From wadii at evenmedia.fr Fri Dec 2 13:56:37 2022 From: wadii at evenmedia.fr (Wadii ELMAJDI | Evenmedia) Date: Fri, 2 Dec 2022 13:56:37 +0000 Subject: [OpenSIPS-Users] Erasing statistics groups from memory Message-ID: Hello, I am using statistics module to store CPS data on demand (via an external variable shv_set), keys are current seconds ,example : opensips-cli -x mi get_statistics cps: { "cps:2:20221202143557": 1, "cps:2:20221202143613": 1, "cps:2:20221202143616": 1 } The data is stored on the server memory, and then reset via a time_route everyday (reset_stat) which updates the counters to 0 but the keys persist on my server. Is there anyway to erase those keys from the memory "cps:2...." Without restarting my server ? I know I can use an external Redis or MYSQL servers for such purpose , but I'm trying as much as possible to keep everything inside opensips. Thank you -------------- next part -------------- An HTML attachment was scrubbed... URL: From johan at democon.be Fri Dec 2 16:22:48 2022 From: johan at democon.be (johan) Date: Fri, 2 Dec 2022 17:22:48 +0100 Subject: [OpenSIPS-Users] reinvite and dialog uniqueness. Message-ID: *1. dialog uniqueness*     Alice     calls     Bob             ->INVITE                 call-id: 1                 from: alice at atlanta;tag=1                 to: bob at boston             <- 200                 call-id: 1                  from: alice at atlanta;tag=1                  to: bob at boston;tag=2             -> ACK                 call-id: 1                  from: alice at atlanta;tag=1                  to: bob at boston;tag=2 Bob does a session refresh             <- INVITE question in on the from tag and the to tag. should it be             <- INVITE                 callid: 1                  from: alice at atlanta;tag=1                  to: bob at boston;tag=2 or             <- INVITE                 call-id: 1                 from: alice at atlanta;tag=1                  to: bob at boston;tag=2 or             are both the same dialog (call-id + the 2 tags) ? *2. reinvite* Secondly, when opensips is used as b2bua scenario as below A                         opensips                     B         -> invite                     ->invite         <-200                        <- 200         -> ack                        ->ack         ...                                ....                                            <-reinvite for keeeping session alive. Is there a way that I can let respond opensips 200 OK on the reinvite without forwarding the request to A ? so I want A                         opensips                     B         -> invite                     ->invite         <-200                        <- 200         -> ack                        ->ack         ...                                ....                                            <-reinvite for keeeping session alive.                                            ->200 instead of A                         opensips                     B         -> invite                     ->invite         <-200                        <- 200         -> ack                        ->ack         ...                                ....                                            <-reinvite for keeeping session alive.         <- reinvite         -> 200                        ->200 wkr, -------------- next part -------------- An HTML attachment was scrubbed... URL: From osas at voipembedded.com Fri Dec 2 16:34:35 2022 From: osas at voipembedded.com (Ovidiu Sas) Date: Fri, 2 Dec 2022 11:34:35 -0500 Subject: [OpenSIPS-Users] reinvite and dialog uniqueness. In-Reply-To: References: Message-ID: The tag belongs to the SIP endpoint and does not change. Take a look here at an example: https://datatracker.ietf.org/doc/html/rfc3665#section-3.7 Regards, Ovidiu Sas On Fri, Dec 2, 2022 at 11:23 AM johan wrote: > > 1. dialog uniqueness > > Alice calls Bob > > ->INVITE > > call-id: 1 > > from: alice at atlanta;tag=1 > > to: bob at boston > > <- 200 > > call-id: 1 > > from: alice at atlanta;tag=1 > > to: bob at boston;tag=2 > > -> ACK > > call-id: 1 > > from: alice at atlanta;tag=1 > > to: bob at boston;tag=2 > > Bob does a session refresh > > <- INVITE > > question in on the from tag and the to tag. > > should it be > > <- INVITE > > callid: 1 > > from: alice at atlanta;tag=1 > > to: bob at boston;tag=2 > > or > > <- INVITE > > call-id: 1 > > from: alice at atlanta;tag=1 > > to: bob at boston;tag=2 > > or > > are both the same dialog (call-id + the 2 tags) ? > > > 2. reinvite > > Secondly, > > when opensips is used as b2bua scenario as below > > > A opensips B > > -> invite ->invite > > <-200 <- 200 > > -> ack ->ack > > ... .... > > <-reinvite for keeeping session alive. > > Is there a way that I can let respond opensips 200 OK on the reinvite without forwarding the request to A ? > > so I want > > A opensips B > > -> invite ->invite > > <-200 <- 200 > > -> ack ->ack > > ... .... > > <-reinvite for keeeping session alive. > > ->200 > > > instead of > > A opensips B > > -> invite ->invite > > <-200 <- 200 > > -> ack ->ack > > ... .... > > <-reinvite for keeeping session alive. > > <- reinvite > > -> 200 ->200 > > > wkr, > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- VoIP Embedded, Inc. http://www.voipembedded.com From johan at democon.be Fri Dec 2 16:40:10 2022 From: johan at democon.be (johan) Date: Fri, 2 Dec 2022 17:40:10 +0100 Subject: [OpenSIPS-Users] reinvite and dialog uniqueness. In-Reply-To: References: Message-ID: <56a1b943-e32b-5bf6-3456-2dab7356b718@democon.be> I realise that I made an error in my drawing. 1. dialog uniqueness Alice calls Bob ->INVITE call-id: 1 from: alice at atlanta;tag=1 to: bob at boston <- 200 call-id: 1 from: alice at atlanta;tag=1 to: bob at boston;tag=2 -> ACK call-id: 1 from: alice at atlanta;tag=1 to: bob at boston;tag=2 Bob does a session refresh <- INVITE question in on the from tag and the to tag. should it be <- INVITE callid: 1 *from: alice at atlanta;tag=1 to: bob at boston;tag=2* or <- INVITE call-id: 1 *to**: alice at atlanta;tag=1 from: bob at boston;tag=2* or are both the same dialog (call-id + the 2 tags) ? On 2/12/2022 17:34, Ovidiu Sas wrote: > The tag belongs to the SIP endpoint and does not change. > Take a look here at an example: > https://datatracker.ietf.org/doc/html/rfc3665#section-3.7 > > Regards, > Ovidiu Sas > > On Fri, Dec 2, 2022 at 11:23 AM johan wrote: >> 1. dialog uniqueness >> >> Alice calls Bob >> >> ->INVITE >> >> call-id: 1 >> >> from: alice at atlanta;tag=1 >> >> to: bob at boston >> >> <- 200 >> >> call-id: 1 >> >> from: alice at atlanta;tag=1 >> >> to: bob at boston;tag=2 >> >> -> ACK >> >> call-id: 1 >> >> from: alice at atlanta;tag=1 >> >> to: bob at boston;tag=2 >> >> Bob does a session refresh >> >> <- INVITE >> >> question in on the from tag and the to tag. >> >> should it be >> >> <- INVITE >> >> callid: 1 >> >> from: alice at atlanta;tag=1 >> >> to: bob at boston;tag=2 >> >> or >> >> <- INVITE >> >> call-id: 1 >> >> from: alice at atlanta;tag=1 >> >> to: bob at boston;tag=2 >> >> or >> >> are both the same dialog (call-id + the 2 tags) ? >> >> >> 2. reinvite >> >> Secondly, >> >> when opensips is used as b2bua scenario as below >> >> >> A opensips B >> >> -> invite ->invite >> >> <-200 <- 200 >> >> -> ack ->ack >> >> ... .... >> >> <-reinvite for keeeping session alive. >> >> Is there a way that I can let respond opensips 200 OK on the reinvite without forwarding the request to A ? >> >> so I want >> >> A opensips B >> >> -> invite ->invite >> >> <-200 <- 200 >> >> -> ack ->ack >> >> ... .... >> >> <-reinvite for keeeping session alive. >> >> ->200 >> >> >> instead of >> >> A opensips B >> >> -> invite ->invite >> >> <-200 <- 200 >> >> -> ack ->ack >> >> ... .... >> >> <-reinvite for keeeping session alive. >> >> <- reinvite >> >> -> 200 ->200 >> >> >> wkr, >> >> >> _______________________________________________ >> 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 Fri Dec 2 16:51:38 2022 From: osas at voipembedded.com (Ovidiu Sas) Date: Fri, 2 Dec 2022 11:51:38 -0500 Subject: [OpenSIPS-Users] reinvite and dialog uniqueness. In-Reply-To: <56a1b943-e32b-5bf6-3456-2dab7356b718@democon.be> References: <56a1b943-e32b-5bf6-3456-2dab7356b718@democon.be> Message-ID: >From has the user that initiated the request. For a reINVITE, if the reINVITE is issued by bob, then From has bob URIs. Take a look at the examples provided in the Session Initiation Protocol (SIP) Basic Call Flow Examples RFC: https://datatracker.ietf.org/doc/html/rfc3665 As for the b2b question, take a look at b2b_sed_reply function: https://opensips.org/docs/modules/3.3.x/b2b_logic.html#func_b2b_send_reply If you need to reply with 200ok to a reINVITE, you need to make sure that the SDP in the reINVITE is the same as the initial SDP and send back the other party's SDP in 200ok. -ovidiu On Fri, Dec 2, 2022 at 11:45 AM johan wrote: > > I realise that I made an error in my drawing. > > 1. dialog uniqueness > > Alice calls Bob > > ->INVITE > > call-id: 1 > > from: alice at atlanta;tag=1 > > to: bob at boston > > <- 200 > > call-id: 1 > > from: alice at atlanta;tag=1 > > to: bob at boston;tag=2 > > -> ACK > > call-id: 1 > > from: alice at atlanta;tag=1 > > to: bob at boston;tag=2 > > Bob does a session refresh > > <- INVITE > > question in on the from tag and the to tag. > > should it be > > <- INVITE > > callid: 1 > > from: alice at atlanta;tag=1 > > to: bob at boston;tag=2 > > or > > <- INVITE > > call-id: 1 > > to: alice at atlanta;tag=1 > > from: bob at boston;tag=2 > > or > > are both the same dialog (call-id + the 2 tags) ? > > > On 2/12/2022 17:34, Ovidiu Sas wrote: > > The tag belongs to the SIP endpoint and does not change. > Take a look here at an example: > https://datatracker.ietf.org/doc/html/rfc3665#section-3.7 > > Regards, > Ovidiu Sas > > On Fri, Dec 2, 2022 at 11:23 AM johan wrote: > > 1. dialog uniqueness > > Alice calls Bob > > ->INVITE > > call-id: 1 > > from: alice at atlanta;tag=1 > > to: bob at boston > > <- 200 > > call-id: 1 > > from: alice at atlanta;tag=1 > > to: bob at boston;tag=2 > > -> ACK > > call-id: 1 > > from: alice at atlanta;tag=1 > > to: bob at boston;tag=2 > > Bob does a session refresh > > <- INVITE > > question in on the from tag and the to tag. > > should it be > > <- INVITE > > callid: 1 > > from: alice at atlanta;tag=1 > > to: bob at boston;tag=2 > > or > > <- INVITE > > call-id: 1 > > from: alice at atlanta;tag=1 > > to: bob at boston;tag=2 > > or > > are both the same dialog (call-id + the 2 tags) ? > > > 2. reinvite > > Secondly, > > when opensips is used as b2bua scenario as below > > > A opensips B > > -> invite ->invite > > <-200 <- 200 > > -> ack ->ack > > ... .... > > <-reinvite for keeeping session alive. > > Is there a way that I can let respond opensips 200 OK on the reinvite without forwarding the request to A ? > > so I want > > A opensips B > > -> invite ->invite > > <-200 <- 200 > > -> ack ->ack > > ... .... > > <-reinvite for keeeping session alive. > > ->200 > > > instead of > > A opensips B > > -> invite ->invite > > <-200 <- 200 > > -> ack ->ack > > ... .... > > <-reinvite for keeeping session alive. > > <- reinvite > > -> 200 ->200 > > > wkr, > > > _______________________________________________ > 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 From johan at democon.be Fri Dec 2 17:35:51 2022 From: johan at democon.be (johan) Date: Fri, 2 Dec 2022 18:35:51 +0100 Subject: [OpenSIPS-Users] reinvite and dialog uniqueness. In-Reply-To: References: <56a1b943-e32b-5bf6-3456-2dab7356b718@democon.be> Message-ID: It would be even better to have somehting without b2b. On 2/12/2022 17:51, Ovidiu Sas wrote: > From has the user that initiated the request. > For a reINVITE, if the reINVITE is issued by bob, then From has bob URIs. > Take a look at the examples provided in the Session Initiation > Protocol (SIP) Basic Call Flow Examples RFC: > https://datatracker.ietf.org/doc/html/rfc3665 > > As for the b2b question, take a look at b2b_sed_reply function: > https://opensips.org/docs/modules/3.3.x/b2b_logic.html#func_b2b_send_reply > If you need to reply with 200ok to a reINVITE, you need to make sure > that the SDP in the reINVITE is the same as the initial SDP and send > back the other party's SDP in 200ok. > > -ovidiu > > On Fri, Dec 2, 2022 at 11:45 AM johan wrote: >> I realise that I made an error in my drawing. >> >> 1. dialog uniqueness >> >> Alice calls Bob >> >> ->INVITE >> >> call-id: 1 >> >> from: alice at atlanta;tag=1 >> >> to: bob at boston >> >> <- 200 >> >> call-id: 1 >> >> from: alice at atlanta;tag=1 >> >> to: bob at boston;tag=2 >> >> -> ACK >> >> call-id: 1 >> >> from: alice at atlanta;tag=1 >> >> to: bob at boston;tag=2 >> >> Bob does a session refresh >> >> <- INVITE >> >> question in on the from tag and the to tag. >> >> should it be >> >> <- INVITE >> >> callid: 1 >> >> from: alice at atlanta;tag=1 >> >> to: bob at boston;tag=2 >> >> or >> >> <- INVITE >> >> call-id: 1 >> >> to: alice at atlanta;tag=1 >> >> from: bob at boston;tag=2 >> >> or >> >> are both the same dialog (call-id + the 2 tags) ? >> >> >> On 2/12/2022 17:34, Ovidiu Sas wrote: >> >> The tag belongs to the SIP endpoint and does not change. >> Take a look here at an example: >> https://datatracker.ietf.org/doc/html/rfc3665#section-3.7 >> >> Regards, >> Ovidiu Sas >> >> On Fri, Dec 2, 2022 at 11:23 AM johan wrote: >> >> 1. dialog uniqueness >> >> Alice calls Bob >> >> ->INVITE >> >> call-id: 1 >> >> from: alice at atlanta;tag=1 >> >> to: bob at boston >> >> <- 200 >> >> call-id: 1 >> >> from: alice at atlanta;tag=1 >> >> to: bob at boston;tag=2 >> >> -> ACK >> >> call-id: 1 >> >> from: alice at atlanta;tag=1 >> >> to: bob at boston;tag=2 >> >> Bob does a session refresh >> >> <- INVITE >> >> question in on the from tag and the to tag. >> >> should it be >> >> <- INVITE >> >> callid: 1 >> >> from: alice at atlanta;tag=1 >> >> to: bob at boston;tag=2 >> >> or >> >> <- INVITE >> >> call-id: 1 >> >> from: alice at atlanta;tag=1 >> >> to: bob at boston;tag=2 >> >> or >> >> are both the same dialog (call-id + the 2 tags) ? >> >> >> 2. reinvite >> >> Secondly, >> >> when opensips is used as b2bua scenario as below >> >> >> A opensips B >> >> -> invite ->invite >> >> <-200 <- 200 >> >> -> ack ->ack >> >> ... .... >> >> <-reinvite for keeeping session alive. >> >> Is there a way that I can let respond opensips 200 OK on the reinvite without forwarding the request to A ? >> >> so I want >> >> A opensips B >> >> -> invite ->invite >> >> <-200 <- 200 >> >> -> ack ->ack >> >> ... .... >> >> <-reinvite for keeeping session alive. >> >> ->200 >> >> >> instead of >> >> A opensips B >> >> -> invite ->invite >> >> <-200 <- 200 >> >> -> ack ->ack >> >> ... .... >> >> <-reinvite for keeeping session alive. >> >> <- reinvite >> >> -> 200 ->200 >> >> >> wkr, >> >> >> _______________________________________________ >> 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 osas at voipembedded.com Fri Dec 2 18:03:00 2022 From: osas at voipembedded.com (Ovidiu Sas) Date: Fri, 2 Dec 2022 13:03:00 -0500 Subject: [OpenSIPS-Users] reinvite and dialog uniqueness. In-Reply-To: References: <56a1b943-e32b-5bf6-3456-2dab7356b718@democon.be> Message-ID: You can try to use topology_hiding and t_reply_with_body(): https://opensips.org/docs/modules/3.3.x/topology_hiding.html https://opensips.org/docs/modules/3.3.x/tm.html#func_t_reply_with_body Never tried it so I don't know if this will work. The question is if the CSeq number will be updated for the leg receiving the reINVITE. -ovidiu On Fri, Dec 2, 2022 at 12:36 PM johan wrote: > > It would be even better to have somehting without b2b. > > On 2/12/2022 17:51, Ovidiu Sas wrote: > > From has the user that initiated the request. > > For a reINVITE, if the reINVITE is issued by bob, then From has bob URIs. > > Take a look at the examples provided in the Session Initiation > > Protocol (SIP) Basic Call Flow Examples RFC: > > https://datatracker.ietf.org/doc/html/rfc3665 > > > > As for the b2b question, take a look at b2b_sed_reply function: > > https://opensips.org/docs/modules/3.3.x/b2b_logic.html#func_b2b_send_reply > > If you need to reply with 200ok to a reINVITE, you need to make sure > > that the SDP in the reINVITE is the same as the initial SDP and send > > back the other party's SDP in 200ok. > > > > -ovidiu > > > > On Fri, Dec 2, 2022 at 11:45 AM johan wrote: > >> I realise that I made an error in my drawing. > >> > >> 1. dialog uniqueness > >> > >> Alice calls Bob > >> > >> ->INVITE > >> > >> call-id: 1 > >> > >> from: alice at atlanta;tag=1 > >> > >> to: bob at boston > >> > >> <- 200 > >> > >> call-id: 1 > >> > >> from: alice at atlanta;tag=1 > >> > >> to: bob at boston;tag=2 > >> > >> -> ACK > >> > >> call-id: 1 > >> > >> from: alice at atlanta;tag=1 > >> > >> to: bob at boston;tag=2 > >> > >> Bob does a session refresh > >> > >> <- INVITE > >> > >> question in on the from tag and the to tag. > >> > >> should it be > >> > >> <- INVITE > >> > >> callid: 1 > >> > >> from: alice at atlanta;tag=1 > >> > >> to: bob at boston;tag=2 > >> > >> or > >> > >> <- INVITE > >> > >> call-id: 1 > >> > >> to: alice at atlanta;tag=1 > >> > >> from: bob at boston;tag=2 > >> > >> or > >> > >> are both the same dialog (call-id + the 2 tags) ? > >> > >> > >> On 2/12/2022 17:34, Ovidiu Sas wrote: > >> > >> The tag belongs to the SIP endpoint and does not change. > >> Take a look here at an example: > >> https://datatracker.ietf.org/doc/html/rfc3665#section-3.7 > >> > >> Regards, > >> Ovidiu Sas > >> > >> On Fri, Dec 2, 2022 at 11:23 AM johan wrote: > >> > >> 1. dialog uniqueness > >> > >> Alice calls Bob > >> > >> ->INVITE > >> > >> call-id: 1 > >> > >> from: alice at atlanta;tag=1 > >> > >> to: bob at boston > >> > >> <- 200 > >> > >> call-id: 1 > >> > >> from: alice at atlanta;tag=1 > >> > >> to: bob at boston;tag=2 > >> > >> -> ACK > >> > >> call-id: 1 > >> > >> from: alice at atlanta;tag=1 > >> > >> to: bob at boston;tag=2 > >> > >> Bob does a session refresh > >> > >> <- INVITE > >> > >> question in on the from tag and the to tag. > >> > >> should it be > >> > >> <- INVITE > >> > >> callid: 1 > >> > >> from: alice at atlanta;tag=1 > >> > >> to: bob at boston;tag=2 > >> > >> or > >> > >> <- INVITE > >> > >> call-id: 1 > >> > >> from: alice at atlanta;tag=1 > >> > >> to: bob at boston;tag=2 > >> > >> or > >> > >> are both the same dialog (call-id + the 2 tags) ? > >> > >> > >> 2. reinvite > >> > >> Secondly, > >> > >> when opensips is used as b2bua scenario as below > >> > >> > >> A opensips B > >> > >> -> invite ->invite > >> > >> <-200 <- 200 > >> > >> -> ack ->ack > >> > >> ... .... > >> > >> <-reinvite for keeeping session alive. > >> > >> Is there a way that I can let respond opensips 200 OK on the reinvite without forwarding the request to A ? > >> > >> so I want > >> > >> A opensips B > >> > >> -> invite ->invite > >> > >> <-200 <- 200 > >> > >> -> ack ->ack > >> > >> ... .... > >> > >> <-reinvite for keeeping session alive. > >> > >> ->200 > >> > >> > >> instead of > >> > >> A opensips B > >> > >> -> invite ->invite > >> > >> <-200 <- 200 > >> > >> -> ack ->ack > >> > >> ... .... > >> > >> <-reinvite for keeeping session alive. > >> > >> <- reinvite > >> > >> -> 200 ->200 > >> > >> > >> wkr, > >> > >> > >> _______________________________________________ > >> 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 -- VoIP Embedded, Inc. http://www.voipembedded.com From ahmed.rehan at gmail.com Mon Dec 5 10:16:47 2022 From: ahmed.rehan at gmail.com (Ahmed Rehan) Date: Mon, 5 Dec 2022 15:16:47 +0500 Subject: [OpenSIPS-Users] Calling external API Message-ID: Hi All I m just wandering is there a way where we can call an external API in routing script to GET some values to use in opensips routing script . -- Regards Ahmed Rehan -------------- next part -------------- An HTML attachment was scrubbed... URL: From clorisice at 163.com Thu Dec 1 02:32:16 2022 From: clorisice at 163.com (wl) Date: Thu, 1 Dec 2022 10:32:16 +0800 (CST) Subject: [OpenSIPS-Users] open3.3 has no opensipsctlrc? Message-ID: <29097287.9b8.184cb870d14.Coremail.clorisice@163.com> hello: I am using opensip-3.3.3 but after configure, make, make install , I cannot fnd the previous config file 'opensipsctlrc' at prefix_dir/etc/opensips/. why? is this file cancelled on new version? and I am trying to connect newest opensips to newest freeswitch, and I refer this article: https://www.opensips.org/Documentation/Tutorials-OpenSIPSF but it is based on opensips-1.8 ,is it obsolete to new version since I cannot find opensipsctlrc file agian , I dont't know how to integrate newest opensips with freeswitch, can you provide new reference article for me? thanks a lot best reagarda looking forward your reply! -------------- next part -------------- An HTML attachment was scrubbed... URL: From john.denoo at telenetgroup.be Sun Dec 4 18:03:19 2022 From: john.denoo at telenetgroup.be (Denoo John) Date: Sun, 4 Dec 2022 18:03:19 +0000 Subject: [OpenSIPS-Users] reinvite and dialog uniqueness. In-Reply-To: References: Message-ID: Johan, In regards to the first question I think the Invite and the Re-Invite have only two differences. The Invite doesn’t has a tag in the To-field, where the Re-Invite contains To:…;tag=2 In addition the Via-header contains a new branch-id because it’s a new transaction. For the second question, I have no experience with OpenSIPS. Best regards, John From: johan Sent: vrijdag 2 december 2022 17:23 To: OpenSIPS users mailling list Subject: reinvite and dialog uniqueness. 1. dialog uniqueness Alice calls Bob ->INVITE call-id: 1 from: alice at atlanta;tag=1 to: bob at boston <- 200 call-id: 1 from: alice at atlanta;tag=1 to: bob at boston;tag=2 -> ACK call-id: 1 from: alice at atlanta;tag=1 to: bob at boston;tag=2 Bob does a session refresh <- INVITE question in on the from tag and the to tag. should it be <- INVITE callid: 1 from: alice at atlanta;tag=1 to: bob at boston;tag=2 or <- INVITE call-id: 1 from: alice at atlanta;tag=1 to: bob at boston;tag=2 or are both the same dialog (call-id + the 2 tags) ? 2. reinvite Secondly, when opensips is used as b2bua scenario as below A opensips B -> invite ->invite <-200 <- 200 -> ack ->ack ... .... <-reinvite for keeeping session alive. Is there a way that I can let respond opensips 200 OK on the reinvite without forwarding the request to A ? so I want A opensips B -> invite ->invite <-200 <- 200 -> ack ->ack ... .... <-reinvite for keeeping session alive. ->200 instead of A opensips B -> invite ->invite <-200 <- 200 -> ack ->ack ... .... <-reinvite for keeeping session alive. <- reinvite -> 200 ->200 wkr, Confidential - Limited *************************************************************** Dit e-mail bericht inclusief eventuele ingesloten bestanden kan informatie bevatten die vertrouwelijk is en/of beschermd door intellectuele eigendomsrechten. Dit bericht is uitsluitend bestemd voor de geadresseerde(n). Elk gebruik van de informatie vervat in dit bericht (waaronder de volledige of gedeeltelijke reproductie of verspreiding onder elke vorm) door andere personen dan de geadresseerde(n) is verboden. 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URL: From alain.bieuzent at free.fr Mon Dec 5 11:25:28 2022 From: alain.bieuzent at free.fr (Alain Bieuzent) Date: Mon, 05 Dec 2022 12:25:28 +0100 Subject: [OpenSIPS-Users] Calling external API In-Reply-To: References: Message-ID: Yes, look at REST_CLIENT Module : https://opensips.org/html/docs/modules/3.2.x/rest_client.html De : Users au nom de Ahmed Rehan Répondre à : OpenSIPS users mailling list Date : lundi 5 décembre 2022 à 11:19 À : OpenSIPS users mailling list Objet : [OpenSIPS-Users] Calling external API Hi All I m just wandering is there a way where we can call an external API in routing script to GET some values to use in opensips routing script . -- 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 ahmed.rehan at gmail.com Mon Dec 5 11:42:26 2022 From: ahmed.rehan at gmail.com (Ahmed Rehan) Date: Mon, 5 Dec 2022 16:42:26 +0500 Subject: [OpenSIPS-Users] Calling external API In-Reply-To: References: Message-ID: Thanks Alain for help On Mon, Dec 5, 2022 at 4:25 PM Alain Bieuzent wrote: > Yes, look at REST_CLIENT Module : > https://opensips.org/html/docs/modules/3.2.x/rest_client.html > > > > > > *De : *Users au nom de Ahmed Rehan < > ahmed.rehan at gmail.com> > *Répondre à : *OpenSIPS users mailling list > *Date : *lundi 5 décembre 2022 à 11:19 > *À : *OpenSIPS users mailling list > *Objet : *[OpenSIPS-Users] Calling external API > > > > Hi All > > > > I m just wandering is there a way where we can call an external API in > routing script to GET some values to use in opensips routing script . > > > > > > > -- > > Regards > Ahmed Rehan > > _______________________________________________ Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -- Regards Ahmed Rehan -------------- next part -------------- An HTML attachment was scrubbed... URL: From nutxase at proton.me Mon Dec 5 16:54:55 2022 From: nutxase at proton.me (nutxase) Date: Mon, 05 Dec 2022 16:54:55 +0000 Subject: [OpenSIPS-Users] retry (!mid_registrar_lookup("location")) Message-ID: Hi All I am trying to set mid_registrar to retry looking up the location of the registration if it does not find it the first time as when it receives the invite it runs a push script to wake up the phone and then needs to lookup the new location and send the call but currently it just gives 404 any ideas how to help me? here is the code but it does not seem to be running as expected if (!mid_registrar_lookup("location")) { exec("/etc/opensips/pusher.sh $tu", , $var(out)); t_reply(404, "Not Found"); } if (!mid_registrar_lookup("location")) { xlog("2nd attempt to lookup location"); exec("/etc/opensips/pusher.sh $tu", , $var(out)); t_reply(404, "Not Found"); exit; } t_relay(); exit; } Sent with [Proton Mail](https://proton.me/) secure email. -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Tue Dec 6 11:10:17 2022 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Tue, 6 Dec 2022 13:10:17 +0200 Subject: [OpenSIPS-Users] setflag/isflagset or cache_store/cache_fetch In-Reply-To: References: Message-ID: <9b97bf09-fdb6-0dda-e6df-81f99065a4fe@opensips.org> Hi Alain, Setting a flag is less expensive. But if you want to have call/dialog persistence for the flags, use the dialog flags, not the transaction flags. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com OpenSIPS Bootcamp 5-16 Dec 2022, online https://www.opensips.org/training/OpenSIPS_eBootcamp_2022/ On 11/21/22 5:39 PM, Alain Bieuzent wrote: > > Hi All, > > I need to store information during a call. They are often binary > information (for example which rtpengine was used). > > For the moment I store this information in a local redis database. In > most cases a setflag/isflagset would be enough. > > What are the best practices, which of the two systems consumes the > least CPU? > > Thanks > > Alain > > > _______________________________________________ > 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 Dec 6 11:15:18 2022 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Tue, 6 Dec 2022 13:15:18 +0200 Subject: [OpenSIPS-Users] open3.3 has no opensipsctlrc? In-Reply-To: <172d3181.193c.184b858278e.Coremail.clorisice@163.com> References: <172d3181.193c.184b858278e.Coremail.clorisice@163.com> Message-ID: Hi, OpenSIPS 3.3 does not use opensipsctl anymore, but opensips-cli     https://github.com/OpenSIPS/opensips-cli Best regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com OpenSIPS Bootcamp 5-16 Dec 2022, online https://www.opensips.org/training/OpenSIPS_eBootcamp_2022/ On 11/27/22 11:08 AM, wl wrote: > hello: > I am using opensip-3.3.3  but after configure, make, make install , I > cannot fnd the previous config file 'opensipsctlrc' > at prefix_dir/etc/opensips/. why? is this file cancelled on new version? > and > I am trying to connect newest opensips to newest freeswitch, and I > refer this article: > > https://www.opensips.org/Documentation/Tutorials-OpenSIPSF > > > but it is based on opensips-1.8 ,is it obsolete to new version since I > cannot find opensipsctlrc file agian , I dont't know how to integrate > newest opensips with freeswitch, can you provide new reference article > for me? > thanks a lot > best reagarda > looking forward your reply! > > > _______________________________________________ > 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 Dec 6 11:18:52 2022 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Tue, 6 Dec 2022 13:18:52 +0200 Subject: [OpenSIPS-Users] Inbound PSTN connection failure In-Reply-To: References: Message-ID: Hi Barry, It looks like the is_from_gw() does not match for the OPTIONS. Have you reloaded the Drouting data after adding the GW ? run "opensips-cli -x mi dr_gw_status" to see what opensips knows as GWs and if they match the source IP of that OPTIONS https://opensips.org/html/docs/modules/3.2.x/drouting.html#idp5947120 Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com OpenSIPS Bootcamp 5-16 Dec 2022, online https://www.opensips.org/training/OpenSIPS_eBootcamp_2022/ On 11/27/22 6:03 PM, Barry O. Williams wrote: > Hi, > > I have installed Opensips successfully, have registered SIP stations > and I am able to make station to station calls. I have setup IP > authentication connection with provider Sketel and unable to connect > from Opensips side. Skyetel is sending OPTIONS request and Opensips is > replying with 403 Relay Forbidden/. Gateways have been added in > Opensips CP. > > >  if ( !(is_method("REGISTER")  || is_from_gw() ) ) { > >                 if (is_from_local()) { >                         # authenticate if from local subscriber >                         # authenticate all initial non-REGISTER > request that pretend to be >                         # generated by local subscriber (domain from > FROM URI is local) >                         if (!proxy_authorize("", "subscriber")) { >                                 proxy_challenge("", "auth"); >                                 exit; >                         } >                         if ($au!=$fU) { >                                 send_reply(403,"Forbidden auth ID"); >                                 exit; >                         } > >                         consume_credentials(); >                         # caller authenticated > >                 } else { >                         # if caller is not local, then called number > must be local > >                         if (!is_uri_host_local()) { >                                 send_reply(403,"Relay Forbidden"); >                                 exit; > > Regards, > Barry > > _______________________________________________ > 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 Dec 6 11:21:10 2022 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Tue, 6 Dec 2022 13:21:10 +0200 Subject: [OpenSIPS-Users] Subscribe/Notify In-Reply-To: References: Message-ID: <31898a56-1f42-2ce7-95a8-591f61cbbaba@opensips.org> Hi Kevin, You must record_route() during the SUBSCRIBE requests, so the following NOTIFY requests will also go via your proxy, so thru the FW. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com OpenSIPS Bootcamp 5-16 Dec 2022, online https://www.opensips.org/training/OpenSIPS_eBootcamp_2022/ On 11/21/22 10:20 PM, Kevin Kennedy wrote: > I am testing opensips 3.2 as a loadbalancer for a group of SBCs.  I am > using Mid-Registrar and Dispatcher to load balance to the SBCs.  I am > able to get Registrations and Calls working, but the issue I am having > is with the Subscribe messages not receiving the NOTIFY response.  > Since it is set as a stateful proxy the out of dialog messages are > coming directly to the users public IP.  So Notify is being dropped at > the firewall as its not coming from the same IP address that the phone > is registered to. > > Is there a mechanism to remedy this other than full B2BUA? > If B2BUA is the only path, what is the best way to set this up? > > Thank you. > > > _______________________________________________ > 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 Dec 6 11:25:39 2022 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Tue, 6 Dec 2022 13:25:39 +0200 Subject: [OpenSIPS-Users] Erasing statistics groups from memory In-Reply-To: References: Message-ID: <02012b66-cca9-0cd6-3381-f40efacfcc18@opensips.org> Hi Wadii, No, you cannot purge an existing stat (even if it was dynamically created). Maybe you should use something else for storing the cps data, like the local cache. https://www.opensips.org/Documentation/Script-CoreFunctions-3-2#cache_store https://opensips.org/html/docs/modules/3.2.x/cachedb_local.html Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com OpenSIPS Bootcamp 5-16 Dec 2022, online https://www.opensips.org/training/OpenSIPS_eBootcamp_2022/ On 12/2/22 3:56 PM, Wadii ELMAJDI | Evenmedia wrote: > > Hello, > > I am using statistics module to store CPS data on demand (via an > external variable shv_set), keys are current seconds ,example : > > opensips-cli -x mi get_statistics cps: > > { > > "cps:2:20221202143557": 1, > > "cps:2:20221202143613": 1, > > "cps:2:20221202143616": 1 > > } > > > The data is stored on the server memory, and then reset via a > time_route everyday (reset_stat) which updates the counters to 0 but > the keys persist on my server. > Is there anyway to erase those keys from the memory “cps:2….” Without > restarting my server ? > I know I can use an external Redis or MYSQL servers for such purpose , > but I’m trying as much as possible to keep everything inside opensips. > > Thank you > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From liviu at opensips.org Wed Dec 7 11:58:55 2022 From: liviu at opensips.org (Liviu Chircu) Date: Wed, 7 Dec 2022 13:58:55 +0200 Subject: [OpenSIPS-Users] [Release Freeze] Upcoming OpenSIPS 3.3.3, 3.2.10 and 3.1.13 minor releases Message-ID: Hi, everyone! The 3.3.3, 3.2.10 and 3.1.13 OpenSIPS minor versions are scheduled for release on Wednesday, December 21st (two weeks from now).  This round of minor releases contains a significant number of fixes across various modules done in the past two months (~40 fixes in total!), so make sure to schedule an upgrade and deploy it as soon as possible (especially if you are using db_flatstore with a high rate of traffic!). In preparation for the releases, we are imposing the usual freeze on any significant fixes (as complexity) on these stable branches, in order to ensure a safe window for testing in the days ahead. Happy testing, -- Liviu Chircu www.twitter.com/liviuchircu | www.opensips-solutions.com From nutxase at proton.me Wed Dec 7 17:42:27 2022 From: nutxase at proton.me (nutxase) Date: Wed, 07 Dec 2022 17:42:27 +0000 Subject: [OpenSIPS-Users] mid_registrar Message-ID: Hi Guys How can i make it that if an invite comes in and the location is not found that it will then try look again for the location before sending 404? i have tried if (!mid_registrar_lookup("location")) { sleep (5); if (!mid_registrar_lookup("location")) { t_reply(404, "Not Found"); } } Sent with [Proton Mail](https://proton.me/) secure email. -------------- next part -------------- An HTML attachment was scrubbed... URL: From adierlam at ptgi-ics.com Wed Dec 7 18:07:57 2022 From: adierlam at ptgi-ics.com (Andy Dierlam) Date: Wed, 7 Dec 2022 18:07:57 +0000 Subject: [OpenSIPS-Users] mid_registrar In-Reply-To: References: Message-ID: Try adding xlog entries after your lookups, to verify its hitting where you think it is. separate to two blocks, and add xlog to see if is hitting them both. if (!mid_registrar_lookup("location")) { xlog("did not find phone 1st lookup"); } if (!mid_registrar_lookup("location")) { xlog("2nd lookup did not find phone, sending 404"); t_reply(404, "Not Found"); } ________________________________ From: Users on behalf of nutxase via Users Sent: Wednesday, December 7, 2022 12:42 PM To: OpenSIPS users mailling list Subject: [OpenSIPS-Users] mid_registrar EXTERNAL: Caution, this email originated from outside the organization. Do not click links or open attachments unless you can verify the sender and know the content is safe. Hi Guys How can i make it that if an invite comes in and the location is not found that it will then try look again for the location before sending 404? i have tried if (!mid_registrar_lookup("location")) { sleep (5); if (!mid_registrar_lookup("location")) { t_reply(404, "Not Found"); } } Sent with Proton Mail secure email. CONFIDENTIALITY NOTICE This email is solely for the use of the intended recipient(s) and may contain information and attachments that are confidential, privileged or otherwise sensitive in nature. Please do not disseminate this email or any of its attachments without the permission of the sender. If this email has been received in error, please do not read, copy, use, forward or disclose the email or any of its attachments to others. Instead, immediately notify the sender by replying to this email and then delete it from your system. Transmission of this email is not to be construed as a waiver of any rights of confidentiality or applicable privilege. NO BINDING AGREEMENT This email does not constitute an agreement to conduct transactions by electronic means and does not create any legally binding contract or enforceable obligation in the absence of a fully signed written contract. -------------- next part -------------- An HTML attachment was scrubbed... URL: From john.quick at smartvox.co.uk Wed Dec 7 18:08:27 2022 From: john.quick at smartvox.co.uk (John Quick) Date: Wed, 7 Dec 2022 18:08:27 -0000 Subject: [OpenSIPS-Users] Inbound PSTN connection failure Message-ID: <000401d90a66$eb8b7ef0$c2a27cd0$@smartvox.co.uk> I think there may be a simpler explanation - the logic looks incorrect in this line: if ( !(is_method("REGISTER") || is_from_gw() ) ) { That will result in a "false" result if the request came from a known gateway. I'm not certain what the test is meant to do, but shouldn't it be more like this: if ( !is_method("REGISTER") && is_from_gw() ) { John Quick Smartvox Limited Web: www.smartvox.co.uk From nutxase at proton.me Wed Dec 7 18:09:22 2022 From: nutxase at proton.me (nutxase) Date: Wed, 07 Dec 2022 18:09:22 +0000 Subject: [OpenSIPS-Users] mid_registrar In-Reply-To: References: Message-ID: i did try and doesnt seem to go to the second one :( Sent with [Proton Mail](https://proton.me/) secure email. ------- Original Message ------- On Wednesday, December 7th, 2022 at 5:42 PM, nutxase via Users wrote: > Hi Guys > > How can i make it that if an invite comes in and the location is not found that it will then try look again for the location before sending 404? > > i have tried > > if (!mid_registrar_lookup("location")) { > sleep (5); > if (!mid_registrar_lookup("location")) { > t_reply(404, "Not Found"); > } > } > > Sent with [Proton Mail](https://proton.me/) secure email. -------------- next part -------------- An HTML attachment was scrubbed... URL: From adierlam at ptgi-ics.com Wed Dec 7 18:20:43 2022 From: adierlam at ptgi-ics.com (Andy Dierlam) Date: Wed, 7 Dec 2022 18:20:43 +0000 Subject: [OpenSIPS-Users] mid_registrar In-Reply-To: References: Message-ID: maybe run in debug foreground, so can see what its doing? opensips -D ________________________________ From: Users on behalf of nutxase via Users Sent: Wednesday, December 7, 2022 1:09 PM To: nutxase ; OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] mid_registrar EXTERNAL: Caution, this email originated from outside the organization. Do not click links or open attachments unless you can verify the sender and know the content is safe. i did try and doesnt seem to go to the second one :( Sent with Proton Mail secure email. ------- Original Message ------- On Wednesday, December 7th, 2022 at 5:42 PM, nutxase via Users wrote: Hi Guys How can i make it that if an invite comes in and the location is not found that it will then try look again for the location before sending 404? i have tried if (!mid_registrar_lookup("location")) { sleep (5); if (!mid_registrar_lookup("location")) { t_reply(404, "Not Found"); } } Sent with Proton Mail secure email. CONFIDENTIALITY NOTICE This email is solely for the use of the intended recipient(s) and may contain information and attachments that are confidential, privileged or otherwise sensitive in nature. Please do not disseminate this email or any of its attachments without the permission of the sender. If this email has been received in error, please do not read, copy, use, forward or disclose the email or any of its attachments to others. Instead, immediately notify the sender by replying to this email and then delete it from your system. Transmission of this email is not to be construed as a waiver of any rights of confidentiality or applicable privilege. NO BINDING AGREEMENT This email does not constitute an agreement to conduct transactions by electronic means and does not create any legally binding contract or enforceable obligation in the absence of a fully signed written contract. -------------- next part -------------- An HTML attachment was scrubbed... URL: From barryo.williams at gmail.com Wed Dec 7 19:20:43 2022 From: barryo.williams at gmail.com (Barry O. Williams) Date: Wed, 7 Dec 2022 14:20:43 -0500 Subject: [OpenSIPS-Users] Inbound PSTN connection failure In-Reply-To: References: Message-ID: Hi Bogdan, I have reloaded Drouting, and verified the gateways below. (opensips-cli): mi dr_gw_status { "Gateways": [ { "ID": "skyetel-NE", "IP": "52.60.138.31", "State": "Active" }, { "ID": "skyetel-NW", "IP": "52.41.52.34", "State": "Active" }, { "ID": "skyetel-SE", "IP": "50.17.48.216", "State": "Active" }, { "ID": "skyetel-SW", "IP": "52.8.201.128", "State": "Active" } ] } On Tue, Dec 6, 2022 at 6:18 AM Bogdan-Andrei Iancu wrote: > Hi Barry, > > It looks like the is_from_gw() does not match for the OPTIONS. Have you > reloaded the Drouting data after adding the GW ? run "opensips-cli -x mi > dr_gw_status" to see what opensips knows as GWs and if they match the > source IP of that OPTIONS > > https://opensips.org/html/docs/modules/3.2.x/drouting.html#idp5947120 > > Regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > OpenSIPS Bootcamp 5-16 Dec 2022, online > https://www.opensips.org/training/OpenSIPS_eBootcamp_2022/ > > On 11/27/22 6:03 PM, Barry O. Williams wrote: > > Hi, > > I have installed Opensips successfully, have registered SIP stations and I > am able to make station to station calls. I have setup IP authentication > connection with provider Sketel and unable to connect from Opensips side. > Skyetel is sending OPTIONS request and Opensips is replying with 403 Relay > Forbidden/. Gateways have been added in Opensips CP. > > > if ( !(is_method("REGISTER") || is_from_gw() ) ) { > > if (is_from_local()) { > # authenticate if from local subscriber > # authenticate all initial non-REGISTER request > that pretend to be > # generated by local subscriber (domain from FROM > URI is local) > if (!proxy_authorize("", "subscriber")) { > proxy_challenge("", "auth"); > exit; > } > if ($au!=$fU) { > send_reply(403,"Forbidden auth ID"); > exit; > } > > consume_credentials(); > # caller authenticated > > } else { > # if caller is not local, then called number must > be local > > if (!is_uri_host_local()) { > send_reply(403,"Relay Forbidden"); > exit; > > Regards, > Barry > > _______________________________________________ > Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From barryo.williams at gmail.com Wed Dec 7 19:29:51 2022 From: barryo.williams at gmail.com (Barry O. Williams) Date: Wed, 7 Dec 2022 14:29:51 -0500 Subject: [OpenSIPS-Users] Inbound PSTN connection failure In-Reply-To: <000401d90a66$eb8b7ef0$c2a27cd0$@smartvox.co.uk> References: <000401d90a66$eb8b7ef0$c2a27cd0$@smartvox.co.uk> Message-ID: Hi John, I modified the configuration script by adding "&&" and it throws a parse error, bad config file error when trying to restart Opensips. The script in my configuration is the Residential script I generated. On Wed, Dec 7, 2022 at 1:08 PM John Quick wrote: > I think there may be a simpler explanation - the logic looks incorrect in > this line: > if ( !(is_method("REGISTER") || is_from_gw() ) ) { > > That will result in a "false" result if the request came from a known > gateway. I'm not certain what the test is meant to do, but shouldn't it be > more like this: > if ( !is_method("REGISTER") && is_from_gw() ) { > > John Quick > Smartvox Limited > Web: www.smartvox.co.uk > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From medeanwz at gmail.com Thu Dec 8 11:43:02 2022 From: medeanwz at gmail.com (M S) Date: Thu, 8 Dec 2022 12:43:02 +0100 Subject: [OpenSIPS-Users] mediaproxy Message-ID: I hope this is the right forum for this post, a few things about mediaproxy :) 1. the keepalive disconnects relay from dispatcher with: 'missed 3 keepalive answers in a row. assuming the connection is down.' however the logic behind it (_queued_keepalives += 1 for ping and _queued_keepalives -= 1 for pong) means that lost packets are accumulated, and it is not disconnect after 3 "in a row". I believe it should be _queued_keepalives=0 for pong... otherwise if i lose a ping today and one tomorrow, the next loss disconnects whenever it is. 2. at around 1300 calls (with high CPS), opensips channel to mediadispatcher gets timed out (even with timeout to 1s from 500ms default). further checks showed it is due to relay not answering. separating relay and dispatcher to different machines alleviates the problem. I'm sure this depends on the environment used (cpu/etc) but I'm wondering if other people have different performance results. I have disabled all database and statistics on mediaproxy. 3. when there is a problem (e.g. severed connection between relay and dispatcher) all the open ports on relay remain opened inside OS (netstat shows udp ports inuse) - port leaking. after a few disconnects all available ports are used and it starts throwing errors. Ideas and comments would be appreciated! -------------- next part -------------- An HTML attachment was scrubbed... URL: From ross at rsmck.co.uk Thu Dec 8 11:52:34 2022 From: ross at rsmck.co.uk (Ross McKillop) Date: Thu, 8 Dec 2022 11:52:34 +0000 Subject: [OpenSIPS-Users] b2b_logic BYE processing Message-ID: Hi, I’ve used OpenSIPS for a while but new to the b2b modules. I’m probably missing something really obvious but the BYE from the B leg is not propagated to the A leg when using my custom scenario (it *does* work when simply using the ‘top hiding’ config) My configuration sets the following on the initial INVITE; b2b_server_new(“A"); b2b_client_new(“B", $var(b2b_dest)); b2b_init_request(“b2b_test", "", “b2b_req", "b2b_reply"); Then I have the following in b2b_req and b2b_reply route[b2b_req] { if ($rm == 'BYE') { b2b_send_reply(200, "OK"); b2b_delete_entity(); exit; } b2b_pass_request(); exit; } route[b2b_reply] { b2b_handle_reply(); } I’ve also tried without the ($rm == ‘BYE’) block but the same behaviour. What do I need to call within that to propagate a BYE to the A leg? Best, Ross From ross at rsmck.co.uk Thu Dec 8 11:55:19 2022 From: ross at rsmck.co.uk (Ross McKillop) Date: Thu, 8 Dec 2022 11:55:19 +0000 Subject: [OpenSIPS-Users] b2b_logic BYE processing Message-ID: Hi, I’ve used OpenSIPS for a while but new to the b2b modules. I’m probably missing something really obvious but the BYE from the B leg is not propagated to the A leg when using my custom scenario (it *does* work when simply using the ‘top hiding’ config) My configuration sets the following on the initial INVITE; b2b_server_new(“A"); b2b_client_new(“B", $var(b2b_dest)); b2b_init_request(“b2b_test", "", “b2b_req", "b2b_reply"); Then I have the following in b2b_req and b2b_reply route[b2b_req] { if ($rm == 'BYE') { b2b_send_reply(200, "OK"); b2b_delete_entity(); exit; } b2b_pass_request(); exit; } route[b2b_reply] { b2b_handle_reply(); } I’ve also tried without the ($rm == ‘BYE’) block but the same behaviour. What do I need to call within that to propagate a BYE to the A leg? Best, Ross -------------- next part -------------- An HTML attachment was scrubbed... URL: From denys.pozniak at gmail.com Mon Dec 12 08:59:05 2022 From: denys.pozniak at gmail.com (Denys Pozniak) Date: Mon, 12 Dec 2022 10:59:05 +0200 Subject: [OpenSIPS-Users] Missing package opensips-b2bua-module in the repository for CentOS 7 for OpenSIPS v3.3. Message-ID: Hello! It seems no CentOS 7 package opensips-b2bua-module for OpenSIPS 3.3. There is: https://yum.opensips.org/3.2/releases/el/7/x86_64/ opensips-b2bua-module-3.2.0-1.el7.x86_64.rpm But not here: https://yum.opensips.org/3.3/releases/el/7/x86_64/ -- BR, Denys Pozniak -------------- next part -------------- An HTML attachment was scrubbed... URL: From nick at altmann.pro Mon Dec 12 11:13:42 2022 From: nick at altmann.pro (Nick Altmann) Date: Mon, 12 Dec 2022 12:13:42 +0100 Subject: [OpenSIPS-Users] Missing package opensips-b2bua-module in the repository for CentOS 7 for OpenSIPS v3.3. In-Reply-To: References: Message-ID: Hello, Denys. In 3.3 these modules are included into the main package. -- Nick Hello! > > It seems no CentOS 7 package opensips-b2bua-module for OpenSIPS 3.3. > There is: > https://yum.opensips.org/3.2/releases/el/7/x86_64/ > opensips-b2bua-module-3.2.0-1.el7.x86_64.rpm > > > But not here: > https://yum.opensips.org/3.3/releases/el/7/x86_64/ > > -- > > BR, > Denys Pozniak > > > _______________________________________________ > 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 denys.pozniak at gmail.com Mon Dec 12 13:32:38 2022 From: denys.pozniak at gmail.com (Denys Pozniak) Date: Mon, 12 Dec 2022 15:32:38 +0200 Subject: [OpenSIPS-Users] Missing package opensips-b2bua-module in the repository for CentOS 7 for OpenSIPS v3.3. In-Reply-To: References: Message-ID: Thank you! I must have missed the announcement about this ... пн, 12 дек. 2022 г. в 13:18, Nick Altmann : > Hello, Denys. > > In 3.3 these modules are included into the main package. > > -- > Nick > > Hello! >> >> It seems no CentOS 7 package opensips-b2bua-module for OpenSIPS 3.3. >> There is: >> https://yum.opensips.org/3.2/releases/el/7/x86_64/ >> opensips-b2bua-module-3.2.0-1.el7.x86_64.rpm >> >> >> But not here: >> https://yum.opensips.org/3.3/releases/el/7/x86_64/ >> >> -- >> >> BR, >> Denys Pozniak >> >> >> _______________________________________________ >> 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 > -- BR, Denys Pozniak -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Tue Dec 13 08:04:16 2022 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Tue, 13 Dec 2022 10:04:16 +0200 Subject: [OpenSIPS-Users] Inbound PSTN connection failure In-Reply-To: References: Message-ID: Hi, And for the call you are trying to accepts, what is the source IP of the INVITE ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com OpenSIPS Bootcamp 5-16 Dec 2022, online https://www.opensips.org/training/OpenSIPS_eBootcamp_2022/ On 12/7/22 9:20 PM, Barry O. Williams wrote: > Hi Bogdan, > > I have reloaded Drouting, and verified the gateways below. > > (opensips-cli): mi dr_gw_status > { >     "Gateways": [ >         { >             "ID": "skyetel-NE", >             "IP": "52.60.138.31", >             "State": "Active" >         }, >         { >             "ID": "skyetel-NW", >             "IP": "52.41.52.34", >             "State": "Active" >         }, >         { >             "ID": "skyetel-SE", >             "IP": "50.17.48.216", >             "State": "Active" >         }, >         { >             "ID": "skyetel-SW", >             "IP": "52.8.201.128", >             "State": "Active" >         } >     ] > } > > On Tue, Dec 6, 2022 at 6:18 AM Bogdan-Andrei Iancu > > wrote: > > Hi Barry, > > It looks like the is_from_gw() does not match for the OPTIONS. > Have you reloaded the Drouting data after adding the GW ? run > "opensips-cli -x mi dr_gw_status" to see what opensips knows as > GWs and if they match the source IP of that OPTIONS > > https://opensips.org/html/docs/modules/3.2.x/drouting.html#idp5947120 > > > Regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > OpenSIPS Bootcamp 5-16 Dec 2022, online > https://www.opensips.org/training/OpenSIPS_eBootcamp_2022/ > > On 11/27/22 6:03 PM, Barry O. Williams wrote: >> Hi, >> >> I have installed Opensips successfully, have registered SIP >> stations and I am able to make station to station calls. I have >> setup IP authentication connection with provider Sketel and >> unable to connect from Opensips side. Skyetel is sending OPTIONS >> request and Opensips is replying with 403 Relay Forbidden/. >> Gateways have been added in Opensips CP. >> >> >>  if ( !(is_method("REGISTER")  || is_from_gw() ) ) { >> >>                 if (is_from_local()) { >>                         # authenticate if from local subscriber >>                         # authenticate all initial non-REGISTER >> request that pretend to be >>                         # generated by local subscriber (domain >> from FROM URI is local) >>                         if (!proxy_authorize("", "subscriber")) { >>                                 proxy_challenge("", "auth"); >>                                 exit; >>                         } >>                         if ($au!=$fU) { >> send_reply(403,"Forbidden auth ID"); >>                                 exit; >>                         } >> >>                         consume_credentials(); >>                         # caller authenticated >> >>                 } else { >>                         # if caller is not local, then called >> number must be local >> >>                         if (!is_uri_host_local()) { >> send_reply(403,"Relay Forbidden"); >>                                 exit; >> >> Regards, >> Barry >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -------------- next part -------------- An HTML attachment was scrubbed... URL: From kingsley at dns99.co.uk Tue Dec 13 10:52:50 2022 From: kingsley at dns99.co.uk (Kingsley Tart) Date: Tue, 13 Dec 2022 10:52:50 +0000 Subject: [OpenSIPS-Users] disabling / filtering some logs Message-ID: Hi, Is it possible to filter what gets logged into syslog, with more fine grained detail than just setting the log level? I have essentially this code: if (!validate_dialog()) { if ($rc == -4) { send_reply(481, "Call does not exist"); exit; } else if (!fix_route_dialog()) { xlog("L_WARN", "$cfg_file/$cfg_line @ $Tsm: failed to fix route dialog; rc=$rc\n"); send_reply(488, "Unable to fix dialog"); exit; } I'm getting loads of entries in opensips.log where validate_dialog() fails with rc -2 but the call still works because fix_route_dialog() is successful, eg: ERROR:dialog:dlg_validate_dialog: failed to validate remote contact: dlg=[sip:65.151.23.22:5060] , req=[sip:65.151.23.22:5060;user=phone] I would prefer to not have opensips.log littered with these, but still want to see other errors. Is this possible? Cheers, Kingsley. From andreigrav at gmail.com Tue Dec 13 11:08:28 2022 From: andreigrav at gmail.com (Andrei G.) Date: Tue, 13 Dec 2022 13:08:28 +0200 Subject: [OpenSIPS-Users] special characters on password field Message-ID: Hello, I'm trying to add a user with password containing % character: ~# opensips-cli -i opensips-1 -f /etc/opensips-cli.cfg -x user password 1234 at sip.domain.com '%1234' ERROR: cannot execute query: UPDATE subscriber SET ha1 = '1460228c41f725017708f4f684b7edb1', password = '%1234', ha1b = '580f8f81d95fcec5e7044f9aba97868a' WHERE username = '1234' AND domain = ' sip.domain.com' ERROR: (_mysql_exceptions.ProgrammingError) not enough arguments for format string [SQL: UPDATE subscriber SET ha1 = '1460228c41f725017708f4f684b7edb1', password = '%1234', ha1b = '580f8f81d95fcec5e7044f9aba97868a' WHERE username = '1234' AND domain = ' sip.domain.com'] (Background on this error at: http://sqlalche.me/e/f405) Please advice Thanks Andrei -------------- next part -------------- An HTML attachment was scrubbed... URL: From alberto.rinaudo at gmail.com Tue Dec 13 23:50:30 2022 From: alberto.rinaudo at gmail.com (Alberto) Date: Tue, 13 Dec 2022 23:50:30 +0000 Subject: [OpenSIPS-Users] mid_registrar and vpn and nat Message-ID: Hi, I have a problem with a vpn setup, not really nat, or maybe it's nat... The setup I have is: Phone - openvpn - opensips - public internet - asterisk openvpn subnet: 10.0.0.0/24 openvpn server: 172.172.0.2/24 opensips server: 172.172.0.10/24 In my configuration I have: loadmodule "mid_registrar.so" modparam("mid_registrar", "max_contacts", 1) modparam("mid_registrar", "mode", 0) modparam("mid_registrar", "received_avp", "$avp(rcv)") modparam("mid_registrar", "tcp_persistent_flag", "TCP_PERSIST_REGISTRATIONS") if (nat_uac_test(119)) { setbflag("NAT"); if (is_method("REGISTER")) { fix_nated_register(); } else { fix_nated_contact(); } } if (is_method("REGISTER")) { mid_registrar_save("location", "p0"); ..... } if ($(tu{uri.param,ctid}) != NULL && mid_registrar_lookup("location")) { if (!t_relay()) { send_reply(500, "Internal Error"); } } So, when the phone sends a REGISTER to opensips, the contact header contains the private ip of the vpn (10.0.0.X/24). fix_nated_register is able to detect that the ip in the contact is different from the source ip and sets the received avp. But if I do opensips-cli -x mi ul_dump, I see that the contact still contains the openvpn ip, and the received field contains the correct openvpn ip. This is a problem because when I do the mid_registrar_lookup, $ru is set to the contact, which contains an ip that the opensips server is not aware of. It should instead contain the openvpn server ip (172.172.0.2), that could then route the call to the phone. I tried to change $rd manually, but that just breaks routing for ACK messages. What am I doing wrong? Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: From aviator.nitesh.d at gmail.com Wed Dec 14 22:01:23 2022 From: aviator.nitesh.d at gmail.com (Nitesh Divecha) Date: Wed, 14 Dec 2022 17:01:23 -0500 Subject: [OpenSIPS-Users] - OpenSIPS-CP Message-ID: Hello All, I'm setting up a fresh install and running into issues with OpenSIPS Control Panel 9.3.2. System specs: Ubuntu 22.04.1 LTS OpenSIPS 3.3.2 OpenSIPS Control Panel ( https://github.com/OpenSIPS/opensips-cp/archive/9.3.2.zip) CP was installed successfully and I was able to login with user/pass but when I click on (Users -> User management or Alias management) or any (System -> Address, etc) it throws PHP exceptions: [Wed Dec 14 21:49:29.216426 2022] [php:error] [pid 32857] [client 10.0.0.34:57311] PHP Fatal error: Uncaught Error: Attempt to assign property "user_management" on null in /var/www/html/opensips-cp/config/tools/users/user_management/settings.inc.php:25\nStack trace:\n#0 /var/www/html/opensips-cp/web/common/cfg_comm.php(203): require()\n#1 /var/www/html/opensips-cp/web/common/cfg_comm.php(340): get_params_from_tool()\n#2 /var/www/html/opensips-cp/web/common/cfg_comm.php(332): session_load_from_tool()\n#3 /var/www/html/opensips-cp/web/tools/users/user_management/template/header.php(32): session_load()\n#4 /var/www/html/opensips-cp/web/tools/users/user_management/user_management.php(24): require('...')