "poked" Nick on the issue ;) Best regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com OpenSIPS Bootcamp Pre-Registration https://opensips.org/training/OpenSIPS_Bootcamp/ On 11/20/19 1:28 PM, Adrien Martin wrote: > Hello, > > > It seems there is another server outage with apt and yum repositories. > > > Regards, > > Adrien Martin > > Le 30/09/2019 à 15:34, Bogdan-Andrei Iancu a écrit : >> And now back online, thanks to Nick Altmann !! (some server outage) >> >> Regards, >> >> Bogdan-Andrei Iancu >> >> OpenSIPS Founder and Developer >>    https://www.opensips-solutions.com >> OpenSIPS Summit 2019 >>    https://www.opensips.org/events/Summit-2019Amsterdam/ >> >> On 9/30/19 11:06 AM, Callum Guy wrote: >>> Hopefully just in prep for 3.0.1! >>> >>> On Mon, 30 Sep 2019 at 08:42, Adrien Martin >> > wrote: >>> >>>     Hello, >>> >>> >>>     Both apt.opensips.org and >>>     yum.opensips.org seem down (IPv4 and >>> IPv6). >>> >>> >>>     Regards, >>> >>>     Adrien Martin >>> >>>     _______________________________________________ >>>     Users mailing list >>>     Users at lists.opensips.org >>>     http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >>> >>> >>> *^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 >> >> >> >> _______________________________________________ >> 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 farmorg at gmail.com Wed Nov 20 09:20:49 2019 From: farmorg at gmail.com (Mark Farmer) Date: Wed, 20 Nov 2019 14:20:49 +0000 Subject: [OpenSIPS-Users] fix_nated_sdp() not taking effect In-Reply-To: References: <2fb96c37-2ed0-5ddd-eacc-9bd249c88774@opensips.org> Message-ID: As I suspected and as indicated by Răzvan, calling rtpproxy_offer() a second time in the failure route breaks the SDP (XXX.XXX.XXX.XXX is the public IP) o=- 1574259046 1574259046 IN IP4 XXX.XXX.XXX.XXX10.150.50.51 s=Polycom IP Phone c=IN IP4 XXX.XXX.XXX.XXX10.150.50.51 I am making progress, my rtpproxy_offer/answer now seems to be working as expected but now I have an issue in that because rtpproxy was never engaged in the initial INVITE, RTPproxy is now trying to send audio to the private ip of the original phone. Is there something in particular that I need to do with a 603 reply to handle the audio path establishment? Many thanks for all input on this! Mark. On Wed, 20 Nov 2019 at 09:58, Callum Guy wrote: > It should be acceptable to repeat rtpproxy_offer in the failure route - > perhaps there is an argument that you'd need to disengage the first session > before starting a new one but I don't recall doing that myself. Just to > clarify you are using the direction options to manage the bridge interfaces > used for each scenario? > i.e. rtpproxy_offer("rfie"), rtpproxy_offer("rfei"), rtpproxy_offer("rfee") > etc? > > On Tue, 19 Nov 2019 at 13:54, Mark Farmer wrote: > >> I have been using bridged mode all along. >> I've flipped over to RTPproxy for now, this is my interface >> configuration, 10.147 being the NATed subnet & 10.150 being the internal >> subnet: >> >> -l 10.150.50.51/10.147.52.52 -A 10.150.50.51/XXX.XXX.XXX.XXX <-- PUBLIC >> IP >> >> I think I understand offer/answer params but calling rtpproxy_offer() in >> both the initial routing and again in the failure route breaks the SDP >> which I believe is expected. >> If I don't call rtpproxy_offer for the initial INVITE then the SDP is >> broken for that leg. >> >> Clearly I'm missing something somewhere... >> >> >> >> On Tue, 19 Nov 2019 at 13:28, Callum Guy wrote: >> >>> You might want to read up on bridge mode, it allows you to meet the >>> finely control which interface is presented during the SDP rewrites. >>> >>> All of the information on the various use cases is available in the >>> module docs, I've used both successfully including some pretty complex >>> request routing. >>> >>> The move to offer/answer with interface specifications works great, >>> you'll just have to fire off the offer with different params when in the >>> failure route so it will override your initial public/public selection from >>> the initial invite processing >>> >>> On Tue, 19 Nov 2019, 12:27 Mark Farmer, wrote: >>> >>>> Hi Răzvan >>>> >>>> My OpenSIPS/RTPProxy box has 2 interfaces, public(NAT) - for phones & >>>> internal - for Asterisk. >>>> The issue is that if a call from one registered user to another is >>>> rejected & goes to failure_route() then I send the call to an Asterisk box >>>> for voicemail which is connected via the internal interface. >>>> >>>> When the call is routed to Asterisk, I need the RTP to flow between >>>> RTPproxy & Asterisk on the internal interfaces so I need to have the SDP >>>> correct before it hits Asterisk. RTP to & from the phone needs to use the >>>> public interface. >>>> >>>> Initial media flow: >>>> phone<-->OpenSIPS/RTPproxy<-->phone >>>> >>>> Voicemail media flow: >>>> phone<-->OpenSIPS/RTPproxy<-->Asterisk >>>> >>>> What is the best way to achieve this? >>>> >>>> Many thanks! >>>> Mark. >>>> >>>> >>>> On Mon, 18 Nov 2019 at 12:50, Răzvan Crainea >>>> wrote: >>>> >>>>> Yes, the problem is definitely the fact that you are calling >>>>> `rtpproxy_offer()` for the initial invite. Hence, when you run >>>>> `fix_nated_sdp()`, you're trying to change the same IP once again - >>>>> this >>>>> is not possile in OpenSIPS. >>>>> But I wonder why you need the `fix_nated_sdp()` if you are using >>>>> RTPProxy. Can't you just use the `ip_address`[1] field to advertise >>>>> the >>>>> proper IP int he c= line. >>>>> >>>>> [1] >>>>> >>>>> https://opensips.org/html/docs/modules/3.0.x/rtpproxy.html#func_rtpproxy_offer >>>>> >>>>> Best regards, >>>>> Răzvan >>>>> >>>>> On 11/13/19 1:51 PM, Mark Farmer wrote: >>>>> > Hi everyone >>>>> > >>>>> > In my failure_route I'm routing to an Asterisk box for voicemail & I >>>>> > need to change the SDP c/o parameters to use the correct internal IP >>>>> > address but using fix_nated_sdp() is not taking effect. >>>>> > >>>>> > if (t_check_status("486|408|603")) { >>>>> > xlog("CUSTOM_LOG: User replied $T_reply_code - >>>>> Routing >>>>> > to Asterisk Voicemail service."); >>>>> > prefix("VMR_"); >>>>> > rewritehostport("10.150.50.53:2404 >>>>> > "); >>>>> > force_send_socket(udp:10.150.50.51); >>>>> > fix_nated_sdp(10,"10.150.50.51"); >>>>> > >>>>> > if (!t_relay()) { >>>>> > send_reply(500,"Internal Error"); >>>>> > } >>>>> > exit; >>>>> > } >>>>> > >>>>> > I get the CUSTOM_LOG entry so I know that the route is executing. >>>>> > >>>>> > Maybe I'm doing something wrong with the flags, I've tried: >>>>> > fix_nated_sdp(2,"10.150.50.51"); >>>>> > fix_nated_sdp(8,"10.150.50.51"); >>>>> > fix_nated_sdp(10,"10.150.50.51"); >>>>> > >>>>> > But when I examine the SDP in the resulting invite, the c/o >>>>> parameters >>>>> > are never changed. >>>>> > I'm using rtpengine_offer/answer in the initial routing, could it be >>>>> > related to that? >>>>> > >>>>> > I'm using OpenSIPS 3.0.1 >>>>> > >>>>> > Best regards >>>>> > Mark. >>>>> > >>>>> > >>>>> > >>>>> > _______________________________________________ >>>>> > 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 >>>>> >>>>> _______________________________________________ >>>>> Users mailing list >>>>> Users at lists.opensips.org >>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>>> >>>> >>>> >>>> -- >>>> Mark Farmer >>>> farmorg at gmail.com >>>> _______________________________________________ >>>> Users mailing list >>>> Users at lists.opensips.org >>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>> >>> >>> >>> *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 >>> >> >> >> -- >> Mark Farmer >> farmorg at gmail.com >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> > > > *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 > -- Mark Farmer farmorg at gmail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From johnkiniston at gmail.com Wed Nov 20 16:38:49 2019 From: johnkiniston at gmail.com (John Kiniston) Date: Wed, 20 Nov 2019 14:38:49 -0700 Subject: [OpenSIPS-Users] 3.0.1 source comes with core files. Message-ID: Downloaded the tar from https://opensips.org/pub/opensips/3.0.1/opensips-3.0.1.tar.gz and extracted it. root at pi:/usr/src/opensips-3.0.1# ls -lh core* -rw------- 1 pi pi 642M Jun 10 07:50 core.17567 -rw------- 1 pi pi 642M Jun 7 15:54 core.21927 -- A human being should be able to change a diaper, plan an invasion, butcher a hog, conn a ship, design a building, write a sonnet, balance accounts, build a wall, set a bone, comfort the dying, take orders, give orders, cooperate, act alone, solve equations, analyze a new problem, pitch manure, program a computer, cook a tasty meal, fight efficiently, die gallantly. Specialization is for insects. ---Heinlein -------------- next part -------------- An HTML attachment was scrubbed... URL: From johnkiniston at gmail.com Wed Nov 20 17:11:05 2019 From: johnkiniston at gmail.com (John Kiniston) Date: Wed, 20 Nov 2019 15:11:05 -0700 Subject: [OpenSIPS-Users] Script Entry points Message-ID: I may be miss-remembering that this is a feature of 3.0 Didn't I read that you could specify a route based on what listener receives the traffic? -- A human being should be able to change a diaper, plan an invasion, butcher a hog, conn a ship, design a building, write a sonnet, balance accounts, build a wall, set a bone, comfort the dying, take orders, give orders, cooperate, act alone, solve equations, analyze a new problem, pitch manure, program a computer, cook a tasty meal, fight efficiently, die gallantly. Specialization is for insects. ---Heinlein -------------- next part -------------- An HTML attachment was scrubbed... URL: From 1157983522 at qq.com Wed Nov 20 21:30:11 2019 From: 1157983522 at qq.com (=?utf-8?B?5rGk5LiW56Wl?=) Date: Thu, 21 Nov 2019 10:30:11 +0800 Subject: [OpenSIPS-Users] confusion about the reply route and the failure route Message-ID: <28E3DDCE-8073-4090-86CB-D9C85F9DB046@qq.com> Hi: I'm a little confused about the reply route and the failure route. Both can handle 404/408 responses in invite session. Both default action is to relay back the SIP reply. If I omit two routes in scripts(that meas not arm the reply route by using the t_on_reply("name") function or t_on_failure("name") function) or call t_relay() in these two routing scripts, will there be two responses sent ? From diptesh.patel at ecosmob.com Wed Nov 20 23:26:40 2019 From: diptesh.patel at ecosmob.com (Dipteshkumar Patel) Date: Thu, 21 Nov 2019 09:56:40 +0530 Subject: [OpenSIPS-Users] How to limit parallel calls duration of prepaid customers? In-Reply-To: References: Message-ID: Hello Daniel, Is your solution scalable? If i have clustering architecture with HA. So in my system I have opensips cluster(Dialog Replication). In my system if Active Node is down and the Virtual IP now bind to the another(Passive) node. In this case, will your solution work? Thanks & Regards *Diptesh Patel* Software Developer Ecosmob Technologies Ltd, Ahmedabad Mo:*+919898962659* On Sat, Nov 16, 2019 at 11:30 PM Daniel Zanutti wrote: > Hi Diptesh > > We tried to implement a native prepaid system on Opensips but didn't found > a way to do this natively, so we developed a custom prepaid mechanism to > our solution. > > Our company (http://dazsoft.com) is focused on complete systems but we > can negotiate this specific part if you want. Let me know. > > Regards > > > On Sat, Nov 16, 2019 at 1:50 PM Dipteshkumar Patel < > diptesh.patel at ecosmob.com> wrote: > >> Hello Team, >> >> I want to use opensips as a pbx system. I have prepaid customers so how >> can i manage prepaid calls scheduling based on customers' balance(dialog >> timeout). As specially in case we have parallel calls of a user. I found >> call-control from ag-projects for that. >> >> Can you please suggest any other possible way to implement this feature? >> >> Thanks & Regards >> *Diptesh Patel* >> Software Developer >> Ecosmob Technologies Ltd, >> Ahmedabad >> Mo:*+919898962659* >> >> *Disclaimer* >> In addition to generic Disclaimer which you have agreed on our website, >> any views or opinions presented in this email are solely those of the >> originator and do not necessarily represent those of the Company or its >> sister concerns. Any liability (in negligence, contract or otherwise) >> arising from any third party taking any action, or refraining from taking >> any action on the basis of any of the information contained in this email >> is hereby excluded. >> >> *Confidentiality* >> This communication (including any attachment/s) is intended only for the >> use of the addressee(s) and contains information that is PRIVILEGED AND >> CONFIDENTIAL. Unauthorized reading, dissemination, distribution, or copying >> of this communication is prohibited. Please inform originator if you have >> received it in error. >> >> *Caution for viruses, malware etc.* >> This communication, including any attachments, may not be free of >> viruses, trojans, similar or new contaminants/malware, interceptions or >> interference, and may not be compatible with your systems. You shall carry >> out virus/malware scanning on your own before opening any attachment to >> this e-mail. The sender of this e-mail and Company including its sister >> concerns shall not be liable for any damage that may incur to you as a >> result of viruses, incompleteness of this message, a delay in receipt of >> this message or any other computer problems. >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> > -- *Disclaimer* In addition to generic Disclaimer which you have agreed on our website, any views or opinions presented in this email are solely those of the originator and do not necessarily represent those of the Company or its sister concerns. Any liability (in negligence, contract or otherwise) arising from any third party taking any action, or refraining from taking any action on the basis of any of the information contained in this email is hereby excluded. *Confidentiality* This communication (including any attachment/s) is intended only for the use of the addressee(s) and contains information that is PRIVILEGED AND CONFIDENTIAL. Unauthorized reading, dissemination, distribution, or copying of this communication is prohibited. Please inform originator if you have received it in error. *Caution for viruses, malware etc.* This communication, including any attachments, may not be free of viruses, trojans, similar or new contaminants/malware, interceptions or interference, and may not be compatible with your systems. You shall carry out virus/malware scanning on your own before opening any attachment to this e-mail. The sender of this e-mail and Company including its sister concerns shall not be liable for any damage that may incur to you as a result of viruses, incompleteness of this message, a delay in receipt of this message or any other computer problems.  -------------- next part -------------- An HTML attachment was scrubbed... URL: From razvan at opensips.org Thu Nov 21 03:42:23 2019 From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=) Date: Thu, 21 Nov 2019 10:42:23 +0200 Subject: [OpenSIPS-Users] 3.0.1 source comes with core files. In-Reply-To: References: Message-ID: <9542a67e-31cf-a2d9-91d4-ce881cb838d9@opensips.org> Thanks for pointing out, I've just updated the tar. Cheers, Răzvan On 11/20/19 11:38 PM, John Kiniston wrote: > Downloaded the tar from > https://opensips.org/pub/opensips/3.0.1/opensips-3.0.1.tar.gz and > extracted it. > > root at pi:/usr/src/opensips-3.0.1# ls -lh core* > -rw------- 1 pi pi 642M Jun 10 07:50 core.17567 > -rw------- 1 pi pi 642M Jun  7 15:54 core.21927 > > > -- > A human being should be able to change a diaper, plan an invasion, > butcher a hog, conn a ship, design a building, write a sonnet, balance > accounts, build a wall, set a bone, comfort the dying, take orders, give > orders, cooperate, act alone, solve equations, analyze a new problem, > pitch manure, program a computer, cook a tasty meal, fight efficiently, > die gallantly. Specialization is for insects. > ---Heinlein > > _______________________________________________ > 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 From david.villasmil.work at gmail.com Thu Nov 21 03:43:44 2019 From: david.villasmil.work at gmail.com (David Villasmil) Date: Thu, 21 Nov 2019 08:43:44 +0000 Subject: [OpenSIPS-Users] How to limit parallel calls duration of prepaid customers? In-Reply-To: References: Message-ID: You can control this with sqlops (or another db backend). If the user can't make multiple call you can control this with dialog timeout ( https://kamailio.org/docs/modules/5.2.x/modules/dialog.html#dialog.p.timeout_avp ) You calculate how long max the call can be established and set the timeout avp accordingly. When a call comes in check if the user is in the table (custom table) and if not store the calling user in a table as "in_use = 1". I did this a loooong time ago, but it works fine. Regards, David Villasmil email: david.villasmil.work at gmail.com phone: +34669448337 On Thu, Nov 21, 2019 at 4:27 AM Dipteshkumar Patel < diptesh.patel at ecosmob.com> wrote: > Hello Daniel, > > Is your solution scalable? > If i have clustering architecture with HA. So in my system I have opensips > cluster(Dialog Replication). > In my system if Active Node is down and the Virtual IP now bind to the > another(Passive) node. > In this case, will your solution work? > > Thanks & Regards > *Diptesh Patel* > Software Developer > Ecosmob Technologies Ltd, > Ahmedabad > Mo:*+919898962659* > > > On Sat, Nov 16, 2019 at 11:30 PM Daniel Zanutti > wrote: > >> Hi Diptesh >> >> We tried to implement a native prepaid system on Opensips but didn't >> found a way to do this natively, so we developed a custom prepaid mechanism >> to our solution. >> >> Our company (http://dazsoft.com) is focused on complete systems but we >> can negotiate this specific part if you want. Let me know. >> >> Regards >> >> >> On Sat, Nov 16, 2019 at 1:50 PM Dipteshkumar Patel < >> diptesh.patel at ecosmob.com> wrote: >> >>> Hello Team, >>> >>> I want to use opensips as a pbx system. I have prepaid customers so how >>> can i manage prepaid calls scheduling based on customers' balance(dialog >>> timeout). As specially in case we have parallel calls of a user. I found >>> call-control from ag-projects for that. >>> >>> Can you please suggest any other possible way to implement this feature? >>> >>> Thanks & Regards >>> *Diptesh Patel* >>> Software Developer >>> Ecosmob Technologies Ltd, >>> Ahmedabad >>> Mo:*+919898962659* >>> >>> *Disclaimer* >>> In addition to generic Disclaimer which you have agreed on our website, >>> any views or opinions presented in this email are solely those of the >>> originator and do not necessarily represent those of the Company or its >>> sister concerns. Any liability (in negligence, contract or otherwise) >>> arising from any third party taking any action, or refraining from taking >>> any action on the basis of any of the information contained in this email >>> is hereby excluded. >>> >>> *Confidentiality* >>> This communication (including any attachment/s) is intended only for the >>> use of the addressee(s) and contains information that is PRIVILEGED AND >>> CONFIDENTIAL. Unauthorized reading, dissemination, distribution, or copying >>> of this communication is prohibited. Please inform originator if you have >>> received it in error. >>> >>> *Caution for viruses, malware etc.* >>> This communication, including any attachments, may not be free of >>> viruses, trojans, similar or new contaminants/malware, interceptions or >>> interference, and may not be compatible with your systems. You shall carry >>> out virus/malware scanning on your own before opening any attachment to >>> this e-mail. The sender of this e-mail and Company including its sister >>> concerns shall not be liable for any damage that may incur to you as a >>> result of viruses, incompleteness of this message, a delay in receipt of >>> this message or any other computer problems. >>> _______________________________________________ >>> Users mailing list >>> Users at lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >> > *Disclaimer* > In addition to generic Disclaimer which you have agreed on our website, > any views or opinions presented in this email are solely those of the > originator and do not necessarily represent those of the Company or its > sister concerns. Any liability (in negligence, contract or otherwise) > arising from any third party taking any action, or refraining from taking > any action on the basis of any of the information contained in this email > is hereby excluded. > > *Confidentiality* > This communication (including any attachment/s) is intended only for the > use of the addressee(s) and contains information that is PRIVILEGED AND > CONFIDENTIAL. Unauthorized reading, dissemination, distribution, or copying > of this communication is prohibited. Please inform originator if you have > received it in error. > > *Caution for viruses, malware etc.* > This communication, including any attachments, may not be free of viruses, > trojans, similar or new contaminants/malware, interceptions or > interference, and may not be compatible with your systems. You shall carry > out virus/malware scanning on your own before opening any attachment to > this e-mail. The sender of this e-mail and Company including its sister > concerns shall not be liable for any damage that may incur to you as a > result of viruses, incompleteness of this message, a delay in receipt of > this message or any other computer problems. > _______________________________________________ > 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 Thu Nov 21 03:44:21 2019 From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=) Date: Thu, 21 Nov 2019 10:44:21 +0200 Subject: [OpenSIPS-Users] Script Entry points In-Reply-To: References: Message-ID: Indeed, this was on the list of our TODOs for 3.0, but unfortunately it didn't get much of a score in our polls. Therefore we decided to postpone it for 3.1, hopefully we'll manage to put it there. Cheers, Răzvan On 11/21/19 12:11 AM, John Kiniston wrote: > I may be miss-remembering that this is a feature of 3.0 > > Didn't I read that you could specify a route based on what listener > receives the traffic? > > -- > A human being should be able to change a diaper, plan an invasion, > butcher a hog, conn a ship, design a building, write a sonnet, balance > accounts, build a wall, set a bone, comfort the dying, take orders, give > orders, cooperate, act alone, solve equations, analyze a new problem, > pitch manure, program a computer, cook a tasty meal, fight efficiently, > die gallantly. Specialization is for insects. > ---Heinlein > > _______________________________________________ > 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 From david.villasmil.work at gmail.com Thu Nov 21 03:45:32 2019 From: david.villasmil.work at gmail.com (David Villasmil) Date: Thu, 21 Nov 2019 08:45:32 +0000 Subject: [OpenSIPS-Users] How to limit parallel calls duration of prepaid customers? In-Reply-To: References: Message-ID: Sorry this is opensips: https://opensips.org/html/docs/modules/devel/dialog.html#pv_DLG_timeout Regards, David Villasmil email: david.villasmil.work at gmail.com phone: +34669448337 On Thu, Nov 21, 2019 at 8:43 AM David Villasmil < david.villasmil.work at gmail.com> wrote: > You can control this with sqlops (or another db backend). > If the user can't make multiple call you can control this with dialog > timeout ( > https://kamailio.org/docs/modules/5.2.x/modules/dialog.html#dialog.p.timeout_avp > ) > You calculate how long max the call can be established and set the timeout > avp accordingly. > When a call comes in check if the user is in the table (custom table) and > if not store the calling user in a table as "in_use = 1". > > I did this a loooong time ago, but it works fine. > > > Regards, > > David Villasmil > email: david.villasmil.work at gmail.com > phone: +34669448337 > > > On Thu, Nov 21, 2019 at 4:27 AM Dipteshkumar Patel < > diptesh.patel at ecosmob.com> wrote: > >> Hello Daniel, >> >> Is your solution scalable? >> If i have clustering architecture with HA. So in my system I have >> opensips cluster(Dialog Replication). >> In my system if Active Node is down and the Virtual IP now bind to the >> another(Passive) node. >> In this case, will your solution work? >> >> Thanks & Regards >> *Diptesh Patel* >> Software Developer >> Ecosmob Technologies Ltd, >> Ahmedabad >> Mo:*+919898962659* >> >> >> On Sat, Nov 16, 2019 at 11:30 PM Daniel Zanutti >> wrote: >> >>> Hi Diptesh >>> >>> We tried to implement a native prepaid system on Opensips but didn't >>> found a way to do this natively, so we developed a custom prepaid mechanism >>> to our solution. >>> >>> Our company (http://dazsoft.com) is focused on complete systems but we >>> can negotiate this specific part if you want. Let me know. >>> >>> Regards >>> >>> >>> On Sat, Nov 16, 2019 at 1:50 PM Dipteshkumar Patel < >>> diptesh.patel at ecosmob.com> wrote: >>> >>>> Hello Team, >>>> >>>> I want to use opensips as a pbx system. I have prepaid customers so how >>>> can i manage prepaid calls scheduling based on customers' balance(dialog >>>> timeout). As specially in case we have parallel calls of a user. I found >>>> call-control from ag-projects for that. >>>> >>>> Can you please suggest any other possible way to implement this feature? >>>> >>>> Thanks & Regards >>>> *Diptesh Patel* >>>> Software Developer >>>> Ecosmob Technologies Ltd, >>>> Ahmedabad >>>> Mo:*+919898962659* >>>> >>>> *Disclaimer* >>>> In addition to generic Disclaimer which you have agreed on our website, >>>> any views or opinions presented in this email are solely those of the >>>> originator and do not necessarily represent those of the Company or its >>>> sister concerns. Any liability (in negligence, contract or otherwise) >>>> arising from any third party taking any action, or refraining from taking >>>> any action on the basis of any of the information contained in this email >>>> is hereby excluded. >>>> >>>> *Confidentiality* >>>> This communication (including any attachment/s) is intended only for >>>> the use of the addressee(s) and contains information that is PRIVILEGED AND >>>> CONFIDENTIAL. Unauthorized reading, dissemination, distribution, or copying >>>> of this communication is prohibited. Please inform originator if you have >>>> received it in error. >>>> >>>> *Caution for viruses, malware etc.* >>>> This communication, including any attachments, may not be free of >>>> viruses, trojans, similar or new contaminants/malware, interceptions or >>>> interference, and may not be compatible with your systems. You shall carry >>>> out virus/malware scanning on your own before opening any attachment to >>>> this e-mail. The sender of this e-mail and Company including its sister >>>> concerns shall not be liable for any damage that may incur to you as a >>>> result of viruses, incompleteness of this message, a delay in receipt of >>>> this message or any other computer problems. >>>> _______________________________________________ >>>> Users mailing list >>>> Users at lists.opensips.org >>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>> >>> >> *Disclaimer* >> In addition to generic Disclaimer which you have agreed on our website, >> any views or opinions presented in this email are solely those of the >> originator and do not necessarily represent those of the Company or its >> sister concerns. Any liability (in negligence, contract or otherwise) >> arising from any third party taking any action, or refraining from taking >> any action on the basis of any of the information contained in this email >> is hereby excluded. >> >> *Confidentiality* >> This communication (including any attachment/s) is intended only for the >> use of the addressee(s) and contains information that is PRIVILEGED AND >> CONFIDENTIAL. Unauthorized reading, dissemination, distribution, or copying >> of this communication is prohibited. Please inform originator if you have >> received it in error. >> >> *Caution for viruses, malware etc.* >> This communication, including any attachments, may not be free of >> viruses, trojans, similar or new contaminants/malware, interceptions or >> interference, and may not be compatible with your systems. You shall carry >> out virus/malware scanning on your own before opening any attachment to >> this e-mail. The sender of this e-mail and Company including its sister >> concerns shall not be liable for any damage that may incur to you as a >> result of viruses, incompleteness of this message, a delay in receipt of >> this message or any other computer problems. >> _______________________________________________ >> 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 Thu Nov 21 03:56:28 2019 From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=) Date: Thu, 21 Nov 2019 10:56:28 +0200 Subject: [OpenSIPS-Users] confusion about the reply route and the failure route In-Reply-To: <28E3DDCE-8073-4090-86CB-D9C85F9DB046@qq.com> References: <28E3DDCE-8073-4090-86CB-D9C85F9DB046@qq.com> Message-ID: <93fbe34b-665f-4a90-1a4b-86dd991bee42@opensips.org> Hello! By default, the replies received by OpenSIPS are relayed further, whether you have a reply of failure route or not. The purpose of these two routes is to do any processing on the messages they are processing. Note that the scope of failure route and reply route is a bit different: the reply route is ran for each reply message - this means that if you for example add a header on a 408, that 408 message will contain the header. Moreover, in failure route you can only drop a reply, but you cannot failover. The failure route has nothing to do with replies processing - it is ran in the context of the initial request. This means that if you add a header in failure route, it will not appear in any replies! However, if you run t_relay() in failure route, you will create a new branch - send the request to a new upstream - that request will contain the added header. The only thing that failure route running relates to replies is the fact that if you do a t_relay() in failure route and dispatch the request to a new destination, no reply will be sent downstream. To summarize, if you don't use reply route or failure route, all responses will be sent. The only exception to this rule is when you use parallel forking: send the initial request to multiple destinations in parallel. In such case, according to RFC 3261, OpenSIPS will only relay downstream the result all replies received from upstream - that is the lowest terminal response code. Best regards, Răzvan On 11/21/19 4:30 AM, 汤世祥 wrote: > Hi: > I'm a little confused about the reply route and the failure route. > Both can handle 404/408 responses in invite session. Both default action is to relay back the SIP reply. > If I omit two routes in scripts(that meas not arm the reply route by using the t_on_reply("name") function or t_on_failure("name") function) or call t_relay() in these two routing scripts, will there be two responses sent ? > > _______________________________________________ > 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 From liviu at opensips.org Thu Nov 21 04:05:59 2019 From: liviu at opensips.org (Liviu Chircu) Date: Thu, 21 Nov 2019 11:05:59 +0200 Subject: [OpenSIPS-Users] Script Entry points In-Reply-To: References: Message-ID: <8002e23e-2c91-e268-0d2d-bc113f3306a0@opensips.org> Hi John, That feature was (and probably still is) on the TODO list [1], but it didn't make it to 3.0. Cheers, [1]: https://www.opensips.org/Development/Opensips-3-0-Planning Liviu Chircu OpenSIPS Developer http://www.opensips-solutions.com On 21.11.2019 00:11, John Kiniston wrote: > I may be miss-remembering that this is a feature of 3.0 > > Didn't I read that you could specify a route based on what listener > receives the traffic? > > -- > A human being should be able to change a diaper, plan an invasion, > butcher a hog, conn a ship, design a building, write a sonnet, balance > accounts, build a wall, set a bone, comfort the dying, take orders, > give orders, cooperate, act alone, solve equations, analyze a new > problem, pitch manure, program a computer, cook a tasty meal, fight > efficiently, die gallantly. Specialization is for insects. > ---Heinlein > > _______________________________________________ > 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 Nov 21 09:17:42 2019 From: Ben.Newlin at genesys.com (Ben Newlin) Date: Thu, 21 Nov 2019 14:17:42 +0000 Subject: [OpenSIPS-Users] confusion about the reply route and the failure route In-Reply-To: <93fbe34b-665f-4a90-1a4b-86dd991bee42@opensips.org> References: <28E3DDCE-8073-4090-86CB-D9C85F9DB046@qq.com> <93fbe34b-665f-4a90-1a4b-86dd991bee42@opensips.org> Message-ID: "Moreover, in failure route you can only drop a reply, but you cannot failover." I think this should say in reply route. Ben Newlin On 11/21/19, 3:56 AM, "Users on behalf of Răzvan Crainea" wrote: Hello! By default, the replies received by OpenSIPS are relayed further, whether you have a reply of failure route or not. The purpose of these two routes is to do any processing on the messages they are processing. Note that the scope of failure route and reply route is a bit different: the reply route is ran for each reply message - this means that if you for example add a header on a 408, that 408 message will contain the header. Moreover, in failure route you can only drop a reply, but you cannot failover. The failure route has nothing to do with replies processing - it is ran in the context of the initial request. This means that if you add a header in failure route, it will not appear in any replies! However, if you run t_relay() in failure route, you will create a new branch - send the request to a new upstream - that request will contain the added header. The only thing that failure route running relates to replies is the fact that if you do a t_relay() in failure route and dispatch the request to a new destination, no reply will be sent downstream. To summarize, if you don't use reply route or failure route, all responses will be sent. The only exception to this rule is when you use parallel forking: send the initial request to multiple destinations in parallel. In such case, according to RFC 3261, OpenSIPS will only relay downstream the result all replies received from upstream - that is the lowest terminal response code. Best regards, Răzvan On 11/21/19 4:30 AM, 汤世祥 wrote: > Hi: > I'm a little confused about the reply route and the failure route. > Both can handle 404/408 responses in invite session. Both default action is to relay back the SIP reply. > If I omit two routes in scripts(that meas not arm the reply route by using the t_on_reply("name") function or t_on_failure("name") function) or call t_relay() in these two routing scripts, will there be two responses sent ? > > _______________________________________________ > 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 _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users From razvan at opensips.org Thu Nov 21 09:47:51 2019 From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=) Date: Thu, 21 Nov 2019 16:47:51 +0200 Subject: [OpenSIPS-Users] confusion about the reply route and the failure route In-Reply-To: References: <28E3DDCE-8073-4090-86CB-D9C85F9DB046@qq.com> <93fbe34b-665f-4a90-1a4b-86dd991bee42@opensips.org> Message-ID: <6a69797d-9a9e-4f1a-4e80-631e5ed34112@opensips.org> On 11/21/19 4:17 PM, Ben Newlin wrote: > "Moreover, in failure route you can only drop a reply, but you cannot failover." > > I think this should say in reply route. That is correct, my apologies, you can only drop a response in reply_route. > > Ben Newlin > > On 11/21/19, 3:56 AM, "Users on behalf of Răzvan Crainea" wrote: > > Hello! > > By default, the replies received by OpenSIPS are relayed further, > whether you have a reply of failure route or not. The purpose of these > two routes is to do any processing on the messages they are processing. > Note that the scope of failure route and reply route is a bit different: > the reply route is ran for each reply message - this means that if you > for example add a header on a 408, that 408 message will contain the > header. Moreover, in failure route you can only drop a reply, but you > cannot failover. > The failure route has nothing to do with replies processing - it is ran > in the context of the initial request. This means that if you add a > header in failure route, it will not appear in any replies! However, if > you run t_relay() in failure route, you will create a new branch - send > the request to a new upstream - that request will contain the added > header. The only thing that failure route running relates to replies is > the fact that if you do a t_relay() in failure route and dispatch the > request to a new destination, no reply will be sent downstream. > To summarize, if you don't use reply route or failure route, all > responses will be sent. The only exception to this rule is when you use > parallel forking: send the initial request to multiple destinations in > parallel. In such case, according to RFC 3261, OpenSIPS will only relay > downstream the result all replies received from upstream - that is the > lowest terminal response code. > > Best regards, > Răzvan > > > On 11/21/19 4:30 AM, 汤世祥 wrote: > > Hi: > > I'm a little confused about the reply route and the failure route. > > Both can handle 404/408 responses in invite session. Both default action is to relay back the SIP reply. > > If I omit two routes in scripts(that meas not arm the reply route by using the t_on_reply("name") function or t_on_failure("name") function) or call t_relay() in these two routing scripts, will there be two responses sent ? > > > > _______________________________________________ > > 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 > > _______________________________________________ > 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 From johnkiniston at gmail.com Thu Nov 21 10:56:56 2019 From: johnkiniston at gmail.com (John Kiniston) Date: Thu, 21 Nov 2019 08:56:56 -0700 Subject: [OpenSIPS-Users] Script Entry points In-Reply-To: <8002e23e-2c91-e268-0d2d-bc113f3306a0@opensips.org> References: <8002e23e-2c91-e268-0d2d-bc113f3306a0@opensips.org> Message-ID: Thanks guys! On Thu, Nov 21, 2019 at 2:08 AM Liviu Chircu wrote: > Hi John, > > That feature was (and probably still is) on the TODO list [1], but it > didn't make it to 3.0. > > Cheers, > > [1]: https://www.opensips.org/Development/Opensips-3-0-Planning > > Liviu Chircu > OpenSIPS Developerhttp://www.opensips-solutions.com > > On 21.11.2019 00:11, John Kiniston wrote: > > I may be miss-remembering that this is a feature of 3.0 > > Didn't I read that you could specify a route based on what listener > receives the traffic? > > -- > A human being should be able to change a diaper, plan an invasion, butcher > a hog, conn a ship, design a building, write a sonnet, balance accounts, > build a wall, set a bone, comfort the dying, take orders, give orders, > cooperate, act alone, solve equations, analyze a new problem, pitch manure, > program a computer, cook a tasty meal, fight efficiently, die gallantly. > Specialization is for insects. > ---Heinlein > > _______________________________________________ > 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 > -- A human being should be able to change a diaper, plan an invasion, butcher a hog, conn a ship, design a building, write a sonnet, balance accounts, build a wall, set a bone, comfort the dying, take orders, give orders, cooperate, act alone, solve equations, analyze a new problem, pitch manure, program a computer, cook a tasty meal, fight efficiently, die gallantly. Specialization is for insects. ---Heinlein -------------- next part -------------- An HTML attachment was scrubbed... URL: From 1157983522 at qq.com Thu Nov 21 20:34:10 2019 From: 1157983522 at qq.com (=?utf-8?B?MTE1Nzk4MzUyMg==?=) Date: Fri, 22 Nov 2019 09:34:10 +0800 Subject: [OpenSIPS-Users] =?utf-8?b?5Zue5aSN77yaICBjb25mdXNpb24gYWJvdXQg?= =?utf-8?q?the_reply_route_and_the_failure_route?= Message-ID: Thank you for your  explanation.  The failure routeis ran in the context of the initial request and the route reply work on reply message, this explanation helped me understand a lot of confusion. But I also have another question, for example: when openbsuips received non-2xx response for register or option requet message , if I ignore failure route and reply route in the routing script, will opensips forward one response or two (failure route and reply route all excute its default action) to upstream? if I write reply route in routing script,but it do nothing, and i write failure route that only call t_reply(), will opensips forward one response or two to upstream? ------------------ 原始邮件 ------------------ 发件人: "Ben Newlin" From ag at ag-projects.com Fri Nov 22 10:49:30 2019 From: ag at ag-projects.com (Adrian Georgescu) Date: Fri, 22 Nov 2019 12:49:30 -0300 Subject: [OpenSIPS-Users] How to limit parallel calls duration of prepaid customers? In-Reply-To: References: Message-ID: <1DB1FB11-8597-4485-85CD-532683D547CF@ag-projects.com> Call control does handle the ’special’ case of parallel calls for a user. When balance gets to zero all outgoing calls are being cut add the same time. There is nothing special the module work with N parallel calls, not just 1. Adrian > On 16 Nov 2019, at 13:47, Dipteshkumar Patel wrote: > > Hello Team, > > I want to use opensips as a pbx system. I have prepaid customers so how can i manage prepaid calls scheduling based on customers' balance(dialog timeout). As specially in case we have parallel calls of a user. I found call-control from ag-projects for that. > > Can you please suggest any other possible way to implement this feature? > > Thanks & Regards > Diptesh Patel > Software Developer > Ecosmob Technologies Ltd, > Ahmedabad > Mo:+919898962659 > > Disclaimer > In addition to generic Disclaimer which you have agreed on our website, any views or opinions presented in this email are solely those of the originator and do not necessarily represent those of the Company or its sister concerns. Any liability (in negligence, contract or otherwise) arising from any third party taking any action, or refraining from taking any action on the basis of any of the information contained in this email is hereby excluded. > > Confidentiality > This communication (including any attachment/s) is intended only for the use of the addressee(s) and contains information that is PRIVILEGED AND CONFIDENTIAL. Unauthorized reading, dissemination, distribution, or copying of this communication is prohibited. Please inform originator if you have received it in error. > > Caution for viruses, malware etc. > This communication, including any attachments, may not be free of viruses, trojans, similar or new contaminants/malware, interceptions or interference, and may not be compatible with your systems. You shall carry out virus/malware scanning on your own before opening any attachment to this e-mail. The sender of this e-mail and Company including its sister concerns shall not be liable for any damage that may incur to you as a result of viruses, incompleteness of this message, a delay in receipt of this message or any other computer problems. > _______________________________________________ > 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 Fri Nov 22 11:43:18 2019 From: volga629 at networklab.ca (volga629 at networklab.ca) Date: Fri, 22 Nov 2019 12:43:18 -0400 Subject: [OpenSIPS-Users] lua script Message-ID: <1574440998.351851.2@skillsearch.ca> Hello Everyone, In lua module how is possible call multiply arguments for lua function. Is this valid ? if (lua_exec("convert_function",arg1,arg2)) { } volga629 -------------- next part -------------- An HTML attachment was scrubbed... URL: From ulus_egemen at hotmail.com Fri Nov 22 12:25:24 2019 From: ulus_egemen at hotmail.com (egemen ulus) Date: Fri, 22 Nov 2019 17:25:24 +0000 Subject: [OpenSIPS-Users] =?utf-8?q?Too_Many_Hops_on_GCP=2E_=28R=C4=83zvan?= =?utf-8?q?_Crainea=29?= In-Reply-To: References: Message-ID: Hi Răzvan , Yes it happens during registering. After a while it obviously gives 483 too many hops because of max forward. What you mean when you say that "exiting after 'save()' " Regards, Egemen ________________________________ Gönderen: users-request at lists.opensips.org adına Users Gönderildi: 18 Kasım 2019 Pazartesi 20:00 Kime: users at lists.opensips.org Konu: Users Digest, Vol 136, Issue 33 Send Users mailing list submissions to users at lists.opensips.org To subscribe or unsubscribe via the World Wide Web, visit http://lists.opensips.org/cgi-bin/mailman/listinfo/users or, via email, send a message with subject or body 'help' to users-request at lists.opensips.org You can reach the person managing the list at users-owner at lists.opensips.org When replying, please edit your Subject line so it is more specific than "Re: Contents of Users digest..." Today's Topics: 1. Too Many Hops on GCP. (egemen ulus) 2. Re: Too Many Hops on GCP. (John Burke) 3. Re: Too Many Hops on GCP. (Răzvan Crainea) 4. Re: fix_nated_sdp() not taking effect (Răzvan Crainea) ---------------------------------------------------------------------- Message: 1 Date: Mon, 18 Nov 2019 05:36:38 +0000 From: egemen ulus To: "users at lists.opensips.org" Subject: [OpenSIPS-Users] Too Many Hops on GCP. Message-ID: Content-Type: text/plain; charset="iso-8859-9" Hi, I've installed Opensips 2.4 on a GCP vm, when I try to register my softphone to server, I am getting "483 too many hops" error. I am aware that there is loop on the server side, it sends the packet itself. But could not fix it. I have private and public IP addresses defined on google cloud vm. Here is the my conf and logs: auto_aliases=no listen=udp:10.138.0.3:5060 # private ip address listen=tcp:10.138.0.3:5060 logs: opensips[2671]: Nov 12 18:56:53 [2727] DBG:core:parse_headers: via found, flags=22 opensips[2671]: Nov 12 18:56:53 [2727] DBG:core:parse_via_param: found param type 234, = <34.83.194.202>; state=6 opensips[2671]: Nov 12 18:56:53 [2727] DBG:core:parse_via_param: found param type 232, = ; state=16 opensips[2671]: Nov 12 18:56:53 [2727] DBG:core:parse_via: end of header reached, state=5 thank you for help! Egemen -------------- next part -------------- An HTML attachment was scrubbed... URL: ------------------------------ Message: 2 Date: Mon, 18 Nov 2019 00:38:17 -0500 From: "John Burke" To: Subject: Re: [OpenSIPS-Users] Too Many Hops on GCP. Message-ID: <3069669d235a571fd6eae01072a81360-1574055497 at ops-icewarp.voxtelesys.net> Content-Type: text/plain; charset="utf-8" I will be out of the office until 11/25. For immediate concerns please contact support at voxtelesys.com or 402-403-4435. ------------------------------ Message: 3 Date: Mon, 18 Nov 2019 14:44:51 +0200 From: Răzvan Crainea To: users at lists.opensips.org Subject: Re: [OpenSIPS-Users] Too Many Hops on GCP. Message-ID: <4abd0344-58fd-7ccb-dc83-f9e01bb00821 at opensips.org> Content-Type: text/plain; charset=UTF-8; format=flowed Hi, Egemen! Is the REGISTER looping? Are you exiting `exit;` after `save()`? Best regards, Răzvan On 11/18/19 7:36 AM, egemen ulus wrote: > Hi, > I've installed Opensips 2.4 on a GCP vm, when I try to register my > softphone to server, I am getting "483 too many hops" error. I am aware > that there is loop on the server side, it sends the packet itself. But > could not fix it. I have private and public IP addresses defined on > google cloud vm. > > Here is the my conf and logs: > > /auto_aliases=no > / > /listen=udp:10.138.0.3:5060 # private ip address > / > /listen=tcp:10.138.0.3:5060/ > > logs: > /opensips[2671]: Nov 12 18:56:53 [2727] DBG:core:parse_headers: via > found, flags=22 > / > /opensips[2671]: Nov 12 18:56:53 [2727] DBG:core:parse_via_param: found > param type 234, = <34.83.194.202>; state=6 > / > /opensips[2671]: Nov 12 18:56:53 [2727] DBG:core:parse_via_param: found > param type 232, = ; state=16 > / > /opensips[2671]: Nov 12 18:56:53 [2727] DBG:core:parse_via: end of > header reached, state=5/ > / > / > /thank you for help!/ > /Egemen/ > > > _______________________________________________ > 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 ------------------------------ Message: 4 Date: Mon, 18 Nov 2019 14:48:28 +0200 From: Răzvan Crainea To: users at lists.opensips.org Subject: Re: [OpenSIPS-Users] fix_nated_sdp() not taking effect Message-ID: <2fb96c37-2ed0-5ddd-eacc-9bd249c88774 at opensips.org> Content-Type: text/plain; charset=utf-8; format=flowed Yes, the problem is definitely the fact that you are calling `rtpproxy_offer()` for the initial invite. Hence, when you run `fix_nated_sdp()`, you're trying to change the same IP once again - this is not possile in OpenSIPS. But I wonder why you need the `fix_nated_sdp()` if you are using RTPProxy. Can't you just use the `ip_address`[1] field to advertise the proper IP int he c= line. [1] https://opensips.org/html/docs/modules/3.0.x/rtpproxy.html#func_rtpproxy_offer Best regards, Răzvan On 11/13/19 1:51 PM, Mark Farmer wrote: > Hi everyone > > In my failure_route I'm routing to an Asterisk box for voicemail & I > need to change the SDP c/o parameters to use the correct internal IP > address but using fix_nated_sdp() is not taking effect. > > if (t_check_status("486|408|603")) { > xlog("CUSTOM_LOG: User replied $T_reply_code - Routing > to Asterisk Voicemail service."); > prefix("VMR_"); > rewritehostport("10.150.50.53:2404 > "); > force_send_socket(udp:10.150.50.51); > fix_nated_sdp(10,"10.150.50.51"); > > if (!t_relay()) { > send_reply(500,"Internal Error"); > } > exit; > } > > I get the CUSTOM_LOG entry so I know that the route is executing. > > Maybe I'm doing something wrong with the flags, I've tried: > fix_nated_sdp(2,"10.150.50.51"); > fix_nated_sdp(8,"10.150.50.51"); > fix_nated_sdp(10,"10.150.50.51"); > > But when I examine the SDP in the resulting invite, the c/o parameters > are never changed. > I'm using rtpengine_offer/answer in the initial routing, could it be > related to that? > > I'm using OpenSIPS 3.0.1 > > Best regards > Mark. > > > > _______________________________________________ > 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 ------------------------------ Subject: Digest Footer _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ------------------------------ End of Users Digest, Vol 136, Issue 33 ************************************** -------------- next part -------------- An HTML attachment was scrubbed... URL: From ulus_egemen at hotmail.com Fri Nov 22 12:27:39 2019 From: ulus_egemen at hotmail.com (egemen ulus) Date: Fri, 22 Nov 2019 17:27:39 +0000 Subject: [OpenSIPS-Users] =?utf-8?q?Ynt=3A_Too_Many_Hops_on_GCP=2E_=28R?= =?utf-8?b?xIN6dmFuIENyYWluZWEp?= In-Reply-To: References: , Message-ID: Hi again, Btw, if you are asking for this part, here is it; if (!mf_process_maxfwd_header("10")) { send_reply("483","Too Many Hops"); exit; } Regards Egemen ________________________________ Gönderen: egemen ulus Gönderildi: 22 Kasım 2019 Cuma 20:25 Kime: users at lists.opensips.org Konu: Re: Too Many Hops on GCP. (Răzvan Crainea) Hi Răzvan , Yes it happens during registering. After a while it obviously gives 483 too many hops because of max forward. What you mean when you say that "exiting after 'save()' " Regards, Egemen ________________________________ Gönderen: users-request at lists.opensips.org adına Users Gönderildi: 18 Kasım 2019 Pazartesi 20:00 Kime: users at lists.opensips.org Konu: Users Digest, Vol 136, Issue 33 Send Users mailing list submissions to users at lists.opensips.org To subscribe or unsubscribe via the World Wide Web, visit http://lists.opensips.org/cgi-bin/mailman/listinfo/users or, via email, send a message with subject or body 'help' to users-request at lists.opensips.org You can reach the person managing the list at users-owner at lists.opensips.org When replying, please edit your Subject line so it is more specific than "Re: Contents of Users digest..." Today's Topics: 1. Too Many Hops on GCP. (egemen ulus) 2. Re: Too Many Hops on GCP. (John Burke) 3. Re: Too Many Hops on GCP. (Răzvan Crainea) 4. Re: fix_nated_sdp() not taking effect (Răzvan Crainea) ---------------------------------------------------------------------- Message: 1 Date: Mon, 18 Nov 2019 05:36:38 +0000 From: egemen ulus To: "users at lists.opensips.org" Subject: [OpenSIPS-Users] Too Many Hops on GCP. Message-ID: Content-Type: text/plain; charset="iso-8859-9" Hi, I've installed Opensips 2.4 on a GCP vm, when I try to register my softphone to server, I am getting "483 too many hops" error. I am aware that there is loop on the server side, it sends the packet itself. But could not fix it. I have private and public IP addresses defined on google cloud vm. Here is the my conf and logs: auto_aliases=no listen=udp:10.138.0.3:5060 # private ip address listen=tcp:10.138.0.3:5060 logs: opensips[2671]: Nov 12 18:56:53 [2727] DBG:core:parse_headers: via found, flags=22 opensips[2671]: Nov 12 18:56:53 [2727] DBG:core:parse_via_param: found param type 234, = <34.83.194.202>; state=6 opensips[2671]: Nov 12 18:56:53 [2727] DBG:core:parse_via_param: found param type 232, = ; state=16 opensips[2671]: Nov 12 18:56:53 [2727] DBG:core:parse_via: end of header reached, state=5 thank you for help! Egemen -------------- next part -------------- An HTML attachment was scrubbed... URL: ------------------------------ Message: 2 Date: Mon, 18 Nov 2019 00:38:17 -0500 From: "John Burke" To: Subject: Re: [OpenSIPS-Users] Too Many Hops on GCP. Message-ID: <3069669d235a571fd6eae01072a81360-1574055497 at ops-icewarp.voxtelesys.net> Content-Type: text/plain; charset="utf-8" I will be out of the office until 11/25. For immediate concerns please contact support at voxtelesys.com or 402-403-4435. ------------------------------ Message: 3 Date: Mon, 18 Nov 2019 14:44:51 +0200 From: Răzvan Crainea To: users at lists.opensips.org Subject: Re: [OpenSIPS-Users] Too Many Hops on GCP. Message-ID: <4abd0344-58fd-7ccb-dc83-f9e01bb00821 at opensips.org> Content-Type: text/plain; charset=UTF-8; format=flowed Hi, Egemen! Is the REGISTER looping? Are you exiting `exit;` after `save()`? Best regards, Răzvan On 11/18/19 7:36 AM, egemen ulus wrote: > Hi, > I've installed Opensips 2.4 on a GCP vm, when I try to register my > softphone to server, I am getting "483 too many hops" error. I am aware > that there is loop on the server side, it sends the packet itself. But > could not fix it. I have private and public IP addresses defined on > google cloud vm. > > Here is the my conf and logs: > > /auto_aliases=no > / > /listen=udp:10.138.0.3:5060 # private ip address > / > /listen=tcp:10.138.0.3:5060/ > > logs: > /opensips[2671]: Nov 12 18:56:53 [2727] DBG:core:parse_headers: via > found, flags=22 > / > /opensips[2671]: Nov 12 18:56:53 [2727] DBG:core:parse_via_param: found > param type 234, = <34.83.194.202>; state=6 > / > /opensips[2671]: Nov 12 18:56:53 [2727] DBG:core:parse_via_param: found > param type 232, = ; state=16 > / > /opensips[2671]: Nov 12 18:56:53 [2727] DBG:core:parse_via: end of > header reached, state=5/ > / > / > /thank you for help!/ > /Egemen/ > > > _______________________________________________ > 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 ------------------------------ Message: 4 Date: Mon, 18 Nov 2019 14:48:28 +0200 From: Răzvan Crainea To: users at lists.opensips.org Subject: Re: [OpenSIPS-Users] fix_nated_sdp() not taking effect Message-ID: <2fb96c37-2ed0-5ddd-eacc-9bd249c88774 at opensips.org> Content-Type: text/plain; charset=utf-8; format=flowed Yes, the problem is definitely the fact that you are calling `rtpproxy_offer()` for the initial invite. Hence, when you run `fix_nated_sdp()`, you're trying to change the same IP once again - this is not possile in OpenSIPS. But I wonder why you need the `fix_nated_sdp()` if you are using RTPProxy. Can't you just use the `ip_address`[1] field to advertise the proper IP int he c= line. [1] https://opensips.org/html/docs/modules/3.0.x/rtpproxy.html#func_rtpproxy_offer Best regards, Răzvan On 11/13/19 1:51 PM, Mark Farmer wrote: > Hi everyone > > In my failure_route I'm routing to an Asterisk box for voicemail & I > need to change the SDP c/o parameters to use the correct internal IP > address but using fix_nated_sdp() is not taking effect. > > if (t_check_status("486|408|603")) { > xlog("CUSTOM_LOG: User replied $T_reply_code - Routing > to Asterisk Voicemail service."); > prefix("VMR_"); > rewritehostport("10.150.50.53:2404 > "); > force_send_socket(udp:10.150.50.51); > fix_nated_sdp(10,"10.150.50.51"); > > if (!t_relay()) { > send_reply(500,"Internal Error"); > } > exit; > } > > I get the CUSTOM_LOG entry so I know that the route is executing. > > Maybe I'm doing something wrong with the flags, I've tried: > fix_nated_sdp(2,"10.150.50.51"); > fix_nated_sdp(8,"10.150.50.51"); > fix_nated_sdp(10,"10.150.50.51"); > > But when I examine the SDP in the resulting invite, the c/o parameters > are never changed. > I'm using rtpengine_offer/answer in the initial routing, could it be > related to that? > > I'm using OpenSIPS 3.0.1 > > Best regards > Mark. > > > > _______________________________________________ > 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 ------------------------------ Subject: Digest Footer _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ------------------------------ End of Users Digest, Vol 136, Issue 33 ************************************** -------------- next part -------------- An HTML attachment was scrubbed... URL: From virendra at cloud-connect.in Mon Nov 25 01:10:01 2019 From: virendra at cloud-connect.in (Virendra Bhati) Date: Mon, 25 Nov 2019 11:40:01 +0530 Subject: [OpenSIPS-Users] Users Digest, Vol 135, Issue 41 In-Reply-To: References: Message-ID: Dear Bogdan and Team, Cloud you please reply on this as we tested with another server and this issue is persisting continue. -- Regards Virendra Bhati Lead- Architecture and Software Solutions On Thu, Nov 14, 2019 at 10:37 AM Virendra Bhati wrote: > Dear Team, > Does anyone face such issue of "crash of OS at the time of MariDB > connection"? > -- > Regards > Virendra Bhati > Lead- Architecture and Software Solutions > > > On Wed, Oct 30, 2019 at 9:30 PM wrote: > >> Send Users mailing list submissions to >> users at lists.opensips.org >> >> To subscribe or unsubscribe via the World Wide Web, visit >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> or, via email, send a message with subject or body 'help' to >> users-request at lists.opensips.org >> >> You can reach the person managing the list at >> users-owner at lists.opensips.org >> >> When replying, please edit your Subject line so it is more specific >> than "Re: Contents of Users digest..." >> >> >> Today's Topics: >> >> 1. Opensips crash while connecting mariadb (Virendra Bhati) >> 2. Re: Example of configuration "Full Sharing" Topology with >> NoSQL (Liviu Chircu) >> >> >> ---------------------------------------------------------------------- >> >> Message: 1 >> Date: Wed, 30 Oct 2019 13:26:44 +0530 >> From: Virendra Bhati >> To: users at lists.opensips.org >> Subject: [OpenSIPS-Users] Opensips crash while connecting mariadb >> Message-ID: >> > dTEeKNT7w9fDZMAqZmORTSJGOT5QK3RZAcLmpiWU2vHPL1A at mail.gmail.com> >> Content-Type: text/plain; charset="utf-8" >> >> Dear Team, >> >> We are using opensips 3.0.1 with mariadb 10.4.8 . We are facing issue at >> starting opensips. As opensips crash with below given message: >> >> Oct 15 03:27:17 [22849] DBG:core:find_mod_export: found in >> module db_mysql [/usr/local/lib64/opensips/modules/] >> Oct 15 03:27:17 [22849] DBG:core:db_bind_mod: using db bind api for >> db_mysql >> Oct 15 03:27:17 [22849] DBG:core:db_do_init: connection 0x7fba0e093510 not >> found in pool >> Got ERROR: "InnoDB: Unable to lock /var/lib/mysql/ibdata1 error: 11" >> errno: >> 2000 >> >> I have attached logs and backtrace from core so please provide your >> suggestions. Please note credentials given in db_url are working while >> using from mysql client >> >> >> -- >> Regards >> Virendra Bhati >> Lead- Architecture and Software Solutions >> -------------- next part -------------- >> An HTML attachment was scrubbed... >> URL: < >> http://lists.opensips.org/pipermail/users/attachments/20191030/dfb9da26/attachment-0001.html >> > >> -------------- next part -------------- >> Oct 15 03:27:17 [22849] INFO:core:evi_publish_event: Registered event >> >> Oct 15 03:27:17 [22849] DBG:core:find_cmd_export_t: found in >> module tm [/usr/local/lib64/opensips/modules/] >> Oct 15 03:27:17 [22849] DBG:core:find_cmd_export_t: found in >> module rr [/usr/local/lib64/opensips/modules/] >> Oct 15 03:27:17 [22849] DBG:core:find_mod_export: found in >> module db_mysql [/usr/local/lib64/opensips/modules/] >> Oct 15 03:27:17 [22849] DBG:core:db_bind_mod: using db bind api for >> db_mysql >> Oct 15 03:27:17 [22849] DBG:core:db_do_init: connection 0x7fba0e093510 >> not found in pool >> Got ERROR: "InnoDB: Unable to lock /var/lib/mysql/ibdata1 error: 11" >> errno: 2000 >> Got ERROR: "InnoDB: Unable to lock /var/lib/mysql/ibdata1 error: 11" >> errno: 2000 >> Got ERROR: "InnoDB: Unable to lock /var/lib/mysql/ibdata1 error: 11" >> errno: 2000 >> Got ERROR: "InnoDB: Unable to lock /var/lib/mysql/ibdata1 error: 11" >> errno: 2000 >> Got ERROR: "InnoDB: Unable to lock /var/lib/mysql/ibdata1 error: 11" >> errno: 2000 >> Got ERROR: "InnoDB: Unable to lock /var/lib/mysql/ibdata1 error: 11" >> errno: 2000 >> Got ERROR: "InnoDB: Unable to lock /var/lib/mysql/ibdata1 error: 11" >> errno: 2000 >> >> Got ERROR: "InnoDB: Operating system error number 11 in a file >> operation." errno: 2000 >> Got ERROR: "InnoDB: Error number 11 means 'Resource temporarily >> unavailable'" errno: 2000 >> Got ERROR: "InnoDB: Cannot open datafile '/var/lib/mysql/ibdata1'" errno: >> 2000 >> Got ERROR: "InnoDB: Could not open or create the system tablespace. If >> you tried to add new data files to the system tablespace, and it failed >> here, you should now edit innodb_data_file_path in my.cnf back to what it >> was, and remove the new ibdata files InnoDB created in this failed attempt. >> InnoDB only wrote those files full of zeros, but did not yet use them in >> any way. But be careful: do not remove old data files which contain your >> precious data!" errno: 2000 >> Got ERROR: "InnoDB: Plugin initialization aborted with error Cannot open >> a file" errno: 2000 >> Got ERROR: "Plugin 'InnoDB' init function returned error." errno: 2000 >> Got ERROR: "Plugin 'InnoDB' registration as a STORAGE ENGINE failed." >> errno: 2000 >> Got ERROR: "unknown: Can't lock aria control file >> '/var/lib/mysql/aria_log_control' for exclusive use, error: 11. Will retry >> for 30 seconds" errno: 2000 >> >> >> >> ^C^CGot ERROR: "unknown: Got error 'Could not get an exclusive lock; file >> is probably in use by another process' when trying to use aria control file >> '/var/lib/mysql/aria_log_control'" errno: 2000 >> Got ERROR: "Plugin 'Aria' init function returned error." errno: 2000 >> Got ERROR: "Plugin 'Aria' registration as a STORAGE ENGINE failed." >> errno: 2000 >> Got ERROR: "Unknown/unsupported storage engine: InnoDB" errno: 2000 >> DBG:db_mysql:db_mysql_connect: opening connection: >> mysql://xxxx:xxxx at localhost/cc_master >> CRITICAL:core:sig_usr: segfault in attendant (starter) process! >> DBG:core:restore_segv_handler: restoring SIGSEGV handler... >> DBG:core:restore_segv_handler: successfully restored system SIGSEGV >> handler >> ^CSegmentation fault (core dumped) >> >> -------------- next part -------------- >> (gdb) bt full >> #0 intern_plugin_lock (lex=0x0, state_mask=14, rc=0x0) >> at /usr/src/debug/MariaDB-10.3.18/src_0/sql/sql_plugin.cc:948 >> pi = 0x0 >> #1 plugin_thdvar_init (thd=0x28af578) at >> /usr/src/debug/MariaDB-10.3.18/src_0/sql/sql_plugin.cc:3155 >> old_table_plugin = 0x0 >> old_tmp_table_plugin = 0x0 >> old_enforced_table_plugin = 0x0 >> #2 0x00007fc7d58722b1 in THD::init (this=this at entry=0x28af578) >> at /usr/src/debug/MariaDB-10.3.18/src_0/sql/sql_class.cc:1177 >> No locals. >> #3 0x00007fc7d587310a in THD::THD (this=0x28af578, id=, >> is_wsrep_applier=) >> at /usr/src/debug/MariaDB-10.3.18/src_0/sql/sql_class.cc:798 >> tmp = >> #4 0x00007fc7d57f2bdc in create_embedded_thd >> (client_flag=client_flag at entry=-2143837683) >> at /usr/src/debug/MariaDB-10.3.18/src_0/libmysqld/lib_sql.cc:685 >> thd = 0x7fff689c20b0 >> #5 0x00007fc7d57fa1e4 in mysql_real_connect (mysql=0x7fc7da7190b0, >> host=, user=, >> passwd=, db=0x7fc7da717620 "cc_master", port=port at entry=0, >> unix_socket=unix_socket at entry=0x0, >> client_flag=2151129613, client_flag at entry=2147549184) >> at /usr/src/debug/MariaDB-10.3.18/src_0/libmysqld/libmysqld.c:179 >> name_buff = >> "\200\314|\326\307\177\000\000\060!\234h\377\177\000\000 >> !\234h\377\177\000\000\060!\234h\377\177\000\000\200\314|\326\307\177\000\000\001\000\000\000\000\000\000\000\340r\242\000\000\000\000\000p\251\215\000\000\000\000\000\340r\242\000\000\000\000\000\270\"\234h\377\177\000\000 >> vq\332\307\177\000\000\362\344s\333\307\177\000\000\000\000\000\000\000\000\000\000\060\000\000\000\060\000\000\000\060\"\234h\377\177\000\000P!\234h\377\177", >> '\000' , >> "\224*\207\333\307\177\000\000\005+\207\333\307\177\000\000\300\061\253\333\307\177\000\000\200\363\252\333\307\177\000\000c\226u\333\307\177\000\000 >> vq\332\307\177\000\000"... >> #6 0x00007fc7d70e8a63 in db_mysql_connect (ptr=ptr at entry=0x7fc7da717668) >> at my_con.c:105 >> reconnect = 0 '\000' >> ---Type to continue, or q to quit--- >> __FUNCTION__ = "db_mysql_connect" >> #7 0x00007fc7d70e94ff in db_mysql_new_connection (id=0x7fc7da717510) at >> my_con.c:165 >> ptr = 0x7fc7da717668 >> __FUNCTION__ = "db_mysql_new_connection" >> #8 0x00000000005ec4ba in db_do_init (url=, >> new_connection=0x7fc7d70e9417 ) at >> db/db.c:338 >> id = 0x7fc7da717510 >> con = >> res = 0x7fc7da717468 >> con_size = >> __FUNCTION__ = "db_do_init" >> #9 0x00007fc7d01044cd in dlg_connect_db (db_url=db_url at entry=0x7fc7d0340610 >> ) at dlg_db_handler.c:135 >> __FUNCTION__ = "dlg_connect_db" >> #10 0x00007fc7d0104511 in init_dlg_db (db_url=db_url at entry=0x7fc7d0340610 >> , >> dlg_hash_size=, db_update_period=60) at >> dlg_db_handler.c:176 >> __FUNCTION__ = "init_dlg_db" >> #11 0x00007fc7d0100c27 in mod_init () at dialog.c:900 >> n = >> __FUNCTION__ = "mod_init" >> #12 0x000000000050bfa7 in init_mod (m=0x7fc7da6face0, >> skip_others=skip_others at entry=0) at sr_module.c:697 >> dep = >> __FUNCTION__ = "init_mod" >> #13 0x000000000050c028 in init_mod (m=0x7fc7da6fb970, >> skip_others=skip_others at entry=0) at sr_module.c:678 >> dep = >> __FUNCTION__ = "init_mod" >> #14 0x000000000050c028 in init_mod (m=0x7fc7da6fbd28, >> skip_others=skip_others at entry=0) at sr_module.c:678 >> dep = >> __FUNCTION__ = "init_mod" >> ---Type to continue, or q to quit--- >> #15 0x000000000050c028 in init_mod (m=0x7fc7da6fbeb8, >> skip_others=skip_others at entry=0) at sr_module.c:678 >> dep = >> __FUNCTION__ = "init_mod" >> #16 0x000000000050c028 in init_mod (m=0x7fc7da6fc0c0, >> skip_others=skip_others at entry=0) at sr_module.c:678 >> dep = >> __FUNCTION__ = "init_mod" >> #17 0x000000000050c028 in init_mod (m=0x7fc7da6fc340, >> skip_others=skip_others at entry=0) at sr_module.c:678 >> dep = >> __FUNCTION__ = "init_mod" >> #18 0x000000000050c028 in init_mod (m=0x7fc7da6fc4d8, >> skip_others=skip_others at entry=0) at sr_module.c:678 >> dep = >> __FUNCTION__ = "init_mod" >> #19 0x000000000050c028 in init_mod (m=0x7fc7da6fd488, >> skip_others=skip_others at entry=0) at sr_module.c:678 >> dep = >> __FUNCTION__ = "init_mod" >> #20 0x000000000050c028 in init_mod (m=0x7fc7da6fde20, >> skip_others=skip_others at entry=0) at sr_module.c:678 >> dep = >> __FUNCTION__ = "init_mod" >> #21 0x000000000050f45d in init_modules () at sr_module.c:759 >> currentMod = 0x0 >> ret = >> __FUNCTION__ = "init_modules" >> #22 0x0000000000420351 in main (argc=, argv=> out>) at main.c:1421 >> c = >> r = >> tmp = 0x1
>> tmp_len = >> port = >> ---Type to continue, or q to quit--- >> proto = >> protos_no = >> options = 0x67e9f8 >> "f:cCm:M:b:l:n:N:rRvdDFEVhw:t:u:g:p:P:G:W:o:a:k:s:" >> ret = -1 >> seed = 961253727 >> rfd = >> __FUNCTION__ = "main" >> >> >> ------------------------------ >> >> Message: 2 >> Date: Wed, 30 Oct 2019 11:34:19 -0400 >> From: Liviu Chircu >> To: OpenSIPS users mailling list >> Subject: Re: [OpenSIPS-Users] Example of configuration "Full Sharing" >> Topology with NoSQL >> Message-ID: <33579d6f-a9ef-6e4d-5ab9-535450c25714 at opensips.org> >> Content-Type: text/plain; charset=utf-8; format=flowed >> >> Can you rather tell us more about the problem you are trying to solve, >> rather than >> inquiring about various (random) solutions? For example: >> >> * will your platform have 1 POP or multiple POPs? >> * do you want high availability for the user location nodes? >> >> Regards, >> >> -- >> Liviu Chircu >> OpenSIPS Developer >> http://www.opensips-solutions.com >> >> On 10/28/19 10:27 AM, Social Boh wrote: >> > >> > Hello, >> > >> > if i don't want use a SBC how can I known is a node of cluster is up >> > or down? which method do you advise? >> > >> > Thank you >> > >> >> >> >> ------------------------------ >> >> Subject: Digest Footer >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> >> ------------------------------ >> >> End of Users Digest, Vol 135, Issue 41 >> ************************************** >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: From razvan at opensips.org Mon Nov 25 04:44:25 2019 From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=) Date: Mon, 25 Nov 2019 11:44:25 +0200 Subject: [OpenSIPS-Users] =?utf-8?q?Too_Many_Hops_on_GCP=2E_=28R=C4=83zvan?= =?utf-8?q?_Crainea=29?= In-Reply-To: References: Message-ID: <54e843be-63f3-8621-2226-562ffb4baf45@opensips.org> I mean after calling `save()`, you should add an `exit;` Something like if (is_method("REGISTER")) { save(); exit; } Best regards, Răzvan On 11/22/19 7:25 PM, egemen ulus wrote: > Hi Răzvan , > > Yes it happens during registering. After a while it obviously gives 483 > too many hops because of max forward. What you mean when you say that > "exiting after 'save()' " > > Regards, > Egemen > ------------------------------------------------------------------------ > *Gönderen:* users-request at lists.opensips.org > adına Users > > *Gönderildi:* 18 Kasım 2019 Pazartesi 20:00 > *Kime:* users at lists.opensips.org > *Konu:* Users Digest, Vol 136, Issue 33 > Send Users mailing list submissions to >         users at lists.opensips.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > or, via email, send a message with subject or body 'help' to >         users-request at lists.opensips.org > > You can reach the person managing the list at >         users-owner at lists.opensips.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of Users digest..." > > > Today's Topics: > >    1. Too Many Hops on GCP. (egemen ulus) >    2. Re: Too Many Hops on GCP. (John Burke) >    3. Re: Too Many Hops on GCP. (Răzvan Crainea) >    4. Re: fix_nated_sdp() not taking effect (Răzvan Crainea) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Mon, 18 Nov 2019 05:36:38 +0000 > From: egemen ulus > To: "users at lists.opensips.org" > Subject: [OpenSIPS-Users] Too Many Hops on GCP. > Message-ID: > > > > Content-Type: text/plain; charset="iso-8859-9" > > Hi, > I've installed Opensips 2.4 on a GCP vm, when I try to register my > softphone to server, I am getting "483 too many hops" error. I am aware > that there is loop on the server side, it sends the packet itself. But > could not fix it. I have private and public IP addresses defined on > google cloud vm. > > Here is the my conf and logs: > > auto_aliases=no > listen=udp:10.138.0.3:5060  # private ip address > listen=tcp:10.138.0.3:5060 > > logs: > opensips[2671]: Nov 12 18:56:53 [2727] DBG:core:parse_headers: via > found, flags=22 > opensips[2671]: Nov 12 18:56:53 [2727] DBG:core:parse_via_param: found > param type 234, = <34.83.194.202>; state=6 > opensips[2671]: Nov 12 18:56:53 [2727] DBG:core:parse_via_param: found > param type 232, = ; state=16 > opensips[2671]: Nov 12 18:56:53 [2727] DBG:core:parse_via: end of header > reached, state=5 > > thank you for help! > Egemen > > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: > > > ------------------------------ > > Message: 2 > Date: Mon, 18 Nov 2019 00:38:17 -0500 > From: "John Burke" > To: > Subject: Re: [OpenSIPS-Users] Too Many Hops on GCP. > Message-ID: > > <3069669d235a571fd6eae01072a81360-1574055497 at ops-icewarp.voxtelesys.net> > > Content-Type: text/plain; charset="utf-8" > > I will be out of the office until 11/25. For immediate concerns please > contact support at voxtelesys.com or 402-403-4435. > > > > > ------------------------------ > > Message: 3 > Date: Mon, 18 Nov 2019 14:44:51 +0200 > From: Răzvan Crainea > To: users at lists.opensips.org > Subject: Re: [OpenSIPS-Users] Too Many Hops on GCP. > Message-ID: <4abd0344-58fd-7ccb-dc83-f9e01bb00821 at opensips.org> > Content-Type: text/plain; charset=UTF-8; format=flowed > > Hi, Egemen! > > Is the REGISTER looping? Are you exiting `exit;` after `save()`? > > Best regards, > Răzvan > > On 11/18/19 7:36 AM, egemen ulus wrote: >> Hi, >> I've installed Opensips 2.4 on a GCP vm, when I try to register my >> softphone to server, I am getting "483 too many hops" error. I am aware >> that there is loop on the server side, it sends the packet itself. But >> could not fix it. I have private and public IP addresses defined on >> google cloud vm. >> >> Here is the my conf and logs: >> >> /auto_aliases=no >> / >> /listen=udp:10.138.0.3:5060  # private ip address >> / >> /listen=tcp:10.138.0.3:5060/ >> >> logs: >> /opensips[2671]: Nov 12 18:56:53 [2727] DBG:core:parse_headers: via >> found, flags=22 >> / >> /opensips[2671]: Nov 12 18:56:53 [2727] DBG:core:parse_via_param: found >> param type 234, = <34.83.194.202>; state=6 >> / >> /opensips[2671]: Nov 12 18:56:53 [2727] DBG:core:parse_via_param: found >> param type 232, = ; state=16 >> / >> /opensips[2671]: Nov 12 18:56:53 [2727] DBG:core:parse_via: end of >> header reached, state=5/ >> / >> / >> /thank you for help!/ >> /Egemen/ >> >> >> _______________________________________________ >> 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 > > > > ------------------------------ > > Message: 4 > Date: Mon, 18 Nov 2019 14:48:28 +0200 > From: Răzvan Crainea > To: users at lists.opensips.org > Subject: Re: [OpenSIPS-Users] fix_nated_sdp() not taking effect > Message-ID: <2fb96c37-2ed0-5ddd-eacc-9bd249c88774 at opensips.org> > Content-Type: text/plain; charset=utf-8; format=flowed > > Yes, the problem is definitely the fact that you are calling > `rtpproxy_offer()` for the initial invite. Hence, when you run > `fix_nated_sdp()`, you're trying to change the same IP once again - this > is not possile in OpenSIPS. > But I wonder why you need the `fix_nated_sdp()` if you are using > RTPProxy. Can't you just use the `ip_address`[1] field to advertise the > proper IP int he c= line. > > [1] > https://opensips.org/html/docs/modules/3.0.x/rtpproxy.html#func_rtpproxy_offer > > Best regards, > Răzvan > > On 11/13/19 1:51 PM, Mark Farmer wrote: >> Hi everyone >> >> In my failure_route I'm routing to an Asterisk box for voicemail & I >> need to change the SDP c/o parameters to use the correct internal IP >> address but using fix_nated_sdp() is not taking effect. >> >> if (t_check_status("486|408|603")) { >>                  xlog("CUSTOM_LOG: User replied $T_reply_code - Routing >> to Asterisk Voicemail service."); >>                  prefix("VMR_"); >>                  rewritehostport("10.150.50.53:2404 >> "); >>                  force_send_socket(udp:10.150.50.51); >>                  fix_nated_sdp(10,"10.150.50.51"); >> >>                  if (!t_relay()) { >>                          send_reply(500,"Internal Error"); >>                  } >>                  exit; >> } >> >> I get the CUSTOM_LOG entry so I know that the route is executing. >> >> Maybe I'm doing something wrong with the flags, I've tried: >> fix_nated_sdp(2,"10.150.50.51"); >> fix_nated_sdp(8,"10.150.50.51"); >> fix_nated_sdp(10,"10.150.50.51"); >> >> But when I examine the SDP in the resulting invite, the c/o parameters >> are never changed. >> I'm using rtpengine_offer/answer in the initial routing, could it be >> related to that? >> >> I'm using OpenSIPS 3.0.1 >> >> Best regards >> Mark. >> >> >> >> _______________________________________________ >> 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 > > > > ------------------------------ > > Subject: Digest Footer > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > ------------------------------ > > End of Users Digest, Vol 136, Issue 33 > ************************************** > > _______________________________________________ > 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 From bogdan at opensips.org Mon Nov 25 06:39:04 2019 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Mon, 25 Nov 2019 13:39:04 +0200 Subject: [OpenSIPS-Users] OpenSIPS Summit 2020 - Registration Open Message-ID: <651e8c88-8f65-8baf-a25c-95b1fdf58929@opensips.org> Registration open OpenSIPS Summit 2020 May 5th-8th, 2020 Amsterdam, The Netherlands *The registration is now open!* Due to the popularity of OpenSIPS, conference attendees are drawn from many areas both technical and non-technical and include CTOs, Lead Engineers and Technical decision makers from small, medium and large enterprises, corporations and organizations worldwide. Don't miss the opportunity and join for the 2020 edition - the /registration is now open /. You may do individual registration or you can opt in for a /*Corporate Package*/ with an attractive discount. And did I mentioned about the /*Early Birds discount*/? Take advantage of it by registering by the end of January 2020! Register now We welcome everyone to join us and to be part of OpenSIPS Summit 2019, as attendee, speaker or sponsor. But do you want to stand out from the crowd? Then be an OpenSIPS Summit 2020 Sponsor - contact our team or email us! * * *Radisson Blu** **Rusland 17, 1012CK Amsterdam, The Netherlands* Meet us again at our familiar Venue, with the usual space and comfort! ** -- 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 Tue Nov 26 07:31:02 2019 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Tue, 26 Nov 2019 14:31:02 +0200 Subject: [OpenSIPS-Users] OpenSIPS @ SIPNOC 2019 Message-ID: <7bd02c95-300c-0379-67e5-38b211441d1d@opensips.org> Hi All On 5th of December, I will talk atSIPNOC 2019 [1] about the new STIR/SHAKEN implementation [2] in OpenSIPS 3.1, about the usage models and the associated risks. [1] https://www.sipforum.org/news-events/sipnoc-2019-overview/ [2] https://github.com/OpenSIPS/opensips/tree/master/modules/stir_shaken 10:45am – 11:15am: The Usage Models and Risks of STIR/SHAKEN, Seen from the Pragmatism of an Implementation. Abstracts: There are many things still to be defined and settled in STIR/SHAKEN from the regulatory perspective. Nevertheless, this presentation wants to bring this topic under scrutiny from the point of view of an implementation in the OpenSIPS SIP Server. So, what are the possible STIR/SHAKEN usage scenarios from the perspective of how the SIP traffic is handled and, more important, how the certificates are managed. While a SIP server may cover the full horizontal of authorization, inspection and verification processes, it is more relevant to see what are the possible models when comes to certificate managing. And definitely there is a need for coexistence between a certificate-agnostic model and a certificate self-managing model, in order to answer to future standardization and usage challenges. ITSP, Telcos and Carries are to deploy and use STIR/SHAKEN  implementations in the real word, so are they fully aware of the security and performance risks introduced by such a service? Well, the exercise of producing such a STIR/SHAKEN implementation is a good way to answer these questions and get yourselves ready. See you in Washington! -- Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com OpenSIPS Bootcamp Pre-Registration https://opensips.org/training/OpenSIPS_Bootcamp/ -------------- next part -------------- An HTML attachment was scrubbed... URL: From jon at hul.me.uk Tue Nov 26 07:52:56 2019 From: jon at hul.me.uk (Jonathan Hulme) Date: Tue, 26 Nov 2019 12:52:56 +0000 Subject: [OpenSIPS-Users] OpenSIPS @ SIPNOC 2019 In-Reply-To: <7bd02c95-300c-0379-67e5-38b211441d1d@opensips.org> References: <7bd02c95-300c-0379-67e5-38b211441d1d@opensips.org> Message-ID: Hi Bogdan,     Firstly good luck, secondly is this being streamed, or any recordings made available afterwards? Regards Jonathan On 26/11/2019 12:31, Bogdan-Andrei Iancu wrote: > Hi All > > On 5th of December, I will talk > atSIPNOC > 2019 [1] about the new STIR/SHAKEN implementation [2] in OpenSIPS 3.1, > about the usage models and the associated risks. > > > [1] https://www.sipforum.org/news-events/sipnoc-2019-overview/ > [2] https://github.com/OpenSIPS/opensips/tree/master/modules/stir_shaken > > > 10:45am – 11:15am: The Usage Models and Risks of STIR/SHAKEN, Seen > from the Pragmatism of an Implementation. > > Abstracts: > There are many things still to be defined and settled in STIR/SHAKEN > from the regulatory perspective. Nevertheless, this presentation wants > to bring this topic under scrutiny from the point of view of an > implementation in the OpenSIPS SIP Server. So, what are the possible > STIR/SHAKEN usage scenarios from the perspective of how the SIP > traffic is handled and, more important, how the certificates are > managed. While a SIP server may cover the full horizontal of > authorization, inspection and verification processes, it is more > relevant to see what are the possible models when comes to certificate > managing. And definitely there is a need for coexistence between a > certificate-agnostic model and a certificate self-managing model, in > order to answer to future standardization and usage challenges. ITSP, > Telcos and Carries are to deploy and use STIR/SHAKEN implementations > in the real word, so are they fully aware of the security and > performance risks introduced by such a service? Well, the exercise of > producing such a STIR/SHAKEN implementation is a good way to answer > these questions and get yourselves ready. > > > See you in Washington! > -- > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > OpenSIPS Bootcamp Pre-Registration > https://opensips.org/training/OpenSIPS_Bootcamp/ > > _______________________________________________ > 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 Johan at democon.be Tue Nov 26 08:02:52 2019 From: Johan at democon.be (Johan De Clercq) Date: Tue, 26 Nov 2019 14:02:52 +0100 Subject: [OpenSIPS-Users] OpenSIPS @ SIPNOC 2019 In-Reply-To: References: <7bd02c95-300c-0379-67e5-38b211441d1d@opensips.org> Message-ID: Sipnoc always share the presentations. Not sure about video. On Tue, 26 Nov 2019, 13:58 Jonathan Hulme, wrote: > Hi Bogdan, > > Firstly good luck, secondly is this being streamed, or any recordings > made available afterwards? > > Regards Jonathan > On 26/11/2019 12:31, Bogdan-Andrei Iancu wrote: > > Hi All > > On 5th of December, I will talk at > > SIPNOC 2019 [1] about the new STIR/SHAKEN implementation [2] in OpenSIPS > 3.1, about the usage models and the associated risks. > > > [1] https://www.sipforum.org/news-events/sipnoc-2019-overview/ > [2] https://github.com/OpenSIPS/opensips/tree/master/modules/stir_shaken > > > 10:45am – 11:15am: The Usage Models and Risks of STIR/SHAKEN, Seen from > the Pragmatism of an Implementation. > > Abstracts: > There are many things still to be defined and settled in STIR/SHAKEN from > the regulatory perspective. Nevertheless, this presentation wants to bring > this topic under scrutiny from the point of view of an implementation in > the OpenSIPS SIP Server. So, what are the possible STIR/SHAKEN usage > scenarios from the perspective of how the SIP traffic is handled and, more > important, how the certificates are managed. While a SIP server may cover > the full horizontal of authorization, inspection and verification > processes, it is more relevant to see what are the possible models when > comes to certificate managing. And definitely there is a need for > coexistence between a certificate-agnostic model and a certificate > self-managing model, in order to answer to future standardization and usage > challenges. ITSP, Telcos and Carries are to deploy and use STIR/SHAKEN > implementations in the real word, so are they fully aware of the security > and performance risks introduced by such a service? Well, the exercise of > producing such a STIR/SHAKEN implementation is a good way to answer these > questions and get yourselves ready. > > > See you in Washington! > > -- > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > OpenSIPS Bootcamp Pre-Registration > https://opensips.org/training/OpenSIPS_Bootcamp/ > > > _______________________________________________ > 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 volga629 at networklab.ca Tue Nov 26 08:24:45 2019 From: volga629 at networklab.ca (volga629 at networklab.ca) Date: Tue, 26 Nov 2019 09:24:45 -0400 Subject: [OpenSIPS-Users] OpenSIPS @ SIPNOC 2019 In-Reply-To: <7bd02c95-300c-0379-67e5-38b211441d1d@opensips.org> References: <7bd02c95-300c-0379-67e5-38b211441d1d@opensips.org> Message-ID: <1574774685.3038.1@skillsearch.ca> Good Luck, very important mission. You might willing bring to the table some improvement for the effort to provide better availability for CA list and peer verification process. If SIPNOC willing to organize web service where all involved parties will be able to register and go through verify process as trusted telco or voip provider. And then expose API from telcos or voip providers end points only ( must have fixed end point ip) to get verification information for CA list or peer information. That will have minimal impact on on application layer and will provide true trusted source of verification process. 1.3.3. ca_list (string) Path to a file containing trusted CA certificates for the verifier. The certificates must be in PEM format, one after another. Example 1.3. Set ca_list parameter ... modparam("stir_shaken", "ca_list", "/stir_certs/ca_list.pem") ... volga629 On Tue, Nov 26, 2019 at 14:31, Bogdan-Andrei Iancu wrote: > Hi All > > On 5th of December, I will talk at > SIPNOC > 2019 [1] about the new STIR/SHAKENimplementation [2] in OpenSIPS 3.1, > about the usage models and the associated risks. > > > [1] > [2] > > > > 10:45am – 11:15am: The Usage Models and Risks of STIR/SHAKEN, Seen > from the Pragmatism of an Implementation. > > Abstracts: > There are many things still to be defined and settled in STIR/SHAKEN > from the regulatory perspective. Nevertheless, this presentation > wants to bring this topic under scrutiny from the point of view of an > implementation in the OpenSIPS SIP Server. So, what are the possible > STIR/SHAKEN usage scenarios from the perspective of how the SIP > traffic is handled and, more important, how the certificates are > managed. While a SIP server may cover the full horizontal of > authorization, inspection and verification processes, it is more > relevant to see what are the possible models when comes to > certificate managing. And definitely there is a need for coexistence > between a certificate-agnostic model and a certificate self-managing > model, in order to answer to future standardization and usage > challenges. ITSP, Telcos and Carries are to deploy and use > STIR/SHAKEN implementations in the real word, so are they fully > aware of the security and performance risks introduced by such a > service? Well, the exercise of producing such a STIR/SHAKEN > implementation is a good way to answer these questions and get > yourselves ready. > > > See you in Washington! > -- > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > > OpenSIPS Bootcamp Pre-Registration > -------------- next part -------------- An HTML attachment was scrubbed... URL: From gmaruzz at gmail.com Tue Nov 26 09:07:31 2019 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Tue, 26 Nov 2019 15:07:31 +0100 Subject: [OpenSIPS-Users] OpenSIPS @ SIPNOC 2019 In-Reply-To: <1574774685.3038.1@skillsearch.ca> References: <7bd02c95-300c-0379-67e5-38b211441d1d@opensips.org> <1574774685.3038.1@skillsearch.ca> Message-ID: CONGRATULATIONS! YAY! On Tue, Nov 26, 2019 at 2:27 PM volga629 via Users wrote: > Good Luck, very important mission. > You might willing bring to the table some improvement for the effort to > provide better availability for CA list and peer verification process. > If SIPNOC willing to organize web service where all involved parties will > be able to register and go through verify process as trusted telco or > voip provider. > And then expose API from telcos or voip providers end points only ( must > have fixed end point ip) to get verification information for CA list or > peer information. > That will have minimal impact on on application layer and will provide > true trusted source of verification process. > > 1.3.3. ca_list (string) > > Path to a file containing trusted CA certificates for the > verifier. The certificates must be in PEM format, one after > another. > > Example 1.3. Set ca_list parameter > ... > modparam("stir_shaken", "ca_list", "/stir_certs/ca_list.pem") > ... > > > > volga629 > > > > > On Tue, Nov 26, 2019 at 14:31, Bogdan-Andrei Iancu > wrote: > > Hi All > > On 5th of December, I will talk at > > SIPNOC 2019 [1] about the new STIR/SHAKEN implementation [2] in OpenSIPS > 3.1, about the usage models and the associated risks. > > > [1] https://www.sipforum.org/news-events/sipnoc-2019-overview/ > [2] https://github.com/OpenSIPS/opensips/tree/master/modules/stir_shaken > > > 10:45am – 11:15am: The Usage Models and Risks of STIR/SHAKEN, Seen from > the Pragmatism of an Implementation. > > Abstracts: > There are many things still to be defined and settled in STIR/SHAKEN from > the regulatory perspective. Nevertheless, this presentation wants to bring > this topic under scrutiny from the point of view of an implementation in > the OpenSIPS SIP Server. So, what are the possible STIR/SHAKEN usage > scenarios from the perspective of how the SIP traffic is handled and, more > important, how the certificates are managed. While a SIP server may cover > the full horizontal of authorization, inspection and verification > processes, it is more relevant to see what are the possible models when > comes to certificate managing. And definitely there is a need for > coexistence between a certificate-agnostic model and a certificate > self-managing model, in order to answer to future standardization and usage > challenges. ITSP, Telcos and Carries are to deploy and use STIR/SHAKEN > implementations in the real word, so are they fully aware of the security > and performance risks introduced by such a service? Well, the exercise of > producing such a STIR/SHAKEN implementation is a good way to answer these > questions and get yourselves ready. > > > See you in Washington! > > -- > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > OpenSIPS Bootcamp Pre-Registration > https://opensips.org/training/OpenSIPS_Bootcamp/ > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -- Sincerely, Giovanni Maruzzelli OpenTelecom.IT cell: +39 347 266 56 18 -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Tue Nov 26 09:10:58 2019 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Tue, 26 Nov 2019 16:10:58 +0200 Subject: [OpenSIPS-Users] OpenSIPS @ SIPNOC 2019 In-Reply-To: References: <7bd02c95-300c-0379-67e5-38b211441d1d@opensips.org> Message-ID: Hi Jonathan, So far I'm not aware of anything like that, but I push the question to the organizers. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com OpenSIPS Bootcamp Pre-Registration https://opensips.org/training/OpenSIPS_Bootcamp/ On 11/26/19 2:52 PM, Jonathan Hulme wrote: > > Hi Bogdan, > >     Firstly good luck, secondly is this being streamed, or any > recordings made available afterwards? > > Regards Jonathan > > On 26/11/2019 12:31, Bogdan-Andrei Iancu wrote: >> Hi All >> >> On 5th of December, I will talk >> atSIPNOC >> 2019 [1] about the new STIR/SHAKEN implementation [2] in OpenSIPS >> 3.1, about the usage models and the associated risks. >> >> >> [1] https://www.sipforum.org/news-events/sipnoc-2019-overview/ >> [2] https://github.com/OpenSIPS/opensips/tree/master/modules/stir_shaken >> >> >> 10:45am – 11:15am: The Usage Models and Risks of STIR/SHAKEN, Seen >> from the Pragmatism of an Implementation. >> >> Abstracts: >> There are many things still to be defined and settled in STIR/SHAKEN >> from the regulatory perspective. Nevertheless, this presentation >> wants to bring this topic under scrutiny from the point of view of an >> implementation in the OpenSIPS SIP Server. So, what are the possible >> STIR/SHAKEN usage scenarios from the perspective of how the SIP >> traffic is handled and, more important, how the certificates are >> managed. While a SIP server may cover the full horizontal of >> authorization, inspection and verification processes, it is more >> relevant to see what are the possible models when comes to >> certificate managing. And definitely there is a need for coexistence >> between a certificate-agnostic model and a certificate self-managing >> model, in order to answer to future standardization and usage >> challenges. ITSP, Telcos and Carries are to deploy and use >> STIR/SHAKEN  implementations in the real word, so are they fully >> aware of the security and performance risks introduced by such a >> service? Well, the exercise of producing such a STIR/SHAKEN >> implementation is a good way to answer these questions and get >> yourselves ready. >> >> >> See you in Washington! >> -- >> Bogdan-Andrei Iancu >> >> OpenSIPS Founder and Developer >> https://www.opensips-solutions.com >> OpenSIPS Bootcamp Pre-Registration >> https://opensips.org/training/OpenSIPS_Bootcamp/ >> >> _______________________________________________ >> 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 info at byphone.eu Tue Nov 26 11:10:05 2019 From: info at byphone.eu (info) Date: Tue, 26 Nov 2019 16:10:05 +0000 Subject: [OpenSIPS-Users] Opensips 3 & HEP Message-ID: <03BC969A-23C4-4CD9-8190-089C61372B82@byphone.eu> Hello, I setup an opensips 3 with tls, I want to be able to easily troubleshot sip registrations, so I setup hep destination : loadmodule "proto_hep.so" listen = hep_udp:10.62.1.252:9060 modparam("proto_hep", "hep_id", "[hep_dst] 10.62.1.252:9061; version=2") modparam("proto_tls", "trace_destination", "hep_dst") modparam("proto_tls", "trace_on", 1) I have nothing with sngrep with the command : sngrep -L udp:10.62.1.252:9061 and when I do a tcpdump on port 9061, I have no traffic even in the case of registration with tls. Any idea ? Best regards Guillaume -------------- next part -------------- An HTML attachment was scrubbed... URL: From Ben.Newlin at genesys.com Tue Nov 26 12:20:42 2019 From: Ben.Newlin at genesys.com (Ben Newlin) Date: Tue, 26 Nov 2019 17:20:42 +0000 Subject: [OpenSIPS-Users] Rate Limit Module Implementation In-Reply-To: References: Message-ID: <9502EF36-5D94-4EB2-9A07-D6BF5C628698@genesys.com> Some testing of this has convinced me that the FEEDBACK algorithm is not actually tracking CPU Load as is claimed by the documentation. I believe it is actually checking CPU Utilization as a percentage. But since the values in the module are not exposed I cannot be sure. The lack of response on this question leads me to believe the rate_limit module, or perhaps just the NETWORK and FEEDBACK algorithms, are not widely used so I may just avoid them. Ben Newlin From: Users on behalf of Ben Newlin Reply-To: OpenSIPS users mailling list Date: Friday, November 15, 2019 at 5:09 PM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] Rate Limit Module Implementation As a follow up to this, when I tried to set the limit for the FEEDBACK algorithm to 0 for testing, I received this error: invalid limit for FEEDBACK algorithm (must be between 0 and 100) Per the documentation this algorithm should be tracking CPU Load, but the range being 0 to 100 makes me wonder if it is actually reporting a percentage, similar to the load statistics exported by OpenSIPS core [1]. Can anyone using this module clarify what unit the limit for this algorithm is expected to be in? The ambiguity is one of the reasons I had wanted to track the values being obtained for a time before actually engaging limited, which is what prompted my original question. [1] - https://www.opensips.org/Documentation/Interface-CoreStatistics-2-4#toc14 Ben Newlin From: Users on behalf of Ben Newlin Reply-To: OpenSIPS users mailling list Date: Thursday, November 14, 2019 at 3:45 PM To: OpenSIPS users mailling list Subject: [OpenSIPS-Users] Rate Limit Module Implementation Hello, We are looking to implement some rate limiting using the module, but I have a few questions and wanted to see if anyone has run into the same issues or has working experience with the module to answer them. The module provides $rl_count and also a counter value in the output of the rl_list command, but both of these values only reflect the number of rl_check calls that were made in the time window. In the case of TAILDROP, RED, and SBT algorithms, the count is directly related to the limit and drop_rate. But for the NETWORK and FEEDBACK algorithms, the counter is essentially meaningless. What we are looking for is a way to track the values these modules are using to make drop decisions, so that we can set our limit value appropriately. But unless I am missing something, there is no way to access these values for either NETWORK or FEEDBACK. Is that correct? Ben Newlin -------------- next part -------------- An HTML attachment was scrubbed... URL: From gmaruzz at gmail.com Tue Nov 26 15:43:28 2019 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Tue, 26 Nov 2019 21:43:28 +0100 Subject: [OpenSIPS-Users] Opensips 3 & HEP In-Reply-To: <03BC969A-23C4-4CD9-8190-089C61372B82@byphone.eu> References: <03BC969A-23C4-4CD9-8190-089C61372B82@byphone.eu> Message-ID: On Tue, Nov 26, 2019 at 5:12 PM info wrote: > Hello, > > > > I setup an opensips 3 with tls, I want to be able to easily troubleshot > sip registrations, so I setup hep destination : > > > > loadmodule "proto_hep.so" > > listen = hep_udp:10.62.1.252:9060 > > modparam("proto_hep", "hep_id", "[hep_dst] 10.62.1.252:9061; version=2") > > > > modparam("proto_tls", "trace_destination", "hep_dst") > > modparam("proto_tls", "trace_on", 1) > > > You only do half part, you must actually trace too... You want to add to your config something like this: #### tracer module loadmodule "tracer.so" modparam("tracer", "trace_on", 1) modparam("tracer", "trace_id", "[tid]uri=hep:hep_dst") ...... ######################################################### ######### only initial requests from this point ######### ######################################################### t_check_trans(); #Duplicate this sip dialog to sngrep if(!is_method("OPTIONS") ) { trace("tid"); } > I have nothing with sngrep with the command : > > > > sngrep -L udp:10.62.1.252:9061 > > > > and when I do a tcpdump on port 9061, I have no traffic even in the case > of registration with tls. > > > > > > Any idea ? > > > > Best regards > > > > Guillaume > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -- Sincerely, Giovanni Maruzzelli OpenTelecom.IT cell: +39 347 266 56 18 -------------- next part -------------- An HTML attachment was scrubbed... URL: From marnog at mac.com Tue Nov 26 15:46:26 2019 From: marnog at mac.com (Marcelo Nogueira) Date: Tue, 26 Nov 2019 21:46:26 +0100 Subject: [OpenSIPS-Users] Mid_Register config for private IP in the register. In-Reply-To: References: Message-ID: <2A78A98F-CC93-43A3-90C9-4D0BCE846FBB@mac.com> HI to all i have config the mid register using he tutorial and all for fine. Just on think. is possible config the mid_register when the register on main_register use the private IP on the USRLOC in main register are tables of location i need be the user at private IP right now are whit Public IP. thank you Marcelo From aconde at perfectpitchtech.com Tue Nov 26 16:43:32 2019 From: aconde at perfectpitchtech.com (Alexandro Conde) Date: Tue, 26 Nov 2019 15:43:32 -0600 Subject: [OpenSIPS-Users] Set AVP variables from Python script ? Message-ID: Hi all, Is there a way to set a AVP variable from a Python script ? I need to authenticate the endpoints with the username & password stored and managed on other system DB, I already have an API that can return the user information, so it will be great  to create a Python script that consume the API and return the username & password on a AVP variable to do the Authorization using the pv_www_authorize command. I prefer to program that on Python but if it's not possible it can be on LUA or Perl Right? (AVP_set & OpenSIPS::AVP) Thanks in Advance ! -- *Alexandro Conde Mtz.* Voip Specialist & Developer aconde at perfectpitchtech.com Facebook Twitter https://perfectpitchtech.com/ -------------- next part -------------- An HTML attachment was scrubbed... URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: janelbkljpnbapfj.png Type: image/png Size: 9612 bytes Desc: not available URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: gbpkpknoedlglipm.gif Type: image/gif Size: 125 bytes Desc: not available URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: kckinaifmblehjja.gif Type: image/gif Size: 248 bytes Desc: not available URL: From vladp at opensips.org Wed Nov 27 05:47:58 2019 From: vladp at opensips.org (Vlad Patrascu) Date: Wed, 27 Nov 2019 12:47:58 +0200 Subject: [OpenSIPS-Users] Set AVP variables from Python script ? In-Reply-To: References: Message-ID: <956e134f-aae3-ceac-f232-1ecefa1f3513@opensips.org> Hi Alexandro, Unfortunately, you cannot do this in Python, but as you've noticed you can use Lua or Perl. Starting from OpenSIPS 3.1 though, it will be possible to call OpenSIPS core script functions from Python, so 'pv_printf()' would facilitate writing AVPs from Python. Regards, Vlad Patrascu OpenSIPS Developer http://www.opensips-solutions.com On 11/26/19 11:43 PM, Alexandro Conde wrote: > > Hi all, > > Is there a way to set a AVP variable from a Python script ? > > I need to authenticate the endpoints with the username & password > stored and managed on other system DB, I already have an API that can > return the user information, so it will be great  to create a Python > script that consume the API and return the username & password on a > AVP variable to do the Authorization using the pv_www_authorize command. > > I prefer to program that on Python but if it's not possible it can be > on LUA or Perl Right? (AVP_set & OpenSIPS::AVP) > > Thanks in Advance ! > > -- > > *Alexandro Conde Mtz.* > Voip Specialist & Developer > aconde at perfectpitchtech.com > Facebook Twitter > > https://perfectpitchtech.com/ > > > _______________________________________________ > 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: janelbkljpnbapfj.png Type: image/png Size: 9612 bytes Desc: not available URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: gbpkpknoedlglipm.gif Type: image/gif Size: 125 bytes Desc: not available URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: kckinaifmblehjja.gif Type: image/gif Size: 248 bytes Desc: not available URL: From mickael at winlux.fr Wed Nov 27 11:10:08 2019 From: mickael at winlux.fr (Mickael Hubert) Date: Wed, 27 Nov 2019 17:10:08 +0100 Subject: [OpenSIPS-Users] Extract data from AVP Message-ID: Hi all, I want to extract the channel ID from a line in SDP (a=channel:d87c363c1b5b4f13 at speechrecog) I can extract this line, but I don't know how can I have only the ID (d87c363c1b5b4f13) Do you have a way for me please ? thanks in advance SDP example: *********** m=application 1544 TCP/MRCPv2 1 a=setup:passive a=connection:new a=channel:d87c363c1b5b4f13 at speechrecog a=cmid:1 m=audio 5002 RTP/AVP 8 101 ************* My conf: My result: "line in the SDP body is a=channel:d87c363c1b5b4f13 at speechrecog" I want only: d87c363c1b5b4f13 in specific avp onreply_route[reply_mrcp] { xlog("L_INFO","$avp(startlog) In REPLY ROUTE MRCP - fu : $fu , si : $si , rs: $rs\n"); if(has_body_part("application/sdp")) { if (search_body("m=application.*TCP\/MRCPv2")) { if(search_body("a=channel:.*")) { $var(i) = 0; $var(whileflag) = 0; while ($var(i) < 10 && $var(whileflag) != 1) { $avp(aline) = $(rb{sdp.line,a,$var(i)}); if($avp(aline)=~"a=channel") { xlog("line in the SDP body is $avp(aline)\n"); $var(whileflag) = 1; } $var(i) = $var(i) + 1; } xlog("var i = $var(i)\n"); } } } } -------------- next part -------------- An HTML attachment was scrubbed... URL: From Ben.Newlin at genesys.com Wed Nov 27 11:50:07 2019 From: Ben.Newlin at genesys.com (Ben Newlin) Date: Wed, 27 Nov 2019 16:50:07 +0000 Subject: [OpenSIPS-Users] Extract data from AVP In-Reply-To: References: Message-ID: Have you tried using the substitution transformation? https://www.opensips.org/Documentation/Script-Tran-2-4#toc82 Ben Newlin From: Users on behalf of Mickael Hubert Reply-To: OpenSIPS users mailling list Date: Wednesday, November 27, 2019 at 11:11 AM To: OpenSIPS users mailling list Subject: [OpenSIPS-Users] Extract data from AVP Hi all, I want to extract the channel ID from a line in SDP (a=channel:d87c363c1b5b4f13 at speechrecog) I can extract this line, but I don't know how can I have only the ID (d87c363c1b5b4f13) Do you have a way for me please ? thanks in advance SDP example: *********** m=application 1544 TCP/MRCPv2 1 a=setup:passive a=connection:new a=channel:d87c363c1b5b4f13 at speechrecog a=cmid:1 m=audio 5002 RTP/AVP 8 101 ************* My conf: My result: "line in the SDP body is a=channel:d87c363c1b5b4f13 at speechrecog" I want only: d87c363c1b5b4f13 in specific avp onreply_route[reply_mrcp] { xlog("L_INFO","$avp(startlog) In REPLY ROUTE MRCP - fu : $fu , si : $si , rs: $rs\n"); if(has_body_part("application/sdp")) { if (search_body("m=application.*TCP\/MRCPv2")) { if(search_body("a=channel:.*")) { $var(i) = 0; $var(whileflag) = 0; while ($var(i) < 10 && $var(whileflag) != 1) { $avp(aline) = $(rb{sdp.line,a,$var(i)}); if($avp(aline)=~"a=channel") { xlog("line in the SDP body is $avp(aline)\n"); $var(whileflag) = 1; } $var(i) = $var(i) + 1; } xlog("var i = $var(i)\n"); } } } } -------------- next part -------------- An HTML attachment was scrubbed... URL: From mickael at winlux.fr Wed Nov 27 13:27:29 2019 From: mickael at winlux.fr (Mickael Hubert) Date: Wed, 27 Nov 2019 19:27:29 +0100 Subject: [OpenSIPS-Users] Extract data from AVP In-Reply-To: References: Message-ID: Thanks Ben, yes I tried this: $var(toto) = $(avp(aline){re.subst,/.*:(.*)/\1/g}); # delete a=channel: xlog("line in the SDP body is $(var(toto){re.subst,/(.*)@.*/\1/g})\n"); xlog("line in the SDP body is $(var(toto){re.subst,/.*@(.*)/\1/g})\n"); It works like a charm, maybe there is a better way ... Le mer. 27 nov. 2019 à 17:51, Ben Newlin a écrit : > Have you tried using the substitution transformation? > > > > https://www.opensips.org/Documentation/Script-Tran-2-4#toc82 > > > > Ben Newlin > > > > *From: *Users on behalf of Mickael > Hubert > *Reply-To: *OpenSIPS users mailling list > *Date: *Wednesday, November 27, 2019 at 11:11 AM > *To: *OpenSIPS users mailling list > *Subject: *[OpenSIPS-Users] Extract data from AVP > > > > Hi all, > > I want to extract the channel ID from a line in SDP > (a=channel:d87c363c1b5b4f13 at speechrecog) > > I can extract this line, but I don't know how can I have only the ID > (d87c363c1b5b4f13) > > > > Do you have a way for me please ? > > > > thanks in advance > > > > SDP example: > > *********** > m=application 1544 TCP/MRCPv2 1 > a=setup:passive > a=connection:new > a=channel:d87c363c1b5b4f13 at speechrecog > a=cmid:1 > m=audio 5002 RTP/AVP 8 101 > ************* > > > > My conf: > > > > My result: "line in the SDP body is a=channel:d87c363c1b5b4f13 at speechrecog > " > > I want only: d87c363c1b5b4f13 in specific avp > > > > onreply_route[reply_mrcp] > > > > { > > > > xlog("L_INFO","$avp(startlog) In REPLY ROUTE MRCP - fu : $fu , si : $si , > rs: $rs\n"); > > > > if(has_body_part("application/sdp")) > > > > { > > > > if (search_body("m=application.*TCP\/MRCPv2")) > > { > > > > if(search_body("a=channel:.*")) { > > > > $var(i) > > = > > 0; > > > > $var(whileflag) > > = > > 0; > > > > while ($var(i) > > < > > 10 > > && > > $var(whileflag) > > != > > 1) > > > > { > > > > $avp(aline) > > = $(rb{sdp.line,a,$var(i)}); > > > > if($avp(aline)=~"a=channel") > > > > { > > > > xlog("line in the SDP body is $avp(aline)\n"); > > > > $var(whileflag) > > = > > 1; > > > > } > > > > $var(i) > > = > > $var(i) > > + > > 1; > > > > } > > > > xlog("var i = $var(i)\n"); > > > > } > > > > } > > > > } > > > > } > > > _______________________________________________ > 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 sxtang at iflytek.com Wed Nov 20 21:28:39 2019 From: sxtang at iflytek.com (sxtang) Date: Thu, 21 Nov 2019 10:28:39 +0800 Subject: [OpenSIPS-Users] confusion about the reply route and the failure route Message-ID: <000001d5a013$62f98480$28ec8d80$@iflytek.com> Hi: * I'm a little confused about the reply route and the failure route. * =Both can handle 404/408 responses in invite session. Both default action is to relay back the SIP reply. If I omit two routes in scripts(that meas not arm the reply route by using the t_on_reply("name") function or t_on_failure("name") function) or call t_relay() in these two routing scripts, will there be two responses sent ? -------------- next part -------------- An HTML attachment was scrubbed... URL: From ulus_egemen at hotmail.com Wed Nov 27 10:41:26 2019 From: ulus_egemen at hotmail.com (egemen ulus) Date: Wed, 27 Nov 2019 15:41:26 +0000 Subject: [OpenSIPS-Users] =?utf-8?q?Too_Many_Hops_on_GCP=2E_=28R=C4=83zvan?= =?utf-8?q?_Crainea=29?= In-Reply-To: References: Message-ID: Hi, I think there is no any missing part in REGISTER part as you see below; if ( !(is_method("REGISTER") ) ) { if (is_from_local()) { # authenticate if from local subscriber # authenticate all initial non-REGISTER request that pretend to be # generated by local subscriber (domain from FROM URI is local) if (!proxy_authorize("", "subscriber")) { proxy_challenge("", "0"); exit; } if (!db_check_from()) { send_reply("403","Forbidden auth ID"); exit; } consume_credentials(); # caller authenticated } else { # if caller is not local, then called number must be local if (!is_uri_host_local()) { send_reply("403","Relay Forbidden"); exit; } } } Regards Egemen ________________________________ Gönderen: users-request at lists.opensips.org adına Users Gönderildi: 25 Kasım 2019 Pazartesi 20:00 Kime: users at lists.opensips.org Konu: Users Digest, Vol 136, Issue 47 Send Users mailing list submissions to users at lists.opensips.org To subscribe or unsubscribe via the World Wide Web, visit http://lists.opensips.org/cgi-bin/mailman/listinfo/users or, via email, send a message with subject or body 'help' to users-request at lists.opensips.org You can reach the person managing the list at users-owner at lists.opensips.org When replying, please edit your Subject line so it is more specific than "Re: Contents of Users digest..." Today's Topics: 1. Re: Too Many Hops on GCP. (Răzvan Crainea) (Răzvan Crainea) 2. OpenSIPS Summit 2020 - Registration Open (Bogdan-Andrei Iancu) ---------------------------------------------------------------------- Message: 1 Date: Mon, 25 Nov 2019 11:44:25 +0200 From: Răzvan Crainea To: users at lists.opensips.org Subject: Re: [OpenSIPS-Users] Too Many Hops on GCP. (Răzvan Crainea) Message-ID: <54e843be-63f3-8621-2226-562ffb4baf45 at opensips.org> Content-Type: text/plain; charset=utf-8; format=flowed I mean after calling `save()`, you should add an `exit;` Something like if (is_method("REGISTER")) { save(); exit; } Best regards, Răzvan On 11/22/19 7:25 PM, egemen ulus wrote: > Hi Răzvan , > > Yes it happens during registering. After a while it obviously gives 483 > too many hops because of max forward. What you mean when you say that > "exiting after 'save()' " > > Regards, > Egemen > ------------------------------------------------------------------------ > *Gönderen:* users-request at lists.opensips.org > adına Users > > *Gönderildi:* 18 Kasım 2019 Pazartesi 20:00 > *Kime:* users at lists.opensips.org > *Konu:* Users Digest, Vol 136, Issue 33 > Send Users mailing list submissions to > users at lists.opensips.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > or, via email, send a message with subject or body 'help' to > users-request at lists.opensips.org > > You can reach the person managing the list at > users-owner at lists.opensips.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of Users digest..." > > > Today's Topics: > > 1. Too Many Hops on GCP. (egemen ulus) > 2. Re: Too Many Hops on GCP. (John Burke) > 3. Re: Too Many Hops on GCP. (Răzvan Crainea) > 4. Re: fix_nated_sdp() not taking effect (Răzvan Crainea) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Mon, 18 Nov 2019 05:36:38 +0000 > From: egemen ulus > To: "users at lists.opensips.org" > Subject: [OpenSIPS-Users] Too Many Hops on GCP. > Message-ID: > > > > Content-Type: text/plain; charset="iso-8859-9" > > Hi, > I've installed Opensips 2.4 on a GCP vm, when I try to register my > softphone to server, I am getting "483 too many hops" error. I am aware > that there is loop on the server side, it sends the packet itself. But > could not fix it. I have private and public IP addresses defined on > google cloud vm. > > Here is the my conf and logs: > > auto_aliases=no > listen=udp:10.138.0.3:5060 # private ip address > listen=tcp:10.138.0.3:5060 > > logs: > opensips[2671]: Nov 12 18:56:53 [2727] DBG:core:parse_headers: via > found, flags=22 > opensips[2671]: Nov 12 18:56:53 [2727] DBG:core:parse_via_param: found > param type 234, = <34.83.194.202>; state=6 > opensips[2671]: Nov 12 18:56:53 [2727] DBG:core:parse_via_param: found > param type 232, = ; state=16 > opensips[2671]: Nov 12 18:56:53 [2727] DBG:core:parse_via: end of header > reached, state=5 > > thank you for help! > Egemen > > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: > > > ------------------------------ > > Message: 2 > Date: Mon, 18 Nov 2019 00:38:17 -0500 > From: "John Burke" > To: > Subject: Re: [OpenSIPS-Users] Too Many Hops on GCP. > Message-ID: > > <3069669d235a571fd6eae01072a81360-1574055497 at ops-icewarp.voxtelesys.net> > > Content-Type: text/plain; charset="utf-8" > > I will be out of the office until 11/25. For immediate concerns please > contact support at voxtelesys.com or 402-403-4435. > > > > > ------------------------------ > > Message: 3 > Date: Mon, 18 Nov 2019 14:44:51 +0200 > From: Răzvan Crainea > To: users at lists.opensips.org > Subject: Re: [OpenSIPS-Users] Too Many Hops on GCP. > Message-ID: <4abd0344-58fd-7ccb-dc83-f9e01bb00821 at opensips.org> > Content-Type: text/plain; charset=UTF-8; format=flowed > > Hi, Egemen! > > Is the REGISTER looping? Are you exiting `exit;` after `save()`? > > Best regards, > Răzvan > > On 11/18/19 7:36 AM, egemen ulus wrote: >> Hi, >> I've installed Opensips 2.4 on a GCP vm, when I try to register my >> softphone to server, I am getting "483 too many hops" error. I am aware >> that there is loop on the server side, it sends the packet itself. But >> could not fix it. I have private and public IP addresses defined on >> google cloud vm. >> >> Here is the my conf and logs: >> >> /auto_aliases=no >> / >> /listen=udp:10.138.0.3:5060 # private ip address >> / >> /listen=tcp:10.138.0.3:5060/ >> >> logs: >> /opensips[2671]: Nov 12 18:56:53 [2727] DBG:core:parse_headers: via >> found, flags=22 >> / >> /opensips[2671]: Nov 12 18:56:53 [2727] DBG:core:parse_via_param: found >> param type 234, = <34.83.194.202>; state=6 >> / >> /opensips[2671]: Nov 12 18:56:53 [2727] DBG:core:parse_via_param: found >> param type 232, = ; state=16 >> / >> /opensips[2671]: Nov 12 18:56:53 [2727] DBG:core:parse_via: end of >> header reached, state=5/ >> / >> / >> /thank you for help!/ >> /Egemen/ >> >> >> _______________________________________________ >> 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 > > > > ------------------------------ > > Message: 4 > Date: Mon, 18 Nov 2019 14:48:28 +0200 > From: Răzvan Crainea > To: users at lists.opensips.org > Subject: Re: [OpenSIPS-Users] fix_nated_sdp() not taking effect > Message-ID: <2fb96c37-2ed0-5ddd-eacc-9bd249c88774 at opensips.org> > Content-Type: text/plain; charset=utf-8; format=flowed > > Yes, the problem is definitely the fact that you are calling > `rtpproxy_offer()` for the initial invite. Hence, when you run > `fix_nated_sdp()`, you're trying to change the same IP once again - this > is not possile in OpenSIPS. > But I wonder why you need the `fix_nated_sdp()` if you are using > RTPProxy. Can't you just use the `ip_address`[1] field to advertise the > proper IP int he c= line. > > [1] > https://opensips.org/html/docs/modules/3.0.x/rtpproxy.html#func_rtpproxy_offer > > Best regards, > Răzvan > > On 11/13/19 1:51 PM, Mark Farmer wrote: >> Hi everyone >> >> In my failure_route I'm routing to an Asterisk box for voicemail & I >> need to change the SDP c/o parameters to use the correct internal IP >> address but using fix_nated_sdp() is not taking effect. >> >> if (t_check_status("486|408|603")) { >> xlog("CUSTOM_LOG: User replied $T_reply_code - Routing >> to Asterisk Voicemail service."); >> prefix("VMR_"); >> rewritehostport("10.150.50.53:2404 >> "); >> force_send_socket(udp:10.150.50.51); >> fix_nated_sdp(10,"10.150.50.51"); >> >> if (!t_relay()) { >> send_reply(500,"Internal Error"); >> } >> exit; >> } >> >> I get the CUSTOM_LOG entry so I know that the route is executing. >> >> Maybe I'm doing something wrong with the flags, I've tried: >> fix_nated_sdp(2,"10.150.50.51"); >> fix_nated_sdp(8,"10.150.50.51"); >> fix_nated_sdp(10,"10.150.50.51"); >> >> But when I examine the SDP in the resulting invite, the c/o parameters >> are never changed. >> I'm using rtpengine_offer/answer in the initial routing, could it be >> related to that? >> >> I'm using OpenSIPS 3.0.1 >> >> Best regards >> Mark. >> >> >> >> _______________________________________________ >> 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 > > > > ------------------------------ > > Subject: Digest Footer > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > ------------------------------ > > End of Users Digest, Vol 136, Issue 33 > ************************************** > > _______________________________________________ > 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 ------------------------------ Message: 2 Date: Mon, 25 Nov 2019 13:39:04 +0200 From: Bogdan-Andrei Iancu To: "users at lists.opensips.org" , developensips , "business at lists.opensips.org" , "news at lists.opensips.org" Subject: [OpenSIPS-Users] OpenSIPS Summit 2020 - Registration Open Message-ID: <651e8c88-8f65-8baf-a25c-95b1fdf58929 at opensips.org> Content-Type: text/plain; charset="utf-8"; Format="flowed" Registration open OpenSIPS Summit 2020 May 5th-8th, 2020 Amsterdam, The Netherlands *The registration is now open!* Due to the popularity of OpenSIPS, conference attendees are drawn from many areas both technical and non-technical and include CTOs, Lead Engineers and Technical decision makers from small, medium and large enterprises, corporations and organizations worldwide. Don't miss the opportunity and join for the 2020 edition - the /registration is now open /. You may do individual registration or you can opt in for a /*Corporate Package*/ with an attractive discount. And did I mentioned about the /*Early Birds discount*/? Take advantage of it by registering by the end of January 2020! Register now We welcome everyone to join us and to be part of OpenSIPS Summit 2019, as attendee, speaker or sponsor. But do you want to stand out from the crowd? Then be an OpenSIPS Summit 2020 Sponsor - contact our team or email us! * * *Radisson Blu** **Rusland 17, 1012CK Amsterdam, The Netherlands* Meet us again at our familiar Venue, with the usual space and comfort! ** -- Bogdan-Andrei Iancu OpenSIPS Founder and Developer http://www.opensips-solutions.com -------------- next part -------------- An HTML attachment was scrubbed... URL: ------------------------------ Subject: Digest Footer _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ------------------------------ End of Users Digest, Vol 136, Issue 47 ************************************** -------------- next part -------------- An HTML attachment was scrubbed... URL: From Ben.Newlin at genesys.com Wed Nov 27 13:36:16 2019 From: Ben.Newlin at genesys.com (Ben Newlin) Date: Wed, 27 Nov 2019 18:36:16 +0000 Subject: [OpenSIPS-Users] Extract data from AVP In-Reply-To: References: Message-ID: Well, you could improve it by combining the two substitutions into one: $var(toto) = $(avp(aline){re.subst,/.*:(.*)@.*/\1/g}); Beyond that if this is something you do often you can use the dialplan module [1] to store the regex as a rule and apply it anywhere you want. It may have been designed for dialplan management but it is basically a database-backed regex module that can be used for anything. [1] https://opensips.org/html/docs/modules/2.4.x/dialplan.html Ben Newlin From: Users on behalf of Mickael Hubert Reply-To: OpenSIPS users mailling list Date: Wednesday, November 27, 2019 at 1:29 PM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] Extract data from AVP Thanks Ben, yes I tried this: $var(toto) = $(avp(aline){re.subst,/.*:(.*)/\1/g}); # delete a=channel: xlog("line in the SDP body is $(var(toto){re.subst,/(.*)@.*/\1/g})\n"); xlog("line in the SDP body is $(var(toto){re.subst,/.*@(.*)/\1/g})\n"); It works like a charm, maybe there is a better way ... Le mer. 27 nov. 2019 à 17:51, Ben Newlin > a écrit : Have you tried using the substitution transformation? https://www.opensips.org/Documentation/Script-Tran-2-4#toc82 Ben Newlin From: Users > on behalf of Mickael Hubert > Reply-To: OpenSIPS users mailling list > Date: Wednesday, November 27, 2019 at 11:11 AM To: OpenSIPS users mailling list > Subject: [OpenSIPS-Users] Extract data from AVP Hi all, I want to extract the channel ID from a line in SDP (a=channel:d87c363c1b5b4f13 at speechrecog) I can extract this line, but I don't know how can I have only the ID (d87c363c1b5b4f13) Do you have a way for me please ? thanks in advance SDP example: *********** m=application 1544 TCP/MRCPv2 1 a=setup:passive a=connection:new a=channel:d87c363c1b5b4f13 at speechrecog a=cmid:1 m=audio 5002 RTP/AVP 8 101 ************* My conf: My result: "line in the SDP body is a=channel:d87c363c1b5b4f13 at speechrecog" I want only: d87c363c1b5b4f13 in specific avp onreply_route[reply_mrcp] { xlog("L_INFO","$avp(startlog) In REPLY ROUTE MRCP - fu : $fu , si : $si , rs: $rs\n"); if(has_body_part("application/sdp")) { if (search_body("m=application.*TCP\/MRCPv2")) { if(search_body("a=channel:.*")) { $var(i) = 0; $var(whileflag) = 0; while ($var(i) < 10 && $var(whileflag) != 1) { $avp(aline) = $(rb{sdp.line,a,$var(i)}); if($avp(aline)=~"a=channel") { xlog("line in the SDP body is $avp(aline)\n"); $var(whileflag) = 1; } $var(i) = $var(i) + 1; } xlog("var i = $var(i)\n"); } } } } _______________________________________________ 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 ahmed.qadeer1989 at gmail.com Thu Nov 28 02:35:54 2019 From: ahmed.qadeer1989 at gmail.com (ahmed qadeer) Date: Thu, 28 Nov 2019 12:35:54 +0500 Subject: [OpenSIPS-Users] UDP workers 100% CPU usage + utimer_ticker: utimer Message-ID: Dear Team, We are facing issue on opensips 2.4.6 as CPU usage is high and average load goes from 80 to 100. Related information: opensips -V version: opensips 2.4.6 (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, sigio_rt, select. git revision: edef893c5 main.c compiled on 14:30:11 Oct 6 2019 with gcc 6.3.0 #################### children = 250 #################### opensipsctl fifo get_statistics load load:load:: 100 ##################### # opensipsctl fifo get_statistics 'shmem:' shmem:total_size:: 8388608000 shmem:used_size:: 255412840 shmem:real_used_size:: 381230072 shmem:max_used_size:: 4255484288 shmem:free_size:: 8007377928 shmem:fragments:: 342799 ###################### libssl version 1.1.1d-1 ###################### Please find below the error logs. Nov 28 12:31:21 debian /usr/local/sbin/opensips[4614]: WARNING:core:utimer_ticker: utimer task already scheduled for 83617750 ms (now 83618740 ms), it may overlap.. Nov 28 12:31:21 debian /usr/local/sbin/opensips[4614]: WARNING:core:utimer_ticker: utimer task already scheduled for 83617750 ms (now 83618840 ms), it may overlap.. Nov 28 12:31:21 debian /usr/local/sbin/opensips[4614]: WARNING:core:timer_ticker: timer task already scheduled for 83616850 ms (now 83618840 ms), it may overlap.. Nov 28 12:31:21 debian /usr/local/sbin/opensips[4614]: WARNING:core:utimer_ticker: utimer task already scheduled for 83617750 ms (now 83618940 ms), it may overlap.. Nov 28 12:31:21 debian /usr/local/sbin/opensips[4614]: WARNING:core:utimer_ticker: utimer task already scheduled for 83617750 ms (now 83619040 ms), it may overlap.. Nov 28 12:31:21 debian /usr/local/sbin/opensips[4614]: WARNING:core:utimer_ticker: utimer task already scheduled for 83617750 ms (now 83619140 ms), it may overlap.. Nov 28 12:31:22 debian /usr/local/sbin/opensips[4614]: WARNING:core:utimer_ticker: utimer task already scheduled for 83617750 ms (now 83619240 ms), it may overlap.. Nov 28 12:31:22 debian /usr/local/sbin/opensips[4614]: WARNING:core:utimer_ticker: utimer task already scheduled for 83617750 ms (now 83619340 ms), it may overlap.. Nov 28 12:31:22 debian /usr/local/sbin/opensips[4614]: WARNING:core:utimer_ticker: utimer task already scheduled for 83617750 ms (now 83619440 ms), it may overlap.. Nov 28 12:31:22 debian /usr/local/sbin/opensips[4614]: WARNING:core:utimer_ticker: utimer task already scheduled for 83617750 ms (now 83619540 ms), it may overlap.. Nov 28 12:31:22 debian /usr/local/sbin/opensips[4614]: WARNING:core:utimer_ticker: utimer task already scheduled for 83617750 ms (now 83619640 ms), it may overlap.. ^C ################################ # strace -p 4614 strace: Process 4614 attached select(0, NULL, NULL, NULL, {tv_sec=0, tv_usec=32938}) = 0 (Timeout) getpid() = 4614 sendto(9, "<132>Nov 28 12:31:44 /usr/local/"..., 175, MSG_NOSIGNAL, NULL, 0) = 175 select(0, NULL, NULL, NULL, {tv_sec=0, tv_usec=100000}) = 0 (Timeout) getpid() = 4614 sendto(9, "<132>Nov 28 12:31:44 /usr/local/"..., 175, MSG_NOSIGNAL, NULL, 0) = 175 select(0, NULL, NULL, NULL, {tv_sec=0, tv_usec=100000}) = 0 (Timeout) getpid() = 4614 sendto(9, "<132>Nov 28 12:31:44 /usr/local/"..., 175, MSG_NOSIGNAL, NULL, 0) = 175 select(0, NULL, NULL, NULL, {tv_sec=0, tv_usec=100000}) = 0 (Timeout) getpid() = 4614 sendto(9, "<132>Nov 28 12:31:44 /usr/local/"..., 175, MSG_NOSIGNAL, NULL, 0) = 175 select(0, NULL, NULL, NULL, {tv_sec=0, tv_usec=100000}) = 0 (Timeout) getpid() = 4614 sendto(9, "<132>Nov 28 12:31:44 /usr/local/"..., 175, MSG_NOSIGNAL, NULL, 0) = 175 select(0, NULL, NULL, NULL, {tv_sec=0, tv_usec=100000}) = 0 (Timeout) getpid() = 4614 sendto(9, "<132>Nov 28 12:31:44 /usr/local/"..., 175, MSG_NOSIGNAL, NULL, 0) = 175 write(4, "(\217\200uh\177\0\0", 8) = 8 write(4, "\260\216\200uh\177\0\0", 8) = 8 write(4, "@\216\200uh\177\0\0", 8) = 8 getpid() = 4614 sendto(9, "<132>Nov 28 12:31:44 /usr/local/"..., 172, MSG_NOSIGNAL, NULL, 0) = 172 write(4, "\200dXuh\177\0\0", 8) = 8 select(0, NULL, NULL, NULL, {tv_sec=0, tv_usec=100000}) = 0 (Timeout) getpid() = 4614 sendto(9, "<132>Nov 28 12:31:44 /usr/local/"..., 175, MSG_NOSIGNAL, NULL, 0) = 175 select(0, NULL, NULL, NULL, {tv_sec=0, tv_usec=100000}) = 0 (Timeout) getpid() = 4614 sendto(9, "<132>Nov 28 12:31:44 /usr/local/"..., 175, MSG_NOSIGNAL, NULL, 0) = 175 select(0, NULL, NULL, NULL, {tv_sec=0, tv_usec=100000}) = 0 (Timeout) getpid() = 4614 sendto(9, "<132>Nov 28 12:31:44 /usr/local/"..., 175, MSG_NOSIGNAL, NULL, 0) = 175 select(0, NULL, NULL, NULL, {tv_sec=0, tv_usec=100000}) = 0 (Timeout) getpid() = 4614 sendto(9, "<132>Nov 28 12:31:44 /usr/local/"..., 175, MSG_NOSIGNAL, NULL, 0) = 175 select(0, NULL, NULL, NULL, {tv_sec=0, tv_usec=100000}^Cstrace: Process 4614 detached ##################################### root 5414 6.6 12.0 8456272 3978616 ? R Nov27 92:27 /usr/local/sbin/opensips -P /var/run/opensips.pid -m 8000 root 5416 6.7 12.1 8456380 3990136 ? S Nov27 94:30 /usr/local/sbin/opensips -P /var/run/opensips.pid -m 8000 root 5417 6.6 12.0 8456272 3983700 ? S Nov27 93:05 /usr/local/sbin/opensips -P /var/run/opensips.pid -m 8000 root 5419 6.7 12.1 8456272 3988628 ? S Nov27 93:46 /usr/local/sbin/opensips -P /var/run/opensips.pid -m 8000 root 5420 6.7 12.0 8456272 3983044 ? R Nov27 94:22 /usr/local/sbin/opensips -P /var/run/opensips.pid -m 8000 root 5421 6.9 12.1 8456272 3985908 ? S Nov27 97:17 /usr/local/sbin/opensips -P /var/run/opensips.pid -m 8000 root 5428 7.5 12.1 8456388 3988480 ? S Nov27 105:44 /usr/local/sbin/opensips -P /var/run/opensips.pid -m 8000 root 5429 8.1 12.0 8456272 3975616 ? R Nov27 113:23 /usr/local/sbin/opensips -P /var/run/opensips.pid -m 8000 root 5430 6.9 12.3 8382540 4063756 ? S Nov27 97:23 /usr/local/sbin/opensips -P /var/run/opensips.pid -m 8000 From info at byphone.eu Thu Nov 28 04:20:50 2019 From: info at byphone.eu (info) Date: Thu, 28 Nov 2019 09:20:50 +0000 Subject: [OpenSIPS-Users] Opensips 3 & HEP In-Reply-To: References: <03BC969A-23C4-4CD9-8190-089C61372B82@byphone.eu> Message-ID: <465AB2A9-96C9-459A-930B-B639D2921C4E@byphone.eu> Hi, Thanks your reply, no hep packets are sent… loadmodule "proto_hep.so" listen = hep_udp:10.62.1.252:9060 modparam("proto_hep", "hep_id", "[hep_dst] 10.62.1.252:9061; version=2") loadmodule "tracer.so" modparam("tracer", "trace_on", 1) modparam("tracer", "trace_id", "[tid]uri=hep:hep_dst") ####### Routing Logic ######## # main request routing logic route{ if(!is_method("OPTIONS") ) { trace("tid"); } ….. And nothing in the log… Thanks De : Users au nom de Giovanni Maruzzelli Répondre à : "gmaruzz at opentelecom.it" , OpenSIPS users mailling list Date : mardi 26 novembre 2019 à 21:46 À : OpenSIPS users mailling list Objet : Re: [OpenSIPS-Users] Opensips 3 & HEP On Tue, Nov 26, 2019 at 5:12 PM info > wrote: Hello, I setup an opensips 3 with tls, I want to be able to easily troubleshot sip registrations, so I setup hep destination : loadmodule "proto_hep.so" listen = hep_udp:10.62.1.252:9060 modparam("proto_hep", "hep_id", "[hep_dst] 10.62.1.252:9061; version=2") modparam("proto_tls", "trace_destination", "hep_dst") modparam("proto_tls", "trace_on", 1) You only do half part, you must actually trace too... You want to add to your config something like this: #### tracer module loadmodule "tracer.so" modparam("tracer", "trace_on", 1) modparam("tracer", "trace_id", "[tid]uri=hep:hep_dst") ...... ######################################################### ######### only initial requests from this point ######### ######################################################### t_check_trans(); #Duplicate this sip dialog to sngrep if(!is_method("OPTIONS") ) { trace("tid"); } I have nothing with sngrep with the command : sngrep -L udp:10.62.1.252:9061 and when I do a tcpdump on port 9061, I have no traffic even in the case of registration with tls. Any idea ? Best regards Guillaume _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Sincerely, Giovanni Maruzzelli OpenTelecom.IT cell: +39 347 266 56 18 -------------- next part -------------- An HTML attachment was scrubbed... URL: From callum.guy at x-on.co.uk Fri Nov 29 10:55:02 2019 From: callum.guy at x-on.co.uk (Callum Guy) Date: Fri, 29 Nov 2019 15:55:02 +0000 Subject: [OpenSIPS-Users] Memory Leak - runtime flags? Message-ID: Hi All, I have recently deployed a new registrar and have been seeing a gradual increase in the memory footprint - enough that I'm having to expand the RAM (its virtualised) to ensure it doesn't run out. You can see a diff of the statistics collected last night at 11pm and today at 3pm here: https://gist.github.com/spacetourist/2103503674e134bd598c7f1e3a82674c/revisions Processes 5-9 are my UDP SIP receiver threads (autoscaled down from an initial footprint of 20 threads). Using 3.0.1 on CentOS 7 8GB RAM (soon to be 32GB!). Currently OpenSIPs is using all the RAM (minus OS usage) and 2GB of swap. Trying to use dialog and dr clustering if that is significant. Alos have NAT pings configured for all registrations (4000 at time of writing). I am using runtime configuration flags of "*-m 2048 -M 4096*" and am concerned that these were (way) too high, I think I've misinterpreted their meaning during initial setup. Is this a ridiculous setting for my environment? Is it just as simple as OpenSIPs being greedy with the memory such that it doesn't bother to free anything while each process free space remaining? Should my -M value * max number of processes fit into my RAM? I guess with an 8GB system that would mean dropping this to "-M 256"? I've done some research into the issue however I haven't found anything else that would be an obvious target so wondered if the community might have some ideas of where I can begin investigations. Many thanks, Callum -- *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 skumar at netlinkvoice.com Fri Nov 29 16:47:06 2019 From: skumar at netlinkvoice.com (Steve Sharad Kumar) Date: Fri, 29 Nov 2019 15:47:06 -0600 Subject: [OpenSIPS-Users] OpenSIPS 2.4 SIPREC no RTP Message-ID: Hi, I hope you are having a good weekend. I am posting this issue because I am struggling to make openSIPS with SIPREC to work. I have tried various ways but still no success for RTP. Here is my network topology - OpenSIPS - 10.10.10.174:5062 as X.X.X.X Orkaudio - 10.10.10.174:5060 and also listening on 127.0.0.1:5060 RTPPROXY - 10.10.10.174:22222 Both UACs are registered to openSIPS via its X.X.X.X public IP. I am able to send the call to orkaudio successfully but after the call disconnects orkaduio shows this message - session callid=B2B.195.839325.1575063266 localparty=200 remoteparty=100 duration=33 has no rtp And whenever openSIPS sends INVITE to orkaudio, Ork audio sends 200OK SDP back and then openSIPS sends ACK but never send any SDP response or any media back. And on that INVITE openSIPS sends 127.0.0.1 this IP in c element of SDP. Here are my rtpproxy settings - modparam("rtpproxy", "rtpproxy_sock", "udp:10.10.10.174:22222") modparam("rtpproxy", "default_set", 1) This is inside relay route - if (isflagset(NAT_FLAG) && has_body("application/sdp")) { rtpproxy_offer("froc","X.X.X.X","1","$var(siprec_rtpproxy_socket)"); xlog("RTPPROXY Sock used is $var(siprec_rtpproxy_socket)"); siprec_start_recording("sip:10.10.10.174:5060 ",,,"$var(siprec_rtpproxy_socket)"); } Please help me to fix this issue and I know I am messing up somewhere in osips config file. I will appreciate any help. Thanks and regards -------------- next part -------------- An HTML attachment was scrubbed... URL: From m.shirazi at gmail.com Sat Nov 30 08:39:01 2019 From: m.shirazi at gmail.com (Mehdi Shirazi) Date: Sat, 30 Nov 2019 17:09:01 +0330 Subject: [OpenSIPS-Users] Save custom usrloc without register Message-ID: Hi Is it possible to make a custom usrloc and save it without register message ? Regards.Shirazi -------------- next part -------------- An HTML attachment was scrubbed... URL: From david.villasmil.work at gmail.com Sat Nov 30 09:04:52 2019 From: david.villasmil.work at gmail.com (David Villasmil) Date: Sat, 30 Nov 2019 14:04:52 +0000 Subject: [OpenSIPS-Users] Save custom usrloc without register In-Reply-To: References: Message-ID: You can use kamctl Kamctl ul add Or Kamctl rpc ul.add Check out the help Hope that helps David On Sat, 30 Nov 2019 at 13:39, Mehdi Shirazi wrote: > Hi > Is it possible to make a custom usrloc and save it without register > message ? > > Regards.Shirazi > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -- Regards, David Villasmil email: david.villasmil.work at gmail.com phone: +34669448337 -------------- next part -------------- An HTML attachment was scrubbed... URL: From m.shirazi at gmail.com Sat Nov 30 10:16:38 2019 From: m.shirazi at gmail.com (Mehdi Shirazi) Date: Sat, 30 Nov 2019 18:46:38 +0330 Subject: [OpenSIPS-Users] Save custom usrloc without register Message-ID: Hi David ! Tank you for answer. Is there any way to do similar function of opensips-cli mi ul_add... from opensips.cfg ? Regards M.Shirazi >You can use kamctl >Kamctl ul add >Or >Kamctl rpc ul.add >Check out the help >Hope that helps >David -------------- next part -------------- An HTML attachment was scrubbed... URL: From Ben.Newlin at genesys.com Sat Nov 30 10:45:14 2019 From: Ben.Newlin at genesys.com (Ben Newlin) Date: Sat, 30 Nov 2019 15:45:14 +0000 Subject: [OpenSIPS-Users] Memory Leak - runtime flags? In-Reply-To: References: Message-ID: <18918680-A59B-4AD0-B3F9-A91D1C399532@genesys.com> Callum, It’s my understanding that OpenSIPS does not release memory back to the OS, but it also pre-allocates all memory at startup into its private pool and then allocates from that internally. Normally shared memory should be significantly higher than package memory. For reference, on our system we run with “-m 1024 -M 64” and that is sufficient for us to process very high traffic volume. We don’t do registration though, so that may affect the sizes you need. You are setting your package memory size to 4G, so that will allocate 4G memory for every package (process) that loads and then 2G for shared memory. That will use up all the memory on your machine extremely quickly for sure. The statistics you provided seem like the memory increase is consistent with higher traffic levels on the second reading. You can see in your case that all of your “pkmem” processes have an extremely high amount of free memory (~3GB!). But that memory is still allocated from the OS, so you are instructing OpenSIPS to allocate much more than your system memory right at startup. Your shared memory also has just under 2GB free, so you have a lot of headroom there too. Since OpenSIPS pre-allocates, the amount of memory being used by the system overall should be fairly steady; if it is continuously increasing that implies a leak somewhere. IIRC there are a few processes/modules/commands in OpenSIPS or libraries it uses that do allocate memory directly from the system and not from OpenSIPS’ pool. You may need to investigate some of those to find out where your memory is going, or look at other processes/daemons you have running that could be using that memory. Ben Newlin From: Users on behalf of Callum Guy Reply-To: OpenSIPS users mailling list Date: Friday, November 29, 2019 at 10:57 AM To: OpenSIPS users mailling list Subject: [OpenSIPS-Users] Memory Leak - runtime flags? Hi All, I have recently deployed a new registrar and have been seeing a gradual increase in the memory footprint - enough that I'm having to expand the RAM (its virtualised) to ensure it doesn't run out. You can see a diff of the statistics collected last night at 11pm and today at 3pm here: https://gist.github.com/spacetourist/2103503674e134bd598c7f1e3a82674c/revisions Processes 5-9 are my UDP SIP receiver threads (autoscaled down from an initial footprint of 20 threads). Using 3.0.1 on CentOS 7 8GB RAM (soon to be 32GB!). Currently OpenSIPs is using all the RAM (minus OS usage) and 2GB of swap. Trying to use dialog and dr clustering if that is significant. Alos have NAT pings configured for all registrations (4000 at time of writing). I am using runtime configuration flags of "-m 2048 -M 4096" and am concerned that these were (way) too high, I think I've misinterpreted their meaning during initial setup. Is this a ridiculous setting for my environment? Is it just as simple as OpenSIPs being greedy with the memory such that it doesn't bother to free anything while each process free space remaining? Should my -M value * max number of processes fit into my RAM? I guess with an 8GB system that would mean dropping this to "-M 256"? I've done some research into the issue however I haven't found anything else that would be an obvious target so wondered if the community might have some ideas of where I can begin investigations. Many thanks, Callum [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 david.villasmil.work at gmail.com Sat Nov 30 12:47:29 2019 From: david.villasmil.work at gmail.com (David Villasmil) Date: Sat, 30 Nov 2019 17:47:29 +0000 Subject: [OpenSIPS-Users] Save custom usrloc without register In-Reply-To: References: Message-ID: Whoops sorry! https://opensips.org/html/docs/modules/1.8.x/usrloc#id294643 On Sat, 30 Nov 2019 at 15:16, Mehdi Shirazi wrote: > Hi David ! > Tank you for answer. > Is there any way to do similar function of opensips-cli mi ul_add... from > opensips.cfg ? > > Regards > M.Shirazi > > >You can use kamctl > > >Kamctl ul add > > >Or > > >Kamctl rpc ul.add > > >Check out the help > > >Hope that helps > > >David > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -- Regards, David Villasmil email: david.villasmil.work at gmail.com phone: +34669448337 -------------- next part -------------- An HTML attachment was scrubbed... URL: From callum.guy at x-on.co.uk Sat Nov 30 17:51:00 2019 From: callum.guy at x-on.co.uk (Callum Guy) Date: Sat, 30 Nov 2019 22:51:00 +0000 Subject: [OpenSIPS-Users] Memory Leak - runtime flags? In-Reply-To: <18918680-A59B-4AD0-B3F9-A91D1C399532@genesys.com> References: <18918680-A59B-4AD0-B3F9-A91D1C399532@genesys.com> Message-ID: Hi Ben, Thank you for your reply and insight here, very helpful to know you're running a drastically different setting for the package memory. I presumed if it preallocated that I would have seen some issues during testing, hence I've ended up with figures that were intended to provide 75% of the system memory to the application. Memory usage had been creeping up all day at the time of writing however migrations to this platform had been on hold since the initial capture of memory usage although call traffic would have been relatively even during daytime hours where the increase continued. On that basis I'm still concerned that there is an issue with my config causing this growth however I've now increased available memory and restarted so I should have ample time to investigate this week, I'll report back any findings for the community benefit. I will give some serious thought to lowering the package allocation value once I've got to grips with the situation. Usefully this implementation shares a lot of common components to another variant which acts as a pure proxy and does not deal with registrations where I'm not seeing this issue so that will narrow down the search area somewhat. Thanks again for your time, Callum On Sat, 30 Nov 2019, 15:46 Ben Newlin, wrote: > Callum, > > > > It’s my understanding that OpenSIPS does not release memory back to the > OS, but it also pre-allocates all memory at startup into its private pool > and then allocates from that internally. Normally shared memory should be > significantly higher than package memory. For reference, on our system we > run with “-m 1024 -M 64” and that is sufficient for us to process very high > traffic volume. We don’t do registration though, so that may affect the > sizes you need. > > > > You are setting your package memory size to 4G, so that will allocate 4G > memory for every package (process) that loads and then 2G for shared > memory. That will use up all the memory on your machine extremely quickly > for sure. The statistics you provided seem like the memory increase is > consistent with higher traffic levels on the second reading. You can see in > your case that all of your “pkmem” processes have an extremely high amount > of free memory (~3GB!). But that memory is still allocated from the OS, so > you are instructing OpenSIPS to allocate much more than your system memory > right at startup. > > > > Your shared memory also has just under 2GB free, so you have a lot of > headroom there too. Since OpenSIPS pre-allocates, the amount of memory > being used by the system overall should be fairly steady; if it is > continuously increasing that implies a leak somewhere. IIRC there are a few > processes/modules/commands in OpenSIPS or libraries it uses that do > allocate memory directly from the system and not from OpenSIPS’ pool. You > may need to investigate some of those to find out where your memory is > going, or look at other processes/daemons you have running that could be > using that memory. > > > > Ben Newlin > > > > *From: *Users on behalf of Callum Guy < > callum.guy at x-on.co.uk> > *Reply-To: *OpenSIPS users mailling list > *Date: *Friday, November 29, 2019 at 10:57 AM > *To: *OpenSIPS users mailling list > *Subject: *[OpenSIPS-Users] Memory Leak - runtime flags? > > > > Hi All, > > > > I have recently deployed a new registrar and have been seeing a gradual > increase in the memory footprint - enough that I'm having to expand the RAM > (its virtualised) to ensure it doesn't run out. > > > > You can see a diff of the statistics collected last night at 11pm and > today at 3pm here: > https://gist.github.com/spacetourist/2103503674e134bd598c7f1e3a82674c/revisions > > > > Processes 5-9 are my UDP SIP receiver threads (autoscaled down from an > initial footprint of 20 threads). > > > > Using 3.0.1 on CentOS 7 8GB RAM (soon to be 32GB!). Currently OpenSIPs is > using all the RAM (minus OS usage) and 2GB of swap. Trying to use dialog > and dr clustering if that is significant. Alos have NAT pings configured > for all registrations (4000 at time of writing). > > > > I am using runtime configuration flags of "*-m 2048 -M 4096*" and am > concerned that these were (way) too high, I think I've misinterpreted their > meaning during initial setup. Is this a ridiculous setting for my > environment? Is it just as simple as OpenSIPs being greedy with the memory > such that it doesn't bother to free anything while each process free space > remaining? Should my -M value * max number of processes fit into my RAM? I > guess with an 8GB system that would mean dropping this to "-M 256"? > > > > I've done some research into the issue however I haven't found anything > else that would be an obvious target so wondered if the community might > have some ideas of where I can begin investigations. > > > > Many thanks, > > > > Callum > > > > [image: Image removed by sender.] > > > > *0333 332 0000 | www.x-on.co.uk | * *[image: > Image removed by sender.] [image: > Image removed by sender.] [image: 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 > -- *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: