From bogdan at opensips.org Mon May 2 14:05:04 2022 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Mon, 2 May 2022 17:05:04 +0300 Subject: [OpenSIPS-Users] Dispatcher issue in Active -> Probing State. In-Reply-To: References: Message-ID: <25fec283-a32a-4bb0-5c01-123cc3ec58da@opensips.org> Hi, Do you use the default threshold value ? Also, have you enabled log level 4 (debug) and locate the log line for the probing part, to understand OpenSIPS behavior upon probing ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com OpenSIPS eBootcamp 23rd May - 3rd June 2022 https://opensips.org/training/OpenSIPS_eBootcamp_2022/ On 4/22/22 11:15 AM, Vinayak Makwana via Users wrote: > Hi all, >          I having issue with dispatcher module in that when i execute > "opensipsctl dispatcher dump" command at that time i am getting like this > "URI:: sip:1.2.3.4:5060 state=Active > first_hit_counter=0"     but the trunk doesn't respond to the OPTIONS > packet and it needs to change state from Active to probing as per > below configuration but it doesn't work. > > Why does it not get into probing state after ds_ping_interval timeout > and ds_probing_threshhold reached? > I am using OpenSIPS Version : 2.4.11 > > Here's my configuration for dispatcher module > > loadmodule "dispatcher.so" > modparam("dispatcher","db_url","URL") > modparam("dispatcher", "ds_ping_from", "sip:opensips at pinger.com > ") > modparam("dispatcher", "ds_ping_interval", 30) > modparam("dispatcher", "ds_probing_sock", "udp:6.7.8.9:5060 > ") > modparam("dispatcher", "ds_probing_mode", 1) > modparam("dispatcher", "persistent_state", 1) > > Please Provide any suggestion regarding this > > Regards > Vinayak Makwana > > *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 bogdan at opensips.org Mon May 2 14:07:55 2022 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Mon, 2 May 2022 17:07:55 +0300 Subject: [OpenSIPS-Users] Media IP of Caller In-Reply-To: References: Message-ID: Hi Alex, No, there is no dedicated var. But "looking" into SDP is the correct approach. And you will see there ONLY what the device sent you (it may be a private IP, if the device is natted). But the real src ip where the RTP is coming from may be "visible" only to a rtp relay, not to OpenSIPS (it is not reflected in the SIP/SDP part). Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com OpenSIPS eBootcamp 23rd May - 3rd June 2022 https://opensips.org/training/OpenSIPS_eBootcamp_2022/ On 4/23/22 8:41 PM, Alexander Perkins wrote: > Hi All.  I have an interesting question - how can I get the media IP > of the caller? Not the signaling IP, but the media IP. Is there a > variable for that? > > I've tried this: $var(aline) = $(rb{sdp.line,c,1}), but it seems to be > bringing on the caller's private IP address. > >  Any help is appreciated. > > Thank you, > Alex > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Mon May 2 14:25:12 2022 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Mon, 2 May 2022 17:25:12 +0300 Subject: [OpenSIPS-Users] Clustering Presence opensips 3.2/3.3 in K8s environment In-Reply-To: References: <551985e3-ed35-9589-8624-54c1efe1247e@opensips.org> Message-ID: Hi Jonathan, If you set log level to 4 (DBG), do you see any logs when the pinging should be attempted ? Best Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com OpenSIPS eBootcamp 23rd May - 3rd June 2022 https://opensips.org/training/OpenSIPS_eBootcamp_2022/ On 4/27/22 12:30 AM, Jonathan Hunter wrote: > > Hi Bogdan. > > Just to follow up I have been trying to make openSIPS send SIP options > in the same environment to websocket registered users, and I just cant > get it to send even though the branch flags are set. > > I now have these settings defined for NAThelper; > > loadmodule "nathelper.so" > > modparam("nathelper", "natping_interval", 10) > > modparam("nathelper", "ping_nated_only", 0) > > modparam("nathelper", "sipping_method", "OPTIONS") > > modparam("nathelper", "sipping_bflag", "SIPPING_ENABLE") > > modparam("nathelper", "sipping_from", "sip:pinger@") > > modparam("nathelper", "received_avp", "$avp(rcv)") > > modparam("nathelper", "ping_threshold", 5) > > modparam("nathelper", "max_pings_lost", 3) > > modparam("nathelper", "natping_partitions", 4) > > modparam("nathelper", "remove_on_timeout_bflag", "SIPPING_RTO") > > modparam("nathelper", "natping_tcp", 1) > > modparam("nathelper", "cluster_id", 1) > > modparam("nathelper", "cluster_sharing_tag", "node/2=active") > > Also have pinging_mode set for usrloc, and this is in a > federation-cachedb-cluster with 2 opensips containers running > active/active. > > If I use t_new_request I can send a SIP OPTIONS message out, so I > assume I am missing a parameter or its mis configuration? Or could it > be environment, this is a k8s setup. > > Any advice/tips would be great. > > Thank you . > > Jon > > Sent from Mail for > Windows > > *From: *Jonathan Hunter > *Sent: *25 April 2022 15:28 > *To: *Bogdan-Andrei Iancu ; OpenSIPS users > mailling list > *Subject: *Re: [OpenSIPS-Users] Clustering Presence opensips 3.2/3.3 > in K8s environment > > Hi Bogdan, > > Thank you for the reply I can see there are tcp connections but I > don’t seem to get anything. I assume it may well be k8s related?. > > I also cant seem to get NAT ping working, I assume this should work > out to websocket connections as long as flags are set? > > The outputs are; > >   "ID": 1317975555, > >             "Type": "ws", > >             "State": 0, > >             "Remote": "10.10.51.228:35462", > >             "Local": "10.10.2.91:8081", > >             "Lifetime": "2022-04-25 14:11:31", > >             "Alias port": 35462 > >         } > >     ] > > And location shows; > > "AORs": [ > >                 { > >                     "AOR": "61067470a372a031a7495a1a@", > >                     "Contacts": [ > >                         { > >                             "Contact": > "sip:c0r0d0i7 at b0ek39eabrvf.invalid;transport=wss", > >                             "ContactID": "4544061655272656153", > >                             "Expires": 519, > >                             "Q": "", > >                             "Callid": "jbbvtjp2l7bujcom73s3", > >                             "Cseq": 2, > >                             "User-agent": "SIP.js/0.20.0", > >                             "Received": > "sip:10.10.51.228:35462;transport=ws", > >                             "State": "CS_NEW", > >                             "Flags": 0, > >                             "Cflags": "WS_DEVICE SIPPING_RTO > SIPPING_ENABLE", > >                             "Socket": "ws::8081", > >                             "Methods": 5439 > >                         } > >                     ] > >                 } > >             ] > > However I cant seem to also get opensips to send SIP keepalive with > these settings; > > #### NAThelper module > > loadmodule "nathelper.so" > > modparam("nathelper", "received_avp", "$avp(rcv)") > > modparam("nathelper", "natping_tcp",1) > > modparam("nathelper", "natping_interval", 5) > > modparam("nathelper", "sipping_bflag", "SIPPING_ENABLE") > > modparam("nathelper", "remove_on_timeout_bflag", "SIPPING_RTO") > > modparam("nathelper", "sipping_from", "sip:pinger at DOMAIN") > > modparam("nathelper", "max_pings_lost", 2) > > modparam("nathelper", "cluster_id", 9) > > modparam("nathelper", "cluster_sharing_tag", "node/2=active") > > Am I missing something here? > > Many thanks > > Jon > > Sent from Mail > > for Windows > > *From: *Bogdan-Andrei Iancu > *Sent: *18 April 2022 13:59 > *To: *OpenSIPS users mailling list ; > Jonathan Hunter > *Subject: *Re: [OpenSIPS-Users] Clustering Presence opensips 3.2/3.3 > in K8s environment > > Hi Jonathan, > > Maybe the k8s layer (the ingress ??) sticks its tails in there - could > you check at opensips level if the TCP conn is still seen as up ? Use > the mi list_tcp_conns MI function > https://www.opensips.org/Documentation/Interface-CoreMI-3-2#toc4 > > > > Best regard, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > OpenSIPS eBootcamp 23rd May - 3rd June 2022 > https://opensips.org/training/OpenSIPS_eBootcamp_2022/ > > On 4/13/22 6:08 PM, Jonathan Hunter wrote: > > Hi All, > > Has anyone managed to get presence working when using an > active/active opensips setup with k8s ? > > Everything works apart from presence, In particular when a > websocket user disconnects due to a client crash. > > I ideally would want to use event_route[E_CORE_TCP_DISCONNECT] to > then grab the disconnect when it comes in via websockets/tcp, > however I cant seem to get it to trigger. Could this be due to the > underlying hooks OpenSIPS uses to interact with with OS with TCP > or something else? > > As I would use the event route, to then remove the registration > from the location table, as otherwise I have duplicate entries in > both location and the presentity list. > > Is this something anyone else has encountered? > > I have tried using clustering with both presence and pua and have > same issues, whereby after an unwanted disconnect subsequent > NOTIFY messages contain more than one id per entity; > > entity="sip:61067470a372a031a7495a1a at domain" > > > > id="0x7ffe75896760">open xmlns="urn:ietf:params:xml:ns:pidf" > id="0x7ffd0716b390">open > > I need to stop this occurring ideally, any help much appreciated. > > Many thanks > > Jon > > Sent from Mail > > for Windows > > _______________________________________________ > > Users mailing list > > Users at lists.opensips.org > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -------------- next part -------------- An HTML attachment was scrubbed... URL: From vladp at opensips.org Mon May 2 14:26:40 2022 From: vladp at opensips.org (Vlad Patrascu) Date: Mon, 2 May 2022 17:26:40 +0300 Subject: [OpenSIPS-Users] Stir Shaken Verification issue In-Reply-To: References: Message-ID: <82aff749-7ff2-092c-9106-331095d40d2c@opensips.org> Hi Devang, The URL in the info param has nothing to do with the verification itself. I suspect you are somehow not using the proper certificate and/or CA, as the certificate generated by the script you mentioned should not be self signed, as the error indicates. Regards, -- Vlad Patrascu OpenSIPS Core Developer http://www.opensips-solutions.com On 25.04.2022 15:47, Devang Dhandhalya via Users wrote: > Hello All > I am testing STIR/SHAKEN calls using two servers. > calls originating to the first server adding identity header and when > sending calls to the second server for verification service at the > time of verification service i am getting below error . > error :437 , Unsupported Credential , Verification Fails with Return > code :-8 INFO:stir_shaken:verify_callback: certificate validation > failed: self signed certificate INFO:stir_shaken:w_stir_verify: > Invalid certificate > OpenSIPS Version : 3.2.2 I generate certificate using domain which > mapped with those 2 server : > https://github.com/OpenSIPIt/OpenSIPIt_00/blob/master/STIR_SHAKEN/Certgen/gencert.sh > > When the same server generates an identity header and verifies it at > that time not getting an issue call is working fine but when the > identity header generated by server 1 and going to verify it by server > 2 we get this above error. > Is it related to the URL which is in the info param ? When I open that > URL in the browser I am able to see the certificate. > Please suggest a solution for this issue. > Regards > Devang Dhandhalya > > *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 bogdan at opensips.org Mon May 2 14:29:19 2022 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Mon, 2 May 2022 17:29:19 +0300 Subject: [OpenSIPS-Users] Problem with fix_nated_contact In-Reply-To: References: Message-ID: <575521d6-a440-63f4-c516-9062aeb01d3b@opensips.org> Hi, Are there any errors when the "fixing" is done? The presence of a param should not impact here. Regards, Bogdan Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com OpenSIPS eBootcamp 23rd May - 3rd June 2022 https://opensips.org/training/OpenSIPS_eBootcamp_2022/ On 4/29/22 1:43 PM, Yury Kirsanov wrote: > Hi, > I'm using OpenSIPS 3.2.4 and recently run into following issue: > > Imagine simplest proxy setup - OpenSIPS just accepts new packet, for > example INVITE, changes destination using 'sethostport(....)' and then > issues 't_relay()' to forward the packet. Let's ignore replies and so on. > > If I'm doing a 'fix_nated_contact()' before sending this packet I'm > expecting Contact: field to be replaced with a source IP:port as per > manual. And this works if the Contact is in simple form like > 'sip:7777777 at 192.168.29.106:65033 > '. > > But if following Contact comes in OpenSIPS doesn't change it leaving > private IP in the contact: > > 'Contact: sip:7777777 at 192.168.29.106:65033;rinstance=2f59b175103f1088' > > Can you please let me know why is that happening? Thanks! > > Best regards, > Yury. > > _______________________________________________ > 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 y.kirsanov at gmail.com Mon May 2 17:07:47 2022 From: y.kirsanov at gmail.com (Yury Kirsanov) Date: Tue, 3 May 2022 03:07:47 +1000 Subject: [OpenSIPS-Users] Problem with fix_nated_contact In-Reply-To: <575521d6-a440-63f4-c516-9062aeb01d3b@opensips.org> References: <575521d6-a440-63f4-c516-9062aeb01d3b@opensips.org> Message-ID: Hi Bogdan, No, nothing in OpenSIPS logs, unfortunately. Here's another log, I'm doing 'fix_nated_register' in this case at the REGISTER route and doing 'fix_nated_contact()' at the very beginning of my script, just for the testing purpose. May 3 03:00:48 [REGISTER] [123456->123456] Forwarding REGISTER from sip:123456 at domain.com:5060, requested Expries: 60 to main registrar at sip: 172.16.4.22:5060 (84327f479c5d50e1634422f72a0b7619) May 3 03:00:48 [REPLY] [123456->123456] REGISTER 401 Unauthorized FROM 172.16.4.22:5060 (84327f479c5d50e1634422f72a0b7619) May 3 03:00:48 [REGISTER] [123456->123456] Request from 1XX.1XX.1XX.1XX:8001, domain domain.com (84327f479c5d50e1634422f72a0b7619) May 3 03:00:48 [REGISTER] [123456->123456] Forwarding REGISTER from sip:123456 at domain.com:5060, requested Expries: 60 to main registrar at sip: 172.16.4.22:5060 (84327f479c5d50e1634422f72a0b7619) May 3 03:00:48 [REPLY] [123456->123456] REGISTER 200 OK FROM 172.16.4.22:5060 (84327f479c5d50e1634422f72a0b7619) May 3 03:00:48 [REGREPLY] [123456->123456] Reply from 172.16.4.22:5060, code is 200 - OK, saving contact (84327f479c5d50e1634422f72a0b7619) May 3 03:00:48 [EVENT] Inserting contact sip:123456 at 192.168.1.36:8001 (569f6c324981335e0b33daf8fc88ed77) May 3 03:00:51 [OPTIONS] OPTIONS request from 172.16.4.22:5060 to sip:123456 at 172.16.4.254:5060, fu is sip:123456 at 1XX.1XX.1XX.1XX May 3 03:00:51 [OPTIONS] [123456->123456] SIP device sip:123456 at 172.16.4.254 found, relaying to sip:1XX.1XX.1XX.1XX:8001 (76f4319976c85e45b2ff916581912550) No errors in OpenSIPS logs. Here's output of 'opensips-cli -x mi fifo ul_dump': "AORs": [ { "AOR": "123456", "Contacts": [ { "Contact": "sip:123456 at 192.168.1.36:8001", "ContactID": "3713509073413807284", "Expires": 47, "Q": "", "Callid": "6_3941098626", "Cseq": 2, "User-agent": "Yealink SIP-T46G 28.83.0.120", "Received": "sip:1XX.1XX.1XX.1XX:8001", "State": "CS_SYNC", "Flags": 0, "Cflags": "", "Socket": "udp:1XX.1XX.1XX.1XX:5060", "Methods": 16383 } ] } Thanks and best regards, Yury. On Tue, May 3, 2022 at 12:29 AM Bogdan-Andrei Iancu wrote: > Hi, > > Are there any errors when the "fixing" is done? The presence of a param > should not impact here. > > Regards, > Bogdan > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > OpenSIPS eBootcamp 23rd May - 3rd June 2022 > https://opensips.org/training/OpenSIPS_eBootcamp_2022/ > > On 4/29/22 1:43 PM, Yury Kirsanov wrote: > > Hi, > I'm using OpenSIPS 3.2.4 and recently run into following issue: > > Imagine simplest proxy setup - OpenSIPS just accepts new packet, for > example INVITE, changes destination using 'sethostport(....)' and then > issues 't_relay()' to forward the packet. Let's ignore replies and so on. > > If I'm doing a 'fix_nated_contact()' before sending this packet I'm > expecting Contact: field to be replaced with a source IP:port as per > manual. And this works if the Contact is in simple form like ' > sip:7777777 at 192.168.29.106:65033'. > > But if following Contact comes in OpenSIPS doesn't change it leaving > private IP in the contact: > > 'Contact: sip:7777777 at 192.168.29.106:65033;rinstance=2f59b175103f1088' > > Can you please let me know why is that happening? Thanks! > > Best regards, > Yury. > > _______________________________________________ > Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From dan at dcsoftwaresystems.com Mon May 2 22:36:11 2022 From: dan at dcsoftwaresystems.com (Dan Cooper) Date: Mon, 2 May 2022 15:36:11 -0700 Subject: [OpenSIPS-Users] Missing contacts when handling 300 in OnReply route Message-ID: Hello, I'm handling a 300 Multiple Choices reply in an onreply_route as described the Opensips Blog post "Handling SIP Redirect Requests in realtime". The problem I'm seeing is while iterating the contacts through the $ct variable, only one of the two contact records in the SIP message is returned.  Instead it seemingly substitutes the initial contact for one of the returned contacts.  I think I should be getting all the contacts in the message and not the initial contact as I read the documentation. This system is running Opensips v3.2.6 Below is the SIP message (from sngrep), the onreply_route script, and the log output. 2022/05/02 21:12:54.183024 23.101.143.54:5060 -> 192.46.219.150:5060 SIP/2.0 300 Multiple Choices Via: SIP/2.0/UDP 192.46.219.150:5060;branch=z9hG4bKe01e.fcabc446.0;i=881249d1 Via: SIP/2.0/TCP 192.168.49.125:5080;received=67.172.178.64;branch=z9hG4bKBHcBe35Be3KjH Max-Forwards: 69 From: 15074529949 ;tag=jDtDtDmmXaXtQ To: 12187299716 ;tag=6ayH1QZrymXSS Call-ID: 79021022-44ff-123b-42ba-001fc69ce9dd CSeq: 51209194 INVITE Contact: "unknown" ;q=0.129 Contact: "unknown" ;q=0.128 User-Agent: FreeSWITCH-mod_sofia/1.10.5-release-17-25569c1631~64bit Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Reason: Q.850;cause=31;text="NORMAL_UNSPECIFIED" Content-Length: 0 X-FS-Display-Name: 12187299716 X-FS-Display-Number: sip:12187299716 at tfvoip.cloudapp.net Remote-Party-ID: "12187299716" ;party=calling;privacy=off;screen=no onreply_route[RTPENGINE]{ xlog("OnReply RTPENGINE: Status = $rs"); if (t_check_status("300")){ $var(n) = 0; xlog("OnReply RTPENGINE: contact $ct.fields(uri)"); xlog("OnReply RTPENGINE: last contact $(ct.fields(uri)[-1])"); xlog("OnReply RTPENGINE: first contact $(ct.fields(uri)[0])"); xlog("OnReply RTPENGINE: second contact $(ct.fields(uri)[1])"); xlog("OnReply RTPENGINE: third contact $(ct.fields(uri)[2])"); while ($(ct[$var(n)]) != NULL) { xlog("OnReply RTPENGINE: var(n) = $var(n)"); if ($(ct.fields(uri)[$var(n)]) != NULL) { xlog("OnReply RTPENGINE: appending branch = $(ct.fields(uri)[$var(n)]) : var(n) = $var(n)"); append_branch($(ct.fields(uri)[$var(n)])); } $var(n) = $var(n) + 1; } t_inject_branches("msg"); } if (has_body("application/sdp")){ # c flag handles duplicate c= lines in SDP rtpengine_answer("c"); } } May 2 21:12:54 opensipsprod1 /usr/sbin/opensips[30007]: OnReply RTPENGINE: Status = 300 May 2 21:12:54 opensipsprod1 /usr/sbin/opensips[30007]: DBG:tm:t_check_status: checked status is <300> May 2 21:12:54 opensipsprod1 /usr/sbin/opensips[30007]: DBG:core:parse_headers: flags=ffffffffffffffff May 2 21:12:54 opensipsprod1 /usr/sbin/opensips[30007]: OnReply RTPENGINE: contact sip:12187299716 at 23.101.143.54:5060 May 2 21:12:54 opensipsprod1 /usr/sbin/opensips[30007]: DBG:core:parse_headers: flags=ffffffffffffffff May 2 21:12:54 opensipsprod1 /usr/sbin/opensips[30007]: DBG:core:parse_params: Parsing params for:[q=0.128] May 2 21:12:54 opensipsprod1 /usr/sbin/opensips[30007]: OnReply RTPENGINE: last contact sip:12187299716 at 23.101.137.105 May 2 21:12:54 opensipsprod1 /usr/sbin/opensips[30007]: DBG:core:parse_headers: flags=ffffffffffffffff May 2 21:12:54 opensipsprod1 /usr/sbin/opensips[30007]: OnReply RTPENGINE: first contact sip:12187299716 at 23.101.143.54:5060 May 2 21:12:54 opensipsprod1 /usr/sbin/opensips[30007]: DBG:core:parse_headers: flags=ffffffffffffffff May 2 21:12:54 opensipsprod1 /usr/sbin/opensips[30007]: OnReply RTPENGINE: second contact sip:12187299716 at 23.101.137.105 May 2 21:12:54 opensipsprod1 /usr/sbin/opensips[30007]: DBG:core:parse_headers: flags=ffffffffffffffff May 2 21:12:54 opensipsprod1 /usr/sbin/opensips[30007]: OnReply RTPENGINE: third contact May 2 21:12:54 opensipsprod1 /usr/sbin/opensips[30007]: DBG:core:parse_headers: flags=ffffffffffffffff May 2 21:12:54 opensipsprod1 /usr/sbin/opensips[30007]: OnReply RTPENGINE: var(n) = 0 May 2 21:12:54 opensipsprod1 /usr/sbin/opensips[30007]: DBG:core:parse_headers: flags=ffffffffffffffff May 2 21:12:54 opensipsprod1 /usr/sbin/opensips[30007]: DBG:core:parse_headers: flags=ffffffffffffffff May 2 21:12:54 opensipsprod1 /usr/sbin/opensips[30007]: OnReply RTPENGINE: appending branch = sip:12187299716 at 23.101.143.54:5060 : va r(n) = 0 May 2 21:12:54 opensipsprod1 /usr/sbin/opensips[30007]: DBG:core:parse_headers: flags=ffffffffffffffff May 2 21:12:54 opensipsprod1 /usr/sbin/opensips[30007]: DBG:core:parse_headers: flags=ffffffffffffffff May 2 21:12:54 opensipsprod1 /usr/sbin/opensips[30007]: OnReply RTPENGINE: var(n) = 1 May 2 21:12:54 opensipsprod1 /usr/sbin/opensips[30007]: DBG:core:parse_headers: flags=ffffffffffffffff May 2 21:12:54 opensipsprod1 /usr/sbin/opensips[30007]: DBG:core:parse_headers: flags=ffffffffffffffff May 2 21:12:54 opensipsprod1 /usr/sbin/opensips[30007]: OnReply RTPENGINE: appending branch = sip:12187299716 at 23.101.137.105 : var(n) = 1 Thanks for your help Dan Cooper From yannick.lecoent at nexcom.fr Tue May 3 06:46:45 2022 From: yannick.lecoent at nexcom.fr (Yannick LE COENT) Date: Tue, 3 May 2022 08:46:45 +0200 Subject: [OpenSIPS-Users] OpenSIPS : no 407 retransmission In-Reply-To: <93b1545a-b0ed-0826-863b-3dfbf90371c4@nexcom.fr> References: <05c3bf91-3081-8815-e092-513035d82308@nexcom.fr> <93b1545a-b0ed-0826-863b-3dfbf90371c4@nexcom.fr> Message-ID: <9822c158-a891-f2bf-3c68-a3c02b17c671@nexcom.fr> Hello all, Could you tell if there is a way to enable 407 in stateful mode ? Thanks, Yannick Le 30/04/2022 à 18:14, Yannick LE COENT a écrit : > Hello Ben, > > Thanks for your answer. > > This problem occurs when OpenSIPS is not in charge of authenticating > the INVITE request, but this is done downstream. > I've sent this question to know if somebody has already solved this > kind of problem. > > Best regards, > Yannick > > Le 30/04/2022 à 16:15, Ben Newlin a écrit : >> >> I see. Apologies, I misunderstood the problem scenario. >> >> Ben Newlin >> >> *From: *Users on behalf of Yannick >> LE COENT >> *Date: *Saturday, April 30, 2022 at 5:46 AM >> *To: *OpenSIPS users mailling list >> *Subject: *Re: [OpenSIPS-Users] OpenSIPS : no 407 retransmission >> >> *EXTERNAL EMAIL - Please use caution with links and attachments * >> >> ------------------------------------------------------------------------ >> >> Hello Ben, >> >> The 407 is sent upstream, but when it is lost, it is not >> retransmitted by OpenSIPS. >> I do not have this problem with other negative status codes (e.g. 486). >> >> This is clearly explained in >> https://opensips.org/pub/opensips/1.8.6/src/ChangeLog >> >> 2012-03-21 18:36:58  Bogdan-Andrei Iancu, > org> >>     * [8811] : >> >>     TM will no longer do retransmission for the 407/401 replies >> (if no ACK is received) for both local or proxied replies. >> >> According to RFC 3261, retransmitting 407s/401s is probably a bad >> idea: >> >>  26.3.2.4 DoS Protection >> >> At the moment, my only solution is to use forward() instead of >> t_relay() in order to use the stateless mode. >> >> Yannick >> >> >> Yannick, >> >> The default behavior of OpenSIPS is to relay any received responses back upstream. If it is not doing that it would have to be because you are stopping it in the script. Take a look at the documentation for failure_route [1] which explains this. Check your own failure_route in your script; you must be doing something there that is telling OpenSIPS not to relay the 401/407 back upstream. >> >> [1]https://www.opensips.org/Documentation/Script-Routes-2-4#toc3 >> >> Ben Newlin >> >> From: Users on behalf of Yannick LE COENT >> >> Date: Friday, April 29, 2022 at 6:44 PM >> >> To:users at lists.opensips.org >> >> Subject: [OpenSIPS-Users] OpenSIPS : no 407 retransmission >> >> EXTERNAL EMAIL - Please use caution with links and attachments >> >> Hello, >> >> I'm using OpenSIPS as a proxy in front of another proxy. The 2nd proxy >> >> is in charge of authenticating the request. >> >> This is the callflow: >> >> Alice           OpenSIPS          Proxy#2 >> >>    | INVITE         |                | >> >>    |--------------->| INVITE         | >> >>    |      100 Tring |--------------->| >> >>    |<---------------|            407 | >> >>    |                |<---------------| >> >>    |                | ACK            | >> >>    |                |--------------->| >> >>    |            407 |                | >> >>    |     X<---------|                | >> >>    |                |                | >> >> Since OpenSIPS does not retransmit 401/407, the call setup gets stuck. >> >> What can I do ? >> >> If I set auto_100trying=1, that works, but this increases the number of >> >> INVITE retransmissions since 180Ringing are not received instantly. >> >> Do you have any suggestion ? >> >> Thanks, >> >> Yannick >> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Tue May 3 08:25:09 2022 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Tue, 3 May 2022 11:25:09 +0300 Subject: [OpenSIPS-Users] Problem with fix_nated_contact In-Reply-To: References: <575521d6-a440-63f4-c516-9062aeb01d3b@opensips.org> Message-ID: <59bf7f7f-8645-4297-4aea-f6b1c637e642@opensips.org> Hi Yury, For a REGISTER you should use the fix_nated_register() function. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com OpenSIPS eBootcamp 23rd May - 3rd June 2022 https://opensips.org/training/OpenSIPS_eBootcamp_2022/ On 5/2/22 8:07 PM, Yury Kirsanov wrote: > Hi Bogdan, > No, nothing in OpenSIPS logs, unfortunately. > > Here's another log, I'm doing 'fix_nated_register' in this case at the > REGISTER route and doing 'fix_nated_contact()' at the very > beginning of my script, just for the testing purpose. > > May  3 03:00:48 [REGISTER]      [123456->123456] Forwarding REGISTER > from sip:123456 at domain.com:5060 , > requested Expries: 60 to main registrar at sip:172.16.4.22:5060 > (84327f479c5d50e1634422f72a0b7619) > May  3 03:00:48 [REPLY]         [123456->123456] REGISTER 401 > Unauthorized FROM 172.16.4.22:5060 > (84327f479c5d50e1634422f72a0b7619) > May  3 03:00:48 [REGISTER]      [123456->123456] Request from > 1XX.1XX.1XX.1XX:8001, domain domain.com > (84327f479c5d50e1634422f72a0b7619) > May  3 03:00:48 [REGISTER]      [123456->123456] Forwarding REGISTER > from sip:123456 at domain.com:5060 , > requested Expries: 60 to main registrar at sip:172.16.4.22:5060 > (84327f479c5d50e1634422f72a0b7619) > May  3 03:00:48 [REPLY]         [123456->123456] REGISTER 200 OK FROM > 172.16.4.22:5060 > (84327f479c5d50e1634422f72a0b7619) > May  3 03:00:48 [REGREPLY]      [123456->123456] Reply from > 172.16.4.22:5060 , code is 200 - OK, saving > contact (84327f479c5d50e1634422f72a0b7619) > May  3 03:00:48 [EVENT] Inserting contact sip:123456 at 192.168.1.36:8001 > (569f6c324981335e0b33daf8fc88ed77) > May  3 03:00:51 [OPTIONS]       OPTIONS request from 172.16.4.22:5060 > to sip:123456 at 172.16.4.254:5060 > , fu is sip:123456 at 1XX.1XX.1XX.1XX > May  3 03:00:51 [OPTIONS]       [123456->123456] SIP device > sip:123456 at 172.16.4.254 found, > relaying to sip:1XX.1XX.1XX.1XX:8001 (76f4319976c85e45b2ff916581912550) > > No errors in OpenSIPS logs. Here's output of 'opensips-cli -x mi fifo > ul_dump': > >             "AORs": [ >                 { >                     "AOR": "123456", >                     "Contacts": [ >                         { >                             "Contact": "sip:123456 at 192.168.1.36:8001 > ", >                             "ContactID": "3713509073413807284", >                             "Expires": 47, >                             "Q": "", >                             "Callid": "6_3941098626", >                             "Cseq": 2, >                             "User-agent": "Yealink SIP-T46G 28.83.0.120", >                             "Received": "sip:1XX.1XX.1XX.1XX:8001", >                             "State": "CS_SYNC", >                             "Flags": 0, >                             "Cflags": "", >                             "Socket": "udp:1XX.1XX.1XX.1XX:5060", >                             "Methods": 16383 >                         } >                     ] >                 } > > Thanks and best regards, > Yury. > > On Tue, May 3, 2022 at 12:29 AM Bogdan-Andrei Iancu > > wrote: > > Hi, > > Are there any errors when the "fixing" is done? The presence of a > param should not impact here. > > Regards, > Bogdan > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > OpenSIPS eBootcamp 23rd May - 3rd June 2022 > https://opensips.org/training/OpenSIPS_eBootcamp_2022/ > > On 4/29/22 1:43 PM, Yury Kirsanov wrote: >> Hi, >> I'm using OpenSIPS 3.2.4 and recently run into following issue: >> >> Imagine simplest proxy setup - OpenSIPS just accepts new packet, >> for example INVITE, changes destination using 'sethostport(....)' >> and then issues 't_relay()' to forward the packet. Let's ignore >> replies and so on. >> >> If I'm doing a 'fix_nated_contact()' before sending this packet >> I'm expecting Contact: field to be replaced with a source IP:port >> as per manual. And this works if the Contact is in simple form >> like 'sip:7777777 at 192.168.29.106:65033 >> '. >> >> But if following Contact comes in OpenSIPS doesn't change it >> leaving private IP in the contact: >> >> 'Contact: >> sip:7777777 at 192.168.29.106:65033;rinstance=2f59b175103f1088' >> >> Can you please let me know why is that happening? Thanks! >> >> Best regards, >> Yury. >> >> _______________________________________________ >> 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 y.kirsanov at gmail.com Tue May 3 08:40:14 2022 From: y.kirsanov at gmail.com (Yury Kirsanov) Date: Tue, 3 May 2022 18:40:14 +1000 Subject: [OpenSIPS-Users] Problem with fix_nated_contact In-Reply-To: <59bf7f7f-8645-4297-4aea-f6b1c637e642@opensips.org> References: <575521d6-a440-63f4-c516-9062aeb01d3b@opensips.org> <59bf7f7f-8645-4297-4aea-f6b1c637e642@opensips.org> Message-ID: Hi Bogdan, Will fix_nated_register() overwrite results of a fix_nated_contact()? Second question - for 'OPTIONS' where Contact is available - should fix_nated_contact() replace it with the correct one? Where exactly does this function take the value to replace Contact with - from '$avp(received)' param? So it won't do anything if, for example, OPTIONS packet comes from my LAN Asterisk server and reaches the OpenSIPS LAN interface? Even though nat_uac_test(7) would confirm a RFC1918 private address fix_nated_contact() can't do much in this case, is that correct? Thanks a lot for your help! Best regards, Yury. On Tue, May 3, 2022 at 6:25 PM Bogdan-Andrei Iancu wrote: > Hi Yury, > > For a REGISTER you should use the fix_nated_register() function. > > Regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > OpenSIPS eBootcamp 23rd May - 3rd June 2022 > https://opensips.org/training/OpenSIPS_eBootcamp_2022/ > > On 5/2/22 8:07 PM, Yury Kirsanov wrote: > > Hi Bogdan, > No, nothing in OpenSIPS logs, unfortunately. > > Here's another log, I'm doing 'fix_nated_register' in this case at the > REGISTER route and doing 'fix_nated_contact()' at the very beginning of my > script, just for the testing purpose. > > May 3 03:00:48 [REGISTER] [123456->123456] Forwarding REGISTER from > sip:123456 at domain.com:5060, requested Expries: 60 to main registrar at > sip:172.16.4.22:5060 (84327f479c5d50e1634422f72a0b7619) > May 3 03:00:48 [REPLY] [123456->123456] REGISTER 401 Unauthorized > FROM 172.16.4.22:5060 (84327f479c5d50e1634422f72a0b7619) > May 3 03:00:48 [REGISTER] [123456->123456] Request from > 1XX.1XX.1XX.1XX:8001, domain domain.com (84327f479c5d50e1634422f72a0b7619) > May 3 03:00:48 [REGISTER] [123456->123456] Forwarding REGISTER from > sip:123456 at domain.com:5060, requested Expries: 60 to main registrar at > sip:172.16.4.22:5060 (84327f479c5d50e1634422f72a0b7619) > May 3 03:00:48 [REPLY] [123456->123456] REGISTER 200 OK FROM > 172.16.4.22:5060 (84327f479c5d50e1634422f72a0b7619) > May 3 03:00:48 [REGREPLY] [123456->123456] Reply from > 172.16.4.22:5060, code is 200 - OK, saving contact > (84327f479c5d50e1634422f72a0b7619) > May 3 03:00:48 [EVENT] Inserting contact sip:123456 at 192.168.1.36:8001 > (569f6c324981335e0b33daf8fc88ed77) > May 3 03:00:51 [OPTIONS] OPTIONS request from 172.16.4.22:5060 to > sip:123456 at 172.16.4.254:5060, fu is sip:123456 at 1XX.1XX.1XX.1XX > May 3 03:00:51 [OPTIONS] [123456->123456] SIP device > sip:123456 at 172.16.4.254 found, relaying to sip:1XX.1XX.1XX.1XX:8001 > (76f4319976c85e45b2ff916581912550) > > No errors in OpenSIPS logs. Here's output of 'opensips-cli -x mi fifo > ul_dump': > > "AORs": [ > { > "AOR": "123456", > "Contacts": [ > { > "Contact": "sip:123456 at 192.168.1.36:8001", > "ContactID": "3713509073413807284", > "Expires": 47, > "Q": "", > "Callid": "6_3941098626", > "Cseq": 2, > "User-agent": "Yealink SIP-T46G 28.83.0.120", > "Received": "sip:1XX.1XX.1XX.1XX:8001", > "State": "CS_SYNC", > "Flags": 0, > "Cflags": "", > "Socket": "udp:1XX.1XX.1XX.1XX:5060", > "Methods": 16383 > } > ] > } > > Thanks and best regards, > Yury. > > On Tue, May 3, 2022 at 12:29 AM Bogdan-Andrei Iancu > wrote: > >> Hi, >> >> Are there any errors when the "fixing" is done? The presence of a param >> should not impact here. >> >> Regards, >> Bogdan >> >> Bogdan-Andrei Iancu >> >> OpenSIPS Founder and Developer >> https://www.opensips-solutions.com >> OpenSIPS eBootcamp 23rd May - 3rd June 2022 >> https://opensips.org/training/OpenSIPS_eBootcamp_2022/ >> >> On 4/29/22 1:43 PM, Yury Kirsanov wrote: >> >> Hi, >> I'm using OpenSIPS 3.2.4 and recently run into following issue: >> >> Imagine simplest proxy setup - OpenSIPS just accepts new packet, for >> example INVITE, changes destination using 'sethostport(....)' and then >> issues 't_relay()' to forward the packet. Let's ignore replies and so on. >> >> If I'm doing a 'fix_nated_contact()' before sending this packet I'm >> expecting Contact: field to be replaced with a source IP:port as per >> manual. And this works if the Contact is in simple form like ' >> sip:7777777 at 192.168.29.106:65033'. >> >> But if following Contact comes in OpenSIPS doesn't change it leaving >> private IP in the contact: >> >> 'Contact: sip:7777777 at 192.168.29.106:65033;rinstance=2f59b175103f1088' >> >> Can you please let me know why is that happening? Thanks! >> >> Best regards, >> Yury. >> >> _______________________________________________ >> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Tue May 3 08:55:12 2022 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Tue, 3 May 2022 11:55:12 +0300 Subject: [OpenSIPS-Users] Problem with fix_nated_contact In-Reply-To: References: <575521d6-a440-63f4-c516-9062aeb01d3b@opensips.org> <59bf7f7f-8645-4297-4aea-f6b1c637e642@opensips.org> Message-ID: <1e06aab8-2dad-b0f7-0275-23572ed5eac8@opensips.org> See inline Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com OpenSIPS eBootcamp 23rd May - 3rd June 2022 https://opensips.org/training/OpenSIPS_eBootcamp_2022/ On 5/3/22 11:40 AM, Yury Kirsanov wrote: > Hi Bogdan, > Will fix_nated_register() overwrite results of a fix_nated_contact()? no, use either one, either the other, but not both in the same time - see the docs for the nathelper module for details. > > Second question - for 'OPTIONS' where Contact is available - should > fix_nated_contact() replace it with the correct one? yes, if you relay the OPTIONS > Where exactly does this function take the value to replace Contact > with - from '$avp(received)' param? no, it is taken from the network level, the src IP and port. > So it won't do anything if, for example, OPTIONS packet comes from my > LAN Asterisk server and reaches the OpenSIPS LAN interface? the fix_nated_xxxX() does not do any testing, it simply replace the host:port part of the contact with the src IP and port from network level. > Even though nat_uac_test(7) would confirm a RFC1918 private > address fix_nated_contact() can't do much in this case, is that correct? > > Thanks a lot for your help! > > Best regards, > Yury. > > On Tue, May 3, 2022 at 6:25 PM Bogdan-Andrei Iancu > > wrote: > > Hi Yury, > > For a REGISTER you should use the fix_nated_register() function. > > Regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > OpenSIPS eBootcamp 23rd May - 3rd June 2022 > https://opensips.org/training/OpenSIPS_eBootcamp_2022/ > > On 5/2/22 8:07 PM, Yury Kirsanov wrote: >> Hi Bogdan, >> No, nothing in OpenSIPS logs, unfortunately. >> >> Here's another log, I'm doing 'fix_nated_register' in this case >> at the REGISTER route and doing 'fix_nated_contact()' at the very >> beginning of my script, just for the testing purpose. >> >> May  3 03:00:48 [REGISTER]      [123456->123456] Forwarding >> REGISTER from sip:123456 at domain.com:5060 >> , requested Expries: 60 to >> main registrar at sip:172.16.4.22:5060 >> (84327f479c5d50e1634422f72a0b7619) >> May  3 03:00:48 [REPLY]         [123456->123456] REGISTER 401 >> Unauthorized FROM 172.16.4.22:5060 >> (84327f479c5d50e1634422f72a0b7619) >> May  3 03:00:48 [REGISTER]      [123456->123456] Request from >> 1XX.1XX.1XX.1XX:8001, domain domain.com >> (84327f479c5d50e1634422f72a0b7619) >> May  3 03:00:48 [REGISTER]      [123456->123456] Forwarding >> REGISTER from sip:123456 at domain.com:5060 >> , requested Expries: 60 to >> main registrar at sip:172.16.4.22:5060 >> (84327f479c5d50e1634422f72a0b7619) >> May  3 03:00:48 [REPLY]         [123456->123456] REGISTER 200 OK >> FROM 172.16.4.22:5060 >> (84327f479c5d50e1634422f72a0b7619) >> May  3 03:00:48 [REGREPLY]      [123456->123456] Reply from >> 172.16.4.22:5060 , code is 200 - OK, >> saving contact (84327f479c5d50e1634422f72a0b7619) >> May  3 03:00:48 [EVENT] Inserting contact >> sip:123456 at 192.168.1.36:8001 >> >> (569f6c324981335e0b33daf8fc88ed77) >> May  3 03:00:51 [OPTIONS]       OPTIONS request from >> 172.16.4.22:5060 to >> sip:123456 at 172.16.4.254:5060 >> , fu is >> sip:123456 at 1XX.1XX.1XX.1XX >> May  3 03:00:51 [OPTIONS]       [123456->123456] SIP device >> sip:123456 at 172.16.4.254 found, >> relaying to sip:1XX.1XX.1XX.1XX:8001 >> (76f4319976c85e45b2ff916581912550) >> >> No errors in OpenSIPS logs. Here's output of 'opensips-cli -x mi >> fifo ul_dump': >> >>             "AORs": [ >>                 { >>                     "AOR": "123456", >>                     "Contacts": [ >>                         { >>                             "Contact": >> "sip:123456 at 192.168.1.36:8001 ", >>                             "ContactID": "3713509073413807284", >>                             "Expires": 47, >>                             "Q": "", >>                             "Callid": "6_3941098626", >>                             "Cseq": 2, >>                             "User-agent": "Yealink SIP-T46G >> 28.83.0.120", >>                             "Received": "sip:1XX.1XX.1XX.1XX:8001", >>                             "State": "CS_SYNC", >>                             "Flags": 0, >>                             "Cflags": "", >>                             "Socket": "udp:1XX.1XX.1XX.1XX:5060", >>                             "Methods": 16383 >>                         } >>                     ] >>                 } >> >> Thanks and best regards, >> Yury. >> >> On Tue, May 3, 2022 at 12:29 AM Bogdan-Andrei Iancu >> > wrote: >> >> Hi, >> >> Are there any errors when the "fixing" is done? The presence >> of a param should not impact here. >> >> Regards, >> Bogdan >> >> Bogdan-Andrei Iancu >> >> OpenSIPS Founder and Developer >> https://www.opensips-solutions.com >> OpenSIPS eBootcamp 23rd May - 3rd June 2022 >> https://opensips.org/training/OpenSIPS_eBootcamp_2022/ >> >> On 4/29/22 1:43 PM, Yury Kirsanov wrote: >>> Hi, >>> I'm using OpenSIPS 3.2.4 and recently run into following issue: >>> >>> Imagine simplest proxy setup - OpenSIPS just accepts new >>> packet, for example INVITE, changes destination using >>> 'sethostport(....)' and then issues 't_relay()' to forward >>> the packet. Let's ignore replies and so on. >>> >>> If I'm doing a 'fix_nated_contact()' before sending this >>> packet I'm expecting Contact: field to be replaced with a >>> source IP:port as per manual. And this works if the Contact >>> is in simple form like 'sip:7777777 at 192.168.29.106:65033 >>> '. >>> >>> But if following Contact comes in OpenSIPS doesn't change it >>> leaving private IP in the contact: >>> >>> 'Contact: >>> sip:7777777 at 192.168.29.106:65033;rinstance=2f59b175103f1088' >>> >>> Can you please let me know why is that happening? Thanks! >>> >>> Best regards, >>> Yury. >>> >>> _______________________________________________ >>> 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 y.kirsanov at gmail.com Tue May 3 09:04:11 2022 From: y.kirsanov at gmail.com (Yury Kirsanov) Date: Tue, 3 May 2022 19:04:11 +1000 Subject: [OpenSIPS-Users] Problem with fix_nated_contact In-Reply-To: <1e06aab8-2dad-b0f7-0275-23572ed5eac8@opensips.org> References: <575521d6-a440-63f4-c516-9062aeb01d3b@opensips.org> <59bf7f7f-8645-4297-4aea-f6b1c637e642@opensips.org> <1e06aab8-2dad-b0f7-0275-23572ed5eac8@opensips.org> Message-ID: Hi Bogdan, Thanks for clarification, I'll try to monitor this and analyze it further! In regards to 'it simply replace the host:port part of the contact with the src IP and port from network level' for example if request is coming from 172.16.22.4:5060 and Contact is set to 'sip:172.167.22.4:5060', would fix_nated_contact() just replace Contact with the same values? As it doesn't have any 'received' parameter to replace this Contact with? Thanks! On Tue, May 3, 2022 at 6:55 PM Bogdan-Andrei Iancu wrote: > See inline > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > OpenSIPS eBootcamp 23rd May - 3rd June 2022 > https://opensips.org/training/OpenSIPS_eBootcamp_2022/ > > On 5/3/22 11:40 AM, Yury Kirsanov wrote: > > Hi Bogdan, > Will fix_nated_register() overwrite results of a fix_nated_contact()? > > no, use either one, either the other, but not both in the same time - see > the docs for the nathelper module for details. > > > Second question - for 'OPTIONS' where Contact is available - should > fix_nated_contact() replace it with the correct one? > > yes, if you relay the OPTIONS > > Where exactly does this function take the value to replace Contact with - > from '$avp(received)' param? > > no, it is taken from the network level, the src IP and port. > > So it won't do anything if, for example, OPTIONS packet comes from my LAN > Asterisk server and reaches the OpenSIPS LAN interface? > > the fix_nated_xxxX() does not do any testing, it simply replace the > host:port part of the contact with the src IP and port from network level. > > Even though nat_uac_test(7) would confirm a RFC1918 private > address fix_nated_contact() can't do much in this case, is that correct? > > Thanks a lot for your help! > > Best regards, > Yury. > > On Tue, May 3, 2022 at 6:25 PM Bogdan-Andrei Iancu > wrote: > >> Hi Yury, >> >> For a REGISTER you should use the fix_nated_register() function. >> >> Regards, >> >> Bogdan-Andrei Iancu >> >> OpenSIPS Founder and Developer >> https://www.opensips-solutions.com >> OpenSIPS eBootcamp 23rd May - 3rd June 2022 >> https://opensips.org/training/OpenSIPS_eBootcamp_2022/ >> >> On 5/2/22 8:07 PM, Yury Kirsanov wrote: >> >> Hi Bogdan, >> No, nothing in OpenSIPS logs, unfortunately. >> >> Here's another log, I'm doing 'fix_nated_register' in this case at the >> REGISTER route and doing 'fix_nated_contact()' at the very beginning of my >> script, just for the testing purpose. >> >> May 3 03:00:48 [REGISTER] [123456->123456] Forwarding REGISTER from >> sip:123456 at domain.com:5060, requested Expries: 60 to main registrar at >> sip:172.16.4.22:5060 (84327f479c5d50e1634422f72a0b7619) >> May 3 03:00:48 [REPLY] [123456->123456] REGISTER 401 >> Unauthorized FROM 172.16.4.22:5060 (84327f479c5d50e1634422f72a0b7619) >> May 3 03:00:48 [REGISTER] [123456->123456] Request from >> 1XX.1XX.1XX.1XX:8001, domain domain.com >> (84327f479c5d50e1634422f72a0b7619) >> May 3 03:00:48 [REGISTER] [123456->123456] Forwarding REGISTER from >> sip:123456 at domain.com:5060, requested Expries: 60 to main registrar at >> sip:172.16.4.22:5060 (84327f479c5d50e1634422f72a0b7619) >> May 3 03:00:48 [REPLY] [123456->123456] REGISTER 200 OK FROM >> 172.16.4.22:5060 (84327f479c5d50e1634422f72a0b7619) >> May 3 03:00:48 [REGREPLY] [123456->123456] Reply from >> 172.16.4.22:5060, code is 200 - OK, saving contact >> (84327f479c5d50e1634422f72a0b7619) >> May 3 03:00:48 [EVENT] Inserting contact sip:123456 at 192.168.1.36:8001 >> (569f6c324981335e0b33daf8fc88ed77) >> May 3 03:00:51 [OPTIONS] OPTIONS request from 172.16.4.22:5060 to >> sip:123456 at 172.16.4.254:5060, fu is sip:123456 at 1XX.1XX.1XX.1XX >> May 3 03:00:51 [OPTIONS] [123456->123456] SIP device >> sip:123456 at 172.16.4.254 found, relaying to sip:1XX.1XX.1XX.1XX:8001 >> (76f4319976c85e45b2ff916581912550) >> >> No errors in OpenSIPS logs. Here's output of 'opensips-cli -x mi fifo >> ul_dump': >> >> "AORs": [ >> { >> "AOR": "123456", >> "Contacts": [ >> { >> "Contact": "sip:123456 at 192.168.1.36:8001", >> "ContactID": "3713509073413807284", >> "Expires": 47, >> "Q": "", >> "Callid": "6_3941098626", >> "Cseq": 2, >> "User-agent": "Yealink SIP-T46G 28.83.0.120", >> "Received": "sip:1XX.1XX.1XX.1XX:8001", >> "State": "CS_SYNC", >> "Flags": 0, >> "Cflags": "", >> "Socket": "udp:1XX.1XX.1XX.1XX:5060", >> "Methods": 16383 >> } >> ] >> } >> >> Thanks and best regards, >> Yury. >> >> On Tue, May 3, 2022 at 12:29 AM Bogdan-Andrei Iancu >> wrote: >> >>> Hi, >>> >>> Are there any errors when the "fixing" is done? The presence of a param >>> should not impact here. >>> >>> Regards, >>> Bogdan >>> >>> Bogdan-Andrei Iancu >>> >>> OpenSIPS Founder and Developer >>> https://www.opensips-solutions.com >>> OpenSIPS eBootcamp 23rd May - 3rd June 2022 >>> https://opensips.org/training/OpenSIPS_eBootcamp_2022/ >>> >>> On 4/29/22 1:43 PM, Yury Kirsanov wrote: >>> >>> Hi, >>> I'm using OpenSIPS 3.2.4 and recently run into following issue: >>> >>> Imagine simplest proxy setup - OpenSIPS just accepts new packet, for >>> example INVITE, changes destination using 'sethostport(....)' and then >>> issues 't_relay()' to forward the packet. Let's ignore replies and so on. >>> >>> If I'm doing a 'fix_nated_contact()' before sending this packet I'm >>> expecting Contact: field to be replaced with a source IP:port as per >>> manual. And this works if the Contact is in simple form like ' >>> sip:7777777 at 192.168.29.106:65033'. >>> >>> But if following Contact comes in OpenSIPS doesn't change it leaving >>> private IP in the contact: >>> >>> 'Contact: sip:7777777 at 192.168.29.106:65033;rinstance=2f59b175103f1088' >>> >>> Can you please let me know why is that happening? Thanks! >>> >>> Best regards, >>> Yury. >>> >>> _______________________________________________ >>> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >>> >>> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Tue May 3 09:51:59 2022 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Tue, 3 May 2022 12:51:59 +0300 Subject: [OpenSIPS-Users] Problem with fix_nated_contact In-Reply-To: References: <575521d6-a440-63f4-c516-9062aeb01d3b@opensips.org> <59bf7f7f-8645-4297-4aea-f6b1c637e642@opensips.org> <1e06aab8-2dad-b0f7-0275-23572ed5eac8@opensips.org> Message-ID: <64399726-1463-fe73-7edf-68e17fdfe535@opensips.org> Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com OpenSIPS eBootcamp 23rd May - 3rd June 2022 https://opensips.org/training/OpenSIPS_eBootcamp_2022/ On 5/3/22 12:04 PM, Yury Kirsanov wrote: > Hi Bogdan, > Thanks for clarification, I'll try to monitor this and analyze it further! > > In regards to 'it simply replace the host:port part of the contact > with the src IP and port from network level' for example if request is > coming from 172.16.22.4:5060 and Contact is > set to 'sip:172.167.22.4:5060 ', would > fix_nated_contact() just replace Contact with the same values? Yes, correct. > As it doesn't have any 'received' parameter to replace this Contact > with? Thanks! $avp(received) it is an output of the fix_nated_register() function, not an input. > > On Tue, May 3, 2022 at 6:55 PM Bogdan-Andrei Iancu > > wrote: > > See inline > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > OpenSIPS eBootcamp 23rd May - 3rd June 2022 > https://opensips.org/training/OpenSIPS_eBootcamp_2022/ > > On 5/3/22 11:40 AM, Yury Kirsanov wrote: >> Hi Bogdan, >> Will fix_nated_register() overwrite results of a fix_nated_contact()? > no, use either one, either the other, but not both in the same > time - see the docs for the nathelper module for details. >> >> Second question - for 'OPTIONS' where Contact is available - >> should fix_nated_contact() replace it with the correct one? > yes, if you relay the OPTIONS >> Where exactly does this function take the value to replace >> Contact with - from '$avp(received)' param? > no, it is taken from the network level, the src IP and port. >> So it won't do anything if, for example, OPTIONS packet comes >> from my LAN Asterisk server and reaches the OpenSIPS LAN interface? > the fix_nated_xxxX() does not do any testing, it simply replace > the host:port part of the contact with the src IP and port from > network level. >> Even though nat_uac_test(7) would confirm a RFC1918 private >> address fix_nated_contact() can't do much in this case, is that >> correct? >> >> Thanks a lot for your help! >> >> Best regards, >> Yury. >> >> On Tue, May 3, 2022 at 6:25 PM Bogdan-Andrei Iancu >> > wrote: >> >> Hi Yury, >> >> For a REGISTER you should use the fix_nated_register() function. >> >> Regards, >> >> Bogdan-Andrei Iancu >> >> OpenSIPS Founder and Developer >> https://www.opensips-solutions.com >> OpenSIPS eBootcamp 23rd May - 3rd June 2022 >> https://opensips.org/training/OpenSIPS_eBootcamp_2022/ >> >> On 5/2/22 8:07 PM, Yury Kirsanov wrote: >>> Hi Bogdan, >>> No, nothing in OpenSIPS logs, unfortunately. >>> >>> Here's another log, I'm doing 'fix_nated_register' in this >>> case at the REGISTER route and doing 'fix_nated_contact()' >>> at the very beginning of my script, just for the testing >>> purpose. >>> >>> May  3 03:00:48 [REGISTER]  [123456->123456] Forwarding >>> REGISTER from sip:123456 at domain.com:5060 >>> , requested Expries: 60 >>> to main registrar at sip:172.16.4.22:5060 >>> (84327f479c5d50e1634422f72a0b7619) >>> May  3 03:00:48 [REPLY] [123456->123456] REGISTER 401 >>> Unauthorized FROM 172.16.4.22:5060 >>> (84327f479c5d50e1634422f72a0b7619) >>> May  3 03:00:48 [REGISTER]  [123456->123456] Request from >>> 1XX.1XX.1XX.1XX:8001, domain domain.com >>> (84327f479c5d50e1634422f72a0b7619) >>> May  3 03:00:48 [REGISTER]  [123456->123456] Forwarding >>> REGISTER from sip:123456 at domain.com:5060 >>> , requested Expries: 60 >>> to main registrar at sip:172.16.4.22:5060 >>> (84327f479c5d50e1634422f72a0b7619) >>> May  3 03:00:48 [REPLY] [123456->123456] REGISTER 200 OK >>> FROM 172.16.4.22:5060 >>> (84327f479c5d50e1634422f72a0b7619) >>> May  3 03:00:48 [REGREPLY]  [123456->123456] Reply from >>> 172.16.4.22:5060 , code is 200 - >>> OK, saving contact (84327f479c5d50e1634422f72a0b7619) >>> May  3 03:00:48 [EVENT] Inserting contact >>> sip:123456 at 192.168.1.36:8001 >>> >>> (569f6c324981335e0b33daf8fc88ed77) >>> May  3 03:00:51 [OPTIONS]       OPTIONS request from >>> 172.16.4.22:5060 to >>> sip:123456 at 172.16.4.254:5060 >>> , fu is >>> sip:123456 at 1XX.1XX.1XX.1XX >>> May  3 03:00:51 [OPTIONS] [123456->123456] SIP device >>> sip:123456 at 172.16.4.254 >>> found, relaying to sip:1XX.1XX.1XX.1XX:8001 >>> (76f4319976c85e45b2ff916581912550) >>> >>> No errors in OpenSIPS logs. Here's output of 'opensips-cli >>> -x mi fifo ul_dump': >>> >>>             "AORs": [ >>>                 { >>>                     "AOR": "123456", >>>                     "Contacts": [ >>>                         { >>>                             "Contact": >>> "sip:123456 at 192.168.1.36:8001 >>> ", >>>                             "ContactID": "3713509073413807284", >>>                             "Expires": 47, >>>                             "Q": "", >>>                             "Callid": "6_3941098626", >>>                             "Cseq": 2, >>>                             "User-agent": "Yealink SIP-T46G >>> 28.83.0.120", >>>                             "Received": >>> "sip:1XX.1XX.1XX.1XX:8001", >>>                             "State": "CS_SYNC", >>>                             "Flags": 0, >>>                             "Cflags": "", >>>                             "Socket": >>> "udp:1XX.1XX.1XX.1XX:5060", >>>                             "Methods": 16383 >>>                         } >>>                     ] >>>                 } >>> >>> Thanks and best regards, >>> Yury. >>> >>> On Tue, May 3, 2022 at 12:29 AM Bogdan-Andrei Iancu >>> > wrote: >>> >>> Hi, >>> >>> Are there any errors when the "fixing" is done? The >>> presence of a param should not impact here. >>> >>> Regards, >>> Bogdan >>> >>> Bogdan-Andrei Iancu >>> >>> OpenSIPS Founder and Developer >>> https://www.opensips-solutions.com >>> OpenSIPS eBootcamp 23rd May - 3rd June 2022 >>> https://opensips.org/training/OpenSIPS_eBootcamp_2022/ >>> >>> On 4/29/22 1:43 PM, Yury Kirsanov wrote: >>>> Hi, >>>> I'm using OpenSIPS 3.2.4 and recently run into >>>> following issue: >>>> >>>> Imagine simplest proxy setup - OpenSIPS just accepts >>>> new packet, for example INVITE, changes destination >>>> using 'sethostport(....)' and then issues 't_relay()' >>>> to forward the packet. Let's ignore replies and so on. >>>> >>>> If I'm doing a 'fix_nated_contact()' before sending >>>> this packet I'm expecting Contact: field to be replaced >>>> with a source IP:port as per manual. And this works if >>>> the Contact is in simple form like >>>> 'sip:7777777 at 192.168.29.106:65033 >>>> '. >>>> >>>> But if following Contact comes in OpenSIPS doesn't >>>> change it leaving private IP in the contact: >>>> >>>> 'Contact: >>>> sip:7777777 at 192.168.29.106:65033;rinstance=2f59b175103f1088' >>>> >>>> Can you please let me know why is that happening? Thanks! >>>> >>>> Best regards, >>>> Yury. >>>> >>>> _______________________________________________ >>>> Users mailing list >>>> Users at lists.opensips.org >>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Tue May 3 12:28:53 2022 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Tue, 3 May 2022 15:28:53 +0300 Subject: [OpenSIPS-Users] OpenSIPS Bootcamp training 2022 In-Reply-To: <98b76556-f672-470c-5a90-8ac2512df014@opensips.org> References: <98b76556-f672-470c-5a90-8ac2512df014@opensips.org> Message-ID: Tick, tack, only 3 weeks left to the training, first come, first served, seats are limited. https://opensips.org/training/OpenSIPS_eBootcamp_2022/ Best regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com OpenSIPS eBootcamp 23rd May - 3rd June 2022 https://opensips.org/training/OpenSIPS_eBootcamp_2022/ On 3/30/22 10:52 AM, Bogdan-Andrei Iancu wrote: > > > > 23rd May - 03rd June 2022, > > > online, worldwide > > > > *Study smarter, not harder! > * > > Take advantage of the *OpenSIPS Bootcamp* > and improve > your OpenSIPS skills - an in-cloud training, a ten days, 4 hours per > day (40 hours) intensive and practical training, covering > installation, configuration and administration on OpenSIPS. > > All the knowledge transferred to the students will be strongly backed > up by practice sessions where you will get hands-on experience in > handling OpenSIPS. The training is structured to be offer 50% / 50% > between the theoretical and practical sessions. > > Check Syllabus > > > *Early Birds open* > > The Early Bird 10% discount is available for registrations before > /*11th of April 2022*/, so do not miss the opportunity. The number of > seats is limited, so be sure and book a seat now. Keep in mind that a > 10% group discount is also available - grab your work mate and start > learning more OpenSIPS together . > . > > Register Now > > > *Certified training saves time and money* > > OpenSIPS mistakes are easily avoided if you get proper training! > Companies that use OpenSIPS waste time and money when they don't have > a trained engineer on staff. Searching on Google, waiting on IRC, even > the latency in mailing list replies takes it's toll over time. Take > this rare opportunity to train your employees with the project members > themselves. > > > Any questions? do not hesitate to contact us > ! > > ------------------------------------------------------------------------ > You received this email as part of your relationship with the OpenSIPS > Project. > If you do not want to receive any more news, please email to > unsubscribe . > > > -- > Bogdan-Andrei Iancu > OpenSIPS Founder and 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 daniel.zanutti at gmail.com Tue May 3 13:16:54 2022 From: daniel.zanutti at gmail.com (Daniel Zanutti) Date: Tue, 3 May 2022 10:16:54 -0300 Subject: [OpenSIPS-Users] OpenSIPS : no 407 retransmission In-Reply-To: <9822c158-a891-f2bf-3c68-a3c02b17c671@nexcom.fr> References: <05c3bf91-3081-8815-e092-513035d82308@nexcom.fr> <93b1545a-b0ed-0826-863b-3dfbf90371c4@nexcom.fr> <9822c158-a891-f2bf-3c68-a3c02b17c671@nexcom.fr> Message-ID: Generate in Stateful -> www_challenge or proxy_challenge? https://opensips.org/html/docs/modules/3.2.x/auth.html Is this what you are looking for? On Tue, May 3, 2022 at 3:50 AM Yannick LE COENT wrote: > Hello all, > > Could you tell if there is a way to enable 407 in stateful mode ? > > Thanks, > Yannick > > Le 30/04/2022 à 18:14, Yannick LE COENT a écrit : > > Hello Ben, > > Thanks for your answer. > > This problem occurs when OpenSIPS is not in charge of authenticating the > INVITE request, but this is done downstream. > I've sent this question to know if somebody has already solved this kind > of problem. > > Best regards, > Yannick > > Le 30/04/2022 à 16:15, Ben Newlin a écrit : > > I see. Apologies, I misunderstood the problem scenario. > > > > Ben Newlin > > > > *From: *Users > on behalf of Yannick LE COENT > > *Date: *Saturday, April 30, 2022 at 5:46 AM > *To: *OpenSIPS users mailling list > > *Subject: *Re: [OpenSIPS-Users] OpenSIPS : no 407 retransmission > > *EXTERNAL EMAIL - Please use caution with links and attachments * > > > ------------------------------ > > Hello Ben, > > The 407 is sent upstream, but when it is lost, it is not retransmitted by > OpenSIPS. > I do not have this problem with other negative status codes (e.g. 486). > > This is clearly explained in > https://opensips.org/pub/opensips/1.8.6/src/ChangeLog > > 2012-03-21 18:36:58 Bogdan-Andrei Iancu, > * [8811] : > > TM will no longer do retransmission for the 407/401 replies (if no ACK > is received) for both local or proxied replies. > > According to RFC 3261, retransmitting 407s/401s is probably a bad idea: > > 26.3.2.4 DoS Protection > > At the moment, my only solution is to use forward() instead of t_relay() > in order to use the stateless mode. > > Yannick > > > Yannick, > > > > The default behavior of OpenSIPS is to relay any received responses back upstream. If it is not doing that it would have to be because you are stopping it in the script. Take a look at the documentation for failure_route [1] which explains this. Check your own failure_route in your script; you must be doing something there that is telling OpenSIPS not to relay the 401/407 back upstream. > > > > [1] https://www.opensips.org/Documentation/Script-Routes-2-4#toc3 > > > > Ben Newlin > > > > From: Users on behalf of Yannick LE COENT > > Date: Friday, April 29, 2022 at 6:44 PM > > To: users at lists.opensips.org > > Subject: [OpenSIPS-Users] OpenSIPS : no 407 retransmission > > EXTERNAL EMAIL - Please use caution with links and attachments > > > > Hello, > > > > I'm using OpenSIPS as a proxy in front of another proxy. The 2nd proxy > > is in charge of authenticating the request. > > > > This is the callflow: > > > > Alice OpenSIPS Proxy#2 > > | INVITE | | > > |--------------->| INVITE | > > | 100 Tring |--------------->| > > |<---------------| 407 | > > | |<---------------| > > | | ACK | > > | |--------------->| > > | 407 | | > > | X<---------| | > > | | | > > > > Since OpenSIPS does not retransmit 401/407, the call setup gets stuck. > > > > What can I do ? > > If I set auto_100trying=1, that works, but this increases the number of > > INVITE retransmissions since 180Ringing are not received instantly. > > > > Do you have any suggestion ? > > > > Thanks, > > Yannick > > > _______________________________________________ > Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > _______________________________________________ > 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 yannick.lecoent at nexcom.fr Tue May 3 15:14:05 2022 From: yannick.lecoent at nexcom.fr (Yannick LE COENT) Date: Tue, 3 May 2022 17:14:05 +0200 Subject: [OpenSIPS-Users] OpenSIPS : no 407 retransmission In-Reply-To: References: <05c3bf91-3081-8815-e092-513035d82308@nexcom.fr> <93b1545a-b0ed-0826-863b-3dfbf90371c4@nexcom.fr> <9822c158-a891-f2bf-3c68-a3c02b17c671@nexcom.fr> Message-ID: <4ab91cb0-d5a7-4988-0dd7-dfb909e4ebdc@nexcom.fr> Hello Daniel, This is not what I looking for. My OpenSIPS instance is working as a relay between the softphone and another proxy (proxy#2 in the call). So it does not handle authentication. Alice           OpenSIPS          Proxy#2    | INVITE         |                | |--------------->| INVITE         |    |      100 Tring |--------------->| |<---------------|            407 | |                |<---------------| |                | ACK            | |                |--------------->| |            407 |                |    | X<---------|                |    |  (no retrans.) |                | When the 407 is lost between OpenSIPS and Alice, it is not retransmitted by OpenSIPS. I would like to force retransmission. Thanks, Yannick Le 03/05/2022 à 15:16, Daniel Zanutti a écrit : > Generate in Stateful -> www_challenge or proxy_challenge? > https://opensips.org/html/docs/modules/3.2.x/auth.html > > Is this what you are looking for? > > > On Tue, May 3, 2022 at 3:50 AM Yannick LE COENT > wrote: > > Hello all, > > Could you tell if there is a way to enable 407 in stateful mode ? > > Thanks, > Yannick > > Le 30/04/2022 à 18:14, Yannick LE COENT a écrit : >> Hello Ben, >> >> Thanks for your answer. >> >> This problem occurs when OpenSIPS is not in charge of >> authenticating the INVITE request, but this is done downstream. >> I've sent this question to know if somebody has already solved >> this kind of problem. >> >> Best regards, >> Yannick >> >> Le 30/04/2022 à 16:15, Ben Newlin a écrit : >>> >>> I see. Apologies, I misunderstood the problem scenario. >>> >>> Ben Newlin >>> >>> *From: *Users >>> on behalf of Yannick >>> LE COENT >>> >>> *Date: *Saturday, April 30, 2022 at 5:46 AM >>> *To: *OpenSIPS users mailling list >>> >>> *Subject: *Re: [OpenSIPS-Users] OpenSIPS : no 407 retransmission >>> >>> *EXTERNAL EMAIL - Please use caution with links and attachments * >>> >>> ------------------------------------------------------------------------ >>> >>> Hello Ben, >>> >>> The 407 is sent upstream, but when it is lost, it is not >>> retransmitted by OpenSIPS. >>> I do not have this problem with other negative status codes >>> (e.g. 486). >>> >>> This is clearly explained in >>> https://opensips.org/pub/opensips/1.8.6/src/ChangeLog >>> >>> 2012-03-21 18:36:58 Bogdan-Andrei Iancu, >> dot org> >>>  * [8811] : >>> >>>  TM will no longer do retransmission for the 407/401 replies >>> (if no ACK is received) for both local or proxied replies. >>> >>> According to RFC 3261, retransmitting 407s/401s is probably >>> a bad idea: >>> >>>  26.3.2.4 DoS Protection >>> >>> At the moment, my only solution is to use forward() instead of >>> t_relay() in order to use the stateless mode. >>> >>> Yannick >>> >>> >>> Yannick, >>> >>> >>> >>> The default behavior of OpenSIPS is to relay any received responses back upstream. If it is not doing that it would have to be because you are stopping it in the script. Take a look at the documentation for failure_route [1] which explains this. Check your own failure_route in your script; you must be doing something there that is telling OpenSIPS not to relay the 401/407 back upstream. >>> >>> >>> >>> [1]https://www.opensips.org/Documentation/Script-Routes-2-4#toc3 >>> >>> >>> >>> Ben Newlin >>> >>> >>> >>> From: Users on behalf of Yannick LE COENT >>> >>> Date: Friday, April 29, 2022 at 6:44 PM >>> >>> To:users at lists.opensips.org >>> >>> Subject: [OpenSIPS-Users] OpenSIPS : no 407 retransmission >>> >>> EXTERNAL EMAIL - Please use caution with links and attachments >>> >>> >>> >>> Hello, >>> >>> >>> >>> I'm using OpenSIPS as a proxy in front of another proxy. The 2nd proxy >>> >>> is in charge of authenticating the request. >>> >>> >>> >>> This is the callflow: >>> >>> >>> >>> Alice           OpenSIPS          Proxy#2 >>> >>>    | INVITE         |                | >>> >>>    |--------------->| INVITE         | >>> >>>    |      100 Tring |--------------->| >>> >>>    |<---------------|            407 | >>> >>>    |                |<---------------| >>> >>>    |                | ACK            | >>> >>>    |                |--------------->| >>> >>>    |            407 |                | >>> >>>    |     X<---------|                | >>> >>>    |                |                | >>> >>> >>> >>> Since OpenSIPS does not retransmit 401/407, the call setup gets stuck. >>> >>> >>> >>> What can I do ? >>> >>> If I set auto_100trying=1, that works, but this increases the number of >>> >>> INVITE retransmissions since 180Ringing are not received instantly. >>> >>> >>> >>> Do you have any suggestion ? >>> >>> >>> >>> Thanks, >>> >>> Yannick >>> >>> >>> _______________________________________________ >>> Users mailing list >>> Users at lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From devang.dhandhalya at ecosmob.com Wed May 4 08:58:27 2022 From: devang.dhandhalya at ecosmob.com (Devang Dhandhalya) Date: Wed, 4 May 2022 14:28:27 +0530 Subject: [OpenSIPS-Users] TCP connection relay issue Message-ID: Hi All Issue is whenever i am sending Register or INVITE request with TCP in opensips using tcp connection to relay request to Freeswitch server while i want to use UDP socket to send Request from OpenSIPs to Freeswitch , using mid registrar for registering users. OpenSIPS version : 3.2.2 flow :end user -> OpenSIPS -> Freeswitch -> OpenSIPS -> enduser OpenSIPS configuration : tcp_connect_timeout=3000 tcp_connection_lifetime = 3615 socket=udp:192.168.0.1:5060 as 1.2.3.4:5060 socket=tcp:192.168.0.1:5060 as 1.2.3.4:5060 loadmodule "proto_tcp.so" modparam("proto_tcp", "tcp_async", 1) modparam("proto_tcp", "tcp_send_timeout", 3000) modparam("proto_tcp", "tcp_async_local_connect_timeout", 3000) modparam("proto_tcp", "tcp_async_local_write_timeout", 3000) modparam("proto_tcp", "tcp_max_msg_chunks", 8) I used force_send_socket , $fs function to use UDP socket but still OpenSIPS using tcp connection for relay packet to Freeswitch any other functionality which overwrites this core function . In another setup the same configuration but opensips sends requests using a UDP socket. That's strange for me because of the same setup . There is a particular configuration that we use tcp connection / UDP socket to relay packets. Kindly inform me how we send requests using UDP socket instead of tcp connections. Regards Devang Dhandhalya -- *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 spanda at 3clogic.com Wed May 4 09:25:23 2022 From: spanda at 3clogic.com (Sasmita Panda) Date: Wed, 4 May 2022 14:55:23 +0530 Subject: [OpenSIPS-Users] Query regarding AWS document BD using through Opensips . In-Reply-To: <47454D85-990A-4965-8A79-48F3D63CC4A1@missouri-telecom.com> References: <47454D85-990A-4965-8A79-48F3D63CC4A1@missouri-telecom.com> Message-ID: Hi, Its not about the domain length . Its about the tls parameter . When I created a document db without tls that was connected successfully . I think with tls , the configuration is something different which I am missing . Thanks for your suggestion. *Thanks & Regards* *Sasmita Panda* *Senior Network Testing and Software Engineer* *3CLogic , ph:07827611765* On Fri, Apr 29, 2022 at 7:05 PM Kevin Wormington wrote: > Hi Samita, > > I don’t have any experience with AWS but from the error message OpenSIPS > is logging the hostname of the server cannot be resolved. The hostname > also appears to be truncated. Have you tried using the IP address instead > of hostname or making a CNAME dns entry for the host that is shorter? > Perhaps this is some parameter length limit. > > Kevin > > On Apr 29, 2022, at 8:23 AM, Sasmita Panda wrote: > > > > Hi All , > > > > > > I was exploring fullsharing-cachedb-cluster in opensips 3.2 . I have > tested this with single stand alone mongo db instance . Its working > perfectly fine . > > > > I know that AWS document DB is mongodb compatible . So I want to explore > that . Because we are using AWS cloud for our deployment . > > > > I have created a single instance of Document DB cluster . I want to > connect to that from the opensips script . There was no error while > starting opensips . But when opensips tried to write data in the db its > threw an error . > > > > > > ERROR:usrloc:release_urecord: failed to flush AoR > Default_Line_118_6 at p2p-cachedb.xyz.com > > ERROR:cachedb_mongodb:mongo_con_update: last error: 15.13053: No > suitable servers found (`serverselectiontryonce` set): [Failed to resolve > 'docdb-2022-04-27-10-26-28.cluster-cryhhicuxgzu.us-east-1.docdb.amdocdb-2022-04-27-10-26-28.cluster-cryhhicuxgzu.us-east-1.docdb.amazona'] > > ERROR:usrloc:cdb_flush_urecord: cache update query for AoR > hynode2_CallDefault at p2p-cachedb.xyz.com failed! > > > > > > My configuration file looks like below . same for usrloc and db_cachedb > > > > modparam("cachedb_mongodb", "cachedb_url","mongodb:// > > ? > master:opensips3 at docdb-2022.cluster-cryhhicuxgzu.us-east-1.docdb.amazonaws.com:27017/db.test/ > > > ?ssl=true&ssl_ca_certs=/usr/local/src/etc/opensips/rds-combined-ca-bundle.pem&replicaSet=rs0&readPreference=secondaryPreferred&retryWrites=false") > > > > In document db it also its says the same way we can connect to docdb > from an application . > > > > I have tried so many ways to resolve this . But without any luck . > Please help me out if anybody has used a document db through any > application can also reply . > > > > > > Thanks & Regards > > Sasmita Panda > > Senior Network Testing and Software Engineer > > 3CLogic , ph:07827611765 > > _______________________________________________ > > Users mailing list > > Users at lists.opensips.org > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -------------- next part -------------- An HTML attachment was scrubbed... URL: From daniel.zanutti at gmail.com Wed May 4 16:48:08 2022 From: daniel.zanutti at gmail.com (Daniel Zanutti) Date: Wed, 4 May 2022 13:48:08 -0300 Subject: [OpenSIPS-Users] OpenSIPS : no 407 retransmission In-Reply-To: <4ab91cb0-d5a7-4988-0dd7-dfb909e4ebdc@nexcom.fr> References: <05c3bf91-3081-8815-e092-513035d82308@nexcom.fr> <93b1545a-b0ed-0826-863b-3dfbf90371c4@nexcom.fr> <9822c158-a891-f2bf-3c68-a3c02b17c671@nexcom.fr> <4ab91cb0-d5a7-4988-0dd7-dfb909e4ebdc@nexcom.fr> Message-ID: Hi Yannick I think you should not reply with ACK to the 407 from destination. Just forward 407 to origin and wait for ACK. As soon you receive ACK from origin, you forward to destination. It's more like a stateless but I believe it's the only way. Regards On Tue, May 3, 2022 at 12:16 PM Yannick LE COENT wrote: > Hello Daniel, > > This is not what I looking for. > My OpenSIPS instance is working as a relay between the softphone and > another proxy (proxy#2 in the call). > So it does not handle authentication. > > Alice OpenSIPS Proxy#2 > | INVITE | | > |--------------->| INVITE | > | 100 Tring |--------------->| > |<---------------| 407 | > | |<---------------| > | | ACK | > | |--------------->| > | 407 | | > | X<---------| | > | (no retrans.) | | > > When the 407 is lost between OpenSIPS and Alice, it is not retransmitted > by OpenSIPS. > > I would like to force retransmission. > > Thanks, > Yannick > > Le 03/05/2022 à 15:16, Daniel Zanutti a écrit : > > Generate in Stateful -> www_challenge or proxy_challenge? > https://opensips.org/html/docs/modules/3.2.x/auth.html > > Is this what you are looking for? > > > On Tue, May 3, 2022 at 3:50 AM Yannick LE COENT > wrote: > >> Hello all, >> >> Could you tell if there is a way to enable 407 in stateful mode ? >> >> Thanks, >> Yannick >> >> Le 30/04/2022 à 18:14, Yannick LE COENT a écrit : >> >> Hello Ben, >> >> Thanks for your answer. >> >> This problem occurs when OpenSIPS is not in charge of authenticating the >> INVITE request, but this is done downstream. >> I've sent this question to know if somebody has already solved this kind >> of problem. >> >> Best regards, >> Yannick >> >> Le 30/04/2022 à 16:15, Ben Newlin a écrit : >> >> I see. Apologies, I misunderstood the problem scenario. >> >> >> >> Ben Newlin >> >> >> >> *From: *Users >> on behalf of Yannick LE COENT >> >> *Date: *Saturday, April 30, 2022 at 5:46 AM >> *To: *OpenSIPS users mailling list >> >> *Subject: *Re: [OpenSIPS-Users] OpenSIPS : no 407 retransmission >> >> *EXTERNAL EMAIL - Please use caution with links and attachments * >> >> >> ------------------------------ >> >> Hello Ben, >> >> The 407 is sent upstream, but when it is lost, it is not retransmitted by >> OpenSIPS. >> I do not have this problem with other negative status codes (e.g. 486). >> >> This is clearly explained in >> https://opensips.org/pub/opensips/1.8.6/src/ChangeLog >> >> 2012-03-21 18:36:58 Bogdan-Andrei Iancu, >> * [8811] : >> >> TM will no longer do retransmission for the 407/401 replies (if no >> ACK is received) for both local or proxied replies. >> >> According to RFC 3261, retransmitting 407s/401s is probably a bad >> idea: >> >> 26.3.2.4 DoS Protection >> >> At the moment, my only solution is to use forward() instead of t_relay() >> in order to use the stateless mode. >> >> Yannick >> >> >> Yannick, >> >> >> >> The default behavior of OpenSIPS is to relay any received responses back upstream. If it is not doing that it would have to be because you are stopping it in the script. Take a look at the documentation for failure_route [1] which explains this. Check your own failure_route in your script; you must be doing something there that is telling OpenSIPS not to relay the 401/407 back upstream. >> >> >> >> [1] https://www.opensips.org/Documentation/Script-Routes-2-4#toc3 >> >> >> >> Ben Newlin >> >> >> >> From: Users on behalf of Yannick LE COENT >> >> Date: Friday, April 29, 2022 at 6:44 PM >> >> To: users at lists.opensips.org >> >> Subject: [OpenSIPS-Users] OpenSIPS : no 407 retransmission >> >> EXTERNAL EMAIL - Please use caution with links and attachments >> >> >> >> Hello, >> >> >> >> I'm using OpenSIPS as a proxy in front of another proxy. The 2nd proxy >> >> is in charge of authenticating the request. >> >> >> >> This is the callflow: >> >> >> >> Alice OpenSIPS Proxy#2 >> >> | INVITE | | >> >> |--------------->| INVITE | >> >> | 100 Tring |--------------->| >> >> |<---------------| 407 | >> >> | |<---------------| >> >> | | ACK | >> >> | |--------------->| >> >> | 407 | | >> >> | X<---------| | >> >> | | | >> >> >> >> Since OpenSIPS does not retransmit 401/407, the call setup gets stuck. >> >> >> >> What can I do ? >> >> If I set auto_100trying=1, that works, but this increases the number of >> >> INVITE retransmissions since 180Ringing are not received instantly. >> >> >> >> Do you have any suggestion ? >> >> >> >> Thanks, >> >> Yannick >> >> >> _______________________________________________ >> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> >> >> _______________________________________________ >> 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 >> > > _______________________________________________ > 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 yannick.lecoent at nexcom.fr Wed May 4 17:17:18 2022 From: yannick.lecoent at nexcom.fr (Yannick LE COENT) Date: Wed, 4 May 2022 19:17:18 +0200 Subject: [OpenSIPS-Users] OpenSIPS : no 407 retransmission In-Reply-To: References: <05c3bf91-3081-8815-e092-513035d82308@nexcom.fr> <93b1545a-b0ed-0826-863b-3dfbf90371c4@nexcom.fr> <9822c158-a891-f2bf-3c68-a3c02b17c671@nexcom.fr> <4ab91cb0-d5a7-4988-0dd7-dfb909e4ebdc@nexcom.fr> Message-ID: <91bded3c-a223-db64-38d9-8d03a88c059b@nexcom.fr> Hi Daniel, I do not think the ACK is sent by my script. It is sent by the TM module since it is a negative response. Am I wrong ? Thanks, Yannick Le 04/05/2022 à 18:48, Daniel Zanutti a écrit : > Hi Yannick > > I think you should not reply with ACK to the 407 from destination. > Just forward 407 to origin and wait for ACK. As soon you receive ACK > from origin, you forward to destination. > > It's more like a stateless but I believe it's the only way. > > Regards > > On Tue, May 3, 2022 at 12:16 PM Yannick LE COENT > wrote: > > Hello Daniel, > > This is not what I looking for. > My OpenSIPS instance is working as a relay between the softphone > and another proxy (proxy#2 in the call). > So it does not handle authentication. > > Alice           OpenSIPS Proxy#2 >    | INVITE         | | > |--------------->| INVITE         | > |      100 Tring |--------------->| > |<---------------|            407 | > |                |<---------------| > |                | ACK            | > |                |--------------->| > |            407 |                | > |     X<---------|                | >    |  (no retrans.) | | > > When the 407 is lost between OpenSIPS and Alice, it is not > retransmitted by OpenSIPS. > > I would like to force retransmission. > > Thanks, > Yannick > > Le 03/05/2022 à 15:16, Daniel Zanutti a écrit : >> Generate in Stateful -> www_challenge or proxy_challenge? >> https://opensips.org/html/docs/modules/3.2.x/auth.html >> >> Is this what you are looking for? >> >> >> On Tue, May 3, 2022 at 3:50 AM Yannick LE COENT >> wrote: >> >> Hello all, >> >> Could you tell if there is a way to enable 407 in stateful mode ? >> >> Thanks, >> Yannick >> >> Le 30/04/2022 à 18:14, Yannick LE COENT a écrit : >>> Hello Ben, >>> >>> Thanks for your answer. >>> >>> This problem occurs when OpenSIPS is not in charge of >>> authenticating the INVITE request, but this is done downstream. >>> I've sent this question to know if somebody has already >>> solved this kind of problem. >>> >>> Best regards, >>> Yannick >>> >>> Le 30/04/2022 à 16:15, Ben Newlin a écrit : >>>> >>>> I see. Apologies, I misunderstood the problem scenario. >>>> >>>> Ben Newlin >>>> >>>> *From: *Users >>>> on behalf of >>>> Yannick LE COENT >>>> >>>> *Date: *Saturday, April 30, 2022 at 5:46 AM >>>> *To: *OpenSIPS users mailling list >>>> >>>> *Subject: *Re: [OpenSIPS-Users] OpenSIPS : no 407 >>>> retransmission >>>> >>>> *EXTERNAL EMAIL - Please use caution with links and >>>> attachments * >>>> >>>> ------------------------------------------------------------------------ >>>> >>>> Hello Ben, >>>> >>>> The 407 is sent upstream, but when it is lost, it is not >>>> retransmitted by OpenSIPS. >>>> I do not have this problem with other negative status codes >>>> (e.g. 486). >>>> >>>> This is clearly explained in >>>> https://opensips.org/pub/opensips/1.8.6/src/ChangeLog >>>> >>>> 2012-03-21 18:36:58  Bogdan-Andrei Iancu, >>> opensips dot org> >>>>     * [8811] : >>>> >>>>     TM will no longer do retransmission for the 407/401 >>>> replies (if no ACK is received) for both local or >>>> proxied replies. >>>> >>>>     According to RFC 3261, retransmitting 407s/401s is >>>> probably a bad idea: >>>> >>>>     26.3.2.4 DoS Protection >>>> >>>> At the moment, my only solution is to use forward() instead >>>> of t_relay() in order to use the stateless mode. >>>> >>>> Yannick >>>> >>>> >>>> Yannick, >>>> >>>> >>>> >>>> The default behavior of OpenSIPS is to relay any received responses back upstream. If it is not doing that it would have to be because you are stopping it in the script. Take a look at the documentation for failure_route [1] which explains this. Check your own failure_route in your script; you must be doing something there that is telling OpenSIPS not to relay the 401/407 back upstream. >>>> >>>> >>>> >>>> [1]https://www.opensips.org/Documentation/Script-Routes-2-4#toc3 >>>> >>>> >>>> >>>> Ben Newlin >>>> >>>> >>>> >>>> From: Users on behalf of Yannick LE COENT >>>> >>>> Date: Friday, April 29, 2022 at 6:44 PM >>>> >>>> To:users at lists.opensips.org >>>> >>>> Subject: [OpenSIPS-Users] OpenSIPS : no 407 retransmission >>>> >>>> EXTERNAL EMAIL - Please use caution with links and attachments >>>> >>>> >>>> >>>> Hello, >>>> >>>> >>>> >>>> I'm using OpenSIPS as a proxy in front of another proxy. The 2nd proxy >>>> >>>> is in charge of authenticating the request. >>>> >>>> >>>> >>>> This is the callflow: >>>> >>>> >>>> >>>> Alice           OpenSIPS          Proxy#2 >>>> >>>>    | INVITE         |                | >>>> >>>>    |--------------->| INVITE         | >>>> >>>>    |      100 Tring |--------------->| >>>> >>>>    |<---------------|            407 | >>>> >>>>    |                |<---------------| >>>> >>>>    |                | ACK            | >>>> >>>>    |                |--------------->| >>>> >>>>    |            407 |                | >>>> >>>>    |     X<---------|                | >>>> >>>>    |                |                | >>>> >>>> >>>> >>>> Since OpenSIPS does not retransmit 401/407, the call setup gets stuck. >>>> >>>> >>>> >>>> What can I do ? >>>> >>>> If I set auto_100trying=1, that works, but this increases the number of >>>> >>>> INVITE retransmissions since 180Ringing are not received instantly. >>>> >>>> >>>> >>>> Do you have any suggestion ? >>>> >>>> >>>> >>>> Thanks, >>>> >>>> Yannick >>>> >>>> >>>> _______________________________________________ >>>> Users mailing list >>>> Users at lists.opensips.org >>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >>> >>> _______________________________________________ >>> Users mailing list >>> Users at lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From daniel.zanutti at gmail.com Wed May 4 17:47:51 2022 From: daniel.zanutti at gmail.com (Daniel Zanutti) Date: Wed, 4 May 2022 14:47:51 -0300 Subject: [OpenSIPS-Users] OpenSIPS : no 407 retransmission In-Reply-To: <91bded3c-a223-db64-38d9-8d03a88c059b@nexcom.fr> References: <05c3bf91-3081-8815-e092-513035d82308@nexcom.fr> <93b1545a-b0ed-0826-863b-3dfbf90371c4@nexcom.fr> <9822c158-a891-f2bf-3c68-a3c02b17c671@nexcom.fr> <4ab91cb0-d5a7-4988-0dd7-dfb909e4ebdc@nexcom.fr> <91bded3c-a223-db64-38d9-8d03a88c059b@nexcom.fr> Message-ID: Well, I don't have your script, cannot help further. If you are proxying, you should proxy every message. Maybe you are mixing stateful and stateless forwarding and are not handling all scenarios? It could be an Opensips problem, but again, don't know what you are doing internally. On Wed, May 4, 2022 at 2:19 PM Yannick LE COENT wrote: > Hi Daniel, > > I do not think the ACK is sent by my script. It is sent by the TM module > since it is a negative response. > Am I wrong ? > > Thanks, > Yannick > > Le 04/05/2022 à 18:48, Daniel Zanutti a écrit : > > Hi Yannick > > I think you should not reply with ACK to the 407 from destination. Just > forward 407 to origin and wait for ACK. As soon you receive ACK from > origin, you forward to destination. > > It's more like a stateless but I believe it's the only way. > > Regards > > On Tue, May 3, 2022 at 12:16 PM Yannick LE COENT < > yannick.lecoent at nexcom.fr> wrote: > >> Hello Daniel, >> >> This is not what I looking for. >> My OpenSIPS instance is working as a relay between the softphone and >> another proxy (proxy#2 in the call). >> So it does not handle authentication. >> >> Alice OpenSIPS Proxy#2 >> | INVITE | | >> |--------------->| INVITE | >> | 100 Tring |--------------->| >> |<---------------| 407 | >> | |<---------------| >> | | ACK | >> | |--------------->| >> | 407 | | >> | X<---------| | >> | (no retrans.) | | >> >> When the 407 is lost between OpenSIPS and Alice, it is not retransmitted >> by OpenSIPS. >> >> I would like to force retransmission. >> >> Thanks, >> Yannick >> >> Le 03/05/2022 à 15:16, Daniel Zanutti a écrit : >> >> Generate in Stateful -> www_challenge or proxy_challenge? >> https://opensips.org/html/docs/modules/3.2.x/auth.html >> >> Is this what you are looking for? >> >> >> On Tue, May 3, 2022 at 3:50 AM Yannick LE COENT < >> yannick.lecoent at nexcom.fr> wrote: >> >>> Hello all, >>> >>> Could you tell if there is a way to enable 407 in stateful mode ? >>> >>> Thanks, >>> Yannick >>> >>> Le 30/04/2022 à 18:14, Yannick LE COENT a écrit : >>> >>> Hello Ben, >>> >>> Thanks for your answer. >>> >>> This problem occurs when OpenSIPS is not in charge of authenticating the >>> INVITE request, but this is done downstream. >>> I've sent this question to know if somebody has already solved this kind >>> of problem. >>> >>> Best regards, >>> Yannick >>> >>> Le 30/04/2022 à 16:15, Ben Newlin a écrit : >>> >>> I see. Apologies, I misunderstood the problem scenario. >>> >>> >>> >>> Ben Newlin >>> >>> >>> >>> *From: *Users >>> on behalf of Yannick LE COENT >>> >>> *Date: *Saturday, April 30, 2022 at 5:46 AM >>> *To: *OpenSIPS users mailling list >>> >>> *Subject: *Re: [OpenSIPS-Users] OpenSIPS : no 407 retransmission >>> >>> *EXTERNAL EMAIL - Please use caution with links and attachments * >>> >>> >>> ------------------------------ >>> >>> Hello Ben, >>> >>> The 407 is sent upstream, but when it is lost, it is not retransmitted >>> by OpenSIPS. >>> I do not have this problem with other negative status codes (e.g. 486). >>> >>> This is clearly explained in >>> https://opensips.org/pub/opensips/1.8.6/src/ChangeLog >>> >>> 2012-03-21 18:36:58 Bogdan-Andrei Iancu, >>> * [8811] : >>> >>> TM will no longer do retransmission for the 407/401 replies (if no >>> ACK is received) for both local or proxied replies. >>> >>> According to RFC 3261, retransmitting 407s/401s is probably a bad >>> idea: >>> >>> 26.3.2.4 DoS Protection >>> >>> At the moment, my only solution is to use forward() instead of t_relay() >>> in order to use the stateless mode. >>> >>> Yannick >>> >>> >>> Yannick, >>> >>> >>> >>> The default behavior of OpenSIPS is to relay any received responses back upstream. If it is not doing that it would have to be because you are stopping it in the script. Take a look at the documentation for failure_route [1] which explains this. Check your own failure_route in your script; you must be doing something there that is telling OpenSIPS not to relay the 401/407 back upstream. >>> >>> >>> >>> [1] https://www.opensips.org/Documentation/Script-Routes-2-4#toc3 >>> >>> >>> >>> Ben Newlin >>> >>> >>> >>> From: Users on behalf of Yannick LE COENT >>> >>> Date: Friday, April 29, 2022 at 6:44 PM >>> >>> To: users at lists.opensips.org >>> >>> Subject: [OpenSIPS-Users] OpenSIPS : no 407 retransmission >>> >>> EXTERNAL EMAIL - Please use caution with links and attachments >>> >>> >>> >>> Hello, >>> >>> >>> >>> I'm using OpenSIPS as a proxy in front of another proxy. The 2nd proxy >>> >>> is in charge of authenticating the request. >>> >>> >>> >>> This is the callflow: >>> >>> >>> >>> Alice OpenSIPS Proxy#2 >>> >>> | INVITE | | >>> >>> |--------------->| INVITE | >>> >>> | 100 Tring |--------------->| >>> >>> |<---------------| 407 | >>> >>> | |<---------------| >>> >>> | | ACK | >>> >>> | |--------------->| >>> >>> | 407 | | >>> >>> | X<---------| | >>> >>> | | | >>> >>> >>> >>> Since OpenSIPS does not retransmit 401/407, the call setup gets stuck. >>> >>> >>> >>> What can I do ? >>> >>> If I set auto_100trying=1, that works, but this increases the number of >>> >>> INVITE retransmissions since 180Ringing are not received instantly. >>> >>> >>> >>> Do you have any suggestion ? >>> >>> >>> >>> Thanks, >>> >>> Yannick >>> >>> >>> _______________________________________________ >>> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >>> >>> >>> _______________________________________________ >>> 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 >>> >> >> _______________________________________________ >> 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 >> > > _______________________________________________ > 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 ahmed.rehan at gmail.com Thu May 5 19:48:28 2022 From: ahmed.rehan at gmail.com (Ahmed Rehan) Date: Fri, 6 May 2022 00:48:28 +0500 Subject: [OpenSIPS-Users] OpenSIPS Bootcamp training 2022 In-Reply-To: References: <98b76556-f672-470c-5a90-8ac2512df014@opensips.org> Message-ID: Hi Bogdan, Is there any discount for a single participant ? i m from pakistan and very eager to join ebootcamp but need some discount on it as 1700$ is quite expensive for me Regards Ahmed On Tue, May 3, 2022 at 5:29 PM Bogdan-Andrei Iancu wrote: > > Tick, tack, only 3 weeks left to the training, first come, first served, > seats are limited. > > https://opensips.org/training/OpenSIPS_eBootcamp_2022/ > > Best regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > OpenSIPS eBootcamp 23rd May - 3rd June 2022 > https://opensips.org/training/OpenSIPS_eBootcamp_2022/ > > On 3/30/22 10:52 AM, Bogdan-Andrei Iancu wrote: > > 23rd May - 03rd > June 2022, > online, worldwide > > > > > *Study smarter, not harder! * > > Take advantage of the *OpenSIPS Bootcamp* > and improve your > OpenSIPS skills - an in-cloud training, a ten days, 4 hours per day (40 > hours) intensive and practical training, covering installation, > configuration and administration on OpenSIPS. > > All the knowledge transferred to the students will be strongly backed up > by practice sessions where you will get hands-on experience in handling > OpenSIPS. The training is structured to be offer 50% / 50% between the > theoretical and practical sessions. > Check Syllabus > > > *Early Birds open* > > The Early Bird 10% discount is available for registrations before *11th > of April 2022*, so do not miss the opportunity. The number of seats is > limited, so be sure and book a seat now. Keep in mind that a 10% group > discount is also available - grab your work mate and start learning more > OpenSIPS together . > . > Register Now > > > *Certified training saves time and money* > > OpenSIPS mistakes are easily avoided if you get proper training! Companies > that use OpenSIPS waste time and money when they don't have a trained > engineer on staff. Searching on Google, waiting on IRC, even the latency in > mailing list replies takes it's toll over time. Take this rare opportunity > to train your employees with the project members themselves. > > > Any questions? do not hesitate to contact us ! > ------------------------------ > You received this email as part of your relationship with the OpenSIPS > Project. > If you do not want to receive any more news, please email to unsubscribe > . > > -- > Bogdan-Andrei Iancu > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > > > _______________________________________________ > 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 > -- Regards Ahmed Rehan -------------- next part -------------- An HTML attachment was scrubbed... URL: From spanda at 3clogic.com Fri May 6 05:37:40 2022 From: spanda at 3clogic.com (Sasmita Panda) Date: Fri, 6 May 2022 11:07:40 +0530 Subject: [OpenSIPS-Users] OpenSIPS Bootcamp training 2022 In-Reply-To: References: <98b76556-f672-470c-5a90-8ac2512df014@opensips.org> Message-ID: Hi , I have also same concern . I am quite interested in Wednesday1 and Thursday1 session . Is this possible to have some discount ? *Thanks & Regards* *Sasmita Panda* *Senior Network Testing and Software Engineer* *3CLogic , ph:07827611765* On Fri, May 6, 2022 at 1:20 AM Ahmed Rehan wrote: > Hi Bogdan, > > Is there any discount for a single participant ? i m from pakistan and > very eager to join ebootcamp but need some discount on it as 1700$ is quite > expensive for me > > Regards > Ahmed > > On Tue, May 3, 2022 at 5:29 PM Bogdan-Andrei Iancu > wrote: > >> >> Tick, tack, only 3 weeks left to the training, first come, first served, >> seats are limited. >> >> https://opensips.org/training/OpenSIPS_eBootcamp_2022/ >> >> Best regards, >> >> Bogdan-Andrei Iancu >> >> OpenSIPS Founder and Developer >> https://www.opensips-solutions.com >> OpenSIPS eBootcamp 23rd May - 3rd June 2022 >> https://opensips.org/training/OpenSIPS_eBootcamp_2022/ >> >> On 3/30/22 10:52 AM, Bogdan-Andrei Iancu wrote: >> >> 23rd May - 03rd >> June 2022, >> online, worldwide >> >> >> >> >> *Study smarter, not harder! * >> >> Take advantage of the *OpenSIPS Bootcamp* >> and improve your >> OpenSIPS skills - an in-cloud training, a ten days, 4 hours per day (40 >> hours) intensive and practical training, covering installation, >> configuration and administration on OpenSIPS. >> >> All the knowledge transferred to the students will be strongly backed up >> by practice sessions where you will get hands-on experience in handling >> OpenSIPS. The training is structured to be offer 50% / 50% between the >> theoretical and practical sessions. >> Check Syllabus >> >> >> *Early Birds open* >> >> The Early Bird 10% discount is available for registrations before *11th >> of April 2022*, so do not miss the opportunity. The number of seats is >> limited, so be sure and book a seat now. Keep in mind that a 10% group >> discount is also available - grab your work mate and start learning more >> OpenSIPS together . >> . >> Register Now >> >> >> *Certified training saves time and money* >> >> OpenSIPS mistakes are easily avoided if you get proper training! >> Companies that use OpenSIPS waste time and money when they don't have a >> trained engineer on staff. Searching on Google, waiting on IRC, even the >> latency in mailing list replies takes it's toll over time. Take this rare >> opportunity to train your employees with the project members themselves. >> >> >> Any questions? do not hesitate to contact us ! >> ------------------------------ >> You received this email as part of your relationship with the OpenSIPS >> Project. >> If you do not want to receive any more news, please email to unsubscribe >> . >> >> -- >> Bogdan-Andrei Iancu >> OpenSIPS Founder and Developer >> http://www.opensips-solutions.com >> >> >> _______________________________________________ >> 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 >> > > > -- > Regards > Ahmed Rehan > _______________________________________________ > 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 venefax at gmail.com Fri May 6 19:39:46 2022 From: venefax at gmail.com (Saint Michael) Date: Fri, 6 May 2022 15:39:46 -0400 Subject: [OpenSIPS-Users] incompatibility leads to massive CDR loss Message-ID: Dear friends Kindly look at the file attached. I am losing 10% of my CDR because some messages cannot be parsed by Opensips opensips -V version: opensips 3.1.9 (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: 1a71fded7 main.c compiled on 13:37:30 May 2 2022 with gcc 9 I need some paid help generating a patch or fixing this somehow. I normally have a consultant but he may be tied up with the war and is not responding. Yours Federico -------------- next part -------------- An HTML attachment was scrubbed... URL: -------------- next part -------------- May 06 17:26:11 federico opensips[358436]: May 6 17:26:11 [358436] ERROR:core:parse_via: parsed so far: May 06 17:26:11 federico opensips[358436]: May 6 17:26:11 [358436] ERROR:core:get_hdr_field: bad via May 06 17:26:11 federico opensips[358436]: May 6 17:26:11 [358436] ERROR:core:parse_msg: message=;tag=29749fd7 May 06 17:26:11 federico opensips[358436]: t:9990118135950761 ;tag=sbcsipuas_1_C33515_20220506132608132_ucs02sb02 May 06 17:26:11 federico opensips[358436]: i:104678NTA3YjljOWZlNzRmNzgyZGYyZTBjZDc4ODMwMWE0ODY May 06 17:26:11 federico opensips[358436]: m: May 06 17:26:11 federico opensips[358436]: CSeq: 1 INVITE May 06 17:26:11 federico opensips[358436]: Allow: ACK,BYE,CANCEL,INVITE,OPTIONS,NOTIFY,INFO,UPDATE May 06 17:26:11 federico opensips[358436]: Server: sbc_5 May 06 17:26:11 federico opensips[358436]: l:0 May 06 17:26:11 federico opensips[358430]: May 6 17:26:11 [358430] ERROR:core:receive_msg: Unable to parse msg received from [ZZ.ZZ.ZZ.ZZ:5060] May 06 17:26:42 federico opensips[358430]: May 6 17:26:42 [358430] ERROR:core:parse_via_param: invalid char <=> in state 202 May 06 17:26:42 federico opensips[358430]: May 6 17:26:42 [358430] ERROR:core:parse_via_param: parse_via_param May 06 17:26:42 federico opensips[358430]: May 6 17:26:42 [358430] ERROR:core:parse_via: ;tag=29749fd7 May 06 17:26:42 federico opensips[358430]: t:9990118135950761 ;tag=sbcsipuas_1_C33515_20220506132608132_ucs02sb02 May 06 17:26:42 federico opensips[358430]: i:104678NTA3YjljOWZlNzRmNzgyZGYyZTBjZDc4ODMwMWE0ODY May 06 17:26:42 federico opensips[358430]: Record-Route: May 06 17:26:42 federico opensips[358430]: m: May 06 17:26:42 federico opensips[358430]: CSeq: 1 INVITE May 06 17:26:42 federico opensips[358430]: Allow: ACK,BYE,CANCEL,INVITE,OPTIONS,NOTIFY,INFO,UPDATE May 06 17:26:42 federico opensips[358430]: Server: sbc_5 May 06 17:26:42 federico opensips[358430]: c:application/sdp May 06 17:26:42 federico opensips[358430]: l:361 May 06 17:26:42 federico opensips[358430]: May 06 17:26:42 federico opensips[358430]: v=0 May 06 17:26:42 federico opensips[358430]: o=SBCSIPUAS 1993104899 1 IN IP4 206.20.217.23 May 06 17:26:42 federico opensips[358430]: s=SBCSIPUAS SIP STACK v1.0 May 06 17:26:42 federico opensips[358430]: c=IN IP4 206.20.217.23 May 06 17:26:42 federico opensips[358430]: t=0 0 May 06 17:26:42 federico opensips[358430]: m=audio 20556 RTP/AVP 0 101 May 06 17:26:42 federico opensips[358430]: a=rtpmap:0 PCMU/8000 May 06 17:26:42 federico opensips[358430]: a=rtpmap:101 telephone-event/8000 May 06 17:26:42 federico opensips[358430]: a=msi:mavodi-0-15b-13f-4-ffffffff-7a3c0000-5dc6608fbb33e-1168-ffffffffffffffff- at 127.0.0.1-127.0.0.1&&SBIIRV203LVN01-104 May 06 17:26:42 federico opensips[358430]: a=fmtp:101 0-15 May 06 17:26:42 federico opensips[358430]: a=sendrecv May 06 17:26:42 federico opensips[358430]: a=rtcp:36607 May 06 17:26:42 federico opensips[358430]: > May 06 17:26:42 federico opensips[358430]: May 6 17:26:42 [358430] ERROR:core:parse_via: parsed so far: May 06 17:26:42 federico opensips[358430]: May 6 17:26:42 [358430] ERROR:core:get_hdr_field: bad via May 06 17:26:42 federico opensips[358430]: May 6 17:26:42 [358430] ERROR:core:parse_msg: message=;tag=29749fd7 May 06 17:26:42 federico opensips[358430]: t:9990118135950761 ;tag=sbcsipuas_1_C33515_20220506132608132_ucs02sb02 May 06 17:26:42 federico opensips[358430]: i:104678NTA3YjljOWZlNzRmNzgyZGYyZTBjZDc4ODMwMWE0ODY May 06 17:26:42 federico opensips[358430]: Record-Route: May 06 17:26:42 federico opensips[358430]: m: May 06 17:26:42 federico opensips[358430]: CSeq: 1 INVITE May 06 17:26:42 federico opensips[358430]: Allow: ACK,BYE,CANCEL,INVITE,OPTIONS,NOTIFY,INFO,UPDATE May 06 17:26:42 federico opensips[358430]: Server: sbc_5 May 06 17:26:42 federico opensips[358430]: c:application/sdp May 06 17:26:42 federico opensips[358430]: l:361 May 06 17:26:42 federico opensips[358430]: May 06 17:26:42 federico opensips[358430]: v=0 May 06 17:26:42 federico opensips[358430]: o=SBCSIPUAS 1993104899 1 IN IP4 206.20.217.23 May 06 17:26:42 federico opensips[358430]: s=SBCSIPUAS SIP STACK v1.0 May 06 17:26:42 federico opensips[358430]: c=IN IP4 206.20.217.23 May 06 17:26:42 federico opensips[358430]: t=0 0 May 06 17:26:42 federico opensips[358430]: m=audio 20556 RTP/AVP 0 101 May 06 17:26:42 federico opensips[358430]: a=rtpmap:0 PCMU/8000 May 06 17:26:42 federico opensips[358430]: a=rtpmap:101 telephone-event/8000 May 06 17:26:42 federico opensips[358430]: a=msi:mavodi-0-15b-13f-4-ffffffff-7a3c0000-5dc6608fbb33e-1168-ffffffffffffffff- at 127.0.0.1-127.0.0.1&&SBIIRV203LVN01-104 May 06 17:26:42 federico opensips[358430]: a=fmtp:101 0-15 May 06 17:26:42 federico opensips[358430]: a=sendrecv May 06 17:26:42 federico opensips[358430]: a=rtcp:36607 May 06 17:26:42 federico opensips[358430]: > May 06 17:26:42 federico opensips[358430]: May 6 17:26:42 [358430] ERROR:core:receive_msg: Unable to parse msg received from [ZZ.ZZ.ZZ.ZZ:5060] root at federico:/usr/src# From daniel.zanutti at gmail.com Sat May 7 03:13:14 2022 From: daniel.zanutti at gmail.com (Daniel Zanutti) Date: Sat, 7 May 2022 00:13:14 -0300 Subject: [OpenSIPS-Users] incompatibility leads to massive CDR loss In-Reply-To: References: Message-ID: I think this is your problem: branch=z9hG4bK-524287-1---b8aced18b4075aa3 *=49972* You have char "=" inside a string, which is a reserved character and not allowed on a string: https://datatracker.ietf.org/doc/html/rfc3261#section-25.1 Should be something on client of your customer, since you received on 180 ringing, but i'm not sure if you can just solve it. It's violating RFC. On Fri, May 6, 2022 at 4:42 PM Saint Michael wrote: > Dear friends > Kindly look at the file attached. I am losing 10% of my CDR because some > messages cannot be parsed by Opensips > opensips -V > version: opensips 3.1.9 (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: 1a71fded7 > main.c compiled on 13:37:30 May 2 2022 with gcc 9 > > I need some paid help generating a patch or fixing this somehow. > I normally have a consultant but he may be tied up with the war and is not > responding. > > Yours > Federico > _______________________________________________ > 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 venefax at gmail.com Sat May 7 03:24:25 2022 From: venefax at gmail.com (Saint Michael) Date: Fri, 6 May 2022 23:24:25 -0400 Subject: [OpenSIPS-Users] incompatibility leads to massive CDR loss In-Reply-To: References: Message-ID: Thanks. It's my carrier. I have already opened a ticket. Yours Federico On Fri, May 6, 2022 at 11:17 PM Daniel Zanutti wrote: > I think this is your problem: branch=z9hG4bK-524287-1---b8aced18b4075aa3 > *=49972* > > You have char "=" inside a string, which is a reserved character and not > allowed on a string: > https://datatracker.ietf.org/doc/html/rfc3261#section-25.1 > > Should be something on client of your customer, since you received on 180 > ringing, but i'm not sure if you can just solve it. It's violating RFC. > > > On Fri, May 6, 2022 at 4:42 PM Saint Michael wrote: > >> Dear friends >> Kindly look at the file attached. I am losing 10% of my CDR because some >> messages cannot be parsed by Opensips >> opensips -V >> version: opensips 3.1.9 (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: 1a71fded7 >> main.c compiled on 13:37:30 May 2 2022 with gcc 9 >> >> I need some paid help generating a patch or fixing this somehow. >> I normally have a consultant but he may be tied up with the war and is >> not responding. >> >> Yours >> Federico >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -------------- next part -------------- An HTML attachment was scrubbed... URL: From venefax at gmail.com Sat May 7 15:38:38 2022 From: venefax at gmail.com (Saint Michael) Date: Sat, 7 May 2022 11:38:38 -0400 Subject: [OpenSIPS-Users] Errors in the machine's log Message-ID: Dear Friends I see a lot of the unixodbc errors below, which I guess are not real errors, just a reconnect from a dead connection after a few hours. The question is how to avoid logging them with error_level=-1. Is that even possible? ERROR:db_unixodbc:db_unixodbc_submit_query: rv=-1. Query= call asterisk.gatekeeper('48c77ca0077fe1de555db2f40c5a48cd at ZZ.ZZ.ZZ.ZZ ','AAA7830513','AAA4536538','ZZ.ZZ.ZZ.ZZ') ERROR:db_unixodbc:db_unixodbc_extract_error: unixodbc:SQLExecDirect=08S01:1:1160:[ma-3.1.13][10.6.7-MariaDB-1:10.6.7+maria~focal] ERROR:db_unixodbc:reconnect: Attempting DB reconnect ERROR:db_unixodbc:db_unixodbc_submit_query: rv=-1. Query= call asterisk.gatekeeper('336f8bfc3d3e9eab7a624b507e7c7c9f at ZZ.ZZ.ZZ.ZZ ','AAA7830513','AAA3012713','ZZ.ZZ.ZZ.ZZ') ERROR:db_unixodbc:db_unixodbc_extract_error: unixodbc:SQLExecDirect=08S01:1:1160:[ma-3.1.13][10.6.7-MariaDB-1:10.6.7+maria~focal] ERROR:db_unixodbc:reconnect: Attempting DB reconnect ERROR:db_unixodbc:db_unixodbc_submit_query: rv=-1. Query= call asterisk.gatekeeper('06867d7363616c6c044c5b89 at ZZ.ZZ.ZZ.ZZ ','AAA5599174','AAA7284460','ZZ.ZZ.ZZ.ZZ') ERROR:db_unixodbc:db_unixodbc_extract_error: unixodbc:SQLExecDirect=08S01:1:1160:[ma-3.1.13][10.6.7-MariaDB-1:10.6.7+maria~focal] ERROR:db_unixodbc:reconnect: Attempting DB reconnect ERROR:rtpproxy:force_rtp_proxy_body: Failed to get dialog The next question is: I see a lot of ERROR:rtpproxy:force_rtp_proxy_body: incorrect port 0 in reply from rtp proxy SCRIPT: Failed to engage rtpproxy for trunk ZZ.ZZ.ZZ.ZZ - But none of those inbound trunks are using media-proxy, I mean, they are just going through the normal process for a call, where RTP goes straight between the parties. Why do I get these errors? The box has sometimes 10000 call attempts at the same time, and 10 RTP systemd services. many thanks Federico SCRIPT: Failed to engage rtpproxy for trunk ZZ.ZZ.ZZ.ZZ - 6b2197a60ed901941be16a5a1085531a at ZZ.ZZ.ZZ.ZZ ERROR:rtpproxy:force_rtp_proxy_body: Failed to get dialog SCRIPT: Failed to engage rtpproxy for trunk ZZ.ZZ.ZZ.ZZ - 07744b7b64f5805e2a4cadda40dfe1da at ZZ.ZZ.ZZ.ZZ ERROR:rtpproxy:force_rtp_proxy_body: Failed to get dialog SCRIPT: Failed to engage rtpproxy for trunk ZZ.ZZ.ZZ.ZZ - 3740afb80c04f1122b470b044e1cc293 at ZZ.ZZ.ZZ.ZZ ERROR:rtpproxy:force_rtp_proxy_body: incorrect port 0 in reply from rtp proxy SCRIPT: Failed to engage rtpproxy for trunk ZZ.ZZ.ZZ.ZZ - 584e46cd63616c6c044c74b1 at ZZ.ZZ.ZZ.ZZ ERROR:rtpproxy:force_rtp_proxy_body: incorrect port 0 in reply from rtp proxy SCRIPT: Failed to engage rtpproxy for trunk ZZ.ZZ.ZZ.ZZ - 058056e963616c6c044c926e at ZZ.ZZ.ZZ.ZZ ERROR:db_unixodbc:db_unixodbc_submit_query: rv=-1. Query= call asterisk.gatekeeper('7173b32375fb54d41bb96e60041bc970 at ZZ.ZZ.ZZ.ZZ ','AAA7830513','1AAA4991471','ZZ.ZZ.ZZ.ZZ') ERROR:db_unixodbc:db_unixodbc_extract_error: unixodbc:SQLExecDirect=08S01:1:1160:[ma-3.1.13][10.6.7-MariaDB-1:10.6.7+maria~focal] ERROR:db_unixodbc:reconnect: Attempting DB reconnect ERROR:rtpproxy:force_rtp_proxy_body: incorrect port 0 in reply from rtp proxy SCRIPT: Failed to engage rtpproxy for trunk ZZ.ZZ.ZZ.ZZ - 27f9b17b63616c6c044cb47c at ZZ.ZZ.ZZ.ZZ -------------- next part -------------- An HTML attachment was scrubbed... URL: From chester at zigbang.com Tue May 10 15:30:27 2022 From: chester at zigbang.com (Chester Lee) Date: Wed, 11 May 2022 00:30:27 +0900 Subject: [OpenSIPS-Users] opensips federate cluster - 407 response from different node Message-ID: Hello everyone, I'd like to ask about my troubles. I set up a federate cluster with mongodb with opensips 3.2.5. It has 2 nodes - node 1 and node 3. 1. One UAC which is registered on node 1 makes a call to one UAC which is registered on node 3. 2. node 1 lookup cachedb and get to know that callee is on node 3. 3. node 1 sends INVITE to node 3. 4. node 3 responses 407 with different nonce value. 5. node 1 just transfer 407 to tje caller with new nonce value from node 3. 6. The caller sends INVITE with authenticate with nonce value from node 3. But node 1 denies with 407 because the authenticate is not for node 1. How can I make a call between different nodes? Thank you Chester -------------- next part -------------- An HTML attachment was scrubbed... URL: From venefax at gmail.com Tue May 10 17:12:56 2022 From: venefax at gmail.com (Saint Michael) Date: Tue, 10 May 2022 13:12:56 -0400 Subject: [OpenSIPS-Users] Topology Hiding Message-ID: Dear friends I am using opensips 3.1.9, with rtp proxy, and without topology hiding it would not talk to any carrier who has a Sonus box. I need to add topology hiding urgently and my support provider is missing in action. Can somebody provide instructions and code samples? Federico -------------- next part -------------- An HTML attachment was scrubbed... URL: From vasilios.tzanoudakis at voiceland.gr Tue May 10 18:25:44 2022 From: vasilios.tzanoudakis at voiceland.gr (Vasilios Tzanoudakis) Date: Tue, 10 May 2022 21:25:44 +0300 Subject: [OpenSIPS-Users] opensips federate cluster - 407 response from different node In-Reply-To: References: Message-ID: Dear Chester, You should configure your cluster nodes to accept invites from other cluster nodes without doing www or proxy auth. Use cluster_check_addr() https://opensips.org/html/docs/modules/3.2.x/clusterer.html on top of your script to check that the call is from cluster and then do the lookup there to route the call immediately to the user registered;-) Good luck Vasilios Tzanoudakis Στις Τρί, 10 Μαΐ 2022, 18:32 ο χρήστης Chester Lee έγραψε: > Hello everyone, I'd like to ask about my troubles. > > I set up a federate cluster with mongodb with opensips 3.2.5. > > It has 2 nodes - node 1 and node 3. > 1. One UAC which is registered on node 1 makes a call to one UAC which is > registered on node 3. > 2. node 1 lookup cachedb and get to know that callee is on node 3. > 3. node 1 sends INVITE to node 3. > 4. node 3 responses 407 with different nonce value. > 5. node 1 just transfer 407 to tje caller with new nonce value from node 3. > 6. The caller sends INVITE with authenticate with nonce value from node 3. > But node 1 denies with 407 because the authenticate is not for node 1. > > How can I make a call between different nodes? > > Thank you > Chester > _______________________________________________ > 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 voransoy at gmail.com Wed May 11 10:14:31 2022 From: voransoy at gmail.com (Volkan Oransoy) Date: Wed, 11 May 2022 11:14:31 +0100 Subject: [OpenSIPS-Users] Drouting relay issue In-Reply-To: References: Message-ID: <32eada8e-1df1-4116-8ea5-07f994d846e3@Spark> Hi all, I have an interesting issue with one of my test setups. I have a simple routing script which gets the gateway id directly from the header originating from a Freeswith box. The system finds and sets the request URL as anticipated. But even if I can see the request on the proxy, I can't see the traffic on the destination. Interestingly, the same proxy can register to the same destination with uac_registrant as a UAC. And I can receive calls from the same destination. Is there anything missing to route this traffic correctly? Thanks in advance. route[to_gateway] {         if ( route_to_gw($hdr(X-GWID)) ) {                 route(relay);         } } route[relay] {         if (is_method("INVITE")) {                 t_on_branch("per_branch_ops");                 t_on_reply("handle_nat");                 t_on_failure("failure");         }         if (!t_relay()) {                 send_reply(500,"Internal Error");         }         exit; } The database structure is as follows; opensips=# select * from dr_gateways;  id | gwid | type |           address            | strip | pri_prefix | attrs | probe_mode | state | socket | description ----+------+------+------------------------------+-------+------------+-------+------------+-------+--------+-------------   6 | 5    |    2 | sip:testgw.bulutfon.net:5060 |     0 |            |       |          0 |     0 |        | 5 Here is the INVITE request sent to the destination, which fails as in the screenshot. INVITE sip:905551234567 at testgw.bulutfon.net:5060 SIP/2.0 Record-Route: Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK372.d935ecb2.0 Via: SIP/2.0/UDP 1.1.1.1:6080;received=1.1.1.1;rport=6080;branch=z9hG4bK99mtDaevmN7Nj Max-Forwards: 67 From: "+908508850000" ;tag=1jcyr93emrjDQ To: Call-ID: aefb8c60-426b-123b-8ca8-82a722ba4f75 CSeq: 51067506 INVITE Contact: Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 621 Remote-Party-ID: "+908508850000" ;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1651210438 1651210439 IN IP4 1.1.1.1 s=FreeSWITCH c=IN IP4 1.1.1.1 t=0 0 ... -- Volkan Oransoy -------------- next part -------------- An HTML attachment was scrubbed... URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: Screen Shot 2022-04-29 at 15.13.21.png Type: image/png Size: 19396 bytes Desc: not available URL: From voransoy at gmail.com Wed May 11 10:52:04 2022 From: voransoy at gmail.com (Volkan Oransoy) Date: Wed, 11 May 2022 11:52:04 +0100 Subject: [OpenSIPS-Users] Drouting relay issue In-Reply-To: <32eada8e-1df1-4116-8ea5-07f994d846e3@Spark> References: <32eada8e-1df1-4116-8ea5-07f994d846e3@Spark> Message-ID: <486d32ea-275b-479c-a1ee-318143a580a6@Spark> I think the screenshot has been discarded by the mailman. The sip traffic is as follows. The proxy tries to retransmit and fails after three more attempts. ──────────┬───────── ──────────┬───────── ──────────┬─────────  10:47:32.603828 │ INVITE (SDP) │ │  +0.000347 │ ──────────────────────────> │ │  10:47:32.604175 │ 100 Giving it a try │ │  +0.001474 │ <────────────────────────── │ │  10:47:32.605649 │ │ INVITE (SDP) │  +0.490742 │ │ ──────────────────────────> │  10:47:33.096391 │ │ INVITE (SDP) │  +1.001859 │ │ ────────────────────────>>> │  10:47:34.098250 │ │ INVITE (SDP) │  +1.953642 │ │ ────────────────────────>>> │  10:47:36.051892 │ │ INVITE (SDP) │  +1.603582 │ │ ────────────────────────>>> │  10:47:37.655474 │ 408 Request Timeout │ │  +0.001615 │ <────────────────────────── │ │  10:47:37.657089 │ ACK │ │  │ ──────────────────────────> │ │ Volkan Oransoy On 11 May 2022 11:14 +0100, Volkan Oransoy , wrote: > Hi all, > > I have an interesting issue with one of my test setups. I have a simple routing script which gets the gateway id directly from the header originating from a Freeswith box. The system finds and sets the request URL as anticipated. But even if I can see the request on the proxy, I can't see the traffic on the destination. Interestingly, the same proxy can register to the same destination with uac_registrant as a UAC. And I can receive calls from the same destination. Is there anything missing to route this traffic correctly? > > Thanks in advance. > > route[to_gateway] { >         if ( route_to_gw($hdr(X-GWID)) ) { >                 route(relay); >         } > } > route[relay] { >         if (is_method("INVITE")) { >                 t_on_branch("per_branch_ops"); >                 t_on_reply("handle_nat"); >                 t_on_failure("failure"); >         } >         if (!t_relay()) { >                 send_reply(500,"Internal Error"); >         } >         exit; > } > > The database structure is as follows; > > opensips=# select * from dr_gateways; >  id | gwid | type |           address            | strip | pri_prefix | attrs | probe_mode | state | socket | description > ----+------+------+------------------------------+-------+------------+-------+------------+-------+--------+------------- >   6 | 5    |    2 | sip:testgw.bulutfon.net:5060 |     0 |            |       |          0 |     0 |        | 5 > > Here is the INVITE request sent to the destination, which fails as in the screenshot. > > INVITE sip:905551234567 at testgw.bulutfon.net:5060 SIP/2.0 > Record-Route: > Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK372.d935ecb2.0 > Via: SIP/2.0/UDP 1.1.1.1:6080;received=1.1.1.1;rport=6080;branch=z9hG4bK99mtDaevmN7Nj > Max-Forwards: 67 > From: "+908508850000" ;tag=1jcyr93emrjDQ > To: > Call-ID: aefb8c60-426b-123b-8ca8-82a722ba4f75 > CSeq: 51067506 INVITE > Contact: > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY > Supported: timer, path, replaces > Allow-Events: talk, hold, conference, refer > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 621 > Remote-Party-ID: "+908508850000" ;party=calling;screen=yes;privacy=off > > v=0 > o=FreeSWITCH 1651210438 1651210439 IN IP4 1.1.1.1 > s=FreeSWITCH > c=IN IP4 1.1.1.1 > t=0 0 ... > > > -- > > Volkan Oransoy -------------- next part -------------- An HTML attachment was scrubbed... URL: From chester at zigbang.com Wed May 11 11:28:44 2022 From: chester at zigbang.com (Chester Lee) Date: Wed, 11 May 2022 20:28:44 +0900 Subject: [OpenSIPS-Users] opensips federate cluster - 407 response from different node In-Reply-To: References: Message-ID: Dear Vasilios Tzanoudakis and users, Thank you for your answer. I tried to add cluster_check_addr() in my script but I see the same symptom as before. --------------------------------------- route { .... # do lookup with method filtering $var(lookup_flags) = "m"; if (cluster_check_addr(1, "$si")) { xlog("L_NOTICE", "$rm from cluster, doing local lookup only. retcode=$retcode\n"); } else { xlog("L_NOTICE", "$rm from outside, doing global lookup. retcode=$retcode\n"); $var(lookup_flags) = $var(lookup_flags) + "g"; } if (!lookup("location","$var(lookup_flags)")) { if (!db_does_uri_exist("$ru","subscriber")) { send_reply(420,"Bad Extension"); exit; } t_reply(404, "Not Found"); exit; } if (isbflagset("NAT")) setflag("NAT"); # when routing via usrloc, log the missed calls also do_accounting("db","missed"); route(relay); } route[relay] { # for INVITEs enable some additional helper routes if (is_method("INVITE")) { t_on_branch("per_branch_ops"); t_on_reply("handle_nat"); t_on_failure("missed_call"); } if (isflagset("NAT")) { add_rr_param(";nat=yes"); } if (!t_relay()) { send_reply(500,"Internal Error"); } exit; } Or could you check mine attached here? 2022년 5월 11일 (수) 오전 3:28, Vasilios Tzanoudakis < vasilios.tzanoudakis at voiceland.gr>님이 작성: > Dear Chester, > > You should configure your cluster nodes to accept invites from other > cluster nodes without doing www or proxy auth. > > Use cluster_check_addr() > https://opensips.org/html/docs/modules/3.2.x/clusterer.html > > on top of your script > to check that the call is from cluster and then do the lookup there to > route the > call immediately to the user registered;-) > > Good luck > > Vasilios Tzanoudakis > > > Στις Τρί, 10 Μαΐ 2022, 18:32 ο χρήστης Chester Lee > έγραψε: > >> Hello everyone, I'd like to ask about my troubles. >> >> I set up a federate cluster with mongodb with opensips 3.2.5. >> >> It has 2 nodes - node 1 and node 3. >> 1. One UAC which is registered on node 1 makes a call to one UAC which is >> registered on node 3. >> 2. node 1 lookup cachedb and get to know that callee is on node 3. >> 3. node 1 sends INVITE to node 3. >> 4. node 3 responses 407 with different nonce value. >> 5. node 1 just transfer 407 to tje caller with new nonce value from node >> 3. >> 6. The caller sends INVITE with authenticate with nonce value from node >> 3. But node 1 denies with 407 because the authenticate is not for node 1. >> >> How can I make a call between different nodes? >> >> Thank you >> Chester >> _______________________________________________ >> 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 > -- ______ 이 기 원 CTO실 / 매니저 (주)직방 | 010.6479.1321 | chester at zigbang.com -------------- next part -------------- An HTML attachment was scrubbed... URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: opensips_chester.cfg Type: application/octet-stream Size: 11981 bytes Desc: not available URL: From vasilios.tzanoudakis at voiceland.gr Wed May 11 14:05:46 2022 From: vasilios.tzanoudakis at voiceland.gr (Vasilios Tzanoudakis) Date: Wed, 11 May 2022 17:05:46 +0300 Subject: [OpenSIPS-Users] opensips federate cluster - 407 response from different node In-Reply-To: References: Message-ID: Dear Chester, Try to move cluster_check_addr() and also copy lookup() too and put them higher in the script for example before ---> if ( !(is_method("REGISTER") ) ) {... Vasilios On Wed, May 11, 2022 at 2:30 PM Chester Lee wrote: > Dear Vasilios Tzanoudakis and users, > Thank you for your answer. > I tried to add cluster_check_addr() in my script but I see the same > symptom as before. > --------------------------------------- > route { > .... > > # do lookup with method filtering > $var(lookup_flags) = "m"; > > if (cluster_check_addr(1, "$si")) { > xlog("L_NOTICE", "$rm from cluster, doing local lookup only. > retcode=$retcode\n"); > } else { > xlog("L_NOTICE", "$rm from outside, doing global lookup. > retcode=$retcode\n"); > $var(lookup_flags) = $var(lookup_flags) + "g"; > } > if (!lookup("location","$var(lookup_flags)")) { > if (!db_does_uri_exist("$ru","subscriber")) { > send_reply(420,"Bad Extension"); > exit; > } > > t_reply(404, "Not Found"); > exit; > } > > if (isbflagset("NAT")) setflag("NAT"); > > # when routing via usrloc, log the missed calls also > do_accounting("db","missed"); > > route(relay); > } > > > route[relay] { > # for INVITEs enable some additional helper routes > if (is_method("INVITE")) { > t_on_branch("per_branch_ops"); > t_on_reply("handle_nat"); > t_on_failure("missed_call"); > } > > if (isflagset("NAT")) { > add_rr_param(";nat=yes"); > } > > if (!t_relay()) { > send_reply(500,"Internal Error"); > } > exit; > } > > > Or could you check mine attached here? > > > > > > > > > > > > 2022년 5월 11일 (수) 오전 3:28, Vasilios Tzanoudakis < > vasilios.tzanoudakis at voiceland.gr>님이 작성: > >> Dear Chester, >> >> You should configure your cluster nodes to accept invites from other >> cluster nodes without doing www or proxy auth. >> >> Use cluster_check_addr() >> https://opensips.org/html/docs/modules/3.2.x/clusterer.html >> >> on top of your script >> to check that the call is from cluster and then do the lookup there to >> route the >> call immediately to the user registered;-) >> >> Good luck >> >> Vasilios Tzanoudakis >> >> >> Στις Τρί, 10 Μαΐ 2022, 18:32 ο χρήστης Chester Lee >> έγραψε: >> >>> Hello everyone, I'd like to ask about my troubles. >>> >>> I set up a federate cluster with mongodb with opensips 3.2.5. >>> >>> It has 2 nodes - node 1 and node 3. >>> 1. One UAC which is registered on node 1 makes a call to one UAC which >>> is registered on node 3. >>> 2. node 1 lookup cachedb and get to know that callee is on node 3. >>> 3. node 1 sends INVITE to node 3. >>> 4. node 3 responses 407 with different nonce value. >>> 5. node 1 just transfer 407 to tje caller with new nonce value from node >>> 3. >>> 6. The caller sends INVITE with authenticate with nonce value from node >>> 3. But node 1 denies with 407 because the authenticate is not for node 1. >>> >>> How can I make a call between different nodes? >>> >>> Thank you >>> Chester >>> _______________________________________________ >>> 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 >> > > > -- > ______ > > > 이 기 원 CTO실 / 매니저 > (주)직방 | 010.6479.1321 | chester at zigbang.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 daniel.zanutti at gmail.com Wed May 11 14:37:16 2022 From: daniel.zanutti at gmail.com (Daniel Zanutti) Date: Wed, 11 May 2022 11:37:16 -0300 Subject: [OpenSIPS-Users] Topology Hiding In-Reply-To: References: Message-ID: https://www.opensips.org/Documentation/Tutorials-Topology-Hiding On Tue, May 10, 2022 at 2:15 PM Saint Michael wrote: > Dear friends > I am using opensips 3.1.9, with rtp proxy, and without topology hiding it > would not talk to any carrier who has a Sonus box. I need to add topology > hiding urgently and my support provider is missing in action. Can somebody > provide instructions and code samples? > > Federico > > _______________________________________________ > 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 venefax at gmail.com Wed May 11 16:40:40 2022 From: venefax at gmail.com (Saint Michael) Date: Wed, 11 May 2022 12:40:40 -0400 Subject: [OpenSIPS-Users] Topology Hiding In-Reply-To: References: Message-ID: Thanks It's solved. Federico On Wed, May 11, 2022, 10:40 AM Daniel Zanutti wrote: > https://www.opensips.org/Documentation/Tutorials-Topology-Hiding > > On Tue, May 10, 2022 at 2:15 PM Saint Michael wrote: > >> Dear friends >> I am using opensips 3.1.9, with rtp proxy, and without topology hiding it >> would not talk to any carrier who has a Sonus box. I need to add topology >> hiding urgently and my support provider is missing in action. Can somebody >> provide instructions and code samples? >> >> Federico >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -------------- next part -------------- An HTML attachment was scrubbed... URL: From pimenta at inatel.br Wed May 11 19:10:38 2022 From: pimenta at inatel.br (Rodrigo Pimenta Carvalho) Date: Wed, 11 May 2022 19:10:38 +0000 Subject: [OpenSIPS-Users] Question about error 500 only via WIFI Message-ID: Hi. My SIP proxy is an OpenSIPS version 2.4.7. It was 'off' for a while and today I turned it on again for some tests. I usually use my home office local WIFI to connect my softphones to the network and it can be all connected (online) to this SIP proxy. However, if I use the mobile network (LTE/4G) to connect the softphones to the SIP proxy, the OpenSIP returns an error 500. That is: "SIP/2.0 500 Server error occurred (7/TM)". One of the softphones (Zoiper) allows me to use a kind of "proxy PUSH". If I use it, the problem is solved even while using the mobile network. What is a proxy PUSH? Why OpenSIPs return error in a case, but not in the other one? What could I do to avoid using a 'proxy PUSH'? Local WIFI and mobile network come from different carriers. Any hint will be very helpful! Best regards. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL: 979 -------------- next part -------------- An HTML attachment was scrubbed... URL: From spanda at 3clogic.com Thu May 12 07:05:03 2022 From: spanda at 3clogic.com (Sasmita Panda) Date: Thu, 12 May 2022 12:35:03 +0530 Subject: [OpenSIPS-Users] Need some help in clusterer table and its use in opensips 3.2 . Message-ID: Hi All , I have 2 opesips cluster each cluster has 2 opensips node . I want to define all the cluster node information in a single opensips database . My clusterer table looks like below . +----+------------+---------+-----------------------+-------+-----------------+----------+----------+-------+----------------+ | id | cluster_id | node_id | url | state | no_ping_retries | priority | sip_addr | flags | description | +----+------------+---------+-----------------------+-------+-----------------+----------+----------+-------+----------------+ | 1 | 1 | 1 | bin:1.1.1.1:5555 | 1 | 3 | 50 | NULL | seed | Node A | | 2 | 1 | 2 | bin:2.2.2.2:5555 | 1 | 3 | 50 | NULL | seed | Node B | | 3 | 2 | 1 | bin:3.3.3.3:5555 | 1 | 3 | 50 | NULL | NULL | cluster2 Node1 | | 4 | 2 | 2 | bin:4.4.4.4:5555 | 1 | 3 | 50 | NULL | NULL | cluster2 Node2 | +----+------------+---------+-----------------------+-------+-----------------+----------+----------+-------+----------------+ In pensips 3.2 there is no cluster_id parameter to define in the config . In the config I don't want to add the IP in the config . For cluster 2 , when I am defining node 1 , its taking the value of node 1 of cluster 1 . Is this possible anyhow or I have to save the data in a different database ? *Thanks & Regards* *Sasmita Panda* *Senior Network Testing and Software Engineer* *3CLogic , ph:07827611765* -------------- next part -------------- An HTML attachment was scrubbed... URL: From chester at zigbang.com Thu May 12 07:47:45 2022 From: chester at zigbang.com (Chester Lee) Date: Thu, 12 May 2022 16:47:45 +0900 Subject: [OpenSIPS-Users] Need some help in clusterer table and its use in opensips 3.2 . In-Reply-To: References: Message-ID: Hi, You can specify cluster id in the config. please refer to https://opensips.org/docs/modules/3.2.x/clusterer.html#param_my_node_id I hope this helps. Regards Chester 2022년 5월 12일 (목) 오후 4:06, Sasmita Panda 님이 작성: > Hi All , > > I have 2 opesips cluster each cluster has 2 opensips node . I want to > define all the cluster node information in a single opensips database . > > My clusterer table looks like below . > > > +----+------------+---------+-----------------------+-------+-----------------+----------+----------+-------+----------------+ > | id | cluster_id | node_id | url | state | > no_ping_retries | priority | sip_addr | flags | description | > > +----+------------+---------+-----------------------+-------+-----------------+----------+----------+-------+----------------+ > | 1 | 1 | 1 | bin:1.1.1.1:5555 | 1 | 3 > | 50 | NULL | seed | Node A | > | 2 | 1 | 2 | bin:2.2.2.2:5555 | 1 | 3 > | 50 | NULL | seed | Node B | > | 3 | 2 | 1 | bin:3.3.3.3:5555 | 1 | 3 > | 50 | NULL | NULL | cluster2 Node1 | > | 4 | 2 | 2 | bin:4.4.4.4:5555 | 1 | 3 > | 50 | NULL | NULL | cluster2 Node2 | > > +----+------------+---------+-----------------------+-------+-----------------+----------+----------+-------+----------------+ > > In pensips 3.2 there is no cluster_id parameter to define in the config . > In the config I don't want to add the IP in the config . For cluster 2 , > when I am defining node 1 , its taking the value of node 1 of cluster 1 . > > Is this possible anyhow or I have to save the data in a different database > ? > > > *Thanks & Regards* > *Sasmita Panda* > *Senior Network Testing and Software Engineer* > *3CLogic , ph:07827611765* > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -- ______ 이 기 원 CTO실 / 매니저 (주)직방 | 010.6479.1321 | chester at zigbang.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From spanda at 3clogic.com Thu May 12 08:43:47 2022 From: spanda at 3clogic.com (Sasmita Panda) Date: Thu, 12 May 2022 14:13:47 +0530 Subject: [OpenSIPS-Users] Need some help in clusterer table and its use in opensips 3.2 . In-Reply-To: References: Message-ID: Hi , my_node_id parameter is to define that particular node id . But how will I define the cluster_id parameter? I want to differentiate both clusters . I have added my_node_id parameter already . But by default its looking for cluster_id:1 . But in the database I have defined cluster_id :2 . modparam("clusterer", "my_node_id", 1) How will I associate the database and config ? *Thanks & Regards* *Sasmita Panda* *Senior Network Testing and Software Engineer* *3CLogic , ph:07827611765* On Thu, May 12, 2022 at 1:20 PM Chester Lee wrote: > Hi, > > You can specify cluster id in the config. please refer to > https://opensips.org/docs/modules/3.2.x/clusterer.html#param_my_node_id > > I hope this helps. > > Regards > Chester > > > 2022년 5월 12일 (목) 오후 4:06, Sasmita Panda 님이 작성: > >> Hi All , >> >> I have 2 opesips cluster each cluster has 2 opensips node . I want to >> define all the cluster node information in a single opensips database . >> >> My clusterer table looks like below . >> >> >> +----+------------+---------+-----------------------+-------+-----------------+----------+----------+-------+----------------+ >> | id | cluster_id | node_id | url | state | >> no_ping_retries | priority | sip_addr | flags | description | >> >> +----+------------+---------+-----------------------+-------+-----------------+----------+----------+-------+----------------+ >> | 1 | 1 | 1 | bin:1.1.1.1:5555 | 1 | 3 >> | 50 | NULL | seed | Node A | >> | 2 | 1 | 2 | bin:2.2.2.2:5555 | 1 | >> 3 | 50 | NULL | seed | Node B | >> | 3 | 2 | 1 | bin:3.3.3.3:5555 | 1 | 3 >> | 50 | NULL | NULL | cluster2 Node1 | >> | 4 | 2 | 2 | bin:4.4.4.4:5555 | 1 | 3 >> | 50 | NULL | NULL | cluster2 Node2 | >> >> +----+------------+---------+-----------------------+-------+-----------------+----------+----------+-------+----------------+ >> >> In pensips 3.2 there is no cluster_id parameter to define in the config . >> In the config I don't want to add the IP in the config . For cluster 2 , >> when I am defining node 1 , its taking the value of node 1 of cluster 1 . >> >> Is this possible anyhow or I have to save the data in a different >> database ? >> >> >> *Thanks & Regards* >> *Sasmita Panda* >> *Senior Network Testing and Software Engineer* >> *3CLogic , ph:07827611765* >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> > > > -- > ______ > > > 이 기 원 CTO실 / 매니저 > (주)직방 | 010.6479.1321 | chester at zigbang.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 daniel.zanutti at gmail.com Thu May 12 14:57:23 2022 From: daniel.zanutti at gmail.com (Daniel Zanutti) Date: Thu, 12 May 2022 11:57:23 -0300 Subject: [OpenSIPS-Users] Question about error 500 only via WIFI In-Reply-To: References: Message-ID: Olá Rodrigo, tudo bem? Saudações de São Paulo! Opensips doesn't differentiate the network, it will look just to the sip packet. I recommend you sniff through your packets and check what's different. Probably there's somenthing on opensips log you didn't get yet, recommend you take a look there first btw. About push, I think you're enable push notifications on your device, take a look: https://www.zoiper.com/en/support/home/article/205/Zoiper%20Push%20Proxy Regards On Wed, May 11, 2022 at 4:13 PM Rodrigo Pimenta Carvalho wrote: > Hi. > > My SIP proxy is an OpenSIPS version 2.4.7. It was 'off' for a while and > today I turned it on again for some tests. > > I usually use my home office local WIFI to connect my softphones to the > network and it can be all connected (online) to this SIP proxy. > > However, if I use the mobile network (LTE/4G) to connect the softphones to > the SIP proxy, the OpenSIP returns an error 500. That is: "SIP/2.0 500 > Server error occurred (7/TM)". > > One of the softphones (Zoiper) allows me to use a kind of "proxy PUSH". If > I use it, the problem is solved even while using the mobile network. > > What is a proxy PUSH? Why OpenSIPs return error in a case, but not in the > other one? > What could I do to avoid using a 'proxy PUSH'? > > Local WIFI and mobile network come from different carriers. > > Any hint will be very helpful! > > Best regards. > > RODRIGO PIMENTA CARVALHO > Inatel Competence Center > Software > Ph: +55 35 3471 9200 RAMAL: 979 > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -------------- next part -------------- An HTML attachment was scrubbed... URL: From pimenta at inatel.br Thu May 12 18:14:28 2022 From: pimenta at inatel.br (Rodrigo Pimenta Carvalho) Date: Thu, 12 May 2022 18:14:28 +0000 Subject: [OpenSIPS-Users] Question about error 500 only via WIFI In-Reply-To: References: Message-ID: Olá Daniel. Thank you ! I will take a look. Regards. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL: 979 ________________________________ De: Users em nome de Daniel Zanutti Enviado: quinta-feira, 12 de maio de 2022 11:57 Para: OpenSIPS users mailling list Assunto: Re: [OpenSIPS-Users] Question about error 500 only via WIFI Olá Rodrigo, tudo bem? Saudações de São Paulo! Opensips doesn't differentiate the network, it will look just to the sip packet. I recommend you sniff through your packets and check what's different. Probably there's somenthing on opensips log you didn't get yet, recommend you take a look there first btw. About push, I think you're enable push notifications on your device, take a look: https://www.zoiper.com/en/support/home/article/205/Zoiper%20Push%20Proxy Regards On Wed, May 11, 2022 at 4:13 PM Rodrigo Pimenta Carvalho > wrote: Hi. My SIP proxy is an OpenSIPS version 2.4.7. It was 'off' for a while and today I turned it on again for some tests. I usually use my home office local WIFI to connect my softphones to the network and it can be all connected (online) to this SIP proxy. However, if I use the mobile network (LTE/4G) to connect the softphones to the SIP proxy, the OpenSIP returns an error 500. That is: "SIP/2.0 500 Server error occurred (7/TM)". One of the softphones (Zoiper) allows me to use a kind of "proxy PUSH". If I use it, the problem is solved even while using the mobile network. What is a proxy PUSH? Why OpenSIPs return error in a case, but not in the other one? What could I do to avoid using a 'proxy PUSH'? Local WIFI and mobile network come from different carriers. Any hint will be very helpful! Best regards. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL: 979 _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From pimenta at inatel.br Thu May 12 19:54:30 2022 From: pimenta at inatel.br (Rodrigo Pimenta Carvalho) Date: Thu, 12 May 2022 19:54:30 +0000 Subject: [OpenSIPS-Users] Question about error 500 only via WIFI In-Reply-To: References: Message-ID: Hi. I found the error cause. But I still don't know why I have such issue. When I use my Internet Link (WIFI in my home office), the SIP register message is sent correctly. Like this: Session Initiation Protocol (REGISTER) Request-Line: REGISTER sip:54.233.189.46:5060;transport=UDP SIP/2.0 Method: REGISTER Request-URI: sip:54.233.189.46:5060;transport=UDP [Resent Packet: False] Message Header Via: SIP/2.0/UDP 192.168.1.103:5060;branch=z9hG4bK-524287-1---6dbfa766cffddeee;rport Max-Forwards: 70 Contact: To: ;transport=UDP> From: ;tag=98bfc34c Call-ID: H1E0jkwiMniiyT5az1BT7g.. CSeq: 1 REGISTER Expires: 60 Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE User-Agent: Zoiper v2.10.18.1-mod Allow-Events: presence, kpml, talk Content-Length: 0 Opensips got the message above. However, when I use the GSM mobile network (from VIVO) , some service changes the content of the SIP Register message. Like this: Session Initiation Protocol (REGISTER) Request-Line: REGISTER sip:[64:ff9b::36e9:bd2e]:5060;transport=UDP SIP/2.0 Method: REGISTER Request-URI: sip:[64:ff9b::36e9:bd2e]:5060;transport=UDP [Resent Packet: False] Message Header Via: SIP/2.0/UDP [64:ff9b::c000:4];branch=z9hG4bK-524287-1---0a8189adf6c3449a Max-Forwards: 70 Contact: To: From: ;tag=f19aea4d Call-ID: VICBinZsDk5_ZhpHGd__CQ.. CSeq: 1 REGISTER Expires: 60 Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE User-Agent: Zoiper v2.10.18.1-mod Allow-Events: presence, kpml, talk Content-Length: 0 That is why Opensips returns error 500. I guess some service changed IPv4 to something IPv6. Could it be caused by the GSM operator (VIVO) ? What should I investigage to solve this problem? Any hint will be very helpful ! Thanks alot. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL: 979 ________________________________ De: Users em nome de Daniel Zanutti Enviado: quinta-feira, 12 de maio de 2022 11:57 Para: OpenSIPS users mailling list Assunto: Re: [OpenSIPS-Users] Question about error 500 only via WIFI Olá Rodrigo, tudo bem? Saudações de São Paulo! Opensips doesn't differentiate the network, it will look just to the sip packet. I recommend you sniff through your packets and check what's different. Probably there's somenthing on opensips log you didn't get yet, recommend you take a look there first btw. About push, I think you're enable push notifications on your device, take a look: https://www.zoiper.com/en/support/home/article/205/Zoiper%20Push%20Proxy Regards On Wed, May 11, 2022 at 4:13 PM Rodrigo Pimenta Carvalho > wrote: Hi. My SIP proxy is an OpenSIPS version 2.4.7. It was 'off' for a while and today I turned it on again for some tests. I usually use my home office local WIFI to connect my softphones to the network and it can be all connected (online) to this SIP proxy. However, if I use the mobile network (LTE/4G) to connect the softphones to the SIP proxy, the OpenSIP returns an error 500. That is: "SIP/2.0 500 Server error occurred (7/TM)". One of the softphones (Zoiper) allows me to use a kind of "proxy PUSH". If I use it, the problem is solved even while using the mobile network. What is a proxy PUSH? Why OpenSIPs return error in a case, but not in the other one? What could I do to avoid using a 'proxy PUSH'? Local WIFI and mobile network come from different carriers. Any hint will be very helpful! Best regards. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL: 979 _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From daniel.zanutti at gmail.com Fri May 13 14:14:45 2022 From: daniel.zanutti at gmail.com (Daniel Zanutti) Date: Fri, 13 May 2022 11:14:45 -0300 Subject: [OpenSIPS-Users] Question about error 500 only via WIFI In-Reply-To: References: Message-ID: Olá Rodrigo The problem seems to be the IPV6 addresses. Did you look at the logs to know exactly why opensips refused the call? IPV4 is almost out in Brazil, mobile carriers are using ipv6 addresses on mobile devices. This is the issue, but in theory opensips should handle this seamlessly. Do you have IPV6 in your server or the message came on ipv4 address but SIP has ipv6 addresses? I recommend you check logs, the issue should be there. Also, if you plan to allow mobile devices, you should be prepared to use ipv6 on your net. On Thu, May 12, 2022 at 4:57 PM Rodrigo Pimenta Carvalho wrote: > Hi. I found the error cause. But I still don't know why I have such > issue. > > > When I use my Internet Link (WIFI in my home office), the SIP register > message is sent correctly. Like this: > > Session Initiation Protocol (REGISTER) > Request-Line: REGISTER sip:54.233.189.46:5060;transport=UDP SIP/2.0 > Method: REGISTER > Request-URI: sip:54.233.189.46:5060;transport=UDP > [Resent Packet: False] > Message Header > Via: SIP/2.0/UDP 192.168.1.103:5060 > ;branch=z9hG4bK-524287-1---6dbfa766cffddeee;rport > Max-Forwards: 70 > Contact: ;rinstance=1afa98b2b6d17a34;transport=UDP> > To: ;transport=UDP> > From: ;tag=98bfc34c > Call-ID: H1E0jkwiMniiyT5az1BT7g.. > CSeq: 1 REGISTER > Expires: 60 > Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, > INFO, SUBSCRIBE > User-Agent: Zoiper v2.10.18.1-mod > Allow-Events: presence, kpml, talk > Content-Length: 0 > > Opensips got the message above. > > However, when I use the GSM mobile network (from VIVO) , some service > changes the content of the SIP Register message. Like this: > > Session Initiation Protocol (REGISTER) > Request-Line: REGISTER sip:*[64:ff9b::36e9:bd2e]*:5060;transport=UDP > SIP/2.0 > Method: REGISTER > Request-URI: sip:*[64:ff9b::36e9:bd2e]*:5060;transport=UDP > [Resent Packet: False] > Message Header > Via: SIP/2.0/UDP *[64:ff9b::c000:4]* > ;branch=z9hG4bK-524287-1---0a8189adf6c3449a > Max-Forwards: 70 > Contact: :5060;transport=UDP;rinstance=ffaac43f13178e89> > To: > From: ;tag=f19aea4d > Call-ID: VICBinZsDk5_ZhpHGd__CQ.. > CSeq: 1 REGISTER > Expires: 60 > Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, > INFO, SUBSCRIBE > User-Agent: Zoiper v2.10.18.1-mod > Allow-Events: presence, kpml, talk > Content-Length: 0 > > > That is why Opensips returns error 500. I guess some service changed IPv4 > to something IPv6. > Could it be caused by the GSM operator (VIVO) ? > What should I investigage to solve this problem? > > Any hint will be very helpful ! > > Thanks alot. > > RODRIGO PIMENTA CARVALHO > Inatel Competence Center > Software > Ph: +55 35 3471 9200 RAMAL: 979 > > ------------------------------ > *De:* Users em nome de Daniel Zanutti < > daniel.zanutti at gmail.com> > *Enviado:* quinta-feira, 12 de maio de 2022 11:57 > *Para:* OpenSIPS users mailling list > *Assunto:* Re: [OpenSIPS-Users] Question about error 500 only via WIFI > > Olá Rodrigo, tudo bem? Saudações de São Paulo! > > Opensips doesn't differentiate the network, it will look just to the sip > packet. I recommend you sniff through your packets and check what's > different. Probably there's somenthing on opensips log you didn't get yet, > recommend you take a look there first btw. > > About push, I think you're enable push notifications on your device, take > a look: > https://www.zoiper.com/en/support/home/article/205/Zoiper%20Push%20Proxy > > Regards > > > On Wed, May 11, 2022 at 4:13 PM Rodrigo Pimenta Carvalho < > pimenta at inatel.br> wrote: > > Hi. > > My SIP proxy is an OpenSIPS version 2.4.7. It was 'off' for a while and > today I turned it on again for some tests. > > I usually use my home office local WIFI to connect my softphones to the > network and it can be all connected (online) to this SIP proxy. > > However, if I use the mobile network (LTE/4G) to connect the softphones to > the SIP proxy, the OpenSIP returns an error 500. That is: "SIP/2.0 500 > Server error occurred (7/TM)". > > One of the softphones (Zoiper) allows me to use a kind of "proxy PUSH". If > I use it, the problem is solved even while using the mobile network. > > What is a proxy PUSH? Why OpenSIPs return error in a case, but not in the > other one? > What could I do to avoid using a 'proxy PUSH'? > > Local WIFI and mobile network come from different carriers. > > Any hint will be very helpful! > > Best regards. > > RODRIGO PIMENTA CARVALHO > Inatel Competence Center > Software > Ph: +55 35 3471 9200 RAMAL: 979 > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > _______________________________________________ > 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 goatolina at gmail.com Fri May 13 23:30:47 2022 From: goatolina at gmail.com (Ali Alawi) Date: Sat, 14 May 2022 02:30:47 +0300 Subject: [OpenSIPS-Users] Cli and DB path Message-ID: Dear all, When I install opensips3.2 using APT packages, the cli point correctly to mysql (mariadb) through /usr/share/opensips (Everything work fine) However, when installation done using git clone --recursive, the cli point to /usr/share/opensips but in this time the cli doesn't find mysql when i try to: opensips-cli -x database create ERROR: path '/usr/share/opensips' to OpenSIPS DB scripts does not exist! I notice that mysql is resides inside '/usr/local/share/opensips' instead of '/usr/share/opensips' I try to include the corrected path in the default.cfg and also try opensips-cli -o database_schema_path= But I come up with no success Any suggestions please? Regards, Ali -------------- next part -------------- An HTML attachment was scrubbed... URL: From ray at hero.co.nz Sat May 14 02:42:07 2022 From: ray at hero.co.nz (Ray Jackson) Date: Sat, 14 May 2022 14:42:07 +1200 Subject: [OpenSIPS-Users] Maximum number of ReINVITEs option? Message-ID: <3f747271-2af5-2636-17b9-f152b185ba0b@hero.co.nz> Hi all, I have a misbehaving SIP client which is repeatedly sending SIP ReINVITE messages over and over again (in the hundreds) and want to enforce a limit on the number of ReINVITE messages per Call to stop this in it's tracks.  The CSeq is incrementing on each ReINVITE. Are there any config settings to limit the number of ReINVITEs on a single Call/Dialog?  I was going to possibly keep a count of the INVITEs using a cache variable based on the Unique CallID but wanted to check first there wasn't a simpler (built-in) way of doing this? Thanks, Ray From jehanzaib.kiani at gmail.com Sat May 14 02:59:35 2022 From: jehanzaib.kiani at gmail.com (Jehanzaib Younis) Date: Sat, 14 May 2022 14:59:35 +1200 Subject: [OpenSIPS-Users] Maximum number of ReINVITEs option? In-Reply-To: <3f747271-2af5-2636-17b9-f152b185ba0b@hero.co.nz> References: <3f747271-2af5-2636-17b9-f152b185ba0b@hero.co.nz> Message-ID: Hi Ray, I assume the Max-Forwards remains same but the CSeq is increasing ya? Regards, Jehan On Sat, May 14, 2022 at 2:42 PM Ray Jackson wrote: > Hi all, > > I have a misbehaving SIP client which is repeatedly sending SIP ReINVITE > messages over and over again (in the hundreds) and want to enforce a > limit on the number of ReINVITE messages per Call to stop this in it's > tracks. The CSeq is incrementing on each ReINVITE. > > Are there any config settings to limit the number of ReINVITEs on a > single Call/Dialog? I was going to possibly keep a count of the INVITEs > using a cache variable based on the Unique CallID but wanted to check > first there wasn't a simpler (built-in) way of doing this? > > Thanks, > Ray > > > > > _______________________________________________ > 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 ray at hero.co.nz Sat May 14 05:05:56 2022 From: ray at hero.co.nz (Ray Jackson) Date: Sat, 14 May 2022 17:05:56 +1200 Subject: [OpenSIPS-Users] Maximum number of ReINVITEs option? In-Reply-To: References: <3f747271-2af5-2636-17b9-f152b185ba0b@hero.co.nz> Message-ID: Correct, Max-Forwards stays the same on each ReINVITE and the CSeq increments by 1 each time.  All other headers stay the same. I have no idea what is broken with the client to continually ReINVITE in a loop.  We accept the ReINVITE with an OK and we receive an ACK back and then immediately another INVITE and so it goes on until the call is terminated. I'm less interested in working out what is broken on the client end and more interested in protecting our network. Thanks, Ray On 14/05/22 2:59 pm, Jehanzaib Younis wrote: > Hi Ray, > > I assume the Max-Forwards remains same but the CSeq is increasing ya? > > Regards, > Jehan > > > On Sat, May 14, 2022 at 2:42 PM Ray Jackson wrote: > > Hi all, > > I have a misbehaving SIP client which is repeatedly sending SIP > ReINVITE > messages over and over again (in the hundreds) and want to enforce a > limit on the number of ReINVITE messages per Call to stop this in > it's > tracks.  The CSeq is incrementing on each ReINVITE. > > Are there any config settings to limit the number of ReINVITEs on a > single Call/Dialog?  I was going to possibly keep a count of the > INVITEs > using a cache variable based on the Unique CallID but wanted to check > first there wasn't a simpler (built-in) way of doing this? > > Thanks, > Ray > > > > > _______________________________________________ > 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 -- Email Signature Ray Jackson Chief Technology Officer Logo Hero Internet Level 4, 220 Queen Street Auckland t: +649-242-0001 e: ray at hero.co.nz hero.co.nz Facebook icon Twitter icon -------------- next part -------------- An HTML attachment was scrubbed... URL: From johan at democon.be Sat May 14 11:30:33 2022 From: johan at democon.be (Johan De Clercq) Date: Sat, 14 May 2022 11:30:33 +0000 Subject: [OpenSIPS-Users] Maximum number of ReINVITEs option? In-Reply-To: <3f747271-2af5-2636-17b9-f152b185ba0b@hero.co.nz> References: <3f747271-2af5-2636-17b9-f152b185ba0b@hero.co.nz> Message-ID: Fail2ban if they come quickly? Outlook voor iOS downloaden ________________________________ Van: Users namens Ray Jackson Verzonden: Saturday, May 14, 2022 4:42:07 AM Aan: users at lists.opensips.org Onderwerp: [OpenSIPS-Users] Maximum number of ReINVITEs option? Hi all, I have a misbehaving SIP client which is repeatedly sending SIP ReINVITE messages over and over again (in the hundreds) and want to enforce a limit on the number of ReINVITE messages per Call to stop this in it's tracks. The CSeq is incrementing on each ReINVITE. Are there any config settings to limit the number of ReINVITEs on a single Call/Dialog? I was going to possibly keep a count of the INVITEs using a cache variable based on the Unique CallID but wanted to check first there wasn't a simpler (built-in) way of doing this? Thanks, Ray _______________________________________________ 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 jehanzaib.kiani at gmail.com Sun May 15 00:01:59 2022 From: jehanzaib.kiani at gmail.com (Jehanzaib Younis) Date: Sun, 15 May 2022 12:01:59 +1200 Subject: [OpenSIPS-Users] Maximum number of ReINVITEs option? In-Reply-To: References: <3f747271-2af5-2636-17b9-f152b185ba0b@hero.co.nz> Message-ID: Hi Ray, I did not see any built-in way to restrict the number of re-invites. The problem with this buggy client is, if your SBC does not respond the client's reinvite. It might send BYE ;) so even if you restrict the reinvites that might not work well. Regards, Jehan On Sat, May 14, 2022 at 5:06 PM Ray Jackson wrote: > Correct, Max-Forwards stays the same on each ReINVITE and the CSeq > increments by 1 each time. All other headers stay the same. I have no idea > what is broken with the client to continually ReINVITE in a loop. We > accept the ReINVITE with an OK and we receive an ACK back and then > immediately another INVITE and so it goes on until the call is terminated. > > I'm less interested in working out what is broken on the client end and > more interested in protecting our network. > > Thanks, > > Ray > On 14/05/22 2:59 pm, Jehanzaib Younis wrote: > > Hi Ray, > > I assume the Max-Forwards remains same but the CSeq is increasing ya? > > Regards, > Jehan > > > On Sat, May 14, 2022 at 2:42 PM Ray Jackson wrote: > >> Hi all, >> >> I have a misbehaving SIP client which is repeatedly sending SIP ReINVITE >> messages over and over again (in the hundreds) and want to enforce a >> limit on the number of ReINVITE messages per Call to stop this in it's >> tracks. The CSeq is incrementing on each ReINVITE. >> >> Are there any config settings to limit the number of ReINVITEs on a >> single Call/Dialog? I was going to possibly keep a count of the INVITEs >> using a cache variable based on the Unique CallID but wanted to check >> first there wasn't a simpler (built-in) way of doing this? >> >> Thanks, >> Ray >> >> >> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> > > _______________________________________________ > Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users > > -- > Ray Jackson > Chief Technology Officer [image: Logo] > > Hero Internet > Level 4, 220 Queen Street > Auckland > t: +649-242-0001 > e: ray at hero.co.nz > > hero.co.nz [image: Facebook icon] > [image: Twitter icon] > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -------------- next part -------------- An HTML attachment was scrubbed... URL: From razvan at opensips.org Tue May 17 07:48:07 2022 From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=) Date: Tue, 17 May 2022 10:48:07 +0300 Subject: [OpenSIPS-Users] Cli and DB path In-Reply-To: References: Message-ID: <9f57138c-5d40-b8f5-8da2-8fb6cf4e91c2@opensips.org> Hi, Ali! Setting the database_schema_path should do the trick. Can you set it again and provide the logs? Best regards, Răzvan Crainea OpenSIPS Core Developer http://www.opensips-solutions.com On 5/14/22 02:30, Ali Alawi wrote: > Dear all, > > When I install opensips3.2 using APT packages, the cli point correctly > to mysql (mariadb) through /usr/share/opensips (Everything work fine) > > However, when  installation done using git clone --recursive, the cli > point to /usr/share/opensips  but in this time the cli doesn't find > mysql when i try to: > opensips-cli -x database create > ERROR: path '/usr/share/opensips' to OpenSIPS DB scripts does not exist! > > I notice that mysql is resides inside '/usr/local/share/opensips' > instead of '/usr/share/opensips' > I try to include the corrected path in the default.cfg and also try > |opensips-cli -o database_schema_path=| > |But I come up with no success| > |Any suggestions please? | > > Regards, > Ali > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users From hobe69 at hotmail.com Tue May 17 07:55:47 2022 From: hobe69 at hotmail.com (Bela H) Date: Tue, 17 May 2022 07:55:47 +0000 Subject: [OpenSIPS-Users] OpenSIPS CP upgrade to 9.3.2 Message-ID: Hello, What is the best method to upgrade the control panel from 8.3.2 to 9.3.2? I had some extra fields e.g. in CDR viewer and disappeared after 9.3.2. It is in the file /var/www/html/opensips-cp/config/tools/system/cdrviewer/local.inc.php but not visible in the CDR viewer panel only in the detailed view for each call. Also I don’t see this gear icon Bogdan mentioned in the blog: Each tool has its own Settings panel “accessible via the gear-icon in the right side of the tool header”. Cheers, Bela -------------- next part -------------- An HTML attachment was scrubbed... URL: From razvan at opensips.org Tue May 17 08:00:04 2022 From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=) Date: Tue, 17 May 2022 11:00:04 +0300 Subject: [OpenSIPS-Users] Need some help in clusterer table and its use in opensips 3.2 . In-Reply-To: References: Message-ID: Hi, Sasmita! I don't fully understand your use case - you said it is using node 1 in cluster 1 - it is using it for what? A cluster is used for a specific replication feature (i.e. dialog replication, ratelimit pipes replication). When you specify you want to do a specific replication, that's where you specify the cluster (i.e. dialog replication [1]). So what kind of replication feature are you using, that is not properly identifying the nodes? [1] https://opensips.org/docs/modules/3.2.x/dialog.html#param_dialog_replication_cluster Răzvan Crainea OpenSIPS Core Developer http://www.opensips-solutions.com On 5/12/22 11:43, Sasmita Panda wrote: > Hi , > > my_node_id parameter is to define that particular node id . But how will > I define the cluster_id parameter? I want to differentiate both clusters . > I have added my_node_id parameter already . But by default its looking > for cluster_id:1 . But in the database I have defined cluster_id :2 . > > modparam("clusterer", "my_node_id", 1) > > How will I associate the database and config ? > > */Thanks & Regards/* > /Sasmita Panda/ > /Senior Network Testing and Software Engineer/ > /3CLogic , ph:07827611765/ > > > On Thu, May 12, 2022 at 1:20 PM Chester Lee > wrote: > > Hi, > > You can specify cluster id in the config. please refer to > https://opensips.org/docs/modules/3.2.x/clusterer.html#param_my_node_id > > > I hope this helps. > > Regards > Chester > > > 2022년 5월 12일 (목) 오후 4:06, Sasmita Panda >님이 작성: > > Hi All , > > I have 2 opesips cluster each cluster has 2 opensips node . I > want to define all the cluster node information in a single > opensips database . > > My clusterer table looks like below . > > +----+------------+---------+-----------------------+-------+-----------------+----------+----------+-------+----------------+ > | id | cluster_id | node_id | url                   | state | > no_ping_retries | priority | sip_addr | flags | description    | > +----+------------+---------+-----------------------+-------+-----------------+----------+----------+-------+----------------+ > |  1 |          1 |       1 | bin:1.1.1.1:5555 > |     1 |               3 |       50 | > NULL     | seed  | Node A         | > |  2 |          1 |       2 | bin:2.2.2.2:5555 >  |     1 |               3 |       50 | > NULL     | seed  | Node B         | > |  3 |          2 |       1 | bin:3.3.3.3:5555 > |     1 |               3 |       50 | > NULL     | NULL  | cluster2 Node1 | > |  4 |          2 |       2 | bin:4.4.4.4:5555 > |     1 |               3 |       50 | > NULL     | NULL  | cluster2 Node2 | > +----+------------+---------+-----------------------+-------+-----------------+----------+----------+-------+----------------+ > > In pensips 3.2 there is no cluster_id parameter to define in the > config . In the config I don't want to add the IP in the config > . For cluster 2 , when I am defining node 1 , its taking the > value of node 1 of cluster 1 . > > Is this possible anyhow or I have to save the data in a > different database ? > > > */Thanks & Regards/* > /Sasmita Panda/ > /Senior Network Testing and Software Engineer/ > /3CLogic , ph:07827611765/ > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > > -- > ______ > > > 이 기 원 CTO실 / 매니저 > (주)직방 | 010.6479.1321 | chester at zigbang.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 From ideanethelp at gmail.com Tue May 17 08:45:49 2022 From: ideanethelp at gmail.com (ideanet help) Date: Tue, 17 May 2022 20:45:49 +1200 Subject: [OpenSIPS-Users] Wildcard domain certificate for tls Message-ID: Hi Dear community members, I have a few client domains for TLS. Is there any way I can use a wildcard domain like *.mytestdomain.com and use that in the certificate? This way I do not have to create a certificate for each domain. My config works fine with single domain which i am using like this: modparam("tls_mgm", "certificate", "[sip.tls.xx.xx]/etc/letsencrypt/live/sip.tls.xx.xx/cert.pem") I am using wolfssl as tls_library. Opensips version is 3.3 Thank you. -------------- next part -------------- An HTML attachment was scrubbed... URL: From spanda at 3clogic.com Tue May 17 09:43:48 2022 From: spanda at 3clogic.com (Sasmita Panda) Date: Tue, 17 May 2022 15:13:48 +0530 Subject: [OpenSIPS-Users] Need some help in clusterer table and its use in opensips 3.2 . In-Reply-To: References: Message-ID: Hi, I am using location clustering . Only the location table data is getting synched in the cluster . My concern is , when I am saving the node information in clusterer table , is there a way I can define the cluster ID in config ? So , if I have 2 different clusterer then my node could identify itself through cluster_id and node_id combination . In opensips 2.2 , there is a parameter cluster_id to set in config . But in 3.2 this parameter is not present . *Thanks & Regards* *Sasmita Panda* *Senior Network Testing and Software Engineer* *3CLogic , ph:07827611765* On Tue, May 17, 2022 at 1:32 PM Răzvan Crainea wrote: > Hi, Sasmita! > > I don't fully understand your use case - you said it is using node 1 in > cluster 1 - it is using it for what? > A cluster is used for a specific replication feature (i.e. dialog > replication, ratelimit pipes replication). When you specify you want to > do a specific replication, that's where you specify the cluster (i.e. > dialog replication [1]). So what kind of replication feature are you > using, that is not properly identifying the nodes? > > [1] > > https://opensips.org/docs/modules/3.2.x/dialog.html#param_dialog_replication_cluster > > Răzvan Crainea > OpenSIPS Core Developer > http://www.opensips-solutions.com > > On 5/12/22 11:43, Sasmita Panda wrote: > > Hi , > > > > my_node_id parameter is to define that particular node id . But how will > > I define the cluster_id parameter? I want to differentiate both clusters > . > > I have added my_node_id parameter already . But by default its looking > > for cluster_id:1 . But in the database I have defined cluster_id :2 . > > > > modparam("clusterer", "my_node_id", 1) > > > > How will I associate the database and config ? > > > > */Thanks & Regards/* > > /Sasmita Panda/ > > /Senior Network Testing and Software Engineer/ > > /3CLogic , ph:07827611765/ > > > > > > On Thu, May 12, 2022 at 1:20 PM Chester Lee > > wrote: > > > > Hi, > > > > You can specify cluster id in the config. please refer to > > > https://opensips.org/docs/modules/3.2.x/clusterer.html#param_my_node_id > > < > https://opensips.org/docs/modules/3.2.x/clusterer.html#param_my_node_id> > > > > I hope this helps. > > > > Regards > > Chester > > > > > > 2022년 5월 12일 (목) 오후 4:06, Sasmita Panda > >님이 작성: > > > > Hi All , > > > > I have 2 opesips cluster each cluster has 2 opensips node . I > > want to define all the cluster node information in a single > > opensips database . > > > > My clusterer table looks like below . > > > > > +----+------------+---------+-----------------------+-------+-----------------+----------+----------+-------+----------------+ > > | id | cluster_id | node_id | url | state | > > no_ping_retries | priority | sip_addr | flags | description | > > > +----+------------+---------+-----------------------+-------+-----------------+----------+----------+-------+----------------+ > > | 1 | 1 | 1 | bin:1.1.1.1:5555 > > | 1 | 3 | 50 | > > NULL | seed | Node A | > > | 2 | 1 | 2 | bin:2.2.2.2:5555 > > | 1 | 3 | 50 | > > NULL | seed | Node B | > > | 3 | 2 | 1 | bin:3.3.3.3:5555 > > | 1 | 3 | 50 | > > NULL | NULL | cluster2 Node1 | > > | 4 | 2 | 2 | bin:4.4.4.4:5555 > > | 1 | 3 | 50 | > > NULL | NULL | cluster2 Node2 | > > > +----+------------+---------+-----------------------+-------+-----------------+----------+----------+-------+----------------+ > > > > In pensips 3.2 there is no cluster_id parameter to define in the > > config . In the config I don't want to add the IP in the config > > . For cluster 2 , when I am defining node 1 , its taking the > > value of node 1 of cluster 1 . > > > > Is this possible anyhow or I have to save the data in a > > different database ? > > > > > > */Thanks & Regards/* > > /Sasmita Panda/ > > /Senior Network Testing and Software Engineer/ > > /3CLogic , ph:07827611765/ > > _______________________________________________ > > Users mailing list > > Users at lists.opensips.org > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > > > > > > > -- > > ______ > > > > > > 이 기 원 CTO실 / 매니저 > > (주)직방 | 010.6479.1321 | chester at zigbang.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 > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -------------- next part -------------- An HTML attachment was scrubbed... URL: From razvan at opensips.org Tue May 17 10:00:46 2022 From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=) Date: Tue, 17 May 2022 13:00:46 +0300 Subject: [OpenSIPS-Users] Need some help in clusterer table and its use in opensips 3.2 . In-Reply-To: References: Message-ID: <2b51c5d3-dd60-c8ec-d77e-1cdfd416fe4c@opensips.org> No, there is no setting to identify a node based on cluster_id + node_id. Only the node_id is the identifier, so if you are using different servers, you should be using different node ids. Best regards, Răzvan Crainea OpenSIPS Core Developer http://www.opensips-solutions.com On 5/17/22 12:43, Sasmita Panda wrote: > Hi, > > I am using location clustering . > > Only the location table data is getting synched in the cluster . > > My concern is , when I am saving the node information in clusterer table > , is there a way I can define the cluster ID in config ? > So , if I have 2 different clusterer then my node could identify itself > through cluster_id and node_id combination . > > > In opensips 2.2 , there is a parameter cluster_id to set in config . But > in 3.2 this parameter is not present . > > > */Thanks & Regards/* > /Sasmita Panda/ > /Senior Network Testing and Software Engineer/ > /3CLogic , ph:07827611765/ > > > On Tue, May 17, 2022 at 1:32 PM Răzvan Crainea > wrote: > > Hi, Sasmita! > > I don't fully understand your use case - you said it is using node 1 in > cluster 1 - it is using it for what? > A cluster is used for a specific replication feature (i.e. dialog > replication, ratelimit pipes replication). When you specify you want to > do a specific replication, that's where you specify the cluster (i.e. > dialog replication [1]). So what kind of replication feature are you > using, that is not properly identifying the nodes? > > [1] > https://opensips.org/docs/modules/3.2.x/dialog.html#param_dialog_replication_cluster > > > Răzvan Crainea > OpenSIPS Core Developer > http://www.opensips-solutions.com > > On 5/12/22 11:43, Sasmita Panda wrote: > > Hi , > > > > my_node_id parameter is to define that particular node id . But > how will > > I define the cluster_id parameter? I want to differentiate both > clusters . > > I have added my_node_id parameter already . But by default its > looking > > for cluster_id:1 . But in the database I have defined cluster_id :2 . > > > > modparam("clusterer", "my_node_id", 1) > > > > How will I associate the database and config ? > > > > */Thanks & Regards/* > > /Sasmita Panda/ > > /Senior Network Testing and Software Engineer/ > > /3CLogic , ph:07827611765/ > > > > > > On Thu, May 12, 2022 at 1:20 PM Chester Lee > > >> wrote: > > > >     Hi, > > > >     You can specify cluster id in the config. please refer to > > > https://opensips.org/docs/modules/3.2.x/clusterer.html#param_my_node_id > > > >  > > > > >     I hope this helps. > > > >     Regards > >     Chester > > > > > >     2022년 5월 12일 (목) 오후 4:06, Sasmita Panda > > >     >>님이 > 작성: > > > >         Hi All , > > > >         I have 2 opesips cluster each cluster has 2 opensips node . I > >         want to define all the cluster node information in a single > >         opensips database . > > > >         My clusterer table looks like below . > > > > >  +----+------------+---------+-----------------------+-------+-----------------+----------+----------+-------+----------------+ > >         | id | cluster_id | node_id | url                   | state | > >         no_ping_retries | priority | sip_addr | flags | > description    | > > >  +----+------------+---------+-----------------------+-------+-----------------+----------+----------+-------+----------------+ > >         |  1 |          1 |       1 | bin:1.1.1.1:5555 > > >         > |     1 | >           3 |       50 | > >         NULL     | seed  | Node A         | > >         |  2 |          1 |       2 | bin:2.2.2.2:5555 > > >         >  |     1 | >             3 |       50 | > >         NULL     | seed  | Node B         | > >         |  3 |          2 |       1 | bin:3.3.3.3:5555 > > >         > |     1 | >           3 |       50 | > >         NULL     | NULL  | cluster2 Node1 | > >         |  4 |          2 |       2 | bin:4.4.4.4:5555 > > >         > |     1 | >           3 |       50 | > >         NULL     | NULL  | cluster2 Node2 | > > >  +----+------------+---------+-----------------------+-------+-----------------+----------+----------+-------+----------------+ > > > >         In pensips 3.2 there is no cluster_id parameter to define > in the > >         config . In the config I don't want to add the IP in the > config > >         . For cluster 2 , when I am defining node 1 , its taking the > >         value of node 1 of cluster 1 . > > > >         Is this possible anyhow or I have to save the data in a > >         different database ? > > > > > >         */Thanks & Regards/* > >         /Sasmita Panda/ > >         /Senior Network Testing and Software Engineer/ > >         /3CLogic , ph:07827611765/ > >         _______________________________________________ > >         Users mailing list > > Users at lists.opensips.org > > > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > >          > > > > > > > > >     -- > >     ______ > > > > > >     이 기 원 CTO실 / 매니저 > >     (주)직방 | 010.6479.1321 | chester at zigbang.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 > > > _______________________________________________ > 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 voransoy at gmail.com Tue May 17 11:10:14 2022 From: voransoy at gmail.com (Volkan Oransoy) Date: Tue, 17 May 2022 12:10:14 +0100 Subject: [OpenSIPS-Users] Drouting relay issue In-Reply-To: <486d32ea-275b-479c-a1ee-318143a580a6@Spark> References: <32eada8e-1df1-4116-8ea5-07f994d846e3@Spark> <486d32ea-275b-479c-a1ee-318143a580a6@Spark> Message-ID: <91a1ca9f-83ec-4c64-9543-77ce718c4f8c@Spark> Hi all, For further reference, the issue is related to UDP fragmentation. Digitalocean droplet network doesn’t route fragmented packets. I can see only the first part of the fragmented packet, not the subsequent one. So the destination fails with “ICMP ip reassembly time exceeded, length 556” at os network. Cheers Volkan Oransoy On 11 May 2022 11:52 +0100, Volkan Oransoy , wrote: > I think the screenshot has been discarded by the mailman. The sip traffic is as follows. The proxy tries to retransmit and fails after three more attempts. > > ──────────┬───────── ──────────┬───────── ──────────┬───────── >  10:47:32.603828 │ INVITE (SDP) │ │ >  +0.000347 │ ──────────────────────────> │ │ >  10:47:32.604175 │ 100 Giving it a try │ │ >  +0.001474 │ <────────────────────────── │ │ >  10:47:32.605649 │ │ INVITE (SDP) │ >  +0.490742 │ │ ──────────────────────────> │ >  10:47:33.096391 │ │ INVITE (SDP) │ >  +1.001859 │ │ ────────────────────────>>> │ >  10:47:34.098250 │ │ INVITE (SDP) │ >  +1.953642 │ │ ────────────────────────>>> │ >  10:47:36.051892 │ │ INVITE (SDP) │ >  +1.603582 │ │ ────────────────────────>>> │ >  10:47:37.655474 │ 408 Request Timeout │ │ >  +0.001615 │ <────────────────────────── │ │ >  10:47:37.657089 │ ACK │ │ >  │ ──────────────────────────> │ │ > > Volkan Oransoy > On 11 May 2022 11:14 +0100, Volkan Oransoy , wrote: > > Hi all, > > > > I have an interesting issue with one of my test setups. I have a simple routing script which gets the gateway id directly from the header originating from a Freeswith box. The system finds and sets the request URL as anticipated. But even if I can see the request on the proxy, I can't see the traffic on the destination. Interestingly, the same proxy can register to the same destination with uac_registrant as a UAC. And I can receive calls from the same destination. Is there anything missing to route this traffic correctly? > > > > Thanks in advance. > > > > route[to_gateway] { > >         if ( route_to_gw($hdr(X-GWID)) ) { > >                 route(relay); > >         } > > } > > route[relay] { > >         if (is_method("INVITE")) { > >                 t_on_branch("per_branch_ops"); > >                 t_on_reply("handle_nat"); > >                 t_on_failure("failure"); > >         } > >         if (!t_relay()) { > >                 send_reply(500,"Internal Error"); > >         } > >         exit; > > } > > > > The database structure is as follows; > > > > opensips=# select * from dr_gateways; > >  id | gwid | type |           address            | strip | pri_prefix | attrs | probe_mode | state | socket | description > > ----+------+------+------------------------------+-------+------------+-------+------------+-------+--------+------------- > >   6 | 5    |    2 | sip:testgw.bulutfon.net:5060 |     0 |            |       |          0 |     0 |        | 5 > > > > Here is the INVITE request sent to the destination, which fails as in the screenshot. > > > > INVITE sip:905551234567 at testgw.bulutfon.net:5060 SIP/2.0 > > Record-Route: > > Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK372.d935ecb2.0 > > Via: SIP/2.0/UDP 1.1.1.1:6080;received=1.1.1.1;rport=6080;branch=z9hG4bK99mtDaevmN7Nj > > Max-Forwards: 67 > > From: "+908508850000" ;tag=1jcyr93emrjDQ > > To: > > Call-ID: aefb8c60-426b-123b-8ca8-82a722ba4f75 > > CSeq: 51067506 INVITE > > Contact: > > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY > > Supported: timer, path, replaces > > Allow-Events: talk, hold, conference, refer > > Content-Type: application/sdp > > Content-Disposition: session > > Content-Length: 621 > > Remote-Party-ID: "+908508850000" ;party=calling;screen=yes;privacy=off > > > > v=0 > > o=FreeSWITCH 1651210438 1651210439 IN IP4 1.1.1.1 > > s=FreeSWITCH > > c=IN IP4 1.1.1.1 > > t=0 0 ... > > > > > > -- > > > > Volkan Oransoy -------------- next part -------------- An HTML attachment was scrubbed... URL: From jehanzaib.kiani at gmail.com Tue May 17 13:32:07 2022 From: jehanzaib.kiani at gmail.com (Jehanzaib Younis) Date: Wed, 18 May 2022 01:32:07 +1200 Subject: [OpenSIPS-Users] no TLS client domain found error Message-ID: Hi, I am having trouble to send/receive OPTIONS to ms teams. Using the dispatcher module. The socket is defined as tls:*mysbcip*:5061 Looks like when my opensips (3.2.x) tries to send OPTIONS. it is giving me the following error ERROR:proto_tls:proto_tls_conn_init: no TLS client domain found ERROR:core:tcp_conn_create: failed to do proto 3 specific init for conn 0x7f00ef2a85a0 ERROR:core:tcp_async_connect: tcp_conn_create failed ERROR:proto_tls:proto_tls_send: async TCP connect failed ERROR:tm:msg_send: send() to 52.114.76.76:5061 for proto tls/3 failed ERROR:tm:t_uac: attempt to send to 'sip:sip3.pstnhub.microsoft.com:5061;transport:tls' failed I am setting the Contact as Looks like the client domain is used for outgoing TLS connection but no idea which domain i need to add here. The socket is my opensips ip address. Has anyone seen a similar kind of behaviour? Thank you. Regards, Jehanzaib -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Tue May 17 15:01:17 2022 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Tue, 17 May 2022 18:01:17 +0300 Subject: [OpenSIPS-Users] OpenSIPS CP upgrade to 9.3.2 In-Reply-To: References: Message-ID: <0740cc83-a1b4-f384-ffb8-fe53b892ad83@opensips.org> Hi Bela, Does you CDRviewer look like this ? See the gear box in the right upper corner. And be sure that the 9.3.2 version is indeed displayed in the left upper corner. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com OpenSIPS eBootcamp 23rd May - 3rd June 2022 https://opensips.org/training/OpenSIPS_eBootcamp_2022/ On 5/17/22 10:55 AM, Bela H wrote: > > Hello, > > What is the best method to upgrade the control panel from 8.3.2 to 9.3.2? > > I had some extra fields e.g. in CDR viewer and disappeared after > 9.3.2. It is in the file > /var/www/html/opensips-cp/config/tools/system/cdrviewer/local.inc.php > but not visible in the CDR viewer panel only in the detailed view for > each call. > > Also I don’t see this gear icon Bogdan mentioned in the blog: Each > tool has its own Settings panel “accessible via the gear-icon in the > right side of the tool header”. > > Cheers, > > Bela > > > _______________________________________________ > 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: jjfhkbbgkdgdgnie.png Type: image/png Size: 18455 bytes Desc: not available URL: From bogdan at opensips.org Tue May 17 15:02:51 2022 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Tue, 17 May 2022 18:02:51 +0300 Subject: [OpenSIPS-Users] Drouting relay issue In-Reply-To: <91a1ca9f-83ec-4c64-9543-77ce718c4f8c@Spark> References: <32eada8e-1df1-4116-8ea5-07f994d846e3@Spark> <486d32ea-275b-479c-a1ee-318143a580a6@Spark> <91a1ca9f-83ec-4c64-9543-77ce718c4f8c@Spark> Message-ID: Thanks on the follow up here. I guess you need to switch to TCP, right ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com OpenSIPS eBootcamp 23rd May - 3rd June 2022 https://opensips.org/training/OpenSIPS_eBootcamp_2022/ On 5/17/22 2:10 PM, Volkan Oransoy wrote: > Hi all, > > For further reference, the issue is related to UDP fragmentation. > Digitalocean droplet network doesn’t route fragmented packets. I can > see only the first part of the fragmented packet, not the subsequent > one. So the destination fails with “ICMP ip reassembly time exceeded, > length 556” at os network. > > Cheers > > Volkan Oransoy > On 11 May 2022 11:52 +0100, Volkan Oransoy , wrote: >> I think the screenshot has been discarded by the mailman. The sip >> traffic is as follows. The proxy tries to retransmit and fails after >> three more attempts. >> >> ──────────┬───────── ──────────┬───────── ──────────┬───────── >>  10:47:32.603828 │ INVITE (SDP) │ │ >>  +0.000347 │ ──────────────────────────> │ │ >>  10:47:32.604175 │ 100 Giving it a try │ │ >>  +0.001474 │ <────────────────────────── │ │ >>  10:47:32.605649 │ │ INVITE (SDP) │ >>  +0.490742 │ │ ──────────────────────────> │ >>  10:47:33.096391 │ │ INVITE (SDP) │ >>  +1.001859 │ │ ────────────────────────>>> │ >>  10:47:34.098250 │ │ INVITE (SDP) │ >>  +1.953642 │ │ ────────────────────────>>> │ >>  10:47:36.051892 │ │ INVITE (SDP) │ >>  +1.603582 │ │ ────────────────────────>>> │ >>  10:47:37.655474 │ 408 Request Timeout │ │ >>  +0.001615 │ <────────────────────────── │ │ >>  10:47:37.657089 │ ACK │ │ >>  │ ──────────────────────────> │ │ >> >> Volkan Oransoy >> On 11 May 2022 11:14 +0100, Volkan Oransoy , wrote: >>> Hi all, >>> >>> I have an interesting issue with one of my test setups. I have a >>> simple routing script which gets the gateway id directly from the >>> header originating from a Freeswith box. The system finds and sets >>> the request URL as anticipated. But even if I can see the request on >>> the proxy, I can't see the traffic on the destination. >>> Interestingly, the same proxy can register to the same destination >>> with uac_registrant as a UAC. And I can receive calls from the same >>> destination. Is there anything missing to route this traffic correctly? >>> >>> Thanks in advance. >>> >>> route[to_gateway] { >>>         if ( route_to_gw($hdr(X-GWID)) ) { >>>  route(relay); >>>         } >>> } >>> route[relay] { >>>         if (is_method("INVITE")) { >>>  t_on_branch("per_branch_ops"); >>>  t_on_reply("handle_nat"); >>>  t_on_failure("failure"); >>>         } >>>         if (!t_relay()) { >>>  send_reply(500,"Internal Error"); >>>         } >>>         exit; >>> } >>> >>> The database structure is as follows; >>> >>> opensips=# select * from dr_gateways; >>>  id | gwid | type |           address        | strip | pri_prefix | >>> attrs | probe_mode | state | socket | description >>> ----+------+------+------------------------------+-------+------------+-------+------------+-------+--------+------------- >>>   6 | 5    |    2 | sip:testgw.bulutfon.net:5060 >>>  |    0 |            |       |     >>>      0 |     0 |    | 5 >>> >>> Here is the INVITE request sent to the destination, which fails as >>> in the screenshot. >>> >>> INVITE sip:905551234567 at testgw.bulutfon.net:5060 >>>  SIP/2.0 >>> Record-Route: >>> Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK372.d935ecb2.0 >>> Via: SIP/2.0/UDP >>> 1.1.1.1:6080;received=1.1.1.1;rport=6080;branch=z9hG4bK99mtDaevmN7Nj >>> Max-Forwards: 67 >>> From: "+908508850000" >> >;tag=1jcyr93emrjDQ >>> To: >> > >>> Call-ID: aefb8c60-426b-123b-8ca8-82a722ba4f75 >>> CSeq: 51067506 INVITE >>> Contact: >> > >>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, >>> REGISTER, REFER, NOTIFY >>> Supported: timer, path, replaces >>> Allow-Events: talk, hold, conference, refer >>> Content-Type: application/sdp >>> Content-Disposition: session >>> Content-Length: 621 >>> Remote-Party-ID: "+908508850000" >>> >> >;party=calling;screen=yes;privacy=off >>> >>> v=0 >>> o=FreeSWITCH 1651210438 1651210439 IN IP4 1.1.1.1 >>> s=FreeSWITCH >>> c=IN IP4 1.1.1.1 >>> t=0 0 ... >>> >>> >>> -- >>> >>> Volkan Oransoy > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Tue May 17 15:04:01 2022 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Tue, 17 May 2022 18:04:01 +0300 Subject: [OpenSIPS-Users] no TLS client domain found error In-Reply-To: References: Message-ID: <8877c8b2-9ad3-a0d0-3294-5a3d20c748a1@opensips.org> Hi Jehanzaib, What are the TLS client domains you have defined in your tls_mgm module ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com OpenSIPS eBootcamp 23rd May - 3rd June 2022 https://opensips.org/training/OpenSIPS_eBootcamp_2022/ On 5/17/22 4:32 PM, Jehanzaib Younis wrote: > Hi, > > I am having trouble to send/receive OPTIONS to ms teams. > Using the dispatcher module. The socket is defined as tls:*mysbcip*:5061 > Looks like when my opensips (3.2.x) tries to send OPTIONS. it is > giving me the following error > * > * > ERROR:proto_tls:proto_tls_conn_init: no TLS client domain found > ERROR:core:tcp_conn_create: failed to do proto 3 specific init for > conn 0x7f00ef2a85a0 > ERROR:core:tcp_async_connect: tcp_conn_create failed > ERROR:proto_tls:proto_tls_send: async TCP connect failed > ERROR:tm:msg_send: send() to 52.114.76.76:5061 > for proto tls/3 failed > ERROR:tm:t_uac: attempt to send to > 'sip:sip3.pstnhub.microsoft.com:5061;transport:tls' failed > > I am setting the Contact as  > > Looks like the client domain is used for outgoing TLS connection but > no idea which domain i need to add here. The socket is my opensips ip > address. > > Has anyone seen a similar kind of behaviour? > > Thank you. > > Regards, > Jehanzaib > > _______________________________________________ > 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 voransoy at gmail.com Tue May 17 16:46:10 2022 From: voransoy at gmail.com (Volkan Oransoy) Date: Tue, 17 May 2022 17:46:10 +0100 Subject: [OpenSIPS-Users] Drouting relay issue In-Reply-To: References: <32eada8e-1df1-4116-8ea5-07f994d846e3@Spark> <486d32ea-275b-479c-a1ee-318143a580a6@Spark> <91a1ca9f-83ec-4c64-9543-77ce718c4f8c@Spark> Message-ID: <3a9a226b-8ac5-4d9a-93d1-4a39fac4d0dd@Spark> Hi Bogdan Yes, TCP stack works as anticipated. Thanks Volkan Oransoy On 17 May 2022 16:02 +0100, Bogdan-Andrei Iancu , wrote: > Thanks on the follow up here. I guess you need to switch to TCP, right ? > > Regards, > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > OpenSIPS eBootcamp 23rd May - 3rd June 2022 > https://opensips.org/training/OpenSIPS_eBootcamp_2022/ > On 5/17/22 2:10 PM, Volkan Oransoy wrote: > > Hi all, > > > > For further reference, the issue is related to UDP fragmentation. Digitalocean droplet network doesn’t route fragmented packets. I can see only the first part of the fragmented packet, not the subsequent one. So the destination fails with “ICMP ip reassembly time exceeded, length 556” at os network. > > > > Cheers > > > > Volkan Oransoy > > On 11 May 2022 11:52 +0100, Volkan Oransoy , wrote: > > > I think the screenshot has been discarded by the mailman. The sip traffic is as follows. The proxy tries to retransmit and fails after three more attempts. > > > > > > ──────────┬───────── ──────────┬───────── ──────────┬───────── > > >  10:47:32.603828 │ INVITE (SDP) │ │ > > >  +0.000347 │ ──────────────────────────> │ │ > > >  10:47:32.604175 │ 100 Giving it a try │ │ > > >  +0.001474 │ <────────────────────────── │ │ > > >  10:47:32.605649 │ │ INVITE (SDP) │ > > >  +0.490742 │ │ ──────────────────────────> │ > > >  10:47:33.096391 │ │ INVITE (SDP) │ > > >  +1.001859 │ │ ────────────────────────>>> │ > > >  10:47:34.098250 │ │ INVITE (SDP) │ > > >  +1.953642 │ │ ────────────────────────>>> │ > > >  10:47:36.051892 │ │ INVITE (SDP) │ > > >  +1.603582 │ │ ────────────────────────>>> │ > > >  10:47:37.655474 │ 408 Request Timeout │ │ > > >  +0.001615 │ <────────────────────────── │ │ > > >  10:47:37.657089 │ ACK │ │ > > >  │ ──────────────────────────> │ │ > > > > > > Volkan Oransoy > > > On 11 May 2022 11:14 +0100, Volkan Oransoy , wrote: > > > > Hi all, > > > > > > > > I have an interesting issue with one of my test setups. I have a simple routing script which gets the gateway id directly from the header originating from a Freeswith box. The system finds and sets the request URL as anticipated. But even if I can see the request on the proxy, I can't see the traffic on the destination. Interestingly, the same proxy can register to the same destination with uac_registrant as a UAC. And I can receive calls from the same destination. Is there anything missing to route this traffic correctly? > > > > > > > > Thanks in advance. > > > > > > > > route[to_gateway] { > > > >         if ( route_to_gw($hdr(X-GWID)) ) { > > > >                 route(relay); > > > >         } > > > > } > > > > route[relay] { > > > >         if (is_method("INVITE")) { > > > >                 t_on_branch("per_branch_ops"); > > > >                 t_on_reply("handle_nat"); > > > >                 t_on_failure("failure"); > > > >         } > > > >         if (!t_relay()) { > > > >                 send_reply(500,"Internal Error"); > > > >         } > > > >         exit; > > > > } > > > > > > > > The database structure is as follows; > > > > > > > > opensips=# select * from dr_gateways; > > > >  id | gwid | type |           address            | strip | pri_prefix | attrs | probe_mode | state | socket | description > > > > ----+------+------+------------------------------+-------+------------+-------+------------+-------+--------+------------- > > > >   6 | 5    |    2 | sip:testgw.bulutfon.net:5060 |     0 |            |       |          0 |     0 |        | 5 > > > > > > > > Here is the INVITE request sent to the destination, which fails as in the screenshot. > > > > > > > > INVITE sip:905551234567 at testgw.bulutfon.net:5060 SIP/2.0 > > > > Record-Route: > > > > Via: SIP/2.0/UDP 2.2.2.2:5060;branch=z9hG4bK372.d935ecb2.0 > > > > Via: SIP/2.0/UDP 1.1.1.1:6080;received=1.1.1.1;rport=6080;branch=z9hG4bK99mtDaevmN7Nj > > > > Max-Forwards: 67 > > > > From: "+908508850000" ;tag=1jcyr93emrjDQ > > > > To: > > > > Call-ID: aefb8c60-426b-123b-8ca8-82a722ba4f75 > > > > CSeq: 51067506 INVITE > > > > Contact: > > > > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY > > > > Supported: timer, path, replaces > > > > Allow-Events: talk, hold, conference, refer > > > > Content-Type: application/sdp > > > > Content-Disposition: session > > > > Content-Length: 621 > > > > Remote-Party-ID: "+908508850000" ;party=calling;screen=yes;privacy=off > > > > > > > > v=0 > > > > o=FreeSWITCH 1651210438 1651210439 IN IP4 1.1.1.1 > > > > s=FreeSWITCH > > > > c=IN IP4 1.1.1.1 > > > > t=0 0 ... > > > > > > > > > > > > -- > > > > > > > > Volkan Oransoy > > > > _______________________________________________ > > 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.rehan at gmail.com Tue May 17 19:19:37 2022 From: ahmed.rehan at gmail.com (Ahmed Rehan) Date: Wed, 18 May 2022 00:19:37 +0500 Subject: [OpenSIPS-Users] no TLS client domain found error In-Reply-To: References: Message-ID: Hello Jehanzeb You need to add your own domain name as client domain and server domain as well . There will be two set of entries for tls_mgm , one set of entries will be client_domain and one set of entries will be for Server_domain Its flow is like , when an options is sent to MS servers, opensips is acting as a client , and when a reply is sent from MS servers Opensips will act as a server for your domain name hope it clears On Tue, May 17, 2022 at 6:32 PM Jehanzaib Younis wrote: > Hi, > > I am having trouble to send/receive OPTIONS to ms teams. > Using the dispatcher module. The socket is defined as tls:*mysbcip*:5061 > Looks like when my opensips (3.2.x) tries to send OPTIONS. it is giving me > the following error > > ERROR:proto_tls:proto_tls_conn_init: no TLS client domain found > ERROR:core:tcp_conn_create: failed to do proto 3 specific init for conn > 0x7f00ef2a85a0 > ERROR:core:tcp_async_connect: tcp_conn_create failed > ERROR:proto_tls:proto_tls_send: async TCP connect failed > ERROR:tm:msg_send: send() to 52.114.76.76:5061 for proto tls/3 failed > ERROR:tm:t_uac: attempt to send to 'sip:sip3.pstnhub.microsoft.com:5061;transport:tls' > failed > > I am setting the Contact as > > Looks like the client domain is used for outgoing TLS connection but no > idea which domain i need to add here. The socket is my opensips ip address. > > Has anyone seen a similar kind of behaviour? > > Thank you. > > Regards, > Jehanzaib > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -- Regards Ahmed Rehan -------------- next part -------------- An HTML attachment was scrubbed... URL: From jehanzaib.kiani at gmail.com Tue May 17 23:38:51 2022 From: jehanzaib.kiani at gmail.com (Jehanzaib Younis) Date: Wed, 18 May 2022 11:38:51 +1200 Subject: [OpenSIPS-Users] no TLS client domain found error In-Reply-To: <8877c8b2-9ad3-a0d0-3294-5a3d20c748a1@opensips.org> References: <8877c8b2-9ad3-a0d0-3294-5a3d20c748a1@opensips.org> Message-ID: Hi Bogdan, That's the problem, when I try to add the client_domain I get an error. Actually, I have a working config for webrtc but now I am adding a new domain for MS teams direct route. In fact, any other domain gives an error. If I disable MS Teams domain, the opensips do not give an error message and my webrtc client can connect without any issue. loadmodule "tls_mgm.so" modparam("tls_mgm", "tls_library", "wolfssl") #### (WebRTC) Client modparam("tls_mgm", "server_domain", "sip.mywebphone.xx") modparam("tls_mgm", "certificate", "[sip.mywebphone.xx]/etc/letsencrypt/live/sip.mywebphone.xx/cert.pem") modparam("tls_mgm", "private_key", "[sip.mywebphone.xx]/etc/letsencrypt/live/sip.mywebphone.xx/privkey.pem") modparam("tls_mgm", "ca_list", "[sip.mywebphone.xx]/etc/letsencrypt/live/sip.mywebphone.xx/fullchain.pem") modparam("tls_mgm", "ca_dir", "[sip.mywebphone.xx]/etc/letsencrypt/live/sip.mywebphone.xx") modparam("tls_mgm", "tls_method", "[sip.mywebphone.xx]SSLv23") modparam("tls_mgm", "verify_cert", "[sip.mywebphone.xx]1") modparam("tls_mgm", "require_cert", "[sip.mywebphone.xx]1") ### This is for MS-Teams direct route modparam("tls_mgm", "server_domain", "dom1.formsteams.com") modparam("tls_mgm", "client_domain", "dom1.formsteams.com") modparam("tls_mgm", "certificate", "[dom1.formsteams.com ]/etc/letsencrypt/live/dom1.formsteams.com/cert.pem") modparam("tls_mgm", "private_key", "[dom1.formsteams.com ]/etc/letsencrypt/live/dom1.formsteams.com/privkey.pem") modparam("tls_mgm", "ca_list", "[dom1.formsteams.com]/etc/letsencrypt/live/ dom1.formsteams.com/fullchain.pem") modparam("tls_mgm", "ca_dir", "[dom1.formsteams.com]/etc/letsencrypt/live/ dom1.formsteams.com") modparam("tls_mgm", "tls_method", "[dom1.formsteams.com]SSLv23") modparam("tls_mgm", "verify_cert", "[dom1.formsteams.com]1") modparam("tls_mgm", "require_cert", "[dom1.formsteams.com]1") modparam("tls_mgm", "client_sip_domain_avp", "tls_sip_dom") When i enable the MS-Teams direct route domain i get the below error: no certificate for tls domain ' dom1.formsteams.com ' defined Regards, Jehanzaib On Wed, May 18, 2022 at 3:04 AM Bogdan-Andrei Iancu wrote: > Hi Jehanzaib, > > What are the TLS client domains you have defined in your tls_mgm module ? > > Regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > OpenSIPS eBootcamp 23rd May - 3rd June 2022 > https://opensips.org/training/OpenSIPS_eBootcamp_2022/ > > On 5/17/22 4:32 PM, Jehanzaib Younis wrote: > > Hi, > > I am having trouble to send/receive OPTIONS to ms teams. > Using the dispatcher module. The socket is defined as tls:*mysbcip*:5061 > Looks like when my opensips (3.2.x) tries to send OPTIONS. it is giving me > the following error > > ERROR:proto_tls:proto_tls_conn_init: no TLS client domain found > ERROR:core:tcp_conn_create: failed to do proto 3 specific init for conn > 0x7f00ef2a85a0 > ERROR:core:tcp_async_connect: tcp_conn_create failed > ERROR:proto_tls:proto_tls_send: async TCP connect failed > ERROR:tm:msg_send: send() to 52.114.76.76:5061 for proto tls/3 failed > ERROR:tm:t_uac: attempt to send to ' > sip:sip3.pstnhub.microsoft.com:5061;transport:tls' failed > > I am setting the Contact as > > Looks like the client domain is used for outgoing TLS connection but no > idea which domain i need to add here. The socket is my opensips ip address. > > Has anyone seen a similar kind of behaviour? > > Thank you. > > Regards, > Jehanzaib > > _______________________________________________ > Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Wed May 18 06:15:10 2022 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Wed, 18 May 2022 09:15:10 +0300 Subject: [OpenSIPS-Users] no TLS client domain found error In-Reply-To: References: <8877c8b2-9ad3-a0d0-3294-5a3d20c748a1@opensips.org> Message-ID: <72b9b0b1-5520-0818-a5a3-af8ad451fed7@opensips.org> Hi Jehanzaib, The sequence for the MST TLS domains is wrong. For each TLS domain block, you need to start only with a server_domain or client_domain - of course, different names. And for each domain you need you set the matching conditions. See https://opensips.org/html/docs/modules/3.2.x/tls_mgm.html#domains-param Basically something like: modparam("tls_mgm", "server_domain", "formsteams_server") modparam("tls_mgm", "match_ip_address", "[formsteams_server]....") modparam("tls_mgm", "match_sip_domain", "[formsteams_server]....") modparam("tls_mgm", "certificate", "[formsteams_server].....) .... modparam("tls_mgm", "client_domain", "formsteams_client") modparam("tls_mgm", "match_ip_address", "[formsteams_client]....") modparam("tls_mgm", "match_sip_domain", "[formsteams_client]....") modparam("tls_mgm", "certificate", "[formsteams_client].....) .... Best regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com OpenSIPS eBootcamp 23rd May - 3rd June 2022 https://opensips.org/training/OpenSIPS_eBootcamp_2022/ On 5/18/22 2:38 AM, Jehanzaib Younis wrote: > Hi Bogdan, > That's the problem, when I try to add the client_domain I get an > error. Actually, I have a working config for webrtc but now I am > adding a new domain for MS teams direct route. In fact, any other > domain gives an error. If I disable MS Teams domain, the opensips do > not give an error message and my webrtc client can connect without any > issue. > > loadmodule "tls_mgm.so" > modparam("tls_mgm", "tls_library", "wolfssl") > > #### (WebRTC) Client > modparam("tls_mgm", "server_domain", "sip.mywebphone.xx") > modparam("tls_mgm", "certificate", > "[sip.mywebphone.xx]/etc/letsencrypt/live/sip.mywebphone.xx/cert.pem") > modparam("tls_mgm", "private_key", > "[sip.mywebphone.xx]/etc/letsencrypt/live/sip.mywebphone.xx/privkey.pem") > modparam("tls_mgm", "ca_list", > "[sip.mywebphone.xx]/etc/letsencrypt/live/sip.mywebphone.xx/fullchain.pem") > modparam("tls_mgm", "ca_dir", > "[sip.mywebphone.xx]/etc/letsencrypt/live/sip.mywebphone.xx") > modparam("tls_mgm", "tls_method", "[sip.mywebphone.xx]SSLv23") > modparam("tls_mgm", "verify_cert", "[sip.mywebphone.xx]1") > modparam("tls_mgm", "require_cert", "[sip.mywebphone.xx]1") > > ### This is for MS-Teams direct route > modparam("tls_mgm", "server_domain", "dom1.formsteams.com > ") > modparam("tls_mgm", "client_domain", "dom1.formsteams.com > ") > modparam("tls_mgm", "certificate", "[dom1.formsteams.com > ]/etc/letsencrypt/live/dom1.formsteams.com/cert.pem > ") > modparam("tls_mgm", "private_key", "[dom1.formsteams.com > ]/etc/letsencrypt/live/dom1.formsteams.com/privkey.pem > ") > modparam("tls_mgm", "ca_list", "[dom1.formsteams.com > ]/etc/letsencrypt/live/dom1.formsteams.com/fullchain.pem > ") > modparam("tls_mgm", "ca_dir", "[dom1.formsteams.com > ]/etc/letsencrypt/live/dom1.formsteams.com > ") > modparam("tls_mgm", "tls_method", "[dom1.formsteams.com > ]SSLv23") > modparam("tls_mgm", "verify_cert", "[dom1.formsteams.com > ]1") > modparam("tls_mgm", "require_cert", "[dom1.formsteams.com > ]1") > modparam("tls_mgm", "client_sip_domain_avp", "tls_sip_dom") > > When i enable the MS-Teams direct route domain i get the below error: > no certificate for tls domain ' dom1.formsteams.com >  ' defined > > > Regards, > Jehanzaib > > > On Wed, May 18, 2022 at 3:04 AM Bogdan-Andrei Iancu > > wrote: > > Hi Jehanzaib, > > What are the TLS client domains you have defined in your tls_mgm > module ? > > Regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > OpenSIPS eBootcamp 23rd May - 3rd June 2022 > https://opensips.org/training/OpenSIPS_eBootcamp_2022/ > > On 5/17/22 4:32 PM, Jehanzaib Younis wrote: >> Hi, >> >> I am having trouble to send/receive OPTIONS to ms teams. >> Using the dispatcher module. The socket is defined >> as tls:*mysbcip*:5061 >> Looks like when my opensips (3.2.x) tries to send OPTIONS. it is >> giving me the following error >> * >> * >> ERROR:proto_tls:proto_tls_conn_init: no TLS client domain found >> ERROR:core:tcp_conn_create: failed to do proto 3 specific init >> for conn 0x7f00ef2a85a0 >> ERROR:core:tcp_async_connect: tcp_conn_create failed >> ERROR:proto_tls:proto_tls_send: async TCP connect failed >> ERROR:tm:msg_send: send() to 52.114.76.76:5061 >> for proto tls/3 failed >> ERROR:tm:t_uac: attempt to send to >> 'sip:sip3.pstnhub.microsoft.com:5061;transport:tls' failed >> >> I am setting the Contact as >> >> Looks like the client domain is used for outgoing TLS connection >> but no idea which domain i need to add here. The socket is my >> opensips ip address. >> >> Has anyone seen a similar kind of behaviour? >> >> Thank you. >> >> Regards, >> Jehanzaib >> >> _______________________________________________ >> 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 hobe69 at hotmail.com Tue May 17 21:25:27 2022 From: hobe69 at hotmail.com (Bela H) Date: Tue, 17 May 2022 21:25:27 +0000 Subject: [OpenSIPS-Users] OpenSIPS CP upgrade to 9.3.2 In-Reply-To: <0740cc83-a1b4-f384-ffb8-fe53b892ad83@opensips.org> References: <0740cc83-a1b4-f384-ffb8-fe53b892ad83@opensips.org> Message-ID: Hi Bogdan, I have re-installed the CP 9.3.2 but the results are same. Still missing CDR fields in the CDR Viewer and no gear icon anywhere only Users/Alias Management and System/Monit. [cid:image002.png at 01D86A99.33EE85C0] However, in the CDR details I can see the additional fields: [cid:image005.png at 01D86A99.33EE85C0] What did I wrong? Cheers, Bela From: Bogdan-Andrei Iancu Sent: Wednesday, 18 May 2022 03:01 To: OpenSIPS users mailling list; Bela H Subject: Re: [OpenSIPS-Users] OpenSIPS CP upgrade to 9.3.2 Hi Bela, Does you CDRviewer look like this ? [cid:part1.1E9CE55B.28BF7076 at opensips.org] See the gear box in the right upper corner. And be sure that the 9.3.2 version is indeed displayed in the left upper corner. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com OpenSIPS eBootcamp 23rd May - 3rd June 2022 https://opensips.org/training/OpenSIPS_eBootcamp_2022/ On 5/17/22 10:55 AM, Bela H wrote: Hello, What is the best method to upgrade the control panel from 8.3.2 to 9.3.2? I had some extra fields e.g. in CDR viewer and disappeared after 9.3.2. It is in the file /var/www/html/opensips-cp/config/tools/system/cdrviewer/local.inc.php but not visible in the CDR viewer panel only in the detailed view for each call. Also I don’t see this gear icon Bogdan mentioned in the blog: Each tool has its own Settings panel “accessible via the gear-icon in the right side of the tool header”. Cheers, Bela _______________________________________________ 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: jjfhkbbgkdgdgnie.png Type: image/png Size: 18455 bytes Desc: jjfhkbbgkdgdgnie.png URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: 9BE1CA3C97C945809C3B1B43988199E7.png Type: image/png Size: 45526 bytes Desc: 9BE1CA3C97C945809C3B1B43988199E7.png URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: 302052B480C8471E8A3FE4693709167B.png Type: image/png Size: 21364 bytes Desc: 302052B480C8471E8A3FE4693709167B.png URL: From bogdan at opensips.org Wed May 18 06:06:30 2022 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Wed, 18 May 2022 09:06:30 +0300 Subject: [OpenSIPS-Users] OpenSIPS CP upgrade to 9.3.2 In-Reply-To: References: <0740cc83-a1b4-f384-ffb8-fe53b892ad83@opensips.org> Message-ID: Hi Bela, OK, be sure the user you are using to log into CP has the "admin" permission on the cdrviewer tool . Check this via the Admin tools -> Access. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com OpenSIPS eBootcamp 23rd May - 3rd June 2022 https://opensips.org/training/OpenSIPS_eBootcamp_2022/ On 5/18/22 12:25 AM, Bela H wrote: > > Hi Bogdan, > > I have re-installed the CP 9.3.2 but the results are same. Still > missing CDR fields in the CDR Viewer and no gear icon anywhere only > Users/Alias Management and System/Monit. > > However, in the CDR details I can see the additional fields: > > What did I wrong? > > Cheers, > > Bela > > *From: *Bogdan-Andrei Iancu > *Sent: *Wednesday, 18 May 2022 03:01 > *To: *OpenSIPS users mailling list ; > Bela H > *Subject: *Re: [OpenSIPS-Users] OpenSIPS CP upgrade to 9.3.2 > > Hi Bela, > > Does you CDRviewer look like this ? > > > > See the gear box in the right upper corner. > > And be sure that the 9.3.2 version is indeed displayed in the left > upper corner. > > Regards, > > Bogdan-Andrei Iancu > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > OpenSIPS eBootcamp 23rd May - 3rd June 2022 > https://opensips.org/training/OpenSIPS_eBootcamp_2022/ > > On 5/17/22 10:55 AM, Bela H wrote: > > Hello, > > What is the best method to upgrade the control panel from 8.3.2 to > 9.3.2? > > I had some extra fields e.g. in CDR viewer and disappeared after > 9.3.2. It is in the file > /var/www/html/opensips-cp/config/tools/system/cdrviewer/local.inc.php > but not visible in the CDR viewer panel only in the detailed view > for each call. > > Also I don’t see this gear icon Bogdan mentioned in the blog: Each > tool has its own Settings panel “accessible via the gear-icon in > the right side of the tool header”. > > Cheers, > > Bela > > > > _______________________________________________ > > 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: 9BE1CA3C97C945809C3B1B43988199E7.png Type: image/png Size: 45526 bytes Desc: not available URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: 302052B480C8471E8A3FE4693709167B.png Type: image/png Size: 21364 bytes Desc: not available URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: jjfhkbbgkdgdgnie.png Type: image/png Size: 18455 bytes Desc: not available URL: From jehanzaib.kiani at gmail.com Wed May 18 06:39:03 2022 From: jehanzaib.kiani at gmail.com (Jehanzaib Younis) Date: Wed, 18 May 2022 18:39:03 +1200 Subject: [OpenSIPS-Users] How to use old "opensipsctl avp add" with opensips-cli? In-Reply-To: References: Message-ID: Hi Bela, I think you can use (opensips-cli): database add avp command Regards, Jehanzaib On Wed, May 18, 2022 at 6:20 PM Bela H wrote: > Hello, > > I want to set up a call forwarding and followed the instructions from the > "Building telephony systems with OpenSIPS". However, this is a little bit > outdated, the old opensipsctl was replaced by opensips-cli. How can I add > an avp data into a usr_preferences table from opensips-cli? > This was the old format: *opensipsctl avp add A_number callfwd 0 C_number* > > Cheers, > Bela > _______________________________________________ > 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 Wed May 18 07:08:26 2022 From: Johan at democon.be (Johan De Clercq) Date: Wed, 18 May 2022 09:08:26 +0200 Subject: [OpenSIPS-Users] How to use old "opensipsctl avp add" with opensips-cli? In-Reply-To: References: Message-ID: Create your own table in the db. Then specify that as DB table in the module parameters (will avoid hassle when you have to migrate) Op wo 18 mei 2022 om 08:22 schreef Bela H : > Hello, > > I want to set up a call forwarding and followed the instructions from the > "Building telephony systems with OpenSIPS". However, this is a little bit > outdated, the old opensipsctl was replaced by opensips-cli. How can I add > an avp data into a usr_preferences table from opensips-cli? > This was the old format: *opensipsctl avp add A_number callfwd 0 C_number* > > Cheers, > Bela > _______________________________________________ > 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.rehan at gmail.com Wed May 18 08:25:52 2022 From: ahmed.rehan at gmail.com (Ahmed Rehan) Date: Wed, 18 May 2022 13:25:52 +0500 Subject: [OpenSIPS-Users] no TLS client domain found error In-Reply-To: <72b9b0b1-5520-0818-a5a3-af8ad451fed7@opensips.org> References: <8877c8b2-9ad3-a0d0-3294-5a3d20c748a1@opensips.org> <72b9b0b1-5520-0818-a5a3-af8ad451fed7@opensips.org> Message-ID: 1. certificates must be in any subdir of /etc/opensips/tls/ 2. it should be a real files, not symlinks. Check these settings On Wed, May 18, 2022 at 11:15 AM Bogdan-Andrei Iancu wrote: > Hi Jehanzaib, > > The sequence for the MST TLS domains is wrong. > > For each TLS domain block, you need to start only with a server_domain or > client_domain - of course, different names. And for each domain you need > you set the matching conditions. See > https://opensips.org/html/docs/modules/3.2.x/tls_mgm.html#domains-param > > Basically something like: > > modparam("tls_mgm", "server_domain", "formsteams_server") > modparam("tls_mgm", "match_ip_address", "[formsteams_server]....") > modparam("tls_mgm", "match_sip_domain", "[formsteams_server]....") > modparam("tls_mgm", "certificate", "[formsteams_server].....) > .... > > > modparam("tls_mgm", "client_domain", "formsteams_client") > modparam("tls_mgm", "match_ip_address", "[formsteams_client]....") > modparam("tls_mgm", "match_sip_domain", "[formsteams_client]....") > modparam("tls_mgm", "certificate", "[formsteams_client].....) > .... > > > Best regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > OpenSIPS eBootcamp 23rd May - 3rd June 2022 > https://opensips.org/training/OpenSIPS_eBootcamp_2022/ > > On 5/18/22 2:38 AM, Jehanzaib Younis wrote: > > Hi Bogdan, > That's the problem, when I try to add the client_domain I get an error. > Actually, I have a working config for webrtc but now I am adding a new > domain for MS teams direct route. In fact, any other domain gives an error. > If I disable MS Teams domain, the opensips do not give an error message and > my webrtc client can connect without any issue. > > loadmodule "tls_mgm.so" > modparam("tls_mgm", "tls_library", "wolfssl") > > #### (WebRTC) Client > modparam("tls_mgm", "server_domain", "sip.mywebphone.xx") > modparam("tls_mgm", "certificate", > "[sip.mywebphone.xx]/etc/letsencrypt/live/sip.mywebphone.xx/cert.pem") > modparam("tls_mgm", "private_key", > "[sip.mywebphone.xx]/etc/letsencrypt/live/sip.mywebphone.xx/privkey.pem") > modparam("tls_mgm", "ca_list", > "[sip.mywebphone.xx]/etc/letsencrypt/live/sip.mywebphone.xx/fullchain.pem") > modparam("tls_mgm", "ca_dir", > "[sip.mywebphone.xx]/etc/letsencrypt/live/sip.mywebphone.xx") > modparam("tls_mgm", "tls_method", "[sip.mywebphone.xx]SSLv23") > modparam("tls_mgm", "verify_cert", "[sip.mywebphone.xx]1") > modparam("tls_mgm", "require_cert", "[sip.mywebphone.xx]1") > > ### This is for MS-Teams direct route > modparam("tls_mgm", "server_domain", "dom1.formsteams.com") > modparam("tls_mgm", "client_domain", "dom1.formsteams.com") > modparam("tls_mgm", "certificate", "[dom1.formsteams.com > ]/etc/letsencrypt/live/dom1.formsteams.com/cert.pem") > modparam("tls_mgm", "private_key", "[dom1.formsteams.com > ]/etc/letsencrypt/live/dom1.formsteams.com/privkey.pem") > modparam("tls_mgm", "ca_list", "[dom1.formsteams.com > ]/etc/letsencrypt/live/dom1.formsteams.com/fullchain.pem") > modparam("tls_mgm", "ca_dir", "[dom1.formsteams.com]/etc/letsencrypt/live/ > dom1.formsteams.com") > modparam("tls_mgm", "tls_method", "[dom1.formsteams.com]SSLv23") > modparam("tls_mgm", "verify_cert", "[dom1.formsteams.com]1") > modparam("tls_mgm", "require_cert", "[dom1.formsteams.com]1") > modparam("tls_mgm", "client_sip_domain_avp", "tls_sip_dom") > > When i enable the MS-Teams direct route domain i get the below error: > no certificate for tls domain ' dom1.formsteams.com ' defined > > > Regards, > Jehanzaib > > > On Wed, May 18, 2022 at 3:04 AM Bogdan-Andrei Iancu > wrote: > >> Hi Jehanzaib, >> >> What are the TLS client domains you have defined in your tls_mgm module ? >> >> Regards, >> >> Bogdan-Andrei Iancu >> >> OpenSIPS Founder and Developer >> https://www.opensips-solutions.com >> OpenSIPS eBootcamp 23rd May - 3rd June 2022 >> https://opensips.org/training/OpenSIPS_eBootcamp_2022/ >> >> On 5/17/22 4:32 PM, Jehanzaib Younis wrote: >> >> Hi, >> >> I am having trouble to send/receive OPTIONS to ms teams. >> Using the dispatcher module. The socket is defined as tls:*mysbcip*:5061 >> Looks like when my opensips (3.2.x) tries to send OPTIONS. it is giving >> me the following error >> >> ERROR:proto_tls:proto_tls_conn_init: no TLS client domain found >> ERROR:core:tcp_conn_create: failed to do proto 3 specific init for conn >> 0x7f00ef2a85a0 >> ERROR:core:tcp_async_connect: tcp_conn_create failed >> ERROR:proto_tls:proto_tls_send: async TCP connect failed >> ERROR:tm:msg_send: send() to 52.114.76.76:5061 for proto tls/3 failed >> ERROR:tm:t_uac: attempt to send to ' >> sip:sip3.pstnhub.microsoft.com:5061;transport:tls' failed >> >> I am setting the Contact as >> >> Looks like the client domain is used for outgoing TLS connection but no >> idea which domain i need to add here. The socket is my opensips ip address. >> >> Has anyone seen a similar kind of behaviour? >> >> Thank you. >> >> Regards, >> Jehanzaib >> >> _______________________________________________ >> 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 > -- Regards Ahmed Rehan -------------- next part -------------- An HTML attachment was scrubbed... URL: From jehanzaib.kiani at gmail.com Wed May 18 13:01:10 2022 From: jehanzaib.kiani at gmail.com (Jehanzaib Younis) Date: Thu, 19 May 2022 01:01:10 +1200 Subject: [OpenSIPS-Users] no TLS client domain found error In-Reply-To: <72b9b0b1-5520-0818-a5a3-af8ad451fed7@opensips.org> References: <8877c8b2-9ad3-a0d0-3294-5a3d20c748a1@opensips.org> <72b9b0b1-5520-0818-a5a3-af8ad451fed7@opensips.org> Message-ID: Thank you Bogdan, That helped a lot. As you mentioned I need to start only with server_domain or client_domain. Now I changed my config a bit as shown below: #### (WebRTC) Client modparam("tls_mgm", "server_domain", "sip.mywebphone.xx") modparam("tls_mgm", "certificate", "[sip.mywebphone.xx]/etc/letsencrypt/live/sip.mywebphone.xx/cert.pem") modparam("tls_mgm", "private_key", "[sip.mywebphone.xx]/etc/letsencrypt/live/sip.mywebphone.xx/privkey.pem") modparam("tls_mgm", "ca_list", "[sip.mywebphone.xx]/etc/letsencrypt/live/sip.mywebphone.xx/fullchain.pem") modparam("tls_mgm", "ca_dir", "[sip.mywebphone.xx]/etc/letsencrypt/live/sip.mywebphone.xx") modparam("tls_mgm", "tls_method", "[sip.mywebphone.xx]SSLv23") modparam("tls_mgm", "verify_cert", "[sip.mywebphone.xx]1") modparam("tls_mgm", "require_cert", "[sip.mywebphone.xx]1") ### This is for MS-Teams direct route modparam("tls_mgm", "client_domain", "dom1.formsteams.com") modparam("tls_mgm", "certificate", "[dom1.formsteams.com ]/etc/letsencrypt/live/dom1.formsteams.com/cert.pem") modparam("tls_mgm", "private_key", "[dom1.formsteams.com ]/etc/letsencrypt/live/dom1.formsteams.com/privkey.pem") modparam("tls_mgm", "ca_list", "[dom1.formsteams.com]/etc/letsencrypt/live/ dom1.formsteams.com/fullchain.pem") modparam("tls_mgm", "ca_dir", "[dom1.formsteams.com]/etc/letsencrypt/live/ dom1.formsteams.com") modparam("tls_mgm", "tls_method", "[dom1.formsteams.com]SSLv23") modparam("tls_mgm", "verify_cert", "[dom1.formsteams.com]1") modparam("tls_mgm", "require_cert", "[dom1.formsteams.com]1") modparam("tls_mgm", "client_sip_domain_avp", "tls_sip_dom") Looks like the initial handshake is fine when my server sends OPTIONS to MSTeams. There is a bug in the code according to the logs as shown below: opensips[10659]: CRITICAL:core:io_watch_add: #012>>> used fd map fd=142 is not present in fd_array (fd=142,type=19,flags=80000003,data=0x7f825805ceb8)#012#012It seems you have hit a programming bug.#012Please help us make OpenSIPS better by reporting it at https://github.com/OpenSIPS/opensips/issues opensips[10659]: CRITICAL:core:io_watch_add: [TCP_main] check failed after successful fd add (fd=141,type=19,data=0x7f825804fd98,flags=1) already=0 opensips[23993]: NOTICE:tls_wolfssl:verify_callback: depth = 1, verify success opensips[23993]: NOTICE:tls_wolfssl:verify_callback: depth = 0, verify success opensips[23993]: INFO:tls_wolfssl:_wolfssl_tls_async_connect: new TLS connection to 52.114.16.74:5061 established opensips[23993]: NOTICE:tls_wolfssl:verify_callback: depth = 1, verify success opensips[23993]: NOTICE:tls_wolfssl:verify_callback: depth = 0, verify success opensips[23995]: INFO:tls_wolfssl:_wolfssl_tls_async_connect: new TLS connection to 52.114.76.76:5061 established Regards, Jehanzaib On Wed, May 18, 2022 at 6:15 PM Bogdan-Andrei Iancu wrote: > Hi Jehanzaib, > > The sequence for the MST TLS domains is wrong. > > For each TLS domain block, you need to start only with a server_domain or > client_domain - of course, different names. And for each domain you need > you set the matching conditions. See > https://opensips.org/html/docs/modules/3.2.x/tls_mgm.html#domains-param > > Basically something like: > > modparam("tls_mgm", "server_domain", "formsteams_server") > modparam("tls_mgm", "match_ip_address", "[formsteams_server]....") > modparam("tls_mgm", "match_sip_domain", "[formsteams_server]....") > modparam("tls_mgm", "certificate", "[formsteams_server].....) > .... > > > modparam("tls_mgm", "client_domain", "formsteams_client") > modparam("tls_mgm", "match_ip_address", "[formsteams_client]....") > modparam("tls_mgm", "match_sip_domain", "[formsteams_client]....") > modparam("tls_mgm", "certificate", "[formsteams_client].....) > .... > > > Best regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > OpenSIPS eBootcamp 23rd May - 3rd June 2022 > https://opensips.org/training/OpenSIPS_eBootcamp_2022/ > > On 5/18/22 2:38 AM, Jehanzaib Younis wrote: > > Hi Bogdan, > That's the problem, when I try to add the client_domain I get an error. > Actually, I have a working config for webrtc but now I am adding a new > domain for MS teams direct route. In fact, any other domain gives an error. > If I disable MS Teams domain, the opensips do not give an error message and > my webrtc client can connect without any issue. > > loadmodule "tls_mgm.so" > modparam("tls_mgm", "tls_library", "wolfssl") > > #### (WebRTC) Client > modparam("tls_mgm", "server_domain", "sip.mywebphone.xx") > modparam("tls_mgm", "certificate", > "[sip.mywebphone.xx]/etc/letsencrypt/live/sip.mywebphone.xx/cert.pem") > modparam("tls_mgm", "private_key", > "[sip.mywebphone.xx]/etc/letsencrypt/live/sip.mywebphone.xx/privkey.pem") > modparam("tls_mgm", "ca_list", > "[sip.mywebphone.xx]/etc/letsencrypt/live/sip.mywebphone.xx/fullchain.pem") > modparam("tls_mgm", "ca_dir", > "[sip.mywebphone.xx]/etc/letsencrypt/live/sip.mywebphone.xx") > modparam("tls_mgm", "tls_method", "[sip.mywebphone.xx]SSLv23") > modparam("tls_mgm", "verify_cert", "[sip.mywebphone.xx]1") > modparam("tls_mgm", "require_cert", "[sip.mywebphone.xx]1") > > ### This is for MS-Teams direct route > modparam("tls_mgm", "server_domain", "dom1.formsteams.com") > modparam("tls_mgm", "client_domain", "dom1.formsteams.com") > modparam("tls_mgm", "certificate", "[dom1.formsteams.com > ]/etc/letsencrypt/live/dom1.formsteams.com/cert.pem") > modparam("tls_mgm", "private_key", "[dom1.formsteams.com > ]/etc/letsencrypt/live/dom1.formsteams.com/privkey.pem") > modparam("tls_mgm", "ca_list", "[dom1.formsteams.com > ]/etc/letsencrypt/live/dom1.formsteams.com/fullchain.pem") > modparam("tls_mgm", "ca_dir", "[dom1.formsteams.com]/etc/letsencrypt/live/ > dom1.formsteams.com") > modparam("tls_mgm", "tls_method", "[dom1.formsteams.com]SSLv23") > modparam("tls_mgm", "verify_cert", "[dom1.formsteams.com]1") > modparam("tls_mgm", "require_cert", "[dom1.formsteams.com]1") > modparam("tls_mgm", "client_sip_domain_avp", "tls_sip_dom") > > When i enable the MS-Teams direct route domain i get the below error: > no certificate for tls domain ' dom1.formsteams.com ' defined > > > Regards, > Jehanzaib > > > On Wed, May 18, 2022 at 3:04 AM Bogdan-Andrei Iancu > wrote: > >> Hi Jehanzaib, >> >> What are the TLS client domains you have defined in your tls_mgm module ? >> >> Regards, >> >> Bogdan-Andrei Iancu >> >> OpenSIPS Founder and Developer >> https://www.opensips-solutions.com >> OpenSIPS eBootcamp 23rd May - 3rd June 2022 >> https://opensips.org/training/OpenSIPS_eBootcamp_2022/ >> >> On 5/17/22 4:32 PM, Jehanzaib Younis wrote: >> >> Hi, >> >> I am having trouble to send/receive OPTIONS to ms teams. >> Using the dispatcher module. The socket is defined as tls:*mysbcip*:5061 >> Looks like when my opensips (3.2.x) tries to send OPTIONS. it is giving >> me the following error >> >> ERROR:proto_tls:proto_tls_conn_init: no TLS client domain found >> ERROR:core:tcp_conn_create: failed to do proto 3 specific init for conn >> 0x7f00ef2a85a0 >> ERROR:core:tcp_async_connect: tcp_conn_create failed >> ERROR:proto_tls:proto_tls_send: async TCP connect failed >> ERROR:tm:msg_send: send() to 52.114.76.76:5061 for proto tls/3 failed >> ERROR:tm:t_uac: attempt to send to ' >> sip:sip3.pstnhub.microsoft.com:5061;transport:tls' failed >> >> I am setting the Contact as >> >> Looks like the client domain is used for outgoing TLS connection but no >> idea which domain i need to add here. The socket is my opensips ip address. >> >> Has anyone seen a similar kind of behaviour? >> >> Thank you. >> >> Regards, >> Jehanzaib >> >> _______________________________________________ >> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: From razvan at opensips.org Wed May 18 13:04:55 2022 From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=) Date: Wed, 18 May 2022 16:04:55 +0300 Subject: [OpenSIPS-Users] JSON log format In-Reply-To: References: Message-ID: <3b5627e7-fc82-7c88-3709-84d1f0203bee@opensips.org> Hi, Denis! No plans yet, but feel free to open a feature request[1]. This way we can easily keep track of all requests. [1] https://github.com/OpenSIPS/opensips/issues Best regards, Răzvan Crainea OpenSIPS Core Developer http://www.opensips-solutions.com On 3/21/22 14:26, Denis Alekseytsev wrote: > Hi, > > Are there any plans to introduce JSON log format and systemd-journal > integration? > > Thanks, > Xaled > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users From osas at voipembedded.com Wed May 18 13:34:19 2022 From: osas at voipembedded.com (Ovidiu Sas) Date: Wed, 18 May 2022 09:34:19 -0400 Subject: [OpenSIPS-Users] no TLS client domain found error In-Reply-To: References: <8877c8b2-9ad3-a0d0-3294-5a3d20c748a1@opensips.org> <72b9b0b1-5520-0818-a5a3-af8ad451fed7@opensips.org> Message-ID: Please upgrade to the latest version and see if the error persists. If yes, please run the server in debug mode and save the logs so this issue can be investigated properly. Thanks, Ovidiu On Wed, May 18, 2022 at 09:02 Jehanzaib Younis wrote: > Thank you Bogdan, > That helped a lot. As you mentioned I need to start only with > server_domain or client_domain. > Now I changed my config a bit as shown below: > #### (WebRTC) Client > modparam("tls_mgm", "server_domain", "sip.mywebphone.xx") > modparam("tls_mgm", "certificate", > "[sip.mywebphone.xx]/etc/letsencrypt/live/sip.mywebphone.xx/cert.pem") > modparam("tls_mgm", "private_key", > "[sip.mywebphone.xx]/etc/letsencrypt/live/sip.mywebphone.xx/privkey.pem") > modparam("tls_mgm", "ca_list", > "[sip.mywebphone.xx]/etc/letsencrypt/live/sip.mywebphone.xx/fullchain.pem") > modparam("tls_mgm", "ca_dir", > "[sip.mywebphone.xx]/etc/letsencrypt/live/sip.mywebphone.xx") > modparam("tls_mgm", "tls_method", "[sip.mywebphone.xx]SSLv23") > modparam("tls_mgm", "verify_cert", "[sip.mywebphone.xx]1") > modparam("tls_mgm", "require_cert", "[sip.mywebphone.xx]1") > > ### This is for MS-Teams direct route > modparam("tls_mgm", "client_domain", "dom1.formsteams.com") > modparam("tls_mgm", "certificate", "[dom1.formsteams.com > ]/etc/letsencrypt/live/dom1.formsteams.com/cert.pem") > modparam("tls_mgm", "private_key", "[dom1.formsteams.com > ]/etc/letsencrypt/live/dom1.formsteams.com/privkey.pem") > modparam("tls_mgm", "ca_list", "[dom1.formsteams.com > ]/etc/letsencrypt/live/dom1.formsteams.com/fullchain.pem") > modparam("tls_mgm", "ca_dir", "[dom1.formsteams.com]/etc/letsencrypt/live/ > dom1.formsteams.com") > modparam("tls_mgm", "tls_method", "[dom1.formsteams.com]SSLv23") > modparam("tls_mgm", "verify_cert", "[dom1.formsteams.com]1") > modparam("tls_mgm", "require_cert", "[dom1.formsteams.com]1") > modparam("tls_mgm", "client_sip_domain_avp", "tls_sip_dom") > > Looks like the initial handshake is fine when my server sends OPTIONS to > MSTeams. There is a bug in the code according to the logs as shown below: > > opensips[10659]: CRITICAL:core:io_watch_add: #012>>> used fd map fd=142 is > not present in fd_array > (fd=142,type=19,flags=80000003,data=0x7f825805ceb8)#012#012It seems you > have hit a programming bug.#012Please help us make OpenSIPS better by > reporting it at https://github.com/OpenSIPS/opensips/issues > opensips[10659]: CRITICAL:core:io_watch_add: [TCP_main] check failed after > successful fd add (fd=141,type=19,data=0x7f825804fd98,flags=1) already=0 > opensips[23993]: NOTICE:tls_wolfssl:verify_callback: depth = 1, verify > success > opensips[23993]: NOTICE:tls_wolfssl:verify_callback: depth = 0, verify > success > opensips[23993]: INFO:tls_wolfssl:_wolfssl_tls_async_connect: new TLS > connection to 52.114.16.74:5061 established > opensips[23993]: NOTICE:tls_wolfssl:verify_callback: depth = 1, verify > success > opensips[23993]: NOTICE:tls_wolfssl:verify_callback: depth = 0, verify > success > opensips[23995]: INFO:tls_wolfssl:_wolfssl_tls_async_connect: new TLS > connection to 52.114.76.76:5061 established > > > Regards, > Jehanzaib > > > On Wed, May 18, 2022 at 6:15 PM Bogdan-Andrei Iancu > wrote: > >> Hi Jehanzaib, >> >> The sequence for the MST TLS domains is wrong. >> >> For each TLS domain block, you need to start only with a server_domain >> or client_domain - of course, different names. And for each domain you need >> you set the matching conditions. See >> https://opensips.org/html/docs/modules/3.2.x/tls_mgm.html#domains-param >> >> Basically something like: >> >> modparam("tls_mgm", "server_domain", "formsteams_server") >> modparam("tls_mgm", "match_ip_address", "[formsteams_server]....") >> modparam("tls_mgm", "match_sip_domain", "[formsteams_server]....") >> modparam("tls_mgm", "certificate", "[formsteams_server].....) >> .... >> >> >> modparam("tls_mgm", "client_domain", "formsteams_client") >> modparam("tls_mgm", "match_ip_address", "[formsteams_client]....") >> modparam("tls_mgm", "match_sip_domain", "[formsteams_client]....") >> modparam("tls_mgm", "certificate", "[formsteams_client].....) >> .... >> >> >> Best regards, >> >> Bogdan-Andrei Iancu >> >> OpenSIPS Founder and Developer >> https://www.opensips-solutions.com >> OpenSIPS eBootcamp 23rd May - 3rd June 2022 >> https://opensips.org/training/OpenSIPS_eBootcamp_2022/ >> >> On 5/18/22 2:38 AM, Jehanzaib Younis wrote: >> >> Hi Bogdan, >> That's the problem, when I try to add the client_domain I get an error. >> Actually, I have a working config for webrtc but now I am adding a new >> domain for MS teams direct route. In fact, any other domain gives an error. >> If I disable MS Teams domain, the opensips do not give an error message and >> my webrtc client can connect without any issue. >> >> loadmodule "tls_mgm.so" >> modparam("tls_mgm", "tls_library", "wolfssl") >> >> #### (WebRTC) Client >> modparam("tls_mgm", "server_domain", "sip.mywebphone.xx") >> modparam("tls_mgm", "certificate", >> "[sip.mywebphone.xx]/etc/letsencrypt/live/sip.mywebphone.xx/cert.pem") >> modparam("tls_mgm", "private_key", >> "[sip.mywebphone.xx]/etc/letsencrypt/live/sip.mywebphone.xx/privkey.pem") >> modparam("tls_mgm", "ca_list", >> "[sip.mywebphone.xx]/etc/letsencrypt/live/sip.mywebphone.xx/fullchain.pem") >> modparam("tls_mgm", "ca_dir", >> "[sip.mywebphone.xx]/etc/letsencrypt/live/sip.mywebphone.xx") >> modparam("tls_mgm", "tls_method", "[sip.mywebphone.xx]SSLv23") >> modparam("tls_mgm", "verify_cert", "[sip.mywebphone.xx]1") >> modparam("tls_mgm", "require_cert", "[sip.mywebphone.xx]1") >> >> ### This is for MS-Teams direct route >> modparam("tls_mgm", "server_domain", "dom1.formsteams.com") >> modparam("tls_mgm", "client_domain", "dom1.formsteams.com") >> modparam("tls_mgm", "certificate", "[dom1.formsteams.com >> ]/etc/letsencrypt/live/dom1.formsteams.com/cert.pem") >> modparam("tls_mgm", "private_key", "[dom1.formsteams.com >> ]/etc/letsencrypt/live/dom1.formsteams.com/privkey.pem") >> modparam("tls_mgm", "ca_list", "[dom1.formsteams.com >> ]/etc/letsencrypt/live/dom1.formsteams.com/fullchain.pem") >> modparam("tls_mgm", "ca_dir", "[dom1.formsteams.com >> ]/etc/letsencrypt/live/dom1.formsteams.com") >> modparam("tls_mgm", "tls_method", "[dom1.formsteams.com]SSLv23") >> modparam("tls_mgm", "verify_cert", "[dom1.formsteams.com]1") >> modparam("tls_mgm", "require_cert", "[dom1.formsteams.com]1") >> modparam("tls_mgm", "client_sip_domain_avp", "tls_sip_dom") >> >> When i enable the MS-Teams direct route domain i get the below error: >> no certificate for tls domain ' dom1.formsteams.com ' defined >> >> >> Regards, >> Jehanzaib >> >> >> On Wed, May 18, 2022 at 3:04 AM Bogdan-Andrei Iancu >> wrote: >> >>> Hi Jehanzaib, >>> >>> What are the TLS client domains you have defined in your tls_mgm module ? >>> >>> Regards, >>> >>> Bogdan-Andrei Iancu >>> >>> OpenSIPS Founder and Developer >>> https://www.opensips-solutions.com >>> OpenSIPS eBootcamp 23rd May - 3rd June 2022 >>> https://opensips.org/training/OpenSIPS_eBootcamp_2022/ >>> >>> On 5/17/22 4:32 PM, Jehanzaib Younis wrote: >>> >>> Hi, >>> >>> I am having trouble to send/receive OPTIONS to ms teams. >>> Using the dispatcher module. The socket is defined as tls:*mysbcip*:5061 >>> Looks like when my opensips (3.2.x) tries to send OPTIONS. it is giving >>> me the following error >>> >>> ERROR:proto_tls:proto_tls_conn_init: no TLS client domain found >>> ERROR:core:tcp_conn_create: failed to do proto 3 specific init for conn >>> 0x7f00ef2a85a0 >>> ERROR:core:tcp_async_connect: tcp_conn_create failed >>> ERROR:proto_tls:proto_tls_send: async TCP connect failed >>> ERROR:tm:msg_send: send() to 52.114.76.76:5061 for proto tls/3 failed >>> ERROR:tm:t_uac: attempt to send to ' >>> sip:sip3.pstnhub.microsoft.com:5061;transport:tls' failed >>> >>> I am setting the Contact as >>> >>> Looks like the client domain is used for outgoing TLS connection but no >>> idea which domain i need to add here. The socket is my opensips ip address. >>> >>> Has anyone seen a similar kind of behaviour? >>> >>> Thank you. >>> >>> Regards, >>> Jehanzaib >>> >>> _______________________________________________ >>> 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 > -- VoIP Embedded, Inc. http://www.voipembedded.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Wed May 18 15:13:19 2022 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Wed, 18 May 2022 18:13:19 +0300 Subject: [OpenSIPS-Users] [Release] OpenSIPS 3.3.0 major release, beta version Message-ID: <5c0e4ad0-7a75-e5d6-389d-5583a53f4995@opensips.org> Hi there !! It is that time of the year to do our iteration - one more year, one more ambitious roadmap , one more OpenSIPS major release. So, we are all happy to announce the beta release of *OpenSIPS 3.3.0 major version* - and this 3.3 version is all about Messaging with MSRP, about Unified Communications with Contact Centers, about IMS with RCS and Diameter. But here is the shortest possible description of this release; and be aware that it's actually not so short as nothing is short about 3.3 ! Please keep in mind that 3.3.0 is still a beta release, targeting mid July to become fully stable. So, we still have some testing ahead of us :). Many thanks to our awesome community for contributing with ideas, code, patches, tests and reports! Looking for downloading it? See the tarball or the GIT repo . Packages (rpms and debs ) are also already available! Enjoy it, -- Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com OpenSIPS eBootcamp 23rd May - 3rd June 2022 https://opensips.org/training/OpenSIPS_eBootcamp_2022/ -------------- next part -------------- An HTML attachment was scrubbed... URL: From m.shirazi at gmail.com Wed May 18 18:32:03 2022 From: m.shirazi at gmail.com (Mehdi Shirazi) Date: Wed, 18 May 2022 23:02:03 +0430 Subject: [OpenSIPS-Users] SQL Cacher+Galera Cluster Message-ID: Hi I plan to use SQL Cacher with Galera Cluster. After a record changes I want to update the cache. Using standard triggers in mariadb is not possible to run opensips-cli commands. Please tell me your suggestions for the best approach. Regards M.Shirazi -------------- next part -------------- An HTML attachment was scrubbed... URL: From goatolina at gmail.com Wed May 18 21:05:29 2022 From: goatolina at gmail.com (Ali Alawi) Date: Thu, 19 May 2022 00:05:29 +0300 Subject: [OpenSIPS-Users] Cli and DB path In-Reply-To: <9f57138c-5d40-b8f5-8da2-8fb6cf4e91c2@opensips.org> References: <9f57138c-5d40-b8f5-8da2-8fb6cf4e91c2@opensips.org> Message-ID: Dear Răzvan, Thanks for your response, the cli are all set successfully using database_schema_path. Regards, Ali On Tue, May 17, 2022 at 10:52 AM Răzvan Crainea wrote: > Hi, Ali! > > Setting the database_schema_path should do the trick. Can you set it > again and provide the logs? > > Best regards, > > Răzvan Crainea > OpenSIPS Core Developer > http://www.opensips-solutions.com > > On 5/14/22 02:30, Ali Alawi wrote: > > Dear all, > > > > When I install opensips3.2 using APT packages, the cli point correctly > > to mysql (mariadb) through /usr/share/opensips (Everything work fine) > > > > However, when installation done using git clone --recursive, the cli > > point to /usr/share/opensips but in this time the cli doesn't find > > mysql when i try to: > > opensips-cli -x database create > > ERROR: path '/usr/share/opensips' to OpenSIPS DB scripts does not exist! > > > > I notice that mysql is resides inside '/usr/local/share/opensips' > > instead of '/usr/share/opensips' > > I try to include the corrected path in the default.cfg and also try > > |opensips-cli -o database_schema_path=| > > |But I come up with no success| > > |Any suggestions please? | > > > > Regards, > > Ali > > > > _______________________________________________ > > Users mailing list > > Users at lists.opensips.org > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -------------- next part -------------- An HTML attachment was scrubbed... URL: From jehanzaib.kiani at gmail.com Thu May 19 12:07:58 2022 From: jehanzaib.kiani at gmail.com (Jehanzaib Younis) Date: Fri, 20 May 2022 00:07:58 +1200 Subject: [OpenSIPS-Users] no TLS client domain found error In-Reply-To: References: <8877c8b2-9ad3-a0d0-3294-5a3d20c748a1@opensips.org> <72b9b0b1-5520-0818-a5a3-af8ad451fed7@opensips.org> Message-ID: Thanks Ovidiu, I just checked the source code, the same bug is also present in the opensips-3.2.6 branch. I have another issue with 3.2.6. I am not able to compile tls_wolfssl. No issue with 3.3 though. Now I need to check what is causing this. I am getting the following error: make[1]: Entering directory `/usr/src/opensips-3.2/modules/tls_wolfssl' configure: WARNING: unrecognized options: --disable-shared, --enable-static checking whether make supports nested variables... (cached) yes ./configure: line 5259: syntax error near unexpected token `2.4.2' ./configure: line 5259: `LT_PREREQ(2.4.2)' make[1]: *** [lib/lib/libwolfssl.a] Error 2 Regards, Jehanzaib On Thu, May 19, 2022 at 1:35 AM Ovidiu Sas wrote: > Please upgrade to the latest version and see if the error persists. If > yes, please run the server in debug mode and save the logs so this issue > can be investigated properly. > > Thanks, > Ovidiu > > On Wed, May 18, 2022 at 09:02 Jehanzaib Younis > wrote: > >> Thank you Bogdan, >> That helped a lot. As you mentioned I need to start only with >> server_domain or client_domain. >> Now I changed my config a bit as shown below: >> #### (WebRTC) Client >> modparam("tls_mgm", "server_domain", "sip.mywebphone.xx") >> modparam("tls_mgm", "certificate", >> "[sip.mywebphone.xx]/etc/letsencrypt/live/sip.mywebphone.xx/cert.pem") >> modparam("tls_mgm", "private_key", >> "[sip.mywebphone.xx]/etc/letsencrypt/live/sip.mywebphone.xx/privkey.pem") >> modparam("tls_mgm", "ca_list", >> "[sip.mywebphone.xx]/etc/letsencrypt/live/sip.mywebphone.xx/fullchain.pem") >> modparam("tls_mgm", "ca_dir", >> "[sip.mywebphone.xx]/etc/letsencrypt/live/sip.mywebphone.xx") >> modparam("tls_mgm", "tls_method", "[sip.mywebphone.xx]SSLv23") >> modparam("tls_mgm", "verify_cert", "[sip.mywebphone.xx]1") >> modparam("tls_mgm", "require_cert", "[sip.mywebphone.xx]1") >> >> ### This is for MS-Teams direct route >> modparam("tls_mgm", "client_domain", "dom1.formsteams.com") >> modparam("tls_mgm", "certificate", "[dom1.formsteams.com >> ]/etc/letsencrypt/live/dom1.formsteams.com/cert.pem") >> modparam("tls_mgm", "private_key", "[dom1.formsteams.com >> ]/etc/letsencrypt/live/dom1.formsteams.com/privkey.pem") >> modparam("tls_mgm", "ca_list", "[dom1.formsteams.com >> ]/etc/letsencrypt/live/dom1.formsteams.com/fullchain.pem") >> modparam("tls_mgm", "ca_dir", "[dom1.formsteams.com >> ]/etc/letsencrypt/live/dom1.formsteams.com") >> modparam("tls_mgm", "tls_method", "[dom1.formsteams.com]SSLv23") >> modparam("tls_mgm", "verify_cert", "[dom1.formsteams.com]1") >> modparam("tls_mgm", "require_cert", "[dom1.formsteams.com]1") >> modparam("tls_mgm", "client_sip_domain_avp", "tls_sip_dom") >> >> Looks like the initial handshake is fine when my server sends OPTIONS to >> MSTeams. There is a bug in the code according to the logs as shown below: >> >> opensips[10659]: CRITICAL:core:io_watch_add: #012>>> used fd map fd=142 >> is not present in fd_array >> (fd=142,type=19,flags=80000003,data=0x7f825805ceb8)#012#012It seems you >> have hit a programming bug.#012Please help us make OpenSIPS better by >> reporting it at https://github.com/OpenSIPS/opensips/issues >> opensips[10659]: CRITICAL:core:io_watch_add: [TCP_main] check failed >> after successful fd add (fd=141,type=19,data=0x7f825804fd98,flags=1) >> already=0 >> opensips[23993]: NOTICE:tls_wolfssl:verify_callback: depth = 1, verify >> success >> opensips[23993]: NOTICE:tls_wolfssl:verify_callback: depth = 0, verify >> success >> opensips[23993]: INFO:tls_wolfssl:_wolfssl_tls_async_connect: new TLS >> connection to 52.114.16.74:5061 established >> opensips[23993]: NOTICE:tls_wolfssl:verify_callback: depth = 1, verify >> success >> opensips[23993]: NOTICE:tls_wolfssl:verify_callback: depth = 0, verify >> success >> opensips[23995]: INFO:tls_wolfssl:_wolfssl_tls_async_connect: new TLS >> connection to 52.114.76.76:5061 established >> >> >> Regards, >> Jehanzaib >> >> >> On Wed, May 18, 2022 at 6:15 PM Bogdan-Andrei Iancu >> wrote: >> >>> Hi Jehanzaib, >>> >>> The sequence for the MST TLS domains is wrong. >>> >>> For each TLS domain block, you need to start only with a server_domain >>> or client_domain - of course, different names. And for each domain you need >>> you set the matching conditions. See >>> https://opensips.org/html/docs/modules/3.2.x/tls_mgm.html#domains-param >>> >>> Basically something like: >>> >>> modparam("tls_mgm", "server_domain", "formsteams_server") >>> modparam("tls_mgm", "match_ip_address", "[formsteams_server]....") >>> modparam("tls_mgm", "match_sip_domain", "[formsteams_server]....") >>> modparam("tls_mgm", "certificate", "[formsteams_server].....) >>> .... >>> >>> >>> modparam("tls_mgm", "client_domain", "formsteams_client") >>> modparam("tls_mgm", "match_ip_address", "[formsteams_client]....") >>> modparam("tls_mgm", "match_sip_domain", "[formsteams_client]....") >>> modparam("tls_mgm", "certificate", "[formsteams_client].....) >>> .... >>> >>> >>> Best regards, >>> >>> Bogdan-Andrei Iancu >>> >>> OpenSIPS Founder and Developer >>> https://www.opensips-solutions.com >>> OpenSIPS eBootcamp 23rd May - 3rd June 2022 >>> https://opensips.org/training/OpenSIPS_eBootcamp_2022/ >>> >>> On 5/18/22 2:38 AM, Jehanzaib Younis wrote: >>> >>> Hi Bogdan, >>> That's the problem, when I try to add the client_domain I get an error. >>> Actually, I have a working config for webrtc but now I am adding a new >>> domain for MS teams direct route. In fact, any other domain gives an error. >>> If I disable MS Teams domain, the opensips do not give an error message and >>> my webrtc client can connect without any issue. >>> >>> loadmodule "tls_mgm.so" >>> modparam("tls_mgm", "tls_library", "wolfssl") >>> >>> #### (WebRTC) Client >>> modparam("tls_mgm", "server_domain", "sip.mywebphone.xx") >>> modparam("tls_mgm", "certificate", >>> "[sip.mywebphone.xx]/etc/letsencrypt/live/sip.mywebphone.xx/cert.pem") >>> modparam("tls_mgm", "private_key", >>> "[sip.mywebphone.xx]/etc/letsencrypt/live/sip.mywebphone.xx/privkey.pem") >>> modparam("tls_mgm", "ca_list", >>> "[sip.mywebphone.xx]/etc/letsencrypt/live/sip.mywebphone.xx/fullchain.pem") >>> modparam("tls_mgm", "ca_dir", >>> "[sip.mywebphone.xx]/etc/letsencrypt/live/sip.mywebphone.xx") >>> modparam("tls_mgm", "tls_method", "[sip.mywebphone.xx]SSLv23") >>> modparam("tls_mgm", "verify_cert", "[sip.mywebphone.xx]1") >>> modparam("tls_mgm", "require_cert", "[sip.mywebphone.xx]1") >>> >>> ### This is for MS-Teams direct route >>> modparam("tls_mgm", "server_domain", "dom1.formsteams.com") >>> modparam("tls_mgm", "client_domain", "dom1.formsteams.com") >>> modparam("tls_mgm", "certificate", "[dom1.formsteams.com >>> ]/etc/letsencrypt/live/dom1.formsteams.com/cert.pem") >>> modparam("tls_mgm", "private_key", "[dom1.formsteams.com >>> ]/etc/letsencrypt/live/dom1.formsteams.com/privkey.pem") >>> modparam("tls_mgm", "ca_list", "[dom1.formsteams.com >>> ]/etc/letsencrypt/live/dom1.formsteams.com/fullchain.pem") >>> modparam("tls_mgm", "ca_dir", "[dom1.formsteams.com >>> ]/etc/letsencrypt/live/dom1.formsteams.com") >>> modparam("tls_mgm", "tls_method", "[dom1.formsteams.com]SSLv23") >>> modparam("tls_mgm", "verify_cert", "[dom1.formsteams.com]1") >>> modparam("tls_mgm", "require_cert", "[dom1.formsteams.com]1") >>> modparam("tls_mgm", "client_sip_domain_avp", "tls_sip_dom") >>> >>> When i enable the MS-Teams direct route domain i get the below error: >>> no certificate for tls domain ' dom1.formsteams.com ' defined >>> >>> >>> Regards, >>> Jehanzaib >>> >>> >>> On Wed, May 18, 2022 at 3:04 AM Bogdan-Andrei Iancu >>> wrote: >>> >>>> Hi Jehanzaib, >>>> >>>> What are the TLS client domains you have defined in your tls_mgm module >>>> ? >>>> >>>> Regards, >>>> >>>> Bogdan-Andrei Iancu >>>> >>>> OpenSIPS Founder and Developer >>>> https://www.opensips-solutions.com >>>> OpenSIPS eBootcamp 23rd May - 3rd June 2022 >>>> https://opensips.org/training/OpenSIPS_eBootcamp_2022/ >>>> >>>> On 5/17/22 4:32 PM, Jehanzaib Younis wrote: >>>> >>>> Hi, >>>> >>>> I am having trouble to send/receive OPTIONS to ms teams. >>>> Using the dispatcher module. The socket is defined as tls:*mysbcip* >>>> :5061 >>>> Looks like when my opensips (3.2.x) tries to send OPTIONS. it is giving >>>> me the following error >>>> >>>> ERROR:proto_tls:proto_tls_conn_init: no TLS client domain found >>>> ERROR:core:tcp_conn_create: failed to do proto 3 specific init for conn >>>> 0x7f00ef2a85a0 >>>> ERROR:core:tcp_async_connect: tcp_conn_create failed >>>> ERROR:proto_tls:proto_tls_send: async TCP connect failed >>>> ERROR:tm:msg_send: send() to 52.114.76.76:5061 for proto tls/3 failed >>>> ERROR:tm:t_uac: attempt to send to ' >>>> sip:sip3.pstnhub.microsoft.com:5061;transport:tls' failed >>>> >>>> I am setting the Contact as >>>> >>>> Looks like the client domain is used for outgoing TLS connection but no >>>> idea which domain i need to add here. The socket is my opensips ip address. >>>> >>>> Has anyone seen a similar kind of behaviour? >>>> >>>> Thank you. >>>> >>>> Regards, >>>> Jehanzaib >>>> >>>> _______________________________________________ >>>> 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 >> > -- > VoIP Embedded, Inc. > http://www.voipembedded.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 osas at voipembedded.com Thu May 19 15:01:59 2022 From: osas at voipembedded.com (Ovidiu Sas) Date: Thu, 19 May 2022 11:01:59 -0400 Subject: [OpenSIPS-Users] no TLS client domain found error In-Reply-To: References: <8877c8b2-9ad3-a0d0-3294-5a3d20c748a1@opensips.org> <72b9b0b1-5520-0818-a5a3-af8ad451fed7@opensips.org> Message-ID: Set the log_level parameter to 4 and restart opensips. Once the error occurs, collect all the logs from the start (from syslog) and send them to Razvan. There’s bug tracking this issue: https://github.com/OpenSIPS/opensips/issues/2724 For compiling tls_wolfssl, try from a clean clone. -ovidiu On Thu, May 19, 2022 at 08:08 Jehanzaib Younis wrote: > Thanks Ovidiu, > I just checked the source code, the same bug is also present in > the opensips-3.2.6 branch. I have another issue with 3.2.6. I am not able > to compile tls_wolfssl. No issue with 3.3 though. > Now I need to check what is causing this. > > I am getting the following error: > > make[1]: Entering directory `/usr/src/opensips-3.2/modules/tls_wolfssl' > configure: WARNING: unrecognized options: --disable-shared, --enable-static > checking whether make supports nested variables... (cached) yes > ./configure: line 5259: syntax error near unexpected token `2.4.2' > ./configure: line 5259: `LT_PREREQ(2.4.2)' > make[1]: *** [lib/lib/libwolfssl.a] Error 2 > > > > Regards, > Jehanzaib > > > On Thu, May 19, 2022 at 1:35 AM Ovidiu Sas wrote: > >> Please upgrade to the latest version and see if the error persists. If >> yes, please run the server in debug mode and save the logs so this issue >> can be investigated properly. >> >> Thanks, >> Ovidiu >> >> On Wed, May 18, 2022 at 09:02 Jehanzaib Younis >> wrote: >> >>> Thank you Bogdan, >>> That helped a lot. As you mentioned I need to start only with >>> server_domain or client_domain. >>> Now I changed my config a bit as shown below: >>> #### (WebRTC) Client >>> modparam("tls_mgm", "server_domain", "sip.mywebphone.xx") >>> modparam("tls_mgm", "certificate", >>> "[sip.mywebphone.xx]/etc/letsencrypt/live/sip.mywebphone.xx/cert.pem") >>> modparam("tls_mgm", "private_key", >>> "[sip.mywebphone.xx]/etc/letsencrypt/live/sip.mywebphone.xx/privkey.pem") >>> modparam("tls_mgm", "ca_list", >>> "[sip.mywebphone.xx]/etc/letsencrypt/live/sip.mywebphone.xx/fullchain.pem") >>> modparam("tls_mgm", "ca_dir", >>> "[sip.mywebphone.xx]/etc/letsencrypt/live/sip.mywebphone.xx") >>> modparam("tls_mgm", "tls_method", "[sip.mywebphone.xx]SSLv23") >>> modparam("tls_mgm", "verify_cert", "[sip.mywebphone.xx]1") >>> modparam("tls_mgm", "require_cert", "[sip.mywebphone.xx]1") >>> >>> ### This is for MS-Teams direct route >>> modparam("tls_mgm", "client_domain", "dom1.formsteams.com") >>> modparam("tls_mgm", "certificate", "[dom1.formsteams.com >>> ]/etc/letsencrypt/live/dom1.formsteams.com/cert.pem") >>> modparam("tls_mgm", "private_key", "[dom1.formsteams.com >>> ]/etc/letsencrypt/live/dom1.formsteams.com/privkey.pem") >>> modparam("tls_mgm", "ca_list", "[dom1.formsteams.com >>> ]/etc/letsencrypt/live/dom1.formsteams.com/fullchain.pem") >>> modparam("tls_mgm", "ca_dir", "[dom1.formsteams.com >>> ]/etc/letsencrypt/live/dom1.formsteams.com") >>> modparam("tls_mgm", "tls_method", "[dom1.formsteams.com]SSLv23") >>> modparam("tls_mgm", "verify_cert", "[dom1.formsteams.com]1") >>> modparam("tls_mgm", "require_cert", "[dom1.formsteams.com]1") >>> modparam("tls_mgm", "client_sip_domain_avp", "tls_sip_dom") >>> >>> Looks like the initial handshake is fine when my server sends OPTIONS to >>> MSTeams. There is a bug in the code according to the logs as shown below: >>> >>> opensips[10659]: CRITICAL:core:io_watch_add: #012>>> used fd map fd=142 >>> is not present in fd_array >>> (fd=142,type=19,flags=80000003,data=0x7f825805ceb8)#012#012It seems you >>> have hit a programming bug.#012Please help us make OpenSIPS better by >>> reporting it at https://github.com/OpenSIPS/opensips/issues >>> opensips[10659]: CRITICAL:core:io_watch_add: [TCP_main] check failed >>> after successful fd add (fd=141,type=19,data=0x7f825804fd98,flags=1) >>> already=0 >>> opensips[23993]: NOTICE:tls_wolfssl:verify_callback: depth = 1, verify >>> success >>> opensips[23993]: NOTICE:tls_wolfssl:verify_callback: depth = 0, verify >>> success >>> opensips[23993]: INFO:tls_wolfssl:_wolfssl_tls_async_connect: new TLS >>> connection to 52.114.16.74:5061 established >>> opensips[23993]: NOTICE:tls_wolfssl:verify_callback: depth = 1, verify >>> success >>> opensips[23993]: NOTICE:tls_wolfssl:verify_callback: depth = 0, verify >>> success >>> opensips[23995]: INFO:tls_wolfssl:_wolfssl_tls_async_connect: new TLS >>> connection to 52.114.76.76:5061 established >>> >>> >>> Regards, >>> Jehanzaib >>> >>> >>> On Wed, May 18, 2022 at 6:15 PM Bogdan-Andrei Iancu >>> wrote: >>> >>>> Hi Jehanzaib, >>>> >>>> The sequence for the MST TLS domains is wrong. >>>> >>>> For each TLS domain block, you need to start only with a server_domain >>>> or client_domain - of course, different names. And for each domain you need >>>> you set the matching conditions. See >>>> https://opensips.org/html/docs/modules/3.2.x/tls_mgm.html#domains-param >>>> >>>> Basically something like: >>>> >>>> modparam("tls_mgm", "server_domain", "formsteams_server") >>>> modparam("tls_mgm", "match_ip_address", "[formsteams_server]....") >>>> modparam("tls_mgm", "match_sip_domain", "[formsteams_server]....") >>>> modparam("tls_mgm", "certificate", "[formsteams_server].....) >>>> .... >>>> >>>> >>>> modparam("tls_mgm", "client_domain", "formsteams_client") >>>> modparam("tls_mgm", "match_ip_address", "[formsteams_client]....") >>>> modparam("tls_mgm", "match_sip_domain", "[formsteams_client]....") >>>> modparam("tls_mgm", "certificate", "[formsteams_client].....) >>>> .... >>>> >>>> >>>> Best regards, >>>> >>>> Bogdan-Andrei Iancu >>>> >>>> OpenSIPS Founder and Developer >>>> https://www.opensips-solutions.com >>>> OpenSIPS eBootcamp 23rd May - 3rd June 2022 >>>> https://opensips.org/training/OpenSIPS_eBootcamp_2022/ >>>> >>>> On 5/18/22 2:38 AM, Jehanzaib Younis wrote: >>>> >>>> Hi Bogdan, >>>> That's the problem, when I try to add the client_domain I get an error. >>>> Actually, I have a working config for webrtc but now I am adding a new >>>> domain for MS teams direct route. In fact, any other domain gives an error. >>>> If I disable MS Teams domain, the opensips do not give an error message and >>>> my webrtc client can connect without any issue. >>>> >>>> loadmodule "tls_mgm.so" >>>> modparam("tls_mgm", "tls_library", "wolfssl") >>>> >>>> #### (WebRTC) Client >>>> modparam("tls_mgm", "server_domain", "sip.mywebphone.xx") >>>> modparam("tls_mgm", "certificate", >>>> "[sip.mywebphone.xx]/etc/letsencrypt/live/sip.mywebphone.xx/cert.pem") >>>> modparam("tls_mgm", "private_key", >>>> "[sip.mywebphone.xx]/etc/letsencrypt/live/sip.mywebphone.xx/privkey.pem") >>>> modparam("tls_mgm", "ca_list", >>>> "[sip.mywebphone.xx]/etc/letsencrypt/live/sip.mywebphone.xx/fullchain.pem") >>>> modparam("tls_mgm", "ca_dir", >>>> "[sip.mywebphone.xx]/etc/letsencrypt/live/sip.mywebphone.xx") >>>> modparam("tls_mgm", "tls_method", "[sip.mywebphone.xx]SSLv23") >>>> modparam("tls_mgm", "verify_cert", "[sip.mywebphone.xx]1") >>>> modparam("tls_mgm", "require_cert", "[sip.mywebphone.xx]1") >>>> >>>> ### This is for MS-Teams direct route >>>> modparam("tls_mgm", "server_domain", "dom1.formsteams.com") >>>> modparam("tls_mgm", "client_domain", "dom1.formsteams.com") >>>> modparam("tls_mgm", "certificate", "[dom1.formsteams.com >>>> ]/etc/letsencrypt/live/dom1.formsteams.com/cert.pem") >>>> modparam("tls_mgm", "private_key", "[dom1.formsteams.com >>>> ]/etc/letsencrypt/live/dom1.formsteams.com/privkey.pem") >>>> modparam("tls_mgm", "ca_list", "[dom1.formsteams.com >>>> ]/etc/letsencrypt/live/dom1.formsteams.com/fullchain.pem") >>>> modparam("tls_mgm", "ca_dir", "[dom1.formsteams.com >>>> ]/etc/letsencrypt/live/dom1.formsteams.com") >>>> modparam("tls_mgm", "tls_method", "[dom1.formsteams.com]SSLv23") >>>> modparam("tls_mgm", "verify_cert", "[dom1.formsteams.com]1") >>>> modparam("tls_mgm", "require_cert", "[dom1.formsteams.com]1") >>>> modparam("tls_mgm", "client_sip_domain_avp", "tls_sip_dom") >>>> >>>> When i enable the MS-Teams direct route domain i get the below error: >>>> no certificate for tls domain ' dom1.formsteams.com ' defined >>>> >>>> >>>> Regards, >>>> Jehanzaib >>>> >>>> >>>> On Wed, May 18, 2022 at 3:04 AM Bogdan-Andrei Iancu < >>>> bogdan at opensips.org> wrote: >>>> >>>>> Hi Jehanzaib, >>>>> >>>>> What are the TLS client domains you have defined in your tls_mgm >>>>> module ? >>>>> >>>>> Regards, >>>>> >>>>> Bogdan-Andrei Iancu >>>>> >>>>> OpenSIPS Founder and Developer >>>>> https://www.opensips-solutions.com >>>>> OpenSIPS eBootcamp 23rd May - 3rd June 2022 >>>>> https://opensips.org/training/OpenSIPS_eBootcamp_2022/ >>>>> >>>>> On 5/17/22 4:32 PM, Jehanzaib Younis wrote: >>>>> >>>>> Hi, >>>>> >>>>> I am having trouble to send/receive OPTIONS to ms teams. >>>>> Using the dispatcher module. The socket is defined as tls:*mysbcip* >>>>> :5061 >>>>> Looks like when my opensips (3.2.x) tries to send OPTIONS. it is >>>>> giving me the following error >>>>> >>>>> ERROR:proto_tls:proto_tls_conn_init: no TLS client domain found >>>>> ERROR:core:tcp_conn_create: failed to do proto 3 specific init for >>>>> conn 0x7f00ef2a85a0 >>>>> ERROR:core:tcp_async_connect: tcp_conn_create failed >>>>> ERROR:proto_tls:proto_tls_send: async TCP connect failed >>>>> ERROR:tm:msg_send: send() to 52.114.76.76:5061 for proto tls/3 failed >>>>> ERROR:tm:t_uac: attempt to send to ' >>>>> sip:sip3.pstnhub.microsoft.com:5061;transport:tls' failed >>>>> >>>>> I am setting the Contact as >>>>> >>>>> Looks like the client domain is used for outgoing TLS connection but >>>>> no idea which domain i need to add here. The socket is my opensips ip >>>>> address. >>>>> >>>>> Has anyone seen a similar kind of behaviour? >>>>> >>>>> Thank you. >>>>> >>>>> Regards, >>>>> Jehanzaib >>>>> >>>>> _______________________________________________ >>>>> 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 >>> >> -- >> VoIP Embedded, Inc. >> http://www.voipembedded.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 > -- VoIP Embedded, Inc. http://www.voipembedded.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Thu May 19 15:09:59 2022 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Thu, 19 May 2022 18:09:59 +0300 Subject: [OpenSIPS-Users] OpenSIPS Summit 2022 - Athens, Greece Message-ID: OpenSIPS Summit Sept 27th - 30th, 2022 Athens, Greece *Bridging people, bridging technologies, bridging experiences * After 2 years of online, Athens 2022 is the first in-person meeting of the OpenSIPS Community. Of course, we will preserve the online component for both speakers and participants. 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. *Some Great Reasons to Attend* * Access the latest news, knowledge and experience in the VoIP & RTC world * Learn about upcoming 3.3 OpenSIPS release and how you can leverage it * Attend unique presentations and interactive technical workshops * Meet FOSS developers and community to share experience and comments * Get solutions consultancy during the Free Design Clinics * Become an Expert attending the OpenSIPS Advanced Training *Summit Agenda* * Two full days of presentations given by key speakers * Open Discussions with key people from OpenSIPS and other OSS projects * One full day of Interactive Demos and Showcases * One full day of Design Clinics to validate your OpenSIPS deployments * One full day OpenSIPS Training (limited seats!) * Social events in the historical Athens *Attend to learn* - the registration process is already open, for both online and in-person participants. Note that the training and Design Clinics options are available only for the in-person participants. The/*Corporate Package*/ is available with an attractive discount. Register now *Speak to share* - the Call for Papers is open for in-person and online speakers. Our speaker will enjoy free admission to the event, covering lunches and evening events. Submit your paper now *Sponsor to help* - we welcome any help in making the Summit such a great event. Sponsoring is a natural way of saying "Thank you" for the Open Source code you are using within your businesses. Interested? 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 goatolina at gmail.com Thu May 19 19:30:50 2022 From: goatolina at gmail.com (Ali Alawi) Date: Thu, 19 May 2022 22:30:50 +0300 Subject: [OpenSIPS-Users] Opensips active (exited) Message-ID: Dear all, I have installed opensips 3.2 using the git clone --recursive and compile it using make menuconfig, mysql (mariadb) and cli are all set correctly. when I start opensips the status shows no error but only active (exited) as shown below: systemctl status opensips ● opensips.service - LSB: Start the OpenSIPS SIP server Loaded: loaded (/etc/init.d/opensips; generated) Active: active (exited) since Thu 2022-05-19 14:30:15 EDT; 1min 49s ago Docs: man:systemd-sysv-generator(8) Process: 433 ExecStart=/etc/init.d/opensips start (code=exited, status=0/SUCCESS) May 19 14:30:15 debsips systemd[1]: Starting LSB: Start the OpenSIPS SIP server... May 19 14:30:15 debsips systemd[1]: Started LSB: Start the OpenSIPS SIP server. I tried several ways of installation and configuration but the opensips doesn't start properly, I am not able to register clients (even though I have successfully added client and domain from CP). also there is no log recorded in opensips.log. Please give me any advice on this matter. I want to see the status of active (running). Regards, Ali -------------- next part -------------- An HTML attachment was scrubbed... URL: From johan at democon.be Fri May 20 06:38:03 2022 From: johan at democon.be (Johan De Clercq) Date: Fri, 20 May 2022 06:38:03 +0000 Subject: [OpenSIPS-Users] Opensips active (exited) In-Reply-To: References: Message-ID: Check daemon path Outlook voor iOS downloaden ________________________________ Van: Users namens Ali Alawi Verzonden: Thursday, May 19, 2022 9:30:50 PM Aan: OpenSIPS users mailling list Onderwerp: [OpenSIPS-Users] Opensips active (exited) Dear all, I have installed opensips 3.2 using the git clone --recursive and compile it using make menuconfig, mysql (mariadb) and cli are all set correctly. when I start opensips the status shows no error but only active (exited) as shown below: systemctl status opensips ● opensips.service - LSB: Start the OpenSIPS SIP server Loaded: loaded (/etc/init.d/opensips; generated) Active: active (exited) since Thu 2022-05-19 14:30:15 EDT; 1min 49s ago Docs: man:systemd-sysv-generator(8) Process: 433 ExecStart=/etc/init.d/opensips start (code=exited, status=0/SUCCESS) May 19 14:30:15 debsips systemd[1]: Starting LSB: Start the OpenSIPS SIP server... May 19 14:30:15 debsips systemd[1]: Started LSB: Start the OpenSIPS SIP server. I tried several ways of installation and configuration but the opensips doesn't start properly, I am not able to register clients (even though I have successfully added client and domain from CP). also there is no log recorded in opensips.log. Please give me any advice on this matter. I want to see the status of active (running). Regards, Ali -------------- next part -------------- An HTML attachment was scrubbed... URL: From jehanzaib.kiani at gmail.com Sat May 21 02:21:38 2022 From: jehanzaib.kiani at gmail.com (Jehanzaib Younis) Date: Sat, 21 May 2022 14:21:38 +1200 Subject: [OpenSIPS-Users] no TLS client domain found error In-Reply-To: References: <8877c8b2-9ad3-a0d0-3294-5a3d20c748a1@opensips.org> <72b9b0b1-5520-0818-a5a3-af8ad451fed7@opensips.org> Message-ID: Thank you, Ovidiu. I Just posted my logs on github. Regards, Jehanzaib On Fri, May 20, 2022 at 3:02 AM Ovidiu Sas wrote: > Set the log_level parameter to 4 and restart opensips. Once the error > occurs, collect all the logs from the start (from syslog) and send them to > Razvan. > There’s bug tracking this issue: > https://github.com/OpenSIPS/opensips/issues/2724 > > For compiling tls_wolfssl, try from a clean clone. > > -ovidiu > > On Thu, May 19, 2022 at 08:08 Jehanzaib Younis > wrote: > >> Thanks Ovidiu, >> I just checked the source code, the same bug is also present in >> the opensips-3.2.6 branch. I have another issue with 3.2.6. I am not able >> to compile tls_wolfssl. No issue with 3.3 though. >> Now I need to check what is causing this. >> >> I am getting the following error: >> >> make[1]: Entering directory `/usr/src/opensips-3.2/modules/tls_wolfssl' >> configure: WARNING: unrecognized options: --disable-shared, >> --enable-static >> checking whether make supports nested variables... (cached) yes >> ./configure: line 5259: syntax error near unexpected token `2.4.2' >> ./configure: line 5259: `LT_PREREQ(2.4.2)' >> make[1]: *** [lib/lib/libwolfssl.a] Error 2 >> >> >> >> Regards, >> Jehanzaib >> >> >> On Thu, May 19, 2022 at 1:35 AM Ovidiu Sas wrote: >> >>> Please upgrade to the latest version and see if the error persists. If >>> yes, please run the server in debug mode and save the logs so this issue >>> can be investigated properly. >>> >>> Thanks, >>> Ovidiu >>> >>> On Wed, May 18, 2022 at 09:02 Jehanzaib Younis < >>> jehanzaib.kiani at gmail.com> wrote: >>> >>>> Thank you Bogdan, >>>> That helped a lot. As you mentioned I need to start only with >>>> server_domain or client_domain. >>>> Now I changed my config a bit as shown below: >>>> #### (WebRTC) Client >>>> modparam("tls_mgm", "server_domain", "sip.mywebphone.xx") >>>> modparam("tls_mgm", "certificate", >>>> "[sip.mywebphone.xx]/etc/letsencrypt/live/sip.mywebphone.xx/cert.pem") >>>> modparam("tls_mgm", "private_key", >>>> "[sip.mywebphone.xx]/etc/letsencrypt/live/sip.mywebphone.xx/privkey.pem") >>>> modparam("tls_mgm", "ca_list", >>>> "[sip.mywebphone.xx]/etc/letsencrypt/live/sip.mywebphone.xx/fullchain.pem") >>>> modparam("tls_mgm", "ca_dir", >>>> "[sip.mywebphone.xx]/etc/letsencrypt/live/sip.mywebphone.xx") >>>> modparam("tls_mgm", "tls_method", "[sip.mywebphone.xx]SSLv23") >>>> modparam("tls_mgm", "verify_cert", "[sip.mywebphone.xx]1") >>>> modparam("tls_mgm", "require_cert", "[sip.mywebphone.xx]1") >>>> >>>> ### This is for MS-Teams direct route >>>> modparam("tls_mgm", "client_domain", "dom1.formsteams.com") >>>> modparam("tls_mgm", "certificate", "[dom1.formsteams.com >>>> ]/etc/letsencrypt/live/dom1.formsteams.com/cert.pem") >>>> modparam("tls_mgm", "private_key", "[dom1.formsteams.com >>>> ]/etc/letsencrypt/live/dom1.formsteams.com/privkey.pem") >>>> modparam("tls_mgm", "ca_list", "[dom1.formsteams.com >>>> ]/etc/letsencrypt/live/dom1.formsteams.com/fullchain.pem") >>>> modparam("tls_mgm", "ca_dir", "[dom1.formsteams.com >>>> ]/etc/letsencrypt/live/dom1.formsteams.com") >>>> modparam("tls_mgm", "tls_method", "[dom1.formsteams.com]SSLv23") >>>> modparam("tls_mgm", "verify_cert", "[dom1.formsteams.com]1") >>>> modparam("tls_mgm", "require_cert", "[dom1.formsteams.com]1") >>>> modparam("tls_mgm", "client_sip_domain_avp", "tls_sip_dom") >>>> >>>> Looks like the initial handshake is fine when my server sends OPTIONS >>>> to MSTeams. There is a bug in the code according to the logs as shown below: >>>> >>>> opensips[10659]: CRITICAL:core:io_watch_add: #012>>> used fd map fd=142 >>>> is not present in fd_array >>>> (fd=142,type=19,flags=80000003,data=0x7f825805ceb8)#012#012It seems you >>>> have hit a programming bug.#012Please help us make OpenSIPS better by >>>> reporting it at https://github.com/OpenSIPS/opensips/issues >>>> opensips[10659]: CRITICAL:core:io_watch_add: [TCP_main] check failed >>>> after successful fd add (fd=141,type=19,data=0x7f825804fd98,flags=1) >>>> already=0 >>>> opensips[23993]: NOTICE:tls_wolfssl:verify_callback: depth = 1, verify >>>> success >>>> opensips[23993]: NOTICE:tls_wolfssl:verify_callback: depth = 0, verify >>>> success >>>> opensips[23993]: INFO:tls_wolfssl:_wolfssl_tls_async_connect: new TLS >>>> connection to 52.114.16.74:5061 established >>>> opensips[23993]: NOTICE:tls_wolfssl:verify_callback: depth = 1, verify >>>> success >>>> opensips[23993]: NOTICE:tls_wolfssl:verify_callback: depth = 0, verify >>>> success >>>> opensips[23995]: INFO:tls_wolfssl:_wolfssl_tls_async_connect: new TLS >>>> connection to 52.114.76.76:5061 established >>>> >>>> >>>> Regards, >>>> Jehanzaib >>>> >>>> >>>> On Wed, May 18, 2022 at 6:15 PM Bogdan-Andrei Iancu < >>>> bogdan at opensips.org> wrote: >>>> >>>>> Hi Jehanzaib, >>>>> >>>>> The sequence for the MST TLS domains is wrong. >>>>> >>>>> For each TLS domain block, you need to start only with a server_domain >>>>> or client_domain - of course, different names. And for each domain you need >>>>> you set the matching conditions. See >>>>> https://opensips.org/html/docs/modules/3.2.x/tls_mgm.html#domains-param >>>>> >>>>> Basically something like: >>>>> >>>>> modparam("tls_mgm", "server_domain", "formsteams_server") >>>>> modparam("tls_mgm", "match_ip_address", "[formsteams_server]....") >>>>> modparam("tls_mgm", "match_sip_domain", "[formsteams_server]....") >>>>> modparam("tls_mgm", "certificate", "[formsteams_server].....) >>>>> .... >>>>> >>>>> >>>>> modparam("tls_mgm", "client_domain", "formsteams_client") >>>>> modparam("tls_mgm", "match_ip_address", "[formsteams_client]....") >>>>> modparam("tls_mgm", "match_sip_domain", "[formsteams_client]....") >>>>> modparam("tls_mgm", "certificate", "[formsteams_client].....) >>>>> .... >>>>> >>>>> >>>>> Best regards, >>>>> >>>>> Bogdan-Andrei Iancu >>>>> >>>>> OpenSIPS Founder and Developer >>>>> https://www.opensips-solutions.com >>>>> OpenSIPS eBootcamp 23rd May - 3rd June 2022 >>>>> https://opensips.org/training/OpenSIPS_eBootcamp_2022/ >>>>> >>>>> On 5/18/22 2:38 AM, Jehanzaib Younis wrote: >>>>> >>>>> Hi Bogdan, >>>>> That's the problem, when I try to add the client_domain I get an >>>>> error. Actually, I have a working config for webrtc but now I am adding a >>>>> new domain for MS teams direct route. In fact, any other domain gives an >>>>> error. If I disable MS Teams domain, the opensips do not give an >>>>> error message and my webrtc client can connect without any issue. >>>>> >>>>> loadmodule "tls_mgm.so" >>>>> modparam("tls_mgm", "tls_library", "wolfssl") >>>>> >>>>> #### (WebRTC) Client >>>>> modparam("tls_mgm", "server_domain", "sip.mywebphone.xx") >>>>> modparam("tls_mgm", "certificate", >>>>> "[sip.mywebphone.xx]/etc/letsencrypt/live/sip.mywebphone.xx/cert.pem") >>>>> modparam("tls_mgm", "private_key", >>>>> "[sip.mywebphone.xx]/etc/letsencrypt/live/sip.mywebphone.xx/privkey.pem") >>>>> modparam("tls_mgm", "ca_list", >>>>> "[sip.mywebphone.xx]/etc/letsencrypt/live/sip.mywebphone.xx/fullchain.pem") >>>>> modparam("tls_mgm", "ca_dir", >>>>> "[sip.mywebphone.xx]/etc/letsencrypt/live/sip.mywebphone.xx") >>>>> modparam("tls_mgm", "tls_method", "[sip.mywebphone.xx]SSLv23") >>>>> modparam("tls_mgm", "verify_cert", "[sip.mywebphone.xx]1") >>>>> modparam("tls_mgm", "require_cert", "[sip.mywebphone.xx]1") >>>>> >>>>> ### This is for MS-Teams direct route >>>>> modparam("tls_mgm", "server_domain", "dom1.formsteams.com") >>>>> modparam("tls_mgm", "client_domain", "dom1.formsteams.com") >>>>> modparam("tls_mgm", "certificate", "[dom1.formsteams.com >>>>> ]/etc/letsencrypt/live/dom1.formsteams.com/cert.pem") >>>>> modparam("tls_mgm", "private_key", "[dom1.formsteams.com >>>>> ]/etc/letsencrypt/live/dom1.formsteams.com/privkey.pem") >>>>> modparam("tls_mgm", "ca_list", "[dom1.formsteams.com >>>>> ]/etc/letsencrypt/live/dom1.formsteams.com/fullchain.pem") >>>>> modparam("tls_mgm", "ca_dir", "[dom1.formsteams.com >>>>> ]/etc/letsencrypt/live/dom1.formsteams.com") >>>>> modparam("tls_mgm", "tls_method", "[dom1.formsteams.com]SSLv23") >>>>> modparam("tls_mgm", "verify_cert", "[dom1.formsteams.com]1") >>>>> modparam("tls_mgm", "require_cert", "[dom1.formsteams.com]1") >>>>> modparam("tls_mgm", "client_sip_domain_avp", "tls_sip_dom") >>>>> >>>>> When i enable the MS-Teams direct route domain i get the below error: >>>>> no certificate for tls domain ' dom1.formsteams.com ' defined >>>>> >>>>> >>>>> Regards, >>>>> Jehanzaib >>>>> >>>>> >>>>> On Wed, May 18, 2022 at 3:04 AM Bogdan-Andrei Iancu < >>>>> bogdan at opensips.org> wrote: >>>>> >>>>>> Hi Jehanzaib, >>>>>> >>>>>> What are the TLS client domains you have defined in your tls_mgm >>>>>> module ? >>>>>> >>>>>> Regards, >>>>>> >>>>>> Bogdan-Andrei Iancu >>>>>> >>>>>> OpenSIPS Founder and Developer >>>>>> https://www.opensips-solutions.com >>>>>> OpenSIPS eBootcamp 23rd May - 3rd June 2022 >>>>>> https://opensips.org/training/OpenSIPS_eBootcamp_2022/ >>>>>> >>>>>> On 5/17/22 4:32 PM, Jehanzaib Younis wrote: >>>>>> >>>>>> Hi, >>>>>> >>>>>> I am having trouble to send/receive OPTIONS to ms teams. >>>>>> Using the dispatcher module. The socket is defined as tls:*mysbcip* >>>>>> :5061 >>>>>> Looks like when my opensips (3.2.x) tries to send OPTIONS. it is >>>>>> giving me the following error >>>>>> >>>>>> ERROR:proto_tls:proto_tls_conn_init: no TLS client domain found >>>>>> ERROR:core:tcp_conn_create: failed to do proto 3 specific init for >>>>>> conn 0x7f00ef2a85a0 >>>>>> ERROR:core:tcp_async_connect: tcp_conn_create failed >>>>>> ERROR:proto_tls:proto_tls_send: async TCP connect failed >>>>>> ERROR:tm:msg_send: send() to 52.114.76.76:5061 for proto tls/3 failed >>>>>> ERROR:tm:t_uac: attempt to send to ' >>>>>> sip:sip3.pstnhub.microsoft.com:5061;transport:tls' failed >>>>>> >>>>>> I am setting the Contact as >>>>>> >>>>>> Looks like the client domain is used for outgoing TLS connection but >>>>>> no idea which domain i need to add here. The socket is my opensips ip >>>>>> address. >>>>>> >>>>>> Has anyone seen a similar kind of behaviour? >>>>>> >>>>>> Thank you. >>>>>> >>>>>> Regards, >>>>>> Jehanzaib >>>>>> >>>>>> _______________________________________________ >>>>>> 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 >>>> >>> -- >>> VoIP Embedded, Inc. >>> http://www.voipembedded.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 >> > -- > VoIP Embedded, Inc. > http://www.voipembedded.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 wyhc at hotmail.com Sun May 22 15:07:49 2022 From: wyhc at hotmail.com (Wang Wilson) Date: Sun, 22 May 2022 15:07:49 +0000 Subject: [OpenSIPS-Users] TLS Error Message-ID: Hello, I am sending this to follow the issue that was reported on Sep 17 13:13:06 EST 2020. My problem is that I get the same error message, but the path to /etc/opensips/tls/user/user-cert.pem is correct and it is not symlink file. I just start to explore the TLS method for us to support SIP service. What could be the reason for this? Thanks in advance. Regards Wilson ------------------------------------------------------------------------------------------ INFO:core:mod_init: initializing TCP-plain protocol May 22 22:32:45 wilson-VirtualBox /usr/local/opensips/sbin/opensips[7437]: INFO:tls_mgm:mod_init: initializing TLS management May 22 22:32:45 wilson-VirtualBox /usr/local/opensips/sbin/opensips[7437]: INFO:tls_mgm:mod_init: disabling compression due ZLIB problems May 22 22:32:45 wilson-VirtualBox /usr/local/opensips/sbin/opensips[7437]: INFO:tls_mgm:init_tls_dom: Processing TLS domain 'default' May 22 22:32:45 wilson-VirtualBox /usr/local/opensips/sbin/opensips[7437]: NOTICE:tls_mgm:init_tls_dom: No EC curve defined May 22 22:32:45 wilson-VirtualBox /usr/local/opensips/sbin/opensips[7437]: INFO:tls_mgm:get_ssl_ctx_verify_mode: client verification activated. Client certificates are NOT mandatory. May 22 22:32:45 wilson-VirtualBox /usr/local/opensips/sbin/opensips[7437]: NOTICE:tls_mgm:init_tls_dom: no CA dir for tls 'default' defined, using default '/etc/pki/CA/' May 22 22:32:45 wilson-VirtualBox /usr/local/opensips/sbin/opensips[7437]: NOTICE:tls_mgm:init_tls_dom: no crl for tls, using none May 22 22:32:45 wilson-VirtualBox /usr/local/opensips/sbin/opensips[7437]: ERROR:tls_mgm:tls_print_errstack: TLS errstack: error:140AB18E:SSL routines:SSL_CTX_use_certificate:ca md too weak May 22 22:32:45 wilson-VirtualBox /usr/local/opensips/sbin/opensips[7437]: ERROR:tls_mgm:load_certificate: unable to load certificate file '/etc/opensips/tls/user/user-cert.pem' May 22 22:32:45 wilson-VirtualBox /usr/local/opensips/sbin/opensips[7437]: ERROR:tls_mgm:init_tls_domains: Failed to init TLS domain 'default' May 22 22:32:45 wilson-VirtualBox /usr/local/opensips/sbin/opensips[7437]: ERROR:core:init_mod: failed to initialize module tls_mgm May 22 22:32:45 wilson-VirtualBox /usr/local/opensips/sbin/opensips[7437]: ERROR:core:main: error while initializing modules May 22 22:32:45 wilson-VirtualBox /usr/local/opensips/sbin/opensips[7437]: INFO:core:cleanup: cleanup May 22 22:32:45 wilson-VirtualBox /usr/local/opensips/sbin/opensips[7437]: NOTICE:core:main: Exiting.... -------------- next part -------------- An HTML attachment was scrubbed... URL: From ideanethelp at gmail.com Sun May 22 22:53:41 2022 From: ideanethelp at gmail.com (ideanet help) Date: Mon, 23 May 2022 10:53:41 +1200 Subject: [OpenSIPS-Users] TLS Error In-Reply-To: References: Message-ID: Hi Wang, Can you check the user rights of that directory? ls -lrth /etc/opensips/tls/user On Mon, May 23, 2022 at 3:10 AM Wang Wilson wrote: > Hello, > > I am sending this to follow the issue that was reported on *Sep 17 > 13:13:06 EST 2020.* > > > > My problem is that I get the same error message, but the path to > /etc/opensips/tls/user/user-cert.pem is correct and it is not symlink file. > > > > I just start to explore the TLS method for us to support SIP service. What > could be the reason for this? > > > > Thanks in advance. > > > > Regards > > Wilson > > > ------------------------------------------------------------------------------------------ > > INFO:core:mod_init: initializing TCP-plain protocol > > May 22 22:32:45 wilson-VirtualBox /usr/local/opensips/sbin/opensips[7437]: > INFO:tls_mgm:mod_init: initializing TLS management > > May 22 22:32:45 wilson-VirtualBox /usr/local/opensips/sbin/opensips[7437]: > INFO:tls_mgm:mod_init: disabling compression due ZLIB problems > > May 22 22:32:45 wilson-VirtualBox /usr/local/opensips/sbin/opensips[7437]: > INFO:tls_mgm:init_tls_dom: Processing TLS domain 'default' > > May 22 22:32:45 wilson-VirtualBox /usr/local/opensips/sbin/opensips[7437]: > NOTICE:tls_mgm:init_tls_dom: No EC curve defined > > May 22 22:32:45 wilson-VirtualBox /usr/local/opensips/sbin/opensips[7437]: > INFO:tls_mgm:get_ssl_ctx_verify_mode: client verification activated. Client > certificates are NOT mandatory. > > May 22 22:32:45 wilson-VirtualBox /usr/local/opensips/sbin/opensips[7437]: > NOTICE:tls_mgm:init_tls_dom: no CA dir for tls 'default' defined, using > default '/etc/pki/CA/' > > May 22 22:32:45 wilson-VirtualBox /usr/local/opensips/sbin/opensips[7437]: > NOTICE:tls_mgm:init_tls_dom: no crl for tls, using none > > May 22 22:32:45 wilson-VirtualBox /usr/local/opensips/sbin/opensips[7437]: > ERROR:tls_mgm:tls_print_errstack: TLS errstack: error:140AB18E:SSL > routines:SSL_CTX_use_certificate:ca md too weak > > May 22 22:32:45 wilson-VirtualBox /usr/local/opensips/sbin/opensips[7437]: > ERROR:tls_mgm:load_certificate: unable to load certificate file > '/etc/opensips/tls/user/user-cert.pem' > > May 22 22:32:45 wilson-VirtualBox /usr/local/opensips/sbin/opensips[7437]: > ERROR:tls_mgm:init_tls_domains: Failed to init TLS domain 'default' > > May 22 22:32:45 wilson-VirtualBox /usr/local/opensips/sbin/opensips[7437]: > ERROR:core:init_mod: failed to initialize module tls_mgm > > May 22 22:32:45 wilson-VirtualBox /usr/local/opensips/sbin/opensips[7437]: > ERROR:core:main: error while initializing modules > > May 22 22:32:45 wilson-VirtualBox /usr/local/opensips/sbin/opensips[7437]: > INFO:core:cleanup: cleanup > > May 22 22:32:45 wilson-VirtualBox /usr/local/opensips/sbin/opensips[7437]: > NOTICE:core:main: Exiting.... > > > > > _______________________________________________ > 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 wyhc at hotmail.com Mon May 23 02:40:48 2022 From: wyhc at hotmail.com (Wang Wilson) Date: Mon, 23 May 2022 02:40:48 +0000 Subject: [OpenSIPS-Users] TLS Error In-Reply-To: References: Message-ID: This is my folder user rights status, and I am running Opensips3.1 under root user privilege. root at wilson-VirtualBox:/etc/opensips/tls/user# ls -lrth /etc/opensips/tls/user total 20K -rw------- 1 root root 1.7K 5月 23 10:34 user-privkey.pem -rw-r--r-- 1 root root 1.1K 5月 23 10:34 user-cert_req.pem -rw-r--r-- 1 root root 4.2K 5月 23 10:34 user-cert.pem -rw-r--r-- 1 root root 1.3K 5月 23 10:34 user-calist.pem root at wilson-VirtualBox:/etc/opensips/tls/user# Can you tell if there is anything need to pay attention? Regards Wilson ________________________________ From: Users on behalf of ideanet help Sent: Monday, May 23, 2022 6:53:41 AM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] TLS Error Hi Wang, Can you check the user rights of that directory? ls -lrth /etc/opensips/tls/user On Mon, May 23, 2022 at 3:10 AM Wang Wilson > wrote: Hello, I am sending this to follow the issue that was reported on Sep 17 13:13:06 EST 2020. My problem is that I get the same error message, but the path to /etc/opensips/tls/user/user-cert.pem is correct and it is not symlink file. I just start to explore the TLS method for us to support SIP service. What could be the reason for this? Thanks in advance. Regards Wilson ------------------------------------------------------------------------------------------ INFO:core:mod_init: initializing TCP-plain protocol May 22 22:32:45 wilson-VirtualBox /usr/local/opensips/sbin/opensips[7437]: INFO:tls_mgm:mod_init: initializing TLS management May 22 22:32:45 wilson-VirtualBox /usr/local/opensips/sbin/opensips[7437]: INFO:tls_mgm:mod_init: disabling compression due ZLIB problems May 22 22:32:45 wilson-VirtualBox /usr/local/opensips/sbin/opensips[7437]: INFO:tls_mgm:init_tls_dom: Processing TLS domain 'default' May 22 22:32:45 wilson-VirtualBox /usr/local/opensips/sbin/opensips[7437]: NOTICE:tls_mgm:init_tls_dom: No EC curve defined May 22 22:32:45 wilson-VirtualBox /usr/local/opensips/sbin/opensips[7437]: INFO:tls_mgm:get_ssl_ctx_verify_mode: client verification activated. Client certificates are NOT mandatory. May 22 22:32:45 wilson-VirtualBox /usr/local/opensips/sbin/opensips[7437]: NOTICE:tls_mgm:init_tls_dom: no CA dir for tls 'default' defined, using default '/etc/pki/CA/' May 22 22:32:45 wilson-VirtualBox /usr/local/opensips/sbin/opensips[7437]: NOTICE:tls_mgm:init_tls_dom: no crl for tls, using none May 22 22:32:45 wilson-VirtualBox /usr/local/opensips/sbin/opensips[7437]: ERROR:tls_mgm:tls_print_errstack: TLS errstack: error:140AB18E:SSL routines:SSL_CTX_use_certificate:ca md too weak May 22 22:32:45 wilson-VirtualBox /usr/local/opensips/sbin/opensips[7437]: ERROR:tls_mgm:load_certificate: unable to load certificate file '/etc/opensips/tls/user/user-cert.pem' May 22 22:32:45 wilson-VirtualBox /usr/local/opensips/sbin/opensips[7437]: ERROR:tls_mgm:init_tls_domains: Failed to init TLS domain 'default' May 22 22:32:45 wilson-VirtualBox /usr/local/opensips/sbin/opensips[7437]: ERROR:core:init_mod: failed to initialize module tls_mgm May 22 22:32:45 wilson-VirtualBox /usr/local/opensips/sbin/opensips[7437]: ERROR:core:main: error while initializing modules May 22 22:32:45 wilson-VirtualBox /usr/local/opensips/sbin/opensips[7437]: INFO:core:cleanup: cleanup May 22 22:32:45 wilson-VirtualBox /usr/local/opensips/sbin/opensips[7437]: NOTICE:core:main: Exiting.... _______________________________________________ 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 goatolina at gmail.com Tue May 24 20:25:52 2022 From: goatolina at gmail.com (Ali Alawi) Date: Tue, 24 May 2022 23:25:52 +0300 Subject: [OpenSIPS-Users] Opensips active (exited) In-Reply-To: References: Message-ID: Dear Johan, Thanks for your reply. It makes sense. On Fri, May 20, 2022 at 9:41 AM Johan De Clercq wrote: > Check daemon path > > Outlook voor iOS downloaden > ------------------------------ > *Van:* Users namens Ali Alawi < > goatolina at gmail.com> > *Verzonden:* Thursday, May 19, 2022 9:30:50 PM > *Aan:* OpenSIPS users mailling list > *Onderwerp:* [OpenSIPS-Users] Opensips active (exited) > > Dear all, > > I have installed opensips 3.2 using the git clone --recursive and compile > it using make menuconfig, mysql (mariadb) and cli are all set correctly. > > when I start opensips the status shows no error but only active (exited) > as shown below: > > systemctl status opensips > ● opensips.service - LSB: Start the OpenSIPS SIP server > Loaded: loaded (/etc/init.d/opensips; generated) > Active: active (exited) since Thu 2022-05-19 14:30:15 EDT; 1min 49s ago > Docs: man:systemd-sysv-generator(8) > Process: 433 ExecStart=/etc/init.d/opensips start (code=exited, > status=0/SUCCESS) > > May 19 14:30:15 debsips systemd[1]: Starting LSB: Start the OpenSIPS SIP > server... > May 19 14:30:15 debsips systemd[1]: Started LSB: Start the OpenSIPS SIP > server. > > > I tried several ways of installation and configuration but the opensips > doesn't start properly, I am not able to register clients (even though I > have successfully added client and domain from CP). also there is no log > recorded in opensips.log. > > Please give me any advice on this matter. I want to see the status of > active (running). > > Regards, > Ali > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Wed May 25 07:50:20 2022 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Wed, 25 May 2022 10:50:20 +0300 Subject: [OpenSIPS-Users] TLS Error In-Reply-To: References: Message-ID: Hi Wang, A quick googling shows that the problem is with your certificate, being md5 signed - and this is considered a week signature. Check this https://stackoverflow.com/questions/52218876/how-to-fix-ssl-issue-ssl-ctx-use-certificate-ca-md-too-weak-on-python-zeep Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com OpenSIPS Summit 27-30 Sept 2022, Athens https://www.opensips.org/events/Summit-2022Athens/ On 5/23/22 5:40 AM, Wang Wilson wrote: > > This is my folder user rights status, and I am running Opensips3.1 > under root userprivilege. > > root at wilson-VirtualBox:/etc/opensips/tls/user# ls -lrth > /etc/opensips/tls/user > > total 20K > > -rw------- 1 root root 1.7K 5月  23 10:34 user-privkey.pem > > -rw-r--r-- 1 root root 1.1K 5月  23 10:34 user-cert_req.pem > > -rw-r--r-- 1 root root 4.2K 5月  23 10:34 user-cert.pem > > -rw-r--r-- 1 root root 1.3K 5月  23 10:34 user-calist.pem > > root at wilson-VirtualBox:/etc/opensips/tls/user# > > Can you tell if there is anything need to pay attention? > > Regards > > Wilson > > ------------------------------------------------------------------------ > *From:* Users on behalf of ideanet > help > *Sent:* Monday, May 23, 2022 6:53:41 AM > *To:* OpenSIPS users mailling list > *Subject:* Re: [OpenSIPS-Users] TLS Error > Hi Wang, > Can you check the user rights of that directory? ls -lrth > /etc/opensips/tls/user > > > On Mon, May 23, 2022 at 3:10 AM Wang Wilson > wrote: > > Hello, > > I am sending this to follow the issue that was reported on /Sep 17 > 13:13:06 EST 2020./ > > My problem is that I get the same error message, but the path to > /etc/opensips/tls/user/user-cert.pem is correct and it is not > symlink file. > > I just start to explore the TLS method for us to support SIP > service. What could be the reason for this? > > Thanks in advance. > > Regards > > Wilson > > ------------------------------------------------------------------------------------------ > > INFO:core:mod_init: initializing TCP-plain protocol > > May 22 22:32:45 wilson-VirtualBox > /usr/local/opensips/sbin/opensips[7437]: INFO:tls_mgm:mod_init: > initializing TLS management > > May 22 22:32:45 wilson-VirtualBox > /usr/local/opensips/sbin/opensips[7437]: INFO:tls_mgm:mod_init: > disabling compression due ZLIB problems > > May 22 22:32:45 wilson-VirtualBox > /usr/local/opensips/sbin/opensips[7437]: > INFO:tls_mgm:init_tls_dom: Processing TLS domain 'default' > > May 22 22:32:45 wilson-VirtualBox > /usr/local/opensips/sbin/opensips[7437]: > NOTICE:tls_mgm:init_tls_dom: No EC curve defined > > May 22 22:32:45 wilson-VirtualBox > /usr/local/opensips/sbin/opensips[7437]: > INFO:tls_mgm:get_ssl_ctx_verify_mode: client verification > activated. Client certificates are NOT mandatory. > > May 22 22:32:45 wilson-VirtualBox > /usr/local/opensips/sbin/opensips[7437]: > NOTICE:tls_mgm:init_tls_dom: no CA dir for tls 'default' defined, > using default '/etc/pki/CA/' > > May 22 22:32:45 wilson-VirtualBox > /usr/local/opensips/sbin/opensips[7437]: > NOTICE:tls_mgm:init_tls_dom: no crl for tls, using none > > May 22 22:32:45 wilson-VirtualBox > /usr/local/opensips/sbin/opensips[7437]: > ERROR:tls_mgm:tls_print_errstack: TLS errstack: error:140AB18E:SSL > routines:SSL_CTX_use_certificate:ca md too weak > > May 22 22:32:45 wilson-VirtualBox > /usr/local/opensips/sbin/opensips[7437]: > ERROR:tls_mgm:load_certificate: unable to load certificate file > '/etc/opensips/tls/user/user-cert.pem' > > May 22 22:32:45 wilson-VirtualBox > /usr/local/opensips/sbin/opensips[7437]: > ERROR:tls_mgm:init_tls_domains: Failed to init TLS domain 'default' > > May 22 22:32:45 wilson-VirtualBox > /usr/local/opensips/sbin/opensips[7437]: ERROR:core:init_mod: > failed to initialize module tls_mgm > > May 22 22:32:45 wilson-VirtualBox > /usr/local/opensips/sbin/opensips[7437]: ERROR:core:main: error > while initializing modules > > May 22 22:32:45 wilson-VirtualBox > /usr/local/opensips/sbin/opensips[7437]: INFO:core:cleanup: cleanup > > May 22 22:32:45 wilson-VirtualBox > /usr/local/opensips/sbin/opensips[7437]: NOTICE:core:main: Exiting.... > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Wed May 25 07:52:23 2022 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Wed, 25 May 2022 10:52:23 +0300 Subject: [OpenSIPS-Users] no TLS client domain found error In-Reply-To: References: <8877c8b2-9ad3-a0d0-3294-5a3d20c748a1@opensips.org> <72b9b0b1-5520-0818-a5a3-af8ad451fed7@opensips.org> Message-ID: Hi Jehanzaib, For now, to get rid of that issue, just disable the tls_async in your cfg: https://opensips.org/html/docs/modules/3.2.x/proto_tls.html#param_tls_async Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com OpenSIPS Summit 27-30 Sept 2022, Athens https://www.opensips.org/events/Summit-2022Athens/ On 5/21/22 5:21 AM, Jehanzaib Younis wrote: > Thank you, Ovidiu. > I Just posted my logs on github. > > Regards, > Jehanzaib > > > On Fri, May 20, 2022 at 3:02 AM Ovidiu Sas > wrote: > > Set the log_level parameter to 4 and restart opensips. Once the > error occurs, collect all the logs from the start (from syslog) > and send them to Razvan. > There’s bug tracking this issue: > https://github.com/OpenSIPS/opensips/issues/2724 > > > For compiling tls_wolfssl, try from a clean clone. > > -ovidiu > > On Thu, May 19, 2022 at 08:08 Jehanzaib Younis > > wrote: > > Thanks Ovidiu, > I just checked the source code, the same bug is also present > in the opensips-3.2.6 branch. I have another issue with 3.2.6. > I am not able to compile tls_wolfssl. No issue with 3.3 though. > Now I need to check what is causing this. > I am getting the following error: > > make[1]: Entering directory > `/usr/src/opensips-3.2/modules/tls_wolfssl' > configure: WARNING: unrecognized options: --disable-shared, > --enable-static > checking whether make supports nested variables... (cached) yes > ./configure: line 5259: syntax error near unexpected token `2.4.2' > ./configure: line 5259: `LT_PREREQ(2.4.2)' > make[1]: *** [lib/lib/libwolfssl.a] Error 2 > > > > Regards, > Jehanzaib > > > On Thu, May 19, 2022 at 1:35 AM Ovidiu Sas > > wrote: > > Please upgrade to the latest version and see if the error > persists. If yes, please run the server in debug mode and > save the logs so this issue can be investigated properly. > > Thanks, > Ovidiu > > On Wed, May 18, 2022 at 09:02 Jehanzaib Younis > > wrote: > > Thank you Bogdan, > That helped a lot. As you mentioned I need to start > only with server_domain or client_domain. > Now I changed my config a bit as shown below: > #### (WebRTC) Client > modparam("tls_mgm", "server_domain", "sip.mywebphone.xx") > modparam("tls_mgm", "certificate", > "[sip.mywebphone.xx]/etc/letsencrypt/live/sip.mywebphone.xx/cert.pem") > modparam("tls_mgm", "private_key", > "[sip.mywebphone.xx]/etc/letsencrypt/live/sip.mywebphone.xx/privkey.pem") > modparam("tls_mgm", "ca_list", > "[sip.mywebphone.xx]/etc/letsencrypt/live/sip.mywebphone.xx/fullchain.pem") > modparam("tls_mgm", "ca_dir", > "[sip.mywebphone.xx]/etc/letsencrypt/live/sip.mywebphone.xx") > modparam("tls_mgm", "tls_method", > "[sip.mywebphone.xx]SSLv23") > modparam("tls_mgm", "verify_cert", "[sip.mywebphone.xx]1") > modparam("tls_mgm", "require_cert", > "[sip.mywebphone.xx]1") > > ### This is for MS-Teams direct route > modparam("tls_mgm", "client_domain", > "dom1.formsteams.com ") > modparam("tls_mgm", "certificate", > "[dom1.formsteams.com > ]/etc/letsencrypt/live/dom1.formsteams.com/cert.pem > ") > modparam("tls_mgm", "private_key", > "[dom1.formsteams.com > ]/etc/letsencrypt/live/dom1.formsteams.com/privkey.pem > ") > modparam("tls_mgm", "ca_list", "[dom1.formsteams.com > ]/etc/letsencrypt/live/dom1.formsteams.com/fullchain.pem > ") > modparam("tls_mgm", "ca_dir", "[dom1.formsteams.com > ]/etc/letsencrypt/live/dom1.formsteams.com > ") > modparam("tls_mgm", "tls_method", > "[dom1.formsteams.com > ]SSLv23") > modparam("tls_mgm", "verify_cert", > "[dom1.formsteams.com ]1") > modparam("tls_mgm", "require_cert", > "[dom1.formsteams.com ]1") > modparam("tls_mgm", "client_sip_domain_avp", > "tls_sip_dom") > > Looks like the initial handshake is fine when my > server sends OPTIONS to MSTeams. There is a bug in the > code according to the logs as shown below: > > opensips[10659]: CRITICAL:core:io_watch_add: #012>>> > used fd map fd=142 is not present in fd_array > (fd=142,type=19,flags=80000003,data=0x7f825805ceb8)#012#012It > seems you have hit a programming bug.#012Please help > us make OpenSIPS better by reporting it at > https://github.com/OpenSIPS/opensips/issues > > opensips[10659]: CRITICAL:core:io_watch_add: > [TCP_main] check failed after successful fd add > (fd=141,type=19,data=0x7f825804fd98,flags=1) already=0 > opensips[23993]: NOTICE:tls_wolfssl:verify_callback: > depth = 1, verify success > opensips[23993]: NOTICE:tls_wolfssl:verify_callback: > depth = 0, verify success > opensips[23993]: > INFO:tls_wolfssl:_wolfssl_tls_async_connect: new TLS > connection to 52.114.16.74:5061 > established > opensips[23993]: NOTICE:tls_wolfssl:verify_callback: > depth = 1, verify success > opensips[23993]: NOTICE:tls_wolfssl:verify_callback: > depth = 0, verify success > opensips[23995]: > INFO:tls_wolfssl:_wolfssl_tls_async_connect: new TLS > connection to 52.114.76.76:5061 > established > > > Regards, > Jehanzaib > > > On Wed, May 18, 2022 at 6:15 PM Bogdan-Andrei Iancu > > wrote: > > Hi Jehanzaib, > > The sequence for the MST TLS domains is wrong. > > For each TLS domain block, you need to start only > with a server_domain or client_domain - of course, > different names. And for each domain you need you > set the matching conditions. See > https://opensips.org/html/docs/modules/3.2.x/tls_mgm.html#domains-param > > > Basically something like: > > modparam("tls_mgm", "server_domain", > "formsteams_server") > modparam("tls_mgm", "match_ip_address", > "[formsteams_server]....") > modparam("tls_mgm", "match_sip_domain", > "[formsteams_server]....") > modparam("tls_mgm", "certificate", > "[formsteams_server].....) > .... > > > modparam("tls_mgm", "client_domain", > "formsteams_client") > modparam("tls_mgm", "match_ip_address", > "[formsteams_client]....") > modparam("tls_mgm", "match_sip_domain", > "[formsteams_client]....") > modparam("tls_mgm", "certificate", > "[formsteams_client].....) > .... > > > Best regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > OpenSIPS eBootcamp 23rd May - 3rd June 2022 > https://opensips.org/training/OpenSIPS_eBootcamp_2022/ > > On 5/18/22 2:38 AM, Jehanzaib Younis wrote: >> Hi Bogdan, >> That's the problem, when I try to add the >> client_domain I get an error. Actually, I have a >> working config for webrtc but now I am adding a >> new domain for MS teams direct route. In fact, >> any other domain gives an error. If I disable MS >> Teams domain, the opensips do not give an >> error message and my webrtc client can connect >> without any issue. >> >> loadmodule "tls_mgm.so" >> modparam("tls_mgm", "tls_library", "wolfssl") >> >> #### (WebRTC) Client >> modparam("tls_mgm", "server_domain", >> "sip.mywebphone.xx") >> modparam("tls_mgm", "certificate", >> "[sip.mywebphone.xx]/etc/letsencrypt/live/sip.mywebphone.xx/cert.pem") >> modparam("tls_mgm", "private_key", >> "[sip.mywebphone.xx]/etc/letsencrypt/live/sip.mywebphone.xx/privkey.pem") >> modparam("tls_mgm", "ca_list", >> "[sip.mywebphone.xx]/etc/letsencrypt/live/sip.mywebphone.xx/fullchain.pem") >> modparam("tls_mgm", "ca_dir", >> "[sip.mywebphone.xx]/etc/letsencrypt/live/sip.mywebphone.xx") >> modparam("tls_mgm", "tls_method", >> "[sip.mywebphone.xx]SSLv23") >> modparam("tls_mgm", "verify_cert", >> "[sip.mywebphone.xx]1") >> modparam("tls_mgm", "require_cert", >> "[sip.mywebphone.xx]1") >> >> ### This is for MS-Teams direct route >> modparam("tls_mgm", "server_domain", >> "dom1.formsteams.com ") >> modparam("tls_mgm", "client_domain", >> "dom1.formsteams.com ") >> modparam("tls_mgm", "certificate", >> "[dom1.formsteams.com >> ]/etc/letsencrypt/live/dom1.formsteams.com/cert.pem >> ") >> modparam("tls_mgm", "private_key", >> "[dom1.formsteams.com >> ]/etc/letsencrypt/live/dom1.formsteams.com/privkey.pem >> ") >> modparam("tls_mgm", "ca_list", >> "[dom1.formsteams.com >> ]/etc/letsencrypt/live/dom1.formsteams.com/fullchain.pem >> ") >> modparam("tls_mgm", "ca_dir", >> "[dom1.formsteams.com >> ]/etc/letsencrypt/live/dom1.formsteams.com >> ") >> modparam("tls_mgm", "tls_method", >> "[dom1.formsteams.com >> ]SSLv23") >> modparam("tls_mgm", "verify_cert", >> "[dom1.formsteams.com >> ]1") >> modparam("tls_mgm", "require_cert", >> "[dom1.formsteams.com >> ]1") >> modparam("tls_mgm", "client_sip_domain_avp", >> "tls_sip_dom") >> >> When i enable the MS-Teams direct route domain i >> get the below error: >> no certificate for tls domain ' >> dom1.formsteams.com >>  ' defined >> >> >> Regards, >> Jehanzaib >> >> >> On Wed, May 18, 2022 at 3:04 AM Bogdan-Andrei >> Iancu > > wrote: >> >> Hi Jehanzaib, >> >> What are the TLS client domains you have >> defined in your tls_mgm module ? >> >> Regards, >> >> Bogdan-Andrei Iancu >> >> OpenSIPS Founder and Developer >> https://www.opensips-solutions.com >> OpenSIPS eBootcamp 23rd May - 3rd June 2022 >> https://opensips.org/training/OpenSIPS_eBootcamp_2022/ >> >> On 5/17/22 4:32 PM, Jehanzaib Younis wrote: >>> Hi, >>> >>> I am having trouble to send/receive OPTIONS >>> to ms teams. >>> Using the dispatcher module. The socket is >>> defined as tls:*mysbcip*:5061 >>> Looks like when my opensips (3.2.x) tries to >>> send OPTIONS. it is giving me the following >>> error >>> * >>> * >>> ERROR:proto_tls:proto_tls_conn_init: no TLS >>> client domain found >>> ERROR:core:tcp_conn_create: failed to do >>> proto 3 specific init for conn 0x7f00ef2a85a0 >>> ERROR:core:tcp_async_connect: >>> tcp_conn_create failed >>> ERROR:proto_tls:proto_tls_send: async TCP >>> connect failed >>> ERROR:tm:msg_send: send() to >>> 52.114.76.76:5061 >>> for proto tls/3 failed >>> ERROR:tm:t_uac: attempt to send to >>> 'sip:sip3.pstnhub.microsoft.com:5061;transport:tls' >>> failed >>> >>> I am setting the Contact as >>> >>> >>> Looks like the client domain is used for >>> outgoing TLS connection but no idea which >>> domain i need to add here. The socket is my >>> opensips ip address. >>> >>> Has anyone seen a similar kind of behaviour? >>> >>> Thank you. >>> >>> Regards, >>> Jehanzaib >>> >>> _______________________________________________ >>> 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 > > > -- > VoIP Embedded, Inc. > http://www.voipembedded.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 > > > -- > VoIP Embedded, Inc. > http://www.voipembedded.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 -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Wed May 25 07:58:40 2022 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Wed, 25 May 2022 10:58:40 +0300 Subject: [OpenSIPS-Users] SQL Cacher+Galera Cluster In-Reply-To: References: Message-ID: <996a63e2-38fa-29c2-804d-1f827157f053@opensips.org> Hi Mehdi, Just rely on the auto-reloading https://opensips.org/html/docs/modules/3.2.x/sql_cacher.html#param_reload_interval Best regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com OpenSIPS Summit 27-30 Sept 2022, Athens https://www.opensips.org/events/Summit-2022Athens/ On 5/18/22 9:32 PM, Mehdi Shirazi wrote: > Hi > I plan to use SQL Cacher with Galera Cluster. After a record changes I > want to update the cache. Using standard triggers in mariadb is not > possible to run opensips-cli commands. > Please tell me your suggestions for the best approach. > > Regards > M.Shirazi > > _______________________________________________ > 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 wyhc at hotmail.com Thu May 26 05:34:44 2022 From: wyhc at hotmail.com (Wang Wilson) Date: Thu, 26 May 2022 05:34:44 +0000 Subject: [OpenSIPS-Users] TLS Error Message-ID: Dear Bogdan-Andrei Iancu, Thank you for the reply. In fact I re-do the CA generation by following the Opensips TLS setting document (https://opensips.org/html/docs/tutorials/tls-1.4.x). From the request.conf I confirm that “default_md” is set to “sha1”. After I recopy the tls folder to the location /etc/opensips/tls and restart opensips service, it still shows the error message. As for the log message, I like to check with you, if the previous three tls_mgm notice which tell some strange message that create such problem? Regards Wilson Wang May 26 11:49:23 wilson-VirtualBox /usr/local/opensips/sbin/opensips[5103]: NOTICE:tls_mgm:init_tls_dom: No EC curve defined May 26 11:49:23 wilson-VirtualBox /usr/local/opensips/sbin/opensips[5103]: INFO:tls_mgm:get_ssl_ctx_verify_mode: client verification activated. Client certificates are NOT mandatory. May 26 11:49:23 wilson-VirtualBox /usr/local/opensips/sbin/opensips[5103]: NOTICE:tls_mgm:init_tls_dom: no CA dir for tls 'default' defined, using default '/etc/pki/CA/' May 26 11:49:23 wilson-VirtualBox /usr/local/opensips/sbin/opensips[5103]: NOTICE:tls_mgm:init_tls_dom: no crl for tls, using none May 26 11:49:23 wilson-VirtualBox /usr/local/opensips/sbin/opensips[5103]: ERROR:tls_mgm:tls_print_errstack: TLS errstack: error:140AB18E:SSL routines:SSL_CTX_use_certificate:ca md too weak May 26 11:49:23 wilson-VirtualBox /usr/local/opensips/sbin/opensips[5103]: ERROR:tls_mgm:load_certificate: unable to load certificate file '/etc/opensips/tls/user/user-cert.pem' May 26 11:49:23 wilson-VirtualBox /usr/local/opensips/sbin/opensips[5103]: ERROR:tls_mgm:init_tls_domains: Failed to init TLS domain 'default' May 26 11:49:23 wilson-VirtualBox /usr/local/opensips/sbin/opensips[5103]: ERROR:core:init_mod: failed to initialize module tls_mgm -------------- next part -------------- An HTML attachment was scrubbed... URL: From bullehs at gmail.com Thu May 26 08:04:09 2022 From: bullehs at gmail.com (HS) Date: Thu, 26 May 2022 13:04:09 +0500 Subject: [OpenSIPS-Users] No audio on TLS connection - Opensips 3.1 Message-ID: Hi all. Just testing out Opensips 3.1 and run into a confusing issue. Users can connect and make calls successfully on UDP and TCP. However, some users are facing a problem with TLS - they connect fine, but don't hear any audio. Looking at the logs it seems that a private IP is registered. We have tried nat_uac_test 23 and 119 (REGISTER) and 1 and 2 (ONReply). We are using the standard residential script. Any thoughts on what else to test? Thanks for the help. -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Mon May 30 09:35:43 2022 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Mon, 30 May 2022 12:35:43 +0300 Subject: [OpenSIPS-Users] [Blog] Message-ID: Hi all, As OpenSIPS 3.3 focused on messaging, mainly on MSRP / session based messaging, here is a detailed walk thru the entire MSRP stack in OpenSIPS 3.3 and its capabilities, from relaying, to APIs and gatewaying https://blog.opensips.org/2022/05/30/the-msrp-eco-system-in-opensips-3-3/ Enjoy, -- Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com OpenSIPS Summit 27-30 Sept 2022, Athens https://www.opensips.org/events/Summit-2022Athens/ From jehanzaib.kiani at gmail.com Wed May 25 10:34:37 2022 From: jehanzaib.kiani at gmail.com (Jehanzaib Younis) Date: Wed, 25 May 2022 10:34:37 -0000 Subject: [OpenSIPS-Users] no TLS client domain found error In-Reply-To: References: <8877c8b2-9ad3-a0d0-3294-5a3d20c748a1@opensips.org> <72b9b0b1-5520-0818-a5a3-af8ad451fed7@opensips.org> Message-ID: Thank you Bogdan, I will change it. By the way, I noticed that as soon as I added the following before the OPTIONS were sent to MS Teams. There were no CRITICAL:core:io_watch_add bug logs. set_advertised_address("xx.xx.xx.xx"); set_advertised_port("5061"); Regards, Jehanzaib On Wed, May 25, 2022 at 7:52 PM Bogdan-Andrei Iancu wrote: > Hi Jehanzaib, > > For now, to get rid of that issue, just disable the tls_async in your cfg: > > https://opensips.org/html/docs/modules/3.2.x/proto_tls.html#param_tls_async > > Regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > OpenSIPS Summit 27-30 Sept 2022, Athens > https://www.opensips.org/events/Summit-2022Athens/ > > On 5/21/22 5:21 AM, Jehanzaib Younis wrote: > > Thank you, Ovidiu. > I Just posted my logs on github. > > Regards, > Jehanzaib > > > On Fri, May 20, 2022 at 3:02 AM Ovidiu Sas wrote: > >> Set the log_level parameter to 4 and restart opensips. Once the error >> occurs, collect all the logs from the start (from syslog) and send them to >> Razvan. >> There’s bug tracking this issue: >> https://github.com/OpenSIPS/opensips/issues/2724 >> >> For compiling tls_wolfssl, try from a clean clone. >> >> -ovidiu >> >> On Thu, May 19, 2022 at 08:08 Jehanzaib Younis >> wrote: >> >>> Thanks Ovidiu, >>> I just checked the source code, the same bug is also present in >>> the opensips-3.2.6 branch. I have another issue with 3.2.6. I am not able >>> to compile tls_wolfssl. No issue with 3.3 though. >>> Now I need to check what is causing this. >>> >>> I am getting the following error: >>> >>> make[1]: Entering directory `/usr/src/opensips-3.2/modules/tls_wolfssl' >>> configure: WARNING: unrecognized options: --disable-shared, >>> --enable-static >>> checking whether make supports nested variables... (cached) yes >>> ./configure: line 5259: syntax error near unexpected token `2.4.2' >>> ./configure: line 5259: `LT_PREREQ(2.4.2)' >>> make[1]: *** [lib/lib/libwolfssl.a] Error 2 >>> >>> >>> >>> Regards, >>> Jehanzaib >>> >>> >>> On Thu, May 19, 2022 at 1:35 AM Ovidiu Sas >>> wrote: >>> >>>> Please upgrade to the latest version and see if the error persists. If >>>> yes, please run the server in debug mode and save the logs so this issue >>>> can be investigated properly. >>>> >>>> Thanks, >>>> Ovidiu >>>> >>>> On Wed, May 18, 2022 at 09:02 Jehanzaib Younis < >>>> jehanzaib.kiani at gmail.com> wrote: >>>> >>>>> Thank you Bogdan, >>>>> That helped a lot. As you mentioned I need to start only with >>>>> server_domain or client_domain. >>>>> Now I changed my config a bit as shown below: >>>>> #### (WebRTC) Client >>>>> modparam("tls_mgm", "server_domain", "sip.mywebphone.xx") >>>>> modparam("tls_mgm", "certificate", >>>>> "[sip.mywebphone.xx]/etc/letsencrypt/live/sip.mywebphone.xx/cert.pem") >>>>> modparam("tls_mgm", "private_key", >>>>> "[sip.mywebphone.xx]/etc/letsencrypt/live/sip.mywebphone.xx/privkey.pem") >>>>> modparam("tls_mgm", "ca_list", >>>>> "[sip.mywebphone.xx]/etc/letsencrypt/live/sip.mywebphone.xx/fullchain.pem") >>>>> modparam("tls_mgm", "ca_dir", >>>>> "[sip.mywebphone.xx]/etc/letsencrypt/live/sip.mywebphone.xx") >>>>> modparam("tls_mgm", "tls_method", "[sip.mywebphone.xx]SSLv23") >>>>> modparam("tls_mgm", "verify_cert", "[sip.mywebphone.xx]1") >>>>> modparam("tls_mgm", "require_cert", "[sip.mywebphone.xx]1") >>>>> >>>>> ### This is for MS-Teams direct route >>>>> modparam("tls_mgm", "client_domain", "dom1.formsteams.com") >>>>> modparam("tls_mgm", "certificate", "[dom1.formsteams.com >>>>> ]/etc/letsencrypt/live/dom1.formsteams.com/cert.pem") >>>>> modparam("tls_mgm", "private_key", "[dom1.formsteams.com >>>>> ]/etc/letsencrypt/live/dom1.formsteams.com/privkey.pem") >>>>> modparam("tls_mgm", "ca_list", "[dom1.formsteams.com >>>>> ]/etc/letsencrypt/live/dom1.formsteams.com/fullchain.pem") >>>>> modparam("tls_mgm", "ca_dir", "[dom1.formsteams.com >>>>> ]/etc/letsencrypt/live/dom1.formsteams.com") >>>>> modparam("tls_mgm", "tls_method", "[dom1.formsteams.com]SSLv23") >>>>> modparam("tls_mgm", "verify_cert", "[dom1.formsteams.com]1") >>>>> modparam("tls_mgm", "require_cert", "[dom1.formsteams.com]1") >>>>> modparam("tls_mgm", "client_sip_domain_avp", "tls_sip_dom") >>>>> >>>>> Looks like the initial handshake is fine when my server sends OPTIONS >>>>> to MSTeams. There is a bug in the code according to the logs as shown below: >>>>> >>>>> opensips[10659]: CRITICAL:core:io_watch_add: #012>>> used fd map >>>>> fd=142 is not present in fd_array >>>>> (fd=142,type=19,flags=80000003,data=0x7f825805ceb8)#012#012It seems you >>>>> have hit a programming bug.#012Please help us make OpenSIPS better by >>>>> reporting it at https://github.com/OpenSIPS/opensips/issues >>>>> opensips[10659]: CRITICAL:core:io_watch_add: [TCP_main] check failed >>>>> after successful fd add (fd=141,type=19,data=0x7f825804fd98,flags=1) >>>>> already=0 >>>>> opensips[23993]: NOTICE:tls_wolfssl:verify_callback: depth = 1, verify >>>>> success >>>>> opensips[23993]: NOTICE:tls_wolfssl:verify_callback: depth = 0, verify >>>>> success >>>>> opensips[23993]: INFO:tls_wolfssl:_wolfssl_tls_async_connect: new TLS >>>>> connection to 52.114.16.74:5061 established >>>>> opensips[23993]: NOTICE:tls_wolfssl:verify_callback: depth = 1, verify >>>>> success >>>>> opensips[23993]: NOTICE:tls_wolfssl:verify_callback: depth = 0, verify >>>>> success >>>>> opensips[23995]: INFO:tls_wolfssl:_wolfssl_tls_async_connect: new TLS >>>>> connection to 52.114.76.76:5061 established >>>>> >>>>> >>>>> Regards, >>>>> Jehanzaib >>>>> >>>>> >>>>> On Wed, May 18, 2022 at 6:15 PM Bogdan-Andrei Iancu < >>>>> bogdan at opensips.org> wrote: >>>>> >>>>>> Hi Jehanzaib, >>>>>> >>>>>> The sequence for the MST TLS domains is wrong. >>>>>> >>>>>> For each TLS domain block, you need to start only with a server_domain >>>>>> or client_domain - of course, different names. And for each domain you need >>>>>> you set the matching conditions. See >>>>>> https://opensips.org/html/docs/modules/3.2.x/tls_mgm.html#domains-param >>>>>> >>>>>> Basically something like: >>>>>> >>>>>> modparam("tls_mgm", "server_domain", "formsteams_server") >>>>>> modparam("tls_mgm", "match_ip_address", "[formsteams_server]....") >>>>>> modparam("tls_mgm", "match_sip_domain", "[formsteams_server]....") >>>>>> modparam("tls_mgm", "certificate", "[formsteams_server].....) >>>>>> .... >>>>>> >>>>>> >>>>>> modparam("tls_mgm", "client_domain", "formsteams_client") >>>>>> modparam("tls_mgm", "match_ip_address", "[formsteams_client]....") >>>>>> modparam("tls_mgm", "match_sip_domain", "[formsteams_client]....") >>>>>> modparam("tls_mgm", "certificate", "[formsteams_client].....) >>>>>> .... >>>>>> >>>>>> >>>>>> Best regards, >>>>>> >>>>>> Bogdan-Andrei Iancu >>>>>> >>>>>> OpenSIPS Founder and Developer >>>>>> https://www.opensips-solutions.com >>>>>> OpenSIPS eBootcamp 23rd May - 3rd June 2022 >>>>>> https://opensips.org/training/OpenSIPS_eBootcamp_2022/ >>>>>> >>>>>> On 5/18/22 2:38 AM, Jehanzaib Younis wrote: >>>>>> >>>>>> Hi Bogdan, >>>>>> That's the problem, when I try to add the client_domain I get an >>>>>> error. Actually, I have a working config for webrtc but now I am adding a >>>>>> new domain for MS teams direct route. In fact, any other domain gives an >>>>>> error. If I disable MS Teams domain, the opensips do not give an >>>>>> error message and my webrtc client can connect without any issue. >>>>>> >>>>>> loadmodule "tls_mgm.so" >>>>>> modparam("tls_mgm", "tls_library", "wolfssl") >>>>>> >>>>>> #### (WebRTC) Client >>>>>> modparam("tls_mgm", "server_domain", "sip.mywebphone.xx") >>>>>> modparam("tls_mgm", "certificate", >>>>>> "[sip.mywebphone.xx]/etc/letsencrypt/live/sip.mywebphone.xx/cert.pem") >>>>>> modparam("tls_mgm", "private_key", >>>>>> "[sip.mywebphone.xx]/etc/letsencrypt/live/sip.mywebphone.xx/privkey.pem") >>>>>> modparam("tls_mgm", "ca_list", >>>>>> "[sip.mywebphone.xx]/etc/letsencrypt/live/sip.mywebphone.xx/fullchain.pem") >>>>>> modparam("tls_mgm", "ca_dir", >>>>>> "[sip.mywebphone.xx]/etc/letsencrypt/live/sip.mywebphone.xx") >>>>>> modparam("tls_mgm", "tls_method", "[sip.mywebphone.xx]SSLv23") >>>>>> modparam("tls_mgm", "verify_cert", "[sip.mywebphone.xx]1") >>>>>> modparam("tls_mgm", "require_cert", "[sip.mywebphone.xx]1") >>>>>> >>>>>> ### This is for MS-Teams direct route >>>>>> modparam("tls_mgm", "server_domain", "dom1.formsteams.com") >>>>>> modparam("tls_mgm", "client_domain", "dom1.formsteams.com") >>>>>> modparam("tls_mgm", "certificate", "[dom1.formsteams.com >>>>>> ]/etc/letsencrypt/live/dom1.formsteams.com/cert.pem") >>>>>> modparam("tls_mgm", "private_key", "[dom1.formsteams.com >>>>>> ]/etc/letsencrypt/live/dom1.formsteams.com/privkey.pem") >>>>>> modparam("tls_mgm", "ca_list", "[dom1.formsteams.com >>>>>> ]/etc/letsencrypt/live/dom1.formsteams.com/fullchain.pem") >>>>>> modparam("tls_mgm", "ca_dir", "[dom1.formsteams.com >>>>>> ]/etc/letsencrypt/live/dom1.formsteams.com") >>>>>> modparam("tls_mgm", "tls_method", "[dom1.formsteams.com]SSLv23") >>>>>> modparam("tls_mgm", "verify_cert", "[dom1.formsteams.com]1") >>>>>> modparam("tls_mgm", "require_cert", "[dom1.formsteams.com]1") >>>>>> modparam("tls_mgm", "client_sip_domain_avp", "tls_sip_dom") >>>>>> >>>>>> When i enable the MS-Teams direct route domain i get the below error: >>>>>> no certificate for tls domain ' dom1.formsteams.com ' defined >>>>>> >>>>>> >>>>>> Regards, >>>>>> Jehanzaib >>>>>> >>>>>> >>>>>> On Wed, May 18, 2022 at 3:04 AM Bogdan-Andrei Iancu < >>>>>> bogdan at opensips.org> wrote: >>>>>> >>>>>>> Hi Jehanzaib, >>>>>>> >>>>>>> What are the TLS client domains you have defined in your tls_mgm >>>>>>> module ? >>>>>>> >>>>>>> Regards, >>>>>>> >>>>>>> Bogdan-Andrei Iancu >>>>>>> >>>>>>> OpenSIPS Founder and Developer >>>>>>> https://www.opensips-solutions.com >>>>>>> OpenSIPS eBootcamp 23rd May - 3rd June 2022 >>>>>>> https://opensips.org/training/OpenSIPS_eBootcamp_2022/ >>>>>>> >>>>>>> On 5/17/22 4:32 PM, Jehanzaib Younis wrote: >>>>>>> >>>>>>> Hi, >>>>>>> >>>>>>> I am having trouble to send/receive OPTIONS to ms teams. >>>>>>> Using the dispatcher module. The socket is defined as tls:*mysbcip* >>>>>>> :5061 >>>>>>> Looks like when my opensips (3.2.x) tries to send OPTIONS. it is >>>>>>> giving me the following error >>>>>>> >>>>>>> ERROR:proto_tls:proto_tls_conn_init: no TLS client domain found >>>>>>> ERROR:core:tcp_conn_create: failed to do proto 3 specific init for >>>>>>> conn 0x7f00ef2a85a0 >>>>>>> ERROR:core:tcp_async_connect: tcp_conn_create failed >>>>>>> ERROR:proto_tls:proto_tls_send: async TCP connect failed >>>>>>> ERROR:tm:msg_send: send() to 52.114.76.76:5061 for proto tls/3 >>>>>>> failed >>>>>>> ERROR:tm:t_uac: attempt to send to ' >>>>>>> sip:sip3.pstnhub.microsoft.com:5061;transport:tls' failed >>>>>>> >>>>>>> I am setting the Contact as >>>>>>> >>>>>>> Looks like the client domain is used for outgoing TLS connection but >>>>>>> no idea which domain i need to add here. The socket is my opensips ip >>>>>>> address. >>>>>>> >>>>>>> Has anyone seen a similar kind of behaviour? >>>>>>> >>>>>>> Thank you. >>>>>>> >>>>>>> Regards, >>>>>>> Jehanzaib >>>>>>> >>>>>>> _______________________________________________ >>>>>>> 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 >>>>> >>>> -- >>>> VoIP Embedded, Inc. >>>> http://www.voipembedded.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 >>> >> -- >> VoIP Embedded, Inc. >> http://www.voipembedded.com >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> > > _______________________________________________ > Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > -------------- next part -------------- An HTML attachment was scrubbed... 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