\n#5 {main}\n thrown in /var/www/html/opensips-cp/config/tools/users/user_management/settings.inc.php on line 25, referer: http://10.0.0.200/cp/menu.php [Wed Dec 14 21:49:47.057497 2022] [php:warn] [pid 32858] [client 10.0.0.34:57323] PHP Warning: Undefined array key "nodes" in /var/www/html/opensips-cp/config/boxes.load.php on line 50, referer: http://10.0.0.200/cp/menu.php [Wed Dec 14 21:49:47.057553 2022] [php:warn] [pid 32858] [client 10.0.0.34:57323] PHP Warning: Undefined array key "nodes" in /var/www/html/opensips-cp/config/boxes.load.php on line 50, referer: http://10.0.0.200/cp/menu.php [Wed Dec 14 21:49:47.057564 2022] [php:warn] [pid 32858] [client 10.0.0.34:57323] PHP Warning: Undefined array key "nodes" in /var/www/html/opensips-cp/config/boxes.load.php on line 50, referer: http://10.0.0.200/cp/menu.php [Wed Dec 14 21:49:47.057629 2022] [php:error] [pid 32858] [client 10.0.0.34:57323] PHP Fatal error: Uncaught Error: Attempt to assign property "alias_management" on null in /var/www/html/opensips-cp/config/tools/users/alias_management/settings.inc.php:24\nStack trace:\n#0 /var/www/html/opensips-cp/web/common/cfg_comm.php(203): require()\n#1 /var/www/html/opensips-cp/web/common/cfg_comm.php(340): get_params_from_tool()\n#2 /var/www/html/opensips-cp/web/common/cfg_comm.php(332): session_load_from_tool()\n#3 /var/www/html/opensips-cp/web/tools/users/alias_management/template/header.php(33): session_load()\n#4 /var/www/html/opensips-cp/web/tools/users/alias_management/alias_management.php(24): require('...')\n#5 {main}\n thrown in /var/www/html/opensips-cp/config/tools/users/alias_management/settings.inc.php on line 24, referer: http://10.0.0.200/cp/menu.php [Wed Dec 14 21:49:52.209761 2022] [php:warn] [pid 32859] [client 10.0.0.34:57330] PHP Warning: Undefined array key "nodes" in /var/www/html/opensips-cp/config/boxes.load.php on line 50, referer: http://10.0.0.200/cp/menu.php [Wed Dec 14 21:49:52.209814 2022] [php:warn] [pid 32859] [client 10.0.0.34:57330] PHP Warning: Undefined array key "nodes" in /var/www/html/opensips-cp/config/boxes.load.php on line 50, referer: http://10.0.0.200/cp/menu.php [Wed Dec 14 21:49:52.209821 2022] [php:warn] [pid 32859] [client 10.0.0.34:57330] PHP Warning: Undefined array key "nodes" in /var/www/html/opensips-cp/config/boxes.load.php on line 50, referer: http://10.0.0.200/cp/menu.php [Wed Dec 14 21:49:52.209920 2022] [php:error] [pid 32859] [client 10.0.0.34:57330] PHP Fatal error: Uncaught Error: Attempt to assign property "addresses" on null in /var/www/html/opensips-cp/config/tools/system/addresses/settings.inc.php:24\nStack trace:\n#0 /var/www/html/opensips-cp/web/common/cfg_comm.php(203): require()\n#1 /var/www/html/opensips-cp/web/common/cfg_comm.php(340): get_params_from_tool()\n#2 /var/www/html/opensips-cp/web/common/cfg_comm.php(332): session_load_from_tool()\n#3 /var/www/html/opensips-cp/web/tools/system/addresses/template/header.php(32): session_load()\n#4 /var/www/html/opensips-cp/web/tools/system/addresses/address.php(24): require('...')\n#5 {main}\n thrown in /var/www/html/opensips-cp/config/tools/system/addresses/settings.inc.php on line 24, referer: http://10.0.0.200/cp/menu.php Any thoughts or suggestions? Thanks in advance. Cheers, Nitesh -------------- next part -------------- An HTML attachment was scrubbed... URL: From nutxase at proton.me Thu Dec 15 16:35:14 2022 From: nutxase at proton.me (nutxase) Date: Thu, 15 Dec 2022 16:35:14 +0000 Subject: [OpenSIPS-Users] wss and tls Message-ID: Hi All I am trying to get tls working with my letsencrypt cert but i keep getting this error ERROR:tls_openssl:openssl_tls_accept: SSL_ERROR_SYSCALL err=Success(0) ERROR:tls_openssl:openssl_tls_accept: New TLS connection from xxxx:47817 failed to accept Dec 15 16:32:54 [localhost] /usr/sbin/opensips[4373]: ERROR:proto_wss:wss_read_req: cannot fix read connection my config is as follows loadmodule "tls_openssl.so" modparam("tls_mgm", "server_domain", "sip") modparam("tls_mgm", "ca_list", "[sip]/etc/letsencrypt/fullchain.pem") modparam("tls_mgm", "certificate", "[sip]/etc/opensips/tls/cert.pem") modparam("tls_mgm", "private_key", "[sip]/etc/opensips/tls/ckey.pem") modparam("tls_mgm", "require_cert", "[sip]0") modparam("tls_mgm", "tls_method", "[sip]TLSv1") modparam("tls_mgm", "verify_cert", "[sip]0") modparam("tls_mgm", "match_sip_domain", "[sip]*") modparam("tls_mgm", "match_ip_address", "[sip]*") modparam("tls_mgm", "client_domain", "sip1") modparam("tls_mgm", "ca_list", "[sip1]/etc/letsencrypt/fullchain.pem") modparam("tls_mgm", "certificate", "[sip1]/etc/opensips/tls/cert.pem") modparam("tls_mgm", "private_key", "[sip1]/etc/opensips/tls/ckey.pem") modparam("tls_mgm", "require_cert", "[sip1]0") modparam("tls_mgm", "tls_method", "[sip1]TLSv1") modparam("tls_mgm", "verify_cert", "[sip1]0") modparam("tls_mgm", "match_sip_domain", "[sip]*") modparam("tls_mgm", "match_ip_address", "[sip]*") loadmodule "proto_wss.so" modparam("proto_wss", "require_origin", no) loadmodule "proto_ws.so" modparam("proto_ws", "require_origin", no) i have tried wolfssl aswell any ideas :( Sent with [Proton Mail](https://proton.me/) secure email. -------------- next part -------------- An HTML attachment was scrubbed... URL: From aviator.nitesh.d at gmail.com Thu Dec 15 16:52:00 2022 From: aviator.nitesh.d at gmail.com (Nitesh Divecha) Date: Thu, 15 Dec 2022 11:52:00 -0500 Subject: [OpenSIPS-Users] - OpenSIPS-CP In-Reply-To: References: Message-ID: Hello All, Solved this issue by downgrading from PHP8.1 to PHP7.4. On Wed, Dec 14, 2022 at 5:01 PM Nitesh Divecha wrote: > Hello All, > > I'm setting up a fresh install and running into issues with OpenSIPS > Control Panel 9.3.2. > > System specs: > Ubuntu 22.04.1 LTS > OpenSIPS 3.3.2 > OpenSIPS Control Panel ( > https://github.com/OpenSIPS/opensips-cp/archive/9.3.2.zip) > > CP was installed successfully and I was able to login with user/pass but > when I click on (Users -> User management or Alias management) or any > (System -> Address, etc) it throws PHP exceptions: > > [Wed Dec 14 21:49:29.216426 2022] [php:error] [pid 32857] [client > 10.0.0.34:57311] PHP Fatal error: Uncaught Error: Attempt to assign > property "user_management" on null in > /var/www/html/opensips-cp/config/tools/users/user_management/settings.inc.php:25\nStack > trace:\n#0 /var/www/html/opensips-cp/web/common/cfg_comm.php(203): > require()\n#1 /var/www/html/opensips-cp/web/common/cfg_comm.php(340): > get_params_from_tool()\n#2 > /var/www/html/opensips-cp/web/common/cfg_comm.php(332): > session_load_from_tool()\n#3 > /var/www/html/opensips-cp/web/tools/users/user_management/template/header.php(32): > session_load()\n#4 > /var/www/html/opensips-cp/web/tools/users/user_management/user_management.php(24): > require('...')\n#5 {main}\n thrown in > /var/www/html/opensips-cp/config/tools/users/user_management/settings.inc.php > on line 25, referer: http://10.0.0.200/cp/menu.php > [Wed Dec 14 21:49:47.057497 2022] [php:warn] [pid 32858] [client > 10.0.0.34:57323] PHP Warning: Undefined array key "nodes" in > /var/www/html/opensips-cp/config/boxes.load.php on line 50, referer: > http://10.0.0.200/cp/menu.php > [Wed Dec 14 21:49:47.057553 2022] [php:warn] [pid 32858] [client > 10.0.0.34:57323] PHP Warning: Undefined array key "nodes" in > /var/www/html/opensips-cp/config/boxes.load.php on line 50, referer: > http://10.0.0.200/cp/menu.php > [Wed Dec 14 21:49:47.057564 2022] [php:warn] [pid 32858] [client > 10.0.0.34:57323] PHP Warning: Undefined array key "nodes" in > /var/www/html/opensips-cp/config/boxes.load.php on line 50, referer: > http://10.0.0.200/cp/menu.php > [Wed Dec 14 21:49:47.057629 2022] [php:error] [pid 32858] [client > 10.0.0.34:57323] PHP Fatal error: Uncaught Error: Attempt to assign > property "alias_management" on null in > /var/www/html/opensips-cp/config/tools/users/alias_management/settings.inc.php:24\nStack > trace:\n#0 /var/www/html/opensips-cp/web/common/cfg_comm.php(203): > require()\n#1 /var/www/html/opensips-cp/web/common/cfg_comm.php(340): > get_params_from_tool()\n#2 > /var/www/html/opensips-cp/web/common/cfg_comm.php(332): > session_load_from_tool()\n#3 > /var/www/html/opensips-cp/web/tools/users/alias_management/template/header.php(33): > session_load()\n#4 > /var/www/html/opensips-cp/web/tools/users/alias_management/alias_management.php(24): > require('...')\n#5 {main}\n thrown in > /var/www/html/opensips-cp/config/tools/users/alias_management/settings.inc.php > on line 24, referer: http://10.0.0.200/cp/menu.php > [Wed Dec 14 21:49:52.209761 2022] [php:warn] [pid 32859] [client > 10.0.0.34:57330] PHP Warning: Undefined array key "nodes" in > /var/www/html/opensips-cp/config/boxes.load.php on line 50, referer: > http://10.0.0.200/cp/menu.php > [Wed Dec 14 21:49:52.209814 2022] [php:warn] [pid 32859] [client > 10.0.0.34:57330] PHP Warning: Undefined array key "nodes" in > /var/www/html/opensips-cp/config/boxes.load.php on line 50, referer: > http://10.0.0.200/cp/menu.php > [Wed Dec 14 21:49:52.209821 2022] [php:warn] [pid 32859] [client > 10.0.0.34:57330] PHP Warning: Undefined array key "nodes" in > /var/www/html/opensips-cp/config/boxes.load.php on line 50, referer: > http://10.0.0.200/cp/menu.php > [Wed Dec 14 21:49:52.209920 2022] [php:error] [pid 32859] [client > 10.0.0.34:57330] PHP Fatal error: Uncaught Error: Attempt to assign > property "addresses" on null in > /var/www/html/opensips-cp/config/tools/system/addresses/settings.inc.php:24\nStack > trace:\n#0 /var/www/html/opensips-cp/web/common/cfg_comm.php(203): > require()\n#1 /var/www/html/opensips-cp/web/common/cfg_comm.php(340): > get_params_from_tool()\n#2 > /var/www/html/opensips-cp/web/common/cfg_comm.php(332): > session_load_from_tool()\n#3 > /var/www/html/opensips-cp/web/tools/system/addresses/template/header.php(32): > session_load()\n#4 > /var/www/html/opensips-cp/web/tools/system/addresses/address.php(24): > require('...')\n#5 {main}\n thrown in > /var/www/html/opensips-cp/config/tools/system/addresses/settings.inc.php on > line 24, referer: http://10.0.0.200/cp/menu.php > > Any thoughts or suggestions? > > Thanks in advance. > > Cheers, > Nitesh > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From ter.devor at gmail.com Sun Dec 18 17:50:53 2022 From: ter.devor at gmail.com (Terrance Devor) Date: Sun, 18 Dec 2022 12:50:53 -0500 Subject: [OpenSIPS-Users] RTPProxy Docker Image Message-ID: Hello Everyone, Wow! Blast from the past... I am a long time member of this list, been a while. Question, anyone successful in deploying RTPProxy to a dockerized environment? Preferably to a Kubernetes managed environment. Please Help Team :) Kind Regards, Terrance -------------- next part -------------- An HTML attachment was scrubbed... URL: From anubhav.singh at acquirelogix.com Sun Dec 18 19:28:49 2022 From: anubhav.singh at acquirelogix.com (Anubhav Singh) Date: Mon, 19 Dec 2022 00:58:49 +0530 Subject: [OpenSIPS-Users] Help to Make Call to A Gateway Message-ID: Hi, I am new to opensips. Can you please help to create dialplan and routing to make call with prefix '0' though gateway. Regards, Anuhav Singh -------------- next part -------------- An HTML attachment was scrubbed... URL: From wadii at evenmedia.fr Mon Dec 19 14:43:36 2022 From: wadii at evenmedia.fr (Wadii ELMAJDI | Evenmedia) Date: Mon, 19 Dec 2022 14:43:36 +0000 Subject: [OpenSIPS-Users] REPLICATION / SQL_CACHER / Local CacheDB Collection Message-ID: Hello , I am trying to cache an entire MySQL table containing CLIDs/Clients Mapping. My use case is to check pre every call , if the $fU used belongs to the client (via its ip source). I am using SQL CACHER (ondemand = 0) mode module, my opensips box is clustered. the cached_url for SQL_CACHER is a local collection (cachedb_url=local:///clid) I am getting an error that I cannot use an opensips replicatedcache db : Dec 19 15:31:18 primary /usr/sbin/opensips[41751]: ERROR:sql_cacher:db_init_test_conn: Cannot use an OpenSIPS replicated cached Dec 19 15:31:18 primary /usr/sbin/opensips[41751]: ERROR:sql_cacher:mod_init: Failed to validate db conns for cache entry Dec 19 15:31:18 primary /usr/sbin/opensips[41751]: ERROR:core:init_mod: failed to initialize module sql_cacher Dec 19 15:31:18 primary /usr/sbin/opensips[41751]: ERROR:core:main: error while initializing modules I can use cachedb_local module elsewhere, all cached data is replicated via the cluster with no problem. I don’t see why it causes such problem only when using SQLCACHER. Is there anyway to exclude one specific collection from the replication mode ? is there any better approach for my use case ? I know I can use REDIS or MongoDB for such use case, but I’m trying to lower the numbers of servers to monitor 😊 Thank you -------------- next part -------------- An HTML attachment was scrubbed... URL: From wadii at evenmedia.fr Mon Dec 19 16:40:23 2022 From: wadii at evenmedia.fr (Wadii ELMAJDI | Evenmedia) Date: Mon, 19 Dec 2022 16:40:23 +0000 Subject: [OpenSIPS-Users] REPLICATION / SQL_CACHER / Local CacheDB Collection In-Reply-To: References: Message-ID: I ended up using dialplan module with a new partition dedicated to CLID. De : Users De la part de Wadii ELMAJDI | Evenmedia Envoyé : lundi 19 décembre 2022 15:44 À : OpenSIPS users mailling list Objet : [OpenSIPS-Users] REPLICATION / SQL_CACHER / Local CacheDB Collection Hello , I am trying to cache an entire MySQL table containing CLIDs/Clients Mapping. My use case is to check pre every call , if the $fU used belongs to the client (via its ip source). I am using SQL CACHER (ondemand = 0) mode module, my opensips box is clustered. the cached_url for SQL_CACHER is a local collection (cachedb_url=local:///clid) I am getting an error that I cannot use an opensips replicatedcache db : Dec 19 15:31:18 primary /usr/sbin/opensips[41751]: ERROR:sql_cacher:db_init_test_conn: Cannot use an OpenSIPS replicated cached Dec 19 15:31:18 primary /usr/sbin/opensips[41751]: ERROR:sql_cacher:mod_init: Failed to validate db conns for cache entry Dec 19 15:31:18 primary /usr/sbin/opensips[41751]: ERROR:core:init_mod: failed to initialize module sql_cacher Dec 19 15:31:18 primary /usr/sbin/opensips[41751]: ERROR:core:main: error while initializing modules I can use cachedb_local module elsewhere, all cached data is replicated via the cluster with no problem. I don’t see why it causes such problem only when using SQLCACHER. Is there anyway to exclude one specific collection from the replication mode ? is there any better approach for my use case ? I know I can use REDIS or MongoDB for such use case, but I’m trying to lower the numbers of servers to monitor 😊 Thank you -------------- next part -------------- An HTML attachment was scrubbed... URL: From liviu at opensips.org Tue Dec 20 10:06:31 2022 From: liviu at opensips.org (Liviu Chircu) Date: Tue, 20 Dec 2022 12:06:31 +0200 Subject: [OpenSIPS-Users] [Release] OpenSIPS 3.3.3, 3.2.10 and 3.1.13 minor releases In-Reply-To: References: Message-ID: <7cf31038-b6e5-edeb-e8ca-e1dbed9ef555@opensips.org> Hi, everyone! The OpenSIPS 3.3.3, 3.2.10 and 3.1.13 minor releases have just been tagged, packing a significant number of fixes across various modules done in the past two months (~40 fixes in total!), including a rare crash in "db_mysql" after a disconnect and a first failed reconnect on any connection. Detailed changelogs are available on the website^[1][2][3] . Packaging to be available within the next couple of hours. [1]: https://opensips.org/pub/opensips/3.1.13/ChangeLog [2]: https://opensips.org/pub/opensips/3.2.10/ChangeLog [3]: https://opensips.org/pub/opensips/3.3.3/ChangeLog Happy Hacking and Happy Holidays! OpenSIPS Team -------------- next part -------------- An HTML attachment was scrubbed... URL: From ter.devor at gmail.com Tue Dec 20 17:41:37 2022 From: ter.devor at gmail.com (Terrance Devor) Date: Tue, 20 Dec 2022 12:41:37 -0500 Subject: [OpenSIPS-Users] RTPProxy Docker Image In-Reply-To: References: Message-ID: Was it something I said? Terrance On Sun, Dec 18, 2022 at 12:50 PM Terrance Devor wrote: > Hello Everyone, > > Wow! Blast from the past... I am a long time member of this list, been a > while. > > Question, anyone successful in deploying RTPProxy to a dockerized > environment? Preferably to a Kubernetes managed environment. > > Please Help Team :) > > Kind Regards, > Terrance > -------------- next part -------------- An HTML attachment was scrubbed... URL: From brett at nemeroff.com Tue Dec 20 19:43:59 2022 From: brett at nemeroff.com (Brett Nemeroff) Date: Tue, 20 Dec 2022 13:43:59 -0600 Subject: [OpenSIPS-Users] RTPProxy Docker Image In-Reply-To: References: Message-ID: Hello Terrance, I wouldn't really recommend this. RTPProxy is going to use a lot of ports in a very large range. That just doesn't work great in docker, but even worse in K8S. I personally would put the RTPProxy outside of K8S. While you might be able to get it to work, you are likely going against some basic design concepts in containerization. I feel like the tech should propel the solution and not be a hindrance to it. In this case, I'm not sure that K8S is buying you anything of value, but instead creating architectural challenges. I'd love to hear feedback or experiences from others. There's always something to learn :) -Brett On Tue, Dec 20, 2022 at 11:43 AM Terrance Devor wrote: > Was it something I said? > > Terrance > > On Sun, Dec 18, 2022 at 12:50 PM Terrance Devor > wrote: > >> Hello Everyone, >> >> Wow! Blast from the past... I am a long time member of this list, been a >> while. >> >> Question, anyone successful in deploying RTPProxy to a dockerized >> environment? Preferably to a Kubernetes managed environment. >> >> Please Help Team :) >> >> Kind Regards, >> Terrance >> > _______________________________________________ > 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 Tue Dec 20 21:28:34 2022 From: venefax at gmail.com (Saint Michael) Date: Tue, 20 Dec 2022 16:28:34 -0500 Subject: [OpenSIPS-Users] RTPProxy Docker Image In-Reply-To: References: Message-ID: Opensips+ RTPProxy only works fine with plain LXC containers, privileged, which basically have access to all the resources of the box. That is the model I use with great success. On Tue, Dec 20, 2022 at 2:47 PM Brett Nemeroff wrote: > > Hello Terrance, > I wouldn't really recommend this. RTPProxy is going to use a lot of ports in a very large range. That just doesn't work great in docker, but even worse in K8S. > > I personally would put the RTPProxy outside of K8S. While you might be able to get it to work, you are likely going against some basic design concepts in containerization. I feel like the tech should propel the solution and not be a hindrance to it. In this case, I'm not sure that K8S is buying you anything of value, but instead creating architectural challenges. > > I'd love to hear feedback or experiences from others. There's always something to learn :) > -Brett > > On Tue, Dec 20, 2022 at 11:43 AM Terrance Devor wrote: >> >> Was it something I said? >> >> Terrance >> >> On Sun, Dec 18, 2022 at 12:50 PM Terrance Devor wrote: >>> >>> Hello Everyone, >>> >>> Wow! Blast from the past... I am a long time member of this list, been a while. >>> >>> Question, anyone successful in deploying RTPProxy to a dockerized environment? Preferably to a Kubernetes managed environment. >>> >>> Please Help Team :) >>> >>> Kind Regards, >>> Terrance >> >> _______________________________________________ >> 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 ross at rsmck.co.uk Wed Dec 21 11:17:18 2022 From: ross at rsmck.co.uk (Ross McKillop) Date: Wed, 21 Dec 2022 11:17:18 +0000 Subject: [OpenSIPS-Users] b2b_logic modules and transport protocol (TCP/TLS) conversion Message-ID: Hi, I’m hoping I’m missing something obvious here (and explain well enough below what the problem is) - I can get logs etc if required of course. I’m using the OpenSIPS b2bua module(s) in a scenario where one side of the leg is TCP and the other is TLS as follows. Host A —TLS---> OpenSIPS —TCP—> Host B The configuration is something like the below (variables replaced with what they contain for clarity) b2b_server_new(“s”); b2b_client_new(“c”, “sip:user at hostb.example.com:5555;transport=tcp”, “sip:hostb.example.com:5555;transport=tcp”) # have also tried the above without the proxy parameter b2b_init_request(“b2b_ab”, “”, “b2b_ab_request”, “b2b_ab_reply”); If the incoming A leg is also TCP this works as expected, however if the incoming leg is TLS the connection attempt is made to hostb.example.com on port 5061 (default TLS port, ignoring the 5555 specified in the b2b_client_new call) The INVITE RURI is correct (e.g. INVITE sip:user at hostb.example.com:5555;transport=tcp) but the Contact header contains a reference to the TLS listening port (e.g. Contact: ) As this is acting as a B2BUA I would have expected the Contact to be the TCP interface on the opensips, and the INVITE to be sent to the URI (and transport protocol) provided in the b2b_init_request in the config. I’ve also tried setting b2bl_from_spec_param as follows; modparam("b2b_logic", "b2bl_from_spec_param”,”sip:opensips.example.com:5060;transport=tcp”) However, this doesn’t change the observed behaviour. Why is the port and transport selection being ignored, or am I missing something? (All the above is based on opensips 3.3.2, if the behaviour has been changed/corrected in 3.4 then accept my apologies, I’ll be trying that shortly, but wanted to ask here first in case I was missing something obvious!) Any insight anyone can offer would be much appreciated. Best, Ross From david.villasmil.work at gmail.com Wed Dec 21 12:12:05 2022 From: david.villasmil.work at gmail.com (David Villasmil) Date: Wed, 21 Dec 2022 13:12:05 +0100 Subject: [OpenSIPS-Users] kamilio's htable equivalent? Message-ID: Hello folks, I'm trying to find in opensips an equivalent to kamailio's htable module. Opensips' cachedb_local doesn't have autoexpires... Regards, David Villasmil email: david.villasmil.work at gmail.com phone: +34669448337 -------------- next part -------------- An HTML attachment was scrubbed... URL: From daniel.zanutti at gmail.com Wed Dec 21 12:26:17 2022 From: daniel.zanutti at gmail.com (Daniel Zanutti) Date: Wed, 21 Dec 2022 09:26:17 -0300 Subject: [OpenSIPS-Users] kamilio's htable equivalent? In-Reply-To: References: Message-ID: Hey David Did you take a look at core functions of cache? -> https://www.opensips.org/Documentation/Script-CoreFunctions-3-1#toc4 On Wed, Dec 21, 2022 at 9:14 AM David Villasmil < david.villasmil.work at gmail.com> wrote: > Hello folks, > > I'm trying to find in opensips an equivalent to kamailio's htable module. > Opensips' cachedb_local doesn't have autoexpires... > > Regards, > > David Villasmil > email: david.villasmil.work at gmail.com > phone: +34669448337 > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -------------- next part -------------- An HTML attachment was scrubbed... URL: From david.villasmil.work at gmail.com Wed Dec 21 12:52:12 2022 From: david.villasmil.work at gmail.com (David Villasmil) Date: Wed, 21 Dec 2022 13:52:12 +0100 Subject: [OpenSIPS-Users] kamilio's htable equivalent? In-Reply-To: References: Message-ID: Thanks Daniel, Amazingly enough I had! But then completely forgot about it. Thanks!! On Wed, 21 Dec 2022 at 13:26, Daniel Zanutti wrote: > Hey David > > Did you take a look at core functions of cache? -> > https://www.opensips.org/Documentation/Script-CoreFunctions-3-1#toc4 > > > On Wed, Dec 21, 2022 at 9:14 AM David Villasmil < > david.villasmil.work at gmail.com> wrote: > >> Hello folks, >> >> I'm trying to find in opensips an equivalent to kamailio's htable module. >> Opensips' cachedb_local doesn't have autoexpires... >> >> Regards, >> >> David Villasmil >> email: david.villasmil.work at gmail.com >> phone: +34669448337 >> > _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -- Regards, David Villasmil email: david.villasmil.work at gmail.com phone: +34669448337 -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Wed Dec 21 16:30:52 2022 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Wed, 21 Dec 2022 18:30:52 +0200 Subject: [OpenSIPS-Users] Introducing OpenSIPS 3.4 Message-ID: <8d3ede60-a438-9b7e-5d9a-863da9215d60@opensips.org> Dear OpenSIPS'ers, We got to that time of year when we start baking a new OpenSIPS major version. An new year, a new version, a new topic to be addressed. So let me introduce you to the upcoming OpenSIPS 3.4 . For the upcoming OpenSIPS 3.4 release the main focus is on the */consolidation /*topic, from the testing and features perspectives. Shortly said, this mainly (not limited) translates into: * Performance testing and code profiling * Conformity testing with SIP scenarios * More MSRP features (chat, encryption, reports) * More IMS features (IPsec, SIP Outbound) * More Back2Back (better re-bridging ) For the full list with technical description and details, visit : https://www.opensips.org/Development/Opensips-3-4-Planning *IMPORTANT* As community is important to us and we want to align the OpenSIPS roadmap with the needs of our users, be part of the shaping and decision making for the OpenSIPS 3.4 Dev Plan via this *Feature Survey * - any feedback is important and it matters to us. Best regards and enjoy the winter holidays!! -- Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com OpenSIPS Bootcamp 5-16 Dec 2022, online https://www.opensips.org/training/OpenSIPS_eBootcamp_2022/ -------------- next part -------------- An HTML attachment was scrubbed... URL: From ter.devor at gmail.com Thu Dec 22 00:51:33 2022 From: ter.devor at gmail.com (Terrance Devor) Date: Wed, 21 Dec 2022 19:51:33 -0500 Subject: [OpenSIPS-Users] RTPProxy Docker Image In-Reply-To: References: Message-ID: Hello Bret, another approach we are thinking about is put RTPProxy on a VPC. As for Opensips and Asterisk, they can live on a k8s with the understanding that they will not deal with media directly. If anyone can share their experience, I would be interested in hearing from you. Do the RTP guys still follow this mailing list? On Tue, Dec 20, 2022 at 4:31 PM Saint Michael wrote: > Opensips+ RTPProxy only works fine with plain LXC containers, > privileged, which basically have access to all the resources of the > box. > That is the model I use with great success. > > On Tue, Dec 20, 2022 at 2:47 PM Brett Nemeroff wrote: > > > > Hello Terrance, > > I wouldn't really recommend this. RTPProxy is going to use a lot of > ports in a very large range. That just doesn't work great in docker, but > even worse in K8S. > > > > I personally would put the RTPProxy outside of K8S. While you might be > able to get it to work, you are likely going against some basic design > concepts in containerization. I feel like the tech should propel the > solution and not be a hindrance to it. In this case, I'm not sure that K8S > is buying you anything of value, but instead creating architectural > challenges. > > > > I'd love to hear feedback or experiences from others. There's always > something to learn :) > > -Brett > > > > On Tue, Dec 20, 2022 at 11:43 AM Terrance Devor > wrote: > >> > >> Was it something I said? > >> > >> Terrance > >> > >> On Sun, Dec 18, 2022 at 12:50 PM Terrance Devor > wrote: > >>> > >>> Hello Everyone, > >>> > >>> Wow! Blast from the past... I am a long time member of this list, been > a while. > >>> > >>> Question, anyone successful in deploying RTPProxy to a dockerized > environment? Preferably to a Kubernetes managed environment. > >>> > >>> Please Help Team :) > >>> > >>> Kind Regards, > >>> Terrance > >> > >> _______________________________________________ > >> 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 david.villasmil.work at gmail.com Thu Dec 22 00:54:22 2022 From: david.villasmil.work at gmail.com (David Villasmil) Date: Thu, 22 Dec 2022 01:54:22 +0100 Subject: [OpenSIPS-Users] RTPProxy Docker Image In-Reply-To: References: Message-ID: Can you explain more? I.e: params and such? Thanks! On Tue, 20 Dec 2022 at 22:29, Saint Michael wrote: > Opensips+ RTPProxy only works fine with plain LXC containers, > privileged, which basically have access to all the resources of the > box. > That is the model I use with great success. > > On Tue, Dec 20, 2022 at 2:47 PM Brett Nemeroff wrote: > > > > Hello Terrance, > > I wouldn't really recommend this. RTPProxy is going to use a lot of > ports in a very large range. That just doesn't work great in docker, but > even worse in K8S. > > > > I personally would put the RTPProxy outside of K8S. While you might be > able to get it to work, you are likely going against some basic design > concepts in containerization. I feel like the tech should propel the > solution and not be a hindrance to it. In this case, I'm not sure that K8S > is buying you anything of value, but instead creating architectural > challenges. > > > > I'd love to hear feedback or experiences from others. There's always > something to learn :) > > -Brett > > > > On Tue, Dec 20, 2022 at 11:43 AM Terrance Devor > wrote: > >> > >> Was it something I said? > >> > >> Terrance > >> > >> On Sun, Dec 18, 2022 at 12:50 PM Terrance Devor > wrote: > >>> > >>> Hello Everyone, > >>> > >>> Wow! Blast from the past... I am a long time member of this list, been > a while. > >>> > >>> Question, anyone successful in deploying RTPProxy to a dockerized > environment? Preferably to a Kubernetes managed environment. > >>> > >>> Please Help Team :) > >>> > >>> Kind Regards, > >>> Terrance > >> > >> _______________________________________________ > >> 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 > -- Regards, David Villasmil email: david.villasmil.work at gmail.com phone: +34669448337 -------------- next part -------------- An HTML attachment was scrubbed... URL: From ter.devor at gmail.com Thu Dec 22 01:03:07 2022 From: ter.devor at gmail.com (Terrance Devor) Date: Wed, 21 Dec 2022 20:03:07 -0500 Subject: [OpenSIPS-Users] RTPProxy Docker Image In-Reply-To: References: Message-ID: Hello David, Similar to what we have with LXC OpenSIPS - Proxy, Edge Switch, Managing DIDs and Termination routes, CDR, LB to Asterisk Asterisk - PBX, IVR RTPProxy - Media Relay Everything containerized using docker and deployed to our k8s cluster. I would appreciate speaking to anyone that has experience in successfully, or failed, in trying to do this On Wed, Dec 21, 2022 at 7:57 PM David Villasmil < david.villasmil.work at gmail.com> wrote: > Can you explain more? I.e: params and such? > Thanks! > > On Tue, 20 Dec 2022 at 22:29, Saint Michael wrote: > >> Opensips+ RTPProxy only works fine with plain LXC containers, >> privileged, which basically have access to all the resources of the >> box. >> That is the model I use with great success. >> >> On Tue, Dec 20, 2022 at 2:47 PM Brett Nemeroff >> wrote: >> > >> > Hello Terrance, >> > I wouldn't really recommend this. RTPProxy is going to use a lot of >> ports in a very large range. That just doesn't work great in docker, but >> even worse in K8S. >> > >> > I personally would put the RTPProxy outside of K8S. While you might be >> able to get it to work, you are likely going against some basic design >> concepts in containerization. I feel like the tech should propel the >> solution and not be a hindrance to it. In this case, I'm not sure that K8S >> is buying you anything of value, but instead creating architectural >> challenges. >> > >> > I'd love to hear feedback or experiences from others. There's always >> something to learn :) >> > -Brett >> > >> > On Tue, Dec 20, 2022 at 11:43 AM Terrance Devor >> wrote: >> >> >> >> Was it something I said? >> >> >> >> Terrance >> >> >> >> On Sun, Dec 18, 2022 at 12:50 PM Terrance Devor >> wrote: >> >>> >> >>> Hello Everyone, >> >>> >> >>> Wow! Blast from the past... I am a long time member of this list, >> been a while. >> >>> >> >>> Question, anyone successful in deploying RTPProxy to a dockerized >> environment? Preferably to a Kubernetes managed environment. >> >>> >> >>> Please Help Team :) >> >>> >> >>> Kind Regards, >> >>> Terrance >> >> >> >> _______________________________________________ >> >> 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 >> > -- > Regards, > > David Villasmil > email: david.villasmil.work at gmail.com > phone: +34669448337 > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -------------- next part -------------- An HTML attachment was scrubbed... URL: From sobomax at sippysoft.com Thu Dec 22 16:12:34 2022 From: sobomax at sippysoft.com (Maxim Sobolev) Date: Thu, 22 Dec 2022 08:12:34 -0800 Subject: [OpenSIPS-Users] RTPProxy Docker Image In-Reply-To: References: Message-ID: Liviu has done some exploration on getting things handled on Kubernetes. His great presentation is available here: https://youtu.be/JwO0UmauuT4?t=13034 -Max On Wed, Dec 21, 2022, 5:04 PM Terrance Devor wrote: > Hello David, Similar to what we have with LXC > > OpenSIPS - Proxy, Edge Switch, Managing DIDs and Termination routes, CDR, > LB to Asterisk > Asterisk - PBX, IVR > RTPProxy - Media Relay > > Everything containerized using docker and deployed to our k8s cluster. > > I would appreciate speaking to anyone that has experience in successfully, > or failed, in trying to do this > > On Wed, Dec 21, 2022 at 7:57 PM David Villasmil < > david.villasmil.work at gmail.com> wrote: > >> Can you explain more? I.e: params and such? >> Thanks! >> >> On Tue, 20 Dec 2022 at 22:29, Saint Michael wrote: >> >>> Opensips+ RTPProxy only works fine with plain LXC containers, >>> privileged, which basically have access to all the resources of the >>> box. >>> That is the model I use with great success. >>> >>> On Tue, Dec 20, 2022 at 2:47 PM Brett Nemeroff >>> wrote: >>> > >>> > Hello Terrance, >>> > I wouldn't really recommend this. RTPProxy is going to use a lot of >>> ports in a very large range. That just doesn't work great in docker, but >>> even worse in K8S. >>> > >>> > I personally would put the RTPProxy outside of K8S. While you might be >>> able to get it to work, you are likely going against some basic design >>> concepts in containerization. I feel like the tech should propel the >>> solution and not be a hindrance to it. In this case, I'm not sure that K8S >>> is buying you anything of value, but instead creating architectural >>> challenges. >>> > >>> > I'd love to hear feedback or experiences from others. There's always >>> something to learn :) >>> > -Brett >>> > >>> > On Tue, Dec 20, 2022 at 11:43 AM Terrance Devor >>> wrote: >>> >> >>> >> Was it something I said? >>> >> >>> >> Terrance >>> >> >>> >> On Sun, Dec 18, 2022 at 12:50 PM Terrance Devor >>> wrote: >>> >>> >>> >>> Hello Everyone, >>> >>> >>> >>> Wow! Blast from the past... I am a long time member of this list, >>> been a while. >>> >>> >>> >>> Question, anyone successful in deploying RTPProxy to a dockerized >>> environment? Preferably to a Kubernetes managed environment. >>> >>> >>> >>> Please Help Team :) >>> >>> >>> >>> Kind Regards, >>> >>> Terrance >>> >> >>> >> _______________________________________________ >>> >> 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 >>> >> -- >> Regards, >> >> David Villasmil >> email: david.villasmil.work at gmail.com >> phone: +34669448337 >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> > _______________________________________________ > 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 wadii at evenmedia.fr Fri Dec 23 19:42:54 2022 From: wadii at evenmedia.fr (Wadii ELMAJDI | Evenmedia) Date: Fri, 23 Dec 2022 19:42:54 +0000 Subject: [OpenSIPS-Users] RTPPROXY / OPENSIPS Message-ID: hello , i do have a question related to rtpproxy module documentation. The doc describes that rewriting sdp body should happen during either INVITE , 200 OK or ACK. In the case of SDP presence on invite <=> 200 , one should rtpproxy_offer during the invite and rtpproxy_answer during the 200 OK. "Documentation : Rewrites SDP body to ensure that media is passed through an RTP proxy. To be invoked on INVITE for the cases the SDPs are in INVITE and 200 OK and on 200 OK when SDPs are in 200 OK and ACK." But sometimes opensips receives a 183 Session Progress containing SDP before the 200 which i think is related to earlymedia . I think those sdp packets should also be rewritten with the right rtp proxy ip/port. In which case the doc should mention "SDPs are in INVITE and 200 OK /183 Session Progress". My second question : since we can handle most cases with rtpproxy_offer/answer methods, what is the purpose of rtpproxy_engage ? is it to ease the management i described above ? Thank you -------------- next part -------------- An HTML attachment was scrubbed... URL: From efes99999 at gmail.com Fri Dec 23 19:55:24 2022 From: efes99999 at gmail.com (L S) Date: Fri, 23 Dec 2022 14:55:24 -0500 Subject: [OpenSIPS-Users] TLS verify client Message-ID: Hi, We are upgrading from 1.11.5 tls to 3.2.9. In 1.11 we had issues with the client certificate so we had to set the following: # 1.11 parameters tls_verify_server = 1 tls_verify_client = 0 tls_require_client_certificate = 0 TLS works fine for us with those settings. Now we are trying to migrate them to 3.2.9 and having issues. Just wanted to confirm if the following is correct way to migrate those parameters to 3.2? (Just included those parameters - the domains are set up correctly) Server domain modparam("tls_mgm", "verify_cert", "[dom1]0") modparam("tls_mgm", "require_cert", "[dom1]0") Client domain modparam("tls_mgm", "verify_cert", "[dom2]1") modparam("tls_mgm", "require_cert", "[dom2]1") Thanks, Matt -------------- next part -------------- An HTML attachment was scrubbed... URL: From venefax at gmail.com Fri Dec 23 20:22:23 2022 From: venefax at gmail.com (Saint Michael) Date: Fri, 23 Dec 2022 15:22:23 -0500 Subject: [OpenSIPS-Users] RTPPROXY / OPENSIPS In-Reply-To: References: Message-ID: I have a lot of calls that should work with RTPPROXY and instead I get dead air. Maybe this is the issue. On Fri, Dec 23, 2022 at 2:45 PM Wadii ELMAJDI | Evenmedia wrote: > > hello , i do have a question related to rtpproxy module documentation. > > The doc describes that rewriting sdp body should happen during either INVITE , 200 OK or ACK. > In the case of SDP presence on invite <=> 200 , one should rtpproxy_offer during the invite and rtpproxy_answer during the 200 OK. > > "Documentation : Rewrites SDP body to ensure that media is passed through an RTP proxy. To be invoked on INVITE for the cases the SDPs are in INVITE and 200 OK and on 200 OK when SDPs are in 200 OK and ACK." > > But sometimes opensips receives a 183 Session Progress containing SDP before the 200 which i think is related to earlymedia . > I think those sdp packets should also be rewritten with the right rtp proxy ip/port. In which case the doc should mention "SDPs are in INVITE and 200 OK /183 Session Progress". > > My second question : since we can handle most cases with rtpproxy_offer/answer methods, what is the purpose of rtpproxy_engage ? is it to ease the management i described above ? > > Thank you > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users From david.villasmil.work at gmail.com Tue Dec 27 11:35:29 2022 From: david.villasmil.work at gmail.com (David Villasmil) Date: Tue, 27 Dec 2022 12:35:29 +0100 Subject: [OpenSIPS-Users] OpenSIPS Equivalent to Kamailio's cfg_get Message-ID: Hello folks, Is there such a thing? cfgutils seems to be an alternative, but looks like too much for such a simple thing. Regards, David Villasmil email: david.villasmil.work at gmail.com phone: +34669448337 -------------- next part -------------- An HTML attachment was scrubbed... URL: From wadii at evenmedia.fr Tue Dec 27 14:53:53 2022 From: wadii at evenmedia.fr (Wadii ELMAJDI | Evenmedia) Date: Tue, 27 Dec 2022 14:53:53 +0000 Subject: [OpenSIPS-Users] Media server communication with OpenSIPS Message-ID: Hello, I am using Asterisk as a media server behind OpenSIPS. I need to send some additional data to opensips during the call or at the latest during the hangup. For instance a custom hang up reason. Asterisk does not allow neither to send a sequential request withing an ongoing dialog, nor add headers to a BYE method. The current solution I'm using is pushing a dialog var (dlg_push_var) using Call-ID via (mi_http), but that means I rely on a curl http request during my call... Is there any better option for a direct communication between the two boxes during a call ? without any intermediate server (Redis DB ...) Thank you -------------- next part -------------- An HTML attachment was scrubbed... URL: From brett at nemeroff.com Tue Dec 27 15:18:10 2022 From: brett at nemeroff.com (Brett Nemeroff) Date: Tue, 27 Dec 2022 09:18:10 -0600 Subject: [OpenSIPS-Users] Media server communication with OpenSIPS In-Reply-To: References: Message-ID: If your UAC doesn't have the capabilities of sending it in the BYE or an in-dialog request, then you'll have to do something out of band like HTTPS, Redis, etc, which are not bad options. Have you looked into "X-Asterisk-HangupCauseCode" to see if that can help you out? On Tue, Dec 27, 2022 at 8:55 AM Wadii ELMAJDI | Evenmedia < wadii at evenmedia.fr> wrote: > > Hello, > > > I am using Asterisk as a media server behind OpenSIPS. > I need to send some additional data to opensips during the call or at the latest during the hangup. For instance a custom hang up reason. > Asterisk does not allow neither to send a sequential request withing an ongoing dialog, nor add headers to a BYE method. > The current solution I’m using is pushing a dialog var (dlg_push_var) using Call-ID via (mi_http), but that means I rely on a curl http request during my call… > Is there any better option for a direct communication between the two boxes during a call ? without any intermediate server (Redis DB …) > > Thank you > > _______________________________________________ > 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 wadii at evenmedia.fr Wed Dec 28 09:47:05 2022 From: wadii at evenmedia.fr (Wadii ELMAJDI | Evenmedia) Date: Wed, 28 Dec 2022 09:47:05 +0000 Subject: [OpenSIPS-Users] Media server communication with OpenSIPS In-Reply-To: References: Message-ID: Hello, I must send more data than just a hangup code: a duration, a text value.. X-Asterisk-HangupCauseCode is only sent when a called is hung up after being answered. Plus, it isn’t really a custom header. It must be an int value. Same for Q.850 reason header. I will maintain my current solution then (the in-dialog request via mi_http) Thank you De : Users De la part de Brett Nemeroff Envoyé : mardi 27 décembre 2022 16:18 À : OpenSIPS users mailling list Objet : Re: [OpenSIPS-Users] Media server communication with OpenSIPS If your UAC doesn't have the capabilities of sending it in the BYE or an in-dialog request, then you'll have to do something out of band like HTTPS, Redis, etc, which are not bad options. Have you looked into "X-Asterisk-HangupCauseCode" to see if that can help you out? On Tue, Dec 27, 2022 at 8:55 AM Wadii ELMAJDI | Evenmedia > wrote: > > Hello, > > > I am using Asterisk as a media server behind OpenSIPS. > I need to send some additional data to opensips during the call or at the latest during the hangup. For instance a custom hang up reason. > Asterisk does not allow neither to send a sequential request withing an ongoing dialog, nor add headers to a BYE method. > The current solution I’m using is pushing a dialog var (dlg_push_var) using Call-ID via (mi_http), but that means I rely on a curl http request during my call… > Is there any better option for a direct communication between the two boxes during a call ? without any intermediate server (Redis DB …) > > Thank you > > _______________________________________________ > 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 efes99999 at gmail.com Thu Dec 29 14:27:04 2022 From: efes99999 at gmail.com (L S) Date: Thu, 29 Dec 2022 09:27:04 -0500 Subject: [OpenSIPS-Users] Opensips stops responding to TLS Message-ID: Hi, We are in the process of migrating from 1.11.5 tls to 3.2.9, and we are running into an issue with TLS. Opensips stops handling TLS within a few minutes after it is started; e.g. stops responding to Client Hellos. There is no more outgoing TLS traffic from the Opensips server. When we restart Opensips, it goes back to normal for a while, then stops responding to TLS requests again. I don't see any errors in logs. The server runs Centos 7, openssl 1.1.1q. 1.11.5 works fine. Can this be a memory issue? We use S_memory 512 and P_memory 8. Opensips 1.11.5 works fine with the same settings. TCP parameters have their default values. How can we debug this? Any suggestions would be appreciated. Thanks, Matt -------------- next part -------------- An HTML attachment was scrubbed... URL: From nutxase at proton.me Thu Dec 29 16:27:54 2022 From: nutxase at proton.me (nutxase) Date: Thu, 29 Dec 2022 16:27:54 +0000 Subject: [OpenSIPS-Users] suspend transaction Message-ID: <5NoDcGaZIx1exwvRcAMTukXn1jUGfoU2DUSrwqkkxCfns1QBNIBnkW0JftCSmVOKqMCYlSLm6RsvbE92N_eYIYkIUSEUI4_vruWkmqI3JJw=@proton.me> Hi All i notice this is no longer working in 3.3 if (!t_suspend()) { xlog("Failed to suspend transaction\n"); exit; any idea what replaced t_suspend? Sent with [Proton Mail](https://proton.me/) secure email. -------------- next part -------------- An HTML attachment was scrubbed... URL: From greg at switchtel.co.za Thu Dec 29 19:28:40 2022 From: greg at switchtel.co.za (Gregory Massel) Date: Thu, 29 Dec 2022 21:28:40 +0200 Subject: [OpenSIPS-Users] Media server communication with OpenSIPS In-Reply-To: References: Message-ID: <4599151a-f242-c853-d86a-73b479e795b5@switchtel.co.za> You can send custom headers from Asterisk to OpenSIPS using: https://wiki.asterisk.org/wiki/display/AST/Asterisk+18+Function_PJSIP_HEADER If you need Asterisk to receive customer headers from OpenSIPS, you can read custom headers added by OpenSIPS to its 200 response using: https://wiki.asterisk.org/wiki/display/AST/Asterisk+18+Function_PJSIP_RESPONSE_HEADER If you are already midway through a call, you can force an UPDATE or re-INVITE using: https://wiki.asterisk.org/wiki/display/AST/Asterisk+18+Function_PJSIP_SEND_SESSION_REFRESH While I haven't personally tested, I'd expect that you should be able to signal custom information midway through a call with a Set(PJSIP_HEADER(add,X-MyHeader)=myvalue) followed by a Set(PJSIP_SEND_SESSION_REFRESH()=invite). This should then trigger a re-INVITE carrying the new X-MyHeader within it. Similarly, after triggering the a re-INVITE or UPDATE, you should then be able to do a Set(MYVAR=PJSIP_RESPONSE_HEADER(read,X-MyHeader)) to extract what OpenSIPS has sent back to Asterisk in its 200 response. Although more messy, there are also other tricks that you can use, including the CONNECTEDLINE function: https://wiki.asterisk.org/wiki/display/AST/Manipulating+Party+ID+Information https://wiki.asterisk.org/wiki/display/AST/Asterisk+18+Function_CONNECTEDLINE e.g. If you did a Set(CONNECTEDLINE(name)=Information_I_wish_to_share_with_OpenSIPS), then this will, in one step, trigger an UPDATE or re-INVITE carrying the new connected party name in the SIP headers, detail which you can then extract from OpenSIPS. Similarly, you can also use this approach read information that OpenSIPS has sent to Asterisk, with a Set(MYVAR=CONNECTEDLINE(name)). I mention this in case, for whatever reason, the first approach (described above) doesn't work as planned, however, the first approach is definitely preferable as you can carry the information in your own X- header, making it far easier to send/receive multiple headers. > On Tue, Dec 27, 2022 at 8:55 AM Wadii ELMAJDI | Evenmedia > wrote: > > > > Hello, > > > > > > I am using Asterisk as a media server behind OpenSIPS. > > I need to send some additional data to opensips during the call or > at the latest during the hangup. For instance a custom hang up reason. > > Asterisk does not allow neither to send a sequential request withing > an ongoing dialog, nor add headers to a BYE method. > > The current solution I’m using is pushing a dialog var > (dlg_push_var) using Call-ID via (mi_http), but that means I rely on a > curl http request during my call… > > Is there any better option for a direct communication between the > two boxes during a call ? without any intermediate server (Redis DB …) > > > > Thank you > > > > _______________________________________________ > > 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 wadii at evenmedia.fr Thu Dec 29 20:57:06 2022 From: wadii at evenmedia.fr (Wadii ELMAJDI | Evenmedia) Date: Thu, 29 Dec 2022 20:57:06 +0000 Subject: [OpenSIPS-Users] Media server communication with OpenSIPS In-Reply-To: <4599151a-f242-c853-d86a-73b479e795b5@switchtel.co.za> References: <4599151a-f242-c853-d86a-73b479e795b5@switchtel.co.za> Message-ID: Hello, The invite is relayed from OpenSIPS to Asterisk. Asterisk has UAS role here. The data i'm hoping to transfer to OPENSIPS should be sent either with the 200 OK (Asterisk to OpenSIPS), during the call, or via BYE. So neither PJSIP_HEADER nor PJSIP_RESPONSE_HEADER can help with my usecase. I have already used pjsip session refresh to renegotiate media codecs during a call, but i've never tried adding an extraheader to the re-invite/update method. I will try this asap, thank you for the detailed answer. ________________________________ De : Users de la part de Gregory Massel Envoyé : jeudi 29 décembre 2022 20:28 À : users at lists.opensips.org Objet : Re: [OpenSIPS-Users] Media server communication with OpenSIPS You can send custom headers from Asterisk to OpenSIPS using: https://wiki.asterisk.org/wiki/display/AST/Asterisk+18+Function_PJSIP_HEADER If you need Asterisk to receive customer headers from OpenSIPS, you can read custom headers added by OpenSIPS to its 200 response using: https://wiki.asterisk.org/wiki/display/AST/Asterisk+18+Function_PJSIP_RESPONSE_HEADER If you are already midway through a call, you can force an UPDATE or re-INVITE using: https://wiki.asterisk.org/wiki/display/AST/Asterisk+18+Function_PJSIP_SEND_SESSION_REFRESH While I haven't personally tested, I'd expect that you should be able to signal custom information midway through a call with a Set(PJSIP_HEADER(add,X-MyHeader)=myvalue) followed by a Set(PJSIP_SEND_SESSION_REFRESH()=invite). This should then trigger a re-INVITE carrying the new X-MyHeader within it. Similarly, after triggering the a re-INVITE or UPDATE, you should then be able to do a Set(MYVAR=PJSIP_RESPONSE_HEADER(read,X-MyHeader)) to extract what OpenSIPS has sent back to Asterisk in its 200 response. Although more messy, there are also other tricks that you can use, including the CONNECTEDLINE function: https://wiki.asterisk.org/wiki/display/AST/Manipulating+Party+ID+Information https://wiki.asterisk.org/wiki/display/AST/Asterisk+18+Function_CONNECTEDLINE e.g. If you did a Set(CONNECTEDLINE(name)=Information_I_wish_to_share_with_OpenSIPS), then this will, in one step, trigger an UPDATE or re-INVITE carrying the new connected party name in the SIP headers, detail which you can then extract from OpenSIPS. Similarly, you can also use this approach read information that OpenSIPS has sent to Asterisk, with a Set(MYVAR=CONNECTEDLINE(name)). I mention this in case, for whatever reason, the first approach (described above) doesn't work as planned, however, the first approach is definitely preferable as you can carry the information in your own X- header, making it far easier to send/receive multiple headers. On Tue, Dec 27, 2022 at 8:55 AM Wadii ELMAJDI | Evenmedia > wrote: > > Hello, > > > I am using Asterisk as a media server behind OpenSIPS. > I need to send some additional data to opensips during the call or at the latest during the hangup. For instance a custom hang up reason. > Asterisk does not allow neither to send a sequential request withing an ongoing dialog, nor add headers to a BYE method. > The current solution I’m using is pushing a dialog var (dlg_push_var) using Call-ID via (mi_http), but that means I rely on a curl http request during my call… > Is there any better option for a direct communication between the two boxes during a call ? without any intermediate server (Redis DB …) > > Thank you > > _______________________________________________ > 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 efes99999 at gmail.com Thu Dec 29 22:51:26 2022 From: efes99999 at gmail.com (L S) Date: Thu, 29 Dec 2022 17:51:26 -0500 Subject: [OpenSIPS-Users] Opensips stops responding to TLS In-Reply-To: References: Message-ID: Just wanted to add the traffic between the client and Opensips below. It seems Opensips keeps on sending RESET. We have the tcp_max_connections at default value. That value (2048) works fine in 1.11.5. Thanks. client opensips SSL 142 Client Hello client opensips SSL 142 [TCP Retransmission] Client Hello opensips client TCP 54 sips > 5071 [RST] Seq=1 Win=0 Len=0 client opensips SSL 142 [TCP Retransmission] Client Hello opensips client TCP 54 sips > 5064 [RST] Seq=1 Win=0 Len=0 client opensips TCP 74 [TCP Port numbers reused] 5071 > sips [SYN] Seq=0 Win=8192 Len=0 MSS=1460 WS=1 opensips client TCP 54 sips > 5071 [RST, ACK] Seq=1 Ack=1 Win=0 Len=0 client opensips TCP 74 [TCP Port numbers reused] 5064 > sips [SYN] Seq=0 Win=8192 Len=0 MSS=1460 WS=1 opensips client TCP 54 sips > 5064 [RST, ACK] Seq=1 Ack=1 Win=0 Len=0 client opensips TCP 74 [TCP Port numbers reused] 5080 > sips [SYN] Seq=0 Win=8192 Len=0 MSS=1460 WS=1 On Thu, Dec 29, 2022, 9:27 AM L S wrote: > Hi, > > We are in the process of migrating from 1.11.5 tls to 3.2.9, and we are > running into an issue with TLS. > > Opensips stops handling TLS within a few minutes after it is started; e.g. > stops responding to Client Hellos. There is no more outgoing TLS traffic > from the Opensips server. When we restart Opensips, it goes back to normal > for a while, then stops responding to TLS requests again. > > I don't see any errors in logs. > The server runs Centos 7, openssl 1.1.1q. > > 1.11.5 works fine. > > Can this be a memory issue? We use S_memory 512 and P_memory 8. Opensips > 1.11.5 works fine with the same settings. TCP parameters have their default > values. > > How can we debug this? Any suggestions would be appreciated. > > Thanks, > Matt > -------------- next part -------------- An HTML attachment was scrubbed... URL: From efes99999 at gmail.com Fri Dec 30 14:00:22 2022 From: efes99999 at gmail.com (L S) Date: Fri, 30 Dec 2022 09:00:22 -0500 Subject: [OpenSIPS-Users] Opensips stops responding to TLS In-Reply-To: References: Message-ID: One more thing: log_level=4 open_files_limit=32768 At the time Opensips stops responding to TLS, it seems like it stops writing to log file too even though it continues handling the non-TLS SIP. Thanks. On Thu, Dec 29, 2022, 5:51 PM L S wrote: > Just wanted to add the traffic between the client and Opensips below. It > seems Opensips keeps on sending RESET. > > We have the tcp_max_connections at default value. That value (2048) works > fine in 1.11.5. > > Thanks. > > client opensips SSL 142 Client Hello > client opensips SSL 142 [TCP Retransmission] Client Hello > opensips client TCP 54 sips > 5071 [RST] Seq=1 Win=0 Len=0 > client opensips SSL 142 [TCP Retransmission] Client Hello > opensips client TCP 54 sips > 5064 [RST] Seq=1 Win=0 Len=0 > client opensips TCP 74 [TCP Port numbers reused] 5071 > sips [SYN] > Seq=0 Win=8192 Len=0 MSS=1460 WS=1 > opensips client TCP 54 sips > 5071 [RST, ACK] Seq=1 Ack=1 Win=0 Len=0 > client opensips TCP 74 [TCP Port numbers reused] 5064 > sips [SYN] > Seq=0 Win=8192 Len=0 MSS=1460 WS=1 > opensips client TCP 54 sips > 5064 [RST, ACK] Seq=1 Ack=1 Win=0 Len=0 > client opensips TCP 74 [TCP Port numbers reused] 5080 > sips [SYN] > Seq=0 Win=8192 Len=0 MSS=1460 WS=1 > > On Thu, Dec 29, 2022, 9:27 AM L S wrote: > >> Hi, >> >> We are in the process of migrating from 1.11.5 tls to 3.2.9, and we are >> running into an issue with TLS. >> >> Opensips stops handling TLS within a few minutes after it is started; >> e.g. stops responding to Client Hellos. There is no more outgoing TLS >> traffic from the Opensips server. When we restart Opensips, it goes back to >> normal for a while, then stops responding to TLS requests again. >> >> I don't see any errors in logs. >> The server runs Centos 7, openssl 1.1.1q. >> >> 1.11.5 works fine. >> >> Can this be a memory issue? We use S_memory 512 and P_memory 8. Opensips >> 1.11.5 works fine with the same settings. TCP parameters have their default >> values. >> >> How can we debug this? Any suggestions would be appreciated. >> >> Thanks, >> Matt >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: From nutxase at proton.me Fri Dec 30 16:12:54 2022 From: nutxase at proton.me (nutxase) Date: Fri, 30 Dec 2022 16:12:54 +0000 Subject: [OpenSIPS-Users] Suspending a transaction Message-ID: Hi Guys How do i suspend a transaction then run a script then resume it similar to how kamailio does t_suspend? Sent with [Proton Mail](https://proton.me/) secure email. -------------- next part -------------- An HTML attachment was scrubbed... URL: From efes99999 at gmail.com Thu Dec 22 18:58:36 2022 From: efes99999 at gmail.com (L S) Date: Thu, 22 Dec 2022 18:58:36 -0000 Subject: [OpenSIPS-Users] TLS verify client Message-ID: Hi, We are upgrading from 1.11.5 tls to 3.2.9. In 1.11 we had issues with the client certificate so we had to set the following: # 1.11 parameters tls_verify_server = 1 tls_verify_client = 0 tls_require_client_certificate = 0 TLS works fine for us with those settings. Now we are trying to migrate them to 3.2.9 and having issues. Just wanted to confirm if the following is correct way to migrate those parameters to 3.2? (Just included those parameters - the domains are set up correctly) Server domain modparam("tls_mgm", "verify_cert", "[dom1]0") modparam("tls_mgm", "require_cert", "[dom1]0") Client domain modparam("tls_mgm", "verify_cert", "[dom2]1") modparam("tls_mgm", "require_cert", "[dom2]1") Thanks, Matt -------------- next part -------------- An HTML attachment was scrubbed... URL: