From alain.bieuzent at free.fr Thu Apr 2 08:16:35 2020 From: alain.bieuzent at free.fr (Alain Bieuzent) Date: Thu, 02 Apr 2020 10:16:35 +0200 Subject: [OpenSIPS-Users] mid_registrar - AOR mode - Disable parallel forking. Message-ID: <2F415863-4210-4222-9E1E-6904D540AF17@free.fr> Hi, I’m trying to use mid_registrar module between a janus gateway and asterisk box. I have the same janus endpoints registered 2 time opensips mid_registrar module (so only 1 with opensips IP forward to asterisk box) The issue (in my case) that in case of asterisk box send a call to janus (via opensips mid_registrar module), the call is forked to the 2 janus endpoints in parallel I want to limit the incoming to only the last endpoint registered. Is there a parameter to limit this ? Thanks Alain -------------- next part -------------- An HTML attachment was scrubbed... URL: From Johan at democon.be Thu Apr 2 08:47:28 2020 From: Johan at democon.be (Johan De Clercq) Date: Thu, 2 Apr 2020 10:47:28 +0200 Subject: [OpenSIPS-Users] trouble migrating from 3.0 to 3.1 with drouting. In-Reply-To: <7a6ebfea-a36e-f8b6-d799-a0ad1909b0ba@democon.be> References: <7a6ebfea-a36e-f8b6-d799-a0ad1909b0ba@democon.be> Message-ID: Hello, should I open a bug for this on github or how do I need to proceed ? BR, Op di 31 mrt. 2020 om 19:23 schreef johan : > Hi guys when I call do_routing in opensips 3.1. I have : > > Mar 31 16:52:25 hendrix /data/opensips/sbin/opensips[20886]: > callid=hxj~vmgW54: route[drouting]: let's find the group for drouting > based on fU 33757936420 > Mar 31 16:52:25 hendrix /data/opensips/sbin/opensips[20886]: > DBG:core:pv_printf: final buffer length 102 > Mar 31 16:52:25 hendrix /data/opensips/sbin/opensips[20886]: > callid=hxj~vmgW54: route[drouting]: fU 33757936420 does not start with > 32460, we put var(group) 1 to 1 > > Mar 31 16:52:25 hendrix /data/opensips/sbin/opensips[20886]: > DBG:drouting:do_routing: empty routing table > Mar 31 16:52:25 hendrix /data/opensips/sbin/opensips[20886]: > DBG:core:pv_printf: final buffer length 51 > Mar 31 16:52:25 hendrix /data/opensips/sbin/opensips[20886]: > callid=hxj~vmgW54: route[drouting]: drouting failed > > > script part : > > xlog("callid=$ci: route[drouting]: let's find the group for > drouting based on fU $fU"); > $var(group)=""; > if($fU=~"32460.*") > { > $var(group)=2; > xlog("callid=$ci: route[drouting]: fU $fU starts with 32460, we > put var(group) $var(group) to 2"); > } > else > { > $var(group)=1; > xlog("callid=$ci: route[drouting]: fU $fU does not start with > 32460, we put var(group) $var(group) to 1"); > } > if(!do_routing($(var(group){s.int}),,,$var(rule),$var(gw))) > { > xlog("callid=$ci: route[drouting]: drouting failed"); > sl_send_reply(500,"no routes!!!"); > exit; > } > > > > olddb : > > select * from dr_rules; > > +--------+---------+--------+---------+----------+---------+--------+----------------+-------------+ > | ruleid | groupid | prefix | timerec | priority | routeid | gwlist | > attrs | description | > > +--------+---------+--------+---------+----------+---------+--------+----------------+-------------+ > | 4 | 1 | | | 0 | | 32 | > BICS | | > | 7 | 2 | | | 0 | | 32460 | > Belgian mobile | | > > +--------+---------+--------+---------+----------+---------+--------+----------------+-------------+ > > select * from dr_rules; > > +--------+---------+--------+---------+----------+---------+--------+----------------+-------------+ > | ruleid | groupid | prefix | timerec | priority | routeid | gwlist | > attrs | description | > > +--------+---------+--------+---------+----------+---------+--------+----------------+-------------+ > | 4 | 1 | | | 0 | | 32 | A > | | > | 7 | 2 | | | 0 | | 32460 | B > | | > > +--------+---------+--------+---------+----------+---------+--------+----------------+-------------+ > 2 rows in set (0.01 sec) > lect * from dr_gateways > -> ; > > +----+-------+------+----------------------+-------+------------+-------+------------+-------+--------+----------------------------------+ > | id | gwid | type | address | strip | pri_prefix | attrs > | probe_mode | state | socket | description | > > +----+-------+------+----------------------+-------+------------+-------+------------+-------+--------+----------------------------------+ > | 1 | 32 | 2 | 192.168.174.251:5060 | 0 | | | > 0 | 0 | | A | > | 5 | -1 | 1 | 192.168.174.254:5060 | 1 | | | > 0 | 0 | | Inbound from B | > | 7 | 32460 | 1 | 192.168.174.253:5060 | 0 | | | > 0 | 0 | | C| > > +----+-------+------+----------------------+-------+------------+-------+------------+-------+--------+----------------------------------+ > 3 rows in set (0.00 sec) > > select * from dr_groups; > > +----+----------+------------------------+---------+------------------------------+ > | id | username | domain | groupid | > description | > > +----+----------+------------------------+---------+------------------------------+ > | 3 | 1 | abcc| 1 | Default group for | > | 5 | 1 | yourdomain.net | 2 | BICS > mobile | > > +----+----------+------------------------+---------+------------------------------+ > 2 rows in set (0.00 sec) > > new db : > > dr_rules; > > +--------+---------+--------+---------+----------+---------+--------+----------+--------------+----------------+-------------+ > | ruleid | groupid | prefix | timerec | priority | routeid | gwlist | > sort_alg | sort_profile | attrs | description | > > +--------+---------+--------+---------+----------+---------+--------+----------+--------------+----------------+-------------+ > | 4 | 1 | | | 0 | | 32 | > N | 0 | A | | > | 7 | 2 | | | 0 | | 32460 | > N | 0 | B | | > > +--------+---------+--------+---------+----------+---------+--------+----------+--------------+----------------+-------------+ > 2 rows in set (0.00 sec) > select * from dr_gateways; > > +----+-------+------+----------------------+-------+------------+-------+------------+-------+--------+----------------------------------+ > | id | gwid | type | address | strip | pri_prefix | attrs > | probe_mode | state | socket | description | > > +----+-------+------+----------------------+-------+------------+-------+------------+-------+--------+----------------------------------+ > | 1 | 32 | 2 | 192.168.174.251:5060 | 0 | | | > 0 | 0 | | A | > | 5 | -1 | 1 | 192.168.174.254:5060 | 1 | | | > 0 | 0 | | Inbound from B | > | 7 | 32460 | 1 | 192.168.174.253:5060 | 0 | | | > 0 | 0 | | C| > > +----+-------+------+----------------------+-------+------------+-------+------------+-------+--------+----------------------------------+ > 3 rows in set (0.00 sec) > select * from dr_groups; > > +----+----------+------------------------+---------+------------------------------+ > | id | username | domain | groupid | > description | > > +----+----------+------------------------+---------+------------------------------+ > | 3 | 1 | abc | 1 | Default > group for | > | 5 | 1 | yourdomain.net | 2 | BICS > mobile | > > +----+----------+------------------------+---------+------------------------------+ > 2 rows in set (0.00 sec) > > select * from dr_carriers; > Empty set (0.00 sec) > > select * from dr_partitions; > Empty set (0.00 sec) > > > so dr_rules has changed. > > > can somebody please point out what is wrong with my datafill ? > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From liviu at opensips.org Thu Apr 2 08:50:28 2020 From: liviu at opensips.org (Liviu Chircu) Date: Thu, 2 Apr 2020 11:50:28 +0300 Subject: [OpenSIPS-Users] mid_registrar - AOR mode - Disable parallel forking. In-Reply-To: <2F415863-4210-4222-9E1E-6904D540AF17@free.fr> References: <2F415863-4210-4222-9E1E-6904D540AF17@free.fr> Message-ID: <94849880-c5c8-bf23-79cf-8cbb8f555220@opensips.org> On 02.04.2020 11:16, Alain Bieuzent wrote: > > The issue (in my case) that in case of asterisk box send a call to > janus (via opensips mid_registrar module), the call is forked to the 2 > janus endpoints in parallel > > I want to limit the incoming to only the last endpoint registered. > > Is there a parameter to limit this ? > Hi, Alain! This is possible starting with version 3.1, where mid_registrar_save() [1] was enhanced with the "cNN" flag. Best regards, [1]: https://opensips.org/docs/modules/3.1.x/mid_registrar.html#func_mid_registrar_save -- Liviu Chircu www.twitter.com/liviuchircu | www.opensips-solutions.com OpenSIPS Summit, Amsterdam, May 2020 www.opensips.org/events -------------- next part -------------- An HTML attachment was scrubbed... URL: From farmorg at gmail.com Thu Apr 2 13:42:18 2020 From: farmorg at gmail.com (Mark Farmer) Date: Thu, 2 Apr 2020 14:42:18 +0100 Subject: [OpenSIPS-Users] Access to listen/advertised IP Addresses Message-ID: Hi everyone Is there a way to access within the script the listen or 'listen as' IP addresses that are set in Global Parameters? I've been looking at the core variables but I can't see how it could be done. I'd like to reference them without setting a custom variable. I am using OpenSIPS 2.4.7 Many thanks! Mark. -------------- next part -------------- An HTML attachment was scrubbed... URL: From voip.security at protonmail.com Thu Apr 2 16:29:11 2020 From: voip.security at protonmail.com (Sharad Kumar) Date: Thu, 2 Apr 2020 11:29:11 -0500 Subject: [OpenSIPS-Users] Access to listen/advertised IP Addresses In-Reply-To: References: Message-ID: Hi Mark,  If your initial goal is to get the interface IP where request is received then you can try these variables. *$Ri* - reference to IP address of the interface where the request has been received *$Rp* - reference to the port where the message was received -------------- next part -------------- An HTML attachment was scrubbed... URL: From voip.security at protonmail.com Thu Apr 2 16:24:18 2020 From: voip.security at protonmail.com (Sharad Kumar) Date: Thu, 2 Apr 2020 11:24:18 -0500 Subject: [OpenSIPS-Users] Access to listen/advertised IP Addresses In-Reply-To: References: Message-ID: Hi Mark, If your initial goal is to get the listener IP where you received the request then you can try these variables. *$Ri* - reference to IP address of the interface where the request has been received *$Rp* - reference to the port where the message was received -------------- next part -------------- An HTML attachment was scrubbed... URL: From social at bohboh.info Thu Apr 2 16:57:12 2020 From: social at bohboh.info (Social Boh) Date: Thu, 2 Apr 2020 11:57:12 -0500 Subject: [OpenSIPS-Users] OpenSIPs Console Two o more servers configuration. Message-ID: <888899d5-bd1d-12d6-41a2-7c95549753e3@bohboh.info> Hello list, I'm trying to configure two boxes on the OpenSIPs Console trough boxes.global.inc.php without success. Both boxes have the same $boxes[$box_id]['assoc_id']=1; One is local: $boxes[$box_id]['mi']['conn']="json:127.0.0.1:8088/json"; and one is remote: $boxes[$box_id]['mi']['conn']="json:RemoteIP:8080/json"; On the Opensips Console I can see Monit Server of local Box, for example, but no possibility to choose the second box. Any hint, please? Regards -- --- I'm SoCIaL, MayBe From bogdan at opensips.org Thu Apr 2 17:16:55 2020 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Thu, 2 Apr 2020 20:16:55 +0300 Subject: [OpenSIPS-Users] [Blog] BLF reloaded, or a more accurate and detailed approach Message-ID: Hi all, The BLF (or dialoginfo) support has in the upcoming OpenSIPS 3.1 a more accurate and detailed reporting, being able to handle parallel and serial call forking. This is an important milestone on the #class5 path. https://blog.opensips.org/2020/04/02/blf-reloaded-or-a-more-accurate-and-detailed-approach/ Enjoy and happy usage ;) -- Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com From rob.dyck at telus.net Thu Apr 2 18:42:04 2020 From: rob.dyck at telus.net (Robert Dyck) Date: Thu, 02 Apr 2020 11:42:04 -0700 Subject: [OpenSIPS-Users] rtpproxy module not supporting valid payload types Message-ID: <3637827.9PP9qmUYLL@blacky.mylan> Regarding opensips-3.0 Use case - webrtc client behind NAT The rtpproxy module emitted the error message "can't extract media port from the message" ( by the way, very misleading ). In reality extract_mediainfo fails because it could not find a supported payload type in the media description. The payload type in question is "UDP/TLS/RTP/SAVPF". RFC 5764 section 8 introduces four more RTP types. DCCP/TLS/RTP/SAVP and SAVPF UDP/TLS/RTP/SAVP and SAVPF Should rtpproxy.c be extended to support these additional RTP types? Thank you, Rob -------------- next part -------------- An HTML attachment was scrubbed... URL: From sobomax at sippysoft.com Fri Apr 3 06:11:00 2020 From: sobomax at sippysoft.com (Maxim Sobolev) Date: Thu, 2 Apr 2020 23:11:00 -0700 Subject: [OpenSIPS-Users] rtpproxy module not supporting valid payload types In-Reply-To: <3637827.9PP9qmUYLL@blacky.mylan> References: <3637827.9PP9qmUYLL@blacky.mylan> Message-ID: Yes, for sure. As long as the transport is UDP based, the RTPProxy would just work. The change should be trivial, you can get it fixed locally, test and then open a pull request against opensips repo. -Max On Thu., Apr. 2, 2020, 11:43 a.m. Robert Dyck, wrote: > Regarding opensips-3.0 > > Use case - webrtc client behind NAT > > > > The rtpproxy module emitted the error message "can't extract media port > from the message" ( by the way, very misleading ). In reality > extract_mediainfo fails because it could not find a supported payload type > in the media description. The payload type in question is > "UDP/TLS/RTP/SAVPF". > > > > RFC 5764 section 8 introduces four more RTP types. > > DCCP/TLS/RTP/SAVP and SAVPF > > UDP/TLS/RTP/SAVP and SAVPF > > > > Should rtpproxy.c be extended to support these additional RTP types? > > > > Thank you, Rob > _______________________________________________ > 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 barnaby.ritchley at sipsynergy.co.uk Fri Apr 3 11:08:27 2020 From: barnaby.ritchley at sipsynergy.co.uk (Barnaby Ritchley) Date: Fri, 3 Apr 2020 11:08:27 +0000 Subject: [OpenSIPS-Users] siptrace DEB Package Message-ID: <8A383D73-A284-43B7-9DA6-00CF9D719055@sipsynergy.co.uk> Is there a deb package in the apt repository for the siptrace and proto_hep modules? I cant see them and am not sure whether the modules are contained in another package. Barny. The information in this email (and in any attachments sent with it) is confidential. It is intended for the addressee only. Access to this email by anyone else is unintended and unauthorised. Only the addressee may rely on it. If you are not the original addressee, we ask you please to maintain confidentiality. If you have received this email in error please notify us immediately by replying to it, then destroy any copies and delete it from your computer system. Any use, dissemination, forwarding, printing or copying of this email by anyone except the addressee in the normal course of his / her business, is prohibited. We own the copyright in this email and any document created by us and assert the right to be identified as the author of it. Copyright has not been transferred to the addressee. -------------- next part -------------- An HTML attachment was scrubbed... URL: From barnaby.ritchley at sipsynergy.co.uk Fri Apr 3 14:12:03 2020 From: barnaby.ritchley at sipsynergy.co.uk (Barnaby Ritchley) Date: Fri, 3 Apr 2020 14:12:03 +0000 Subject: [OpenSIPS-Users] siptrace DEB Package In-Reply-To: <8A383D73-A284-43B7-9DA6-00CF9D719055@sipsynergy.co.uk> References: <8A383D73-A284-43B7-9DA6-00CF9D719055@sipsynergy.co.uk> Message-ID: Answered my own question by using tracer module. From: Barnaby Ritchley Date: Friday, 3 April 2020 at 12:08 To: "users at lists.opensips.org" Subject: siptrace DEB Package Is there a deb package in the apt repository for the siptrace and proto_hep modules? I cant see them and am not sure whether the modules are contained in another package. Barny. The information in this email (and in any attachments sent with it) is confidential. It is intended for the addressee only. Access to this email by anyone else is unintended and unauthorised. Only the addressee may rely on it. If you are not the original addressee, we ask you please to maintain confidentiality. If you have received this email in error please notify us immediately by replying to it, then destroy any copies and delete it from your computer system. Any use, dissemination, forwarding, printing or copying of this email by anyone except the addressee in the normal course of his / her business, is prohibited. We own the copyright in this email and any document created by us and assert the right to be identified as the author of it. Copyright has not been transferred to the addressee. -------------- next part -------------- An HTML attachment was scrubbed... URL: From kamlesh at worldphone.in Sat Apr 4 18:10:54 2020 From: kamlesh at worldphone.in (Kamlesh .) Date: Sat, 4 Apr 2020 23:40:54 +0530 Subject: [OpenSIPS-Users] problem in call forwarding scenario Message-ID: Dear All, version: opensips 2.4.6 (x86_64/linux) flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535 poll method support: poll, epoll, sigio_rt, select. git revision: edef893c5 main.c compiled on 23:26:34 Dec 14 2019 with gcc 4.8.5 I have a problem with call forwarding on busy. The issue arises when I get an INCOMING call from the gateway and the extension(callee) responds with busy and callee is configured as an external number as forwarding on busy so the forwarded call sends to the same gateway and receives the 183. So in that case the gateway has two provisional responses one is 180 from callee and 183 from the gateway. It is obvious that both the responses have different to tag. So the second response is not accepted by the gateway. Any solution for the issue. Below is the call flow. Regards, Kamlesh -- Disclaimer : This e-mail and any file transmitted with it are for exclusive use of the intended recipient(s) and may contain confidential and privileged information. If you are not the intended recipient,  please contact the sender by replying this e-mail and destroy all copies and original message. Any unauthorized review,use, disclosure, dissemination, forwarding, printing and copying of this email or any action taken in reliance of this e-mail is strictly prohibited and may be unlawful. -------------- next part -------------- An HTML attachment was scrubbed... URL: From masked at vale.ski Sun Apr 5 05:33:46 2020 From: masked at vale.ski (Michael Vale) Date: Sun, 05 Apr 2020 15:33:46 +1000 Subject: [OpenSIPS-Users] using load balancer and lookup together Message-ID: hi, perhaps this can be solved with a failure route and or a check status but i dont know and it would be nice if i could do it without it. no matter how i write the script, either a uac to uac call goes to the load balancer or the load balancer is stuck with a 404 reply from the script or uac to uac works but when one end is not registered it goes to the load balancer instead of getting a 404. i've tried failure routes and get the same problem. here is a snippet. if (!lb_start(1,"pstn")) && (!lookup("location","m",)) { lb_disable_dst(); #route(relay); #send_reply(404,"No user or gateway"); if (lb_start(1,"pstn")) { send_reply(500,"SIPSIPSIPS"); # t_relay(); exit; } # exit; } else if (lookup("location","m")) && (!lb_start(1,"pstn")) { lb_disable_dst(); route(relay); exit; } else if (lb_start(1,"pstn")) && (lookup("location","m")) { lb_disable_dst(); route(relay); exit; } else if (!lookup("location","m")) && (!lb_start(1,"pstn")) { send_reply(404,"Not Found"); exit; } else if (lb_start(1,"pstn")) && (!lookup("location","m")) { # #lb_disable_dst(); if (!lookup("location","m")) { route(relay); exit; } if (lookup("location","m")) { lb_disable_dst(); route(relay); exit; } } thanks in advance, michael. From podguiko at mail.ru Sun Apr 5 10:29:57 2020 From: podguiko at mail.ru (=?UTF-8?B?T2xlZyBQb2RndXlrbw==?=) Date: Sun, 05 Apr 2020 13:29:57 +0300 Subject: [OpenSIPS-Users] =?utf-8?q?t=5Fforward=5Fnonack_failed?= Message-ID: <1586082597.69155217@f471.i.mail.ru>   Opensips works like a proxy. Gets an Invite. In the process of processing it, opensips makes a request via http (rest_client module) and receives a response.  Adds the received information to Invite (as X-header) and sends through the dispatcher module to freeswitch. Everything is working fine. However, there is a scenario in which everything goes a little wrong. Opensips receives an Invite, starts processing it, sends a request via http. At this time, Cancel arrives at Invite. The transaction is being destroyed. Opensips sends 200 Cancelling, and then 487. And here comes the response via HTTP, but since the transaction is no longer there, this call ends with an error 500 no route to destination when the t_relay function is executed. In this case, a message appears in the log / usr / sbin / opensips [5577]: ERROR: tm: w_t_relay: t_forward_nonack failed. How to correctly handle such cases in order to prevent such errors in the logs?   -- Oleg Podguyko -------------- next part -------------- An HTML attachment was scrubbed... URL: From david.villasmil.work at gmail.com Sun Apr 5 10:47:57 2020 From: david.villasmil.work at gmail.com (David Villasmil) Date: Sun, 5 Apr 2020 11:47:57 +0100 Subject: [OpenSIPS-Users] using load balancer and lookup together In-Reply-To: References: Message-ID: Why are you trying to do all at once? Why not first do the lookup https://github.com/davidcsi/kamailio-private-public/blob/a81d7f777a8c5ee2dbb32311f7e6b5a3cf94bf32/kamailio.cfg#L771 and then start load balancing? https://github.com/davidcsi/kamailio-private-public/blob/a81d7f777a8c5ee2dbb32311f7e6b5a3cf94bf32/kamailio.cfg#L1109 Do you have some special need to fulfill? David On Sun, 5 Apr 2020 at 06:34, Michael Vale via Users < users at lists.opensips.org> wrote: > hi, > > perhaps this can be solved with a failure route and or a check status > but i dont know and it would be nice if i could do it without it. > > no matter how i write the script, either a uac to uac call goes to the > load balancer or the load balancer is stuck with a 404 reply from the > script or uac to uac works but when one end is not registered it goes > to the load balancer instead of getting a 404. > > i've tried failure routes and get the same problem. here is a snippet. > > if (!lb_start(1,"pstn")) && (!lookup("location","m",)) { > lb_disable_dst(); > #route(relay); > #send_reply(404,"No user or gateway"); > if (lb_start(1,"pstn")) { > send_reply(500,"SIPSIPSIPS"); > # t_relay(); > exit; > } > # exit; > } else if (lookup("location","m")) && > (!lb_start(1,"pstn")) { > lb_disable_dst(); > route(relay); > exit; > } else if (lb_start(1,"pstn")) && > (lookup("location","m")) { > lb_disable_dst(); > route(relay); > exit; > } else if (!lookup("location","m")) && > (!lb_start(1,"pstn")) { > send_reply(404,"Not Found"); > exit; > } else if (lb_start(1,"pstn")) && > (!lookup("location","m")) { > # #lb_disable_dst(); > if (!lookup("location","m")) { > route(relay); > exit; > } > if (lookup("location","m")) { > lb_disable_dst(); > route(relay); > exit; > } > } > > thanks in advance, > > michael. > > > _______________________________________________ > 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 Ben.Newlin at genesys.com Sun Apr 5 13:11:34 2020 From: Ben.Newlin at genesys.com (Ben Newlin) Date: Sun, 5 Apr 2020 13:11:34 +0000 Subject: [OpenSIPS-Users] t_forward_nonack failed In-Reply-To: <1586082597.69155217@f471.i.mail.ru> References: <1586082597.69155217@f471.i.mail.ru> Message-ID: <00BE023F-DA3A-45EE-988F-BA4891FB9E04@genesys.com> Oleg, At first glance, it seems like t_was_cancelled [1] is exactly what you want. However, it can only be called from onreply_route or failure_route, which makes it unusable for this case. This has caused problems for me as well, as it makes it very hard to detect cancellation when working in asynchronous routes that don’t involve SIP messages. Similarly, t_cancel_branch can only be called from onreply_route, which also causes issues with not being able to manage branches with non-SIP async operations. It may be a good feature request to add the ability to use those functions in async resume routes, but for now they will not work. However, t_relay itself does not ever generate a 500 error back to the client as far as I know. It can only send a 477 response automatically. Are you sure you are not generating the 500 in your script when handling the error from t_relay? The documentation for the function [2] notes that a -3 response from t_relay indicates the request may have already been cancelled. In that case, you can just exit if you know the 487 has already been sent, or you can send a 487 reply yourself. To be honest, I’m not sure where the 487 is coming from in your case since I didn’t think OpenSIPS would automatically respond with a 487 for a cancelled transaction; I have always had to do that from the script myself. So in addition to sending the 500 reply yourself, you may have some code in your script which is also sending the 487. If none of that works, another option would be to set a flag or an avp in the transaction when processing the CANCEL. Then you can check the flag when you receive the HTTP response and if it is set just don’t call t_relay. [1] https://opensips.org/docs/modules/3.0.x/tm.html#func_t_was_cancelled [2] https://opensips.org/docs/modules/3.0.x/tm.html#func_t_relay Ben Newlin From: Users on behalf of Oleg Podguyko via Users Reply-To: Oleg Podguyko , OpenSIPS users mailling list Date: Sunday, April 5, 2020 at 6:31 AM To: users Subject: [OpenSIPS-Users] t_forward_nonack failed Opensips works like a proxy. Gets an Invite. In the process of processing it, opensips makes a request via http (rest_client module) and receives a response. Adds the received information to Invite (as X-header) and sends through the dispatcher module to freeswitch. Everything is working fine. However, there is a scenario in which everything goes a little wrong. Opensips receives an Invite, starts processing it, sends a request via http. At this time, Cancel arrives at Invite. The transaction is being destroyed. Opensips sends 200 Cancelling, and then 487. And here comes the response via HTTP, but since the transaction is no longer there, this call ends with an error 500 no route to destination when the t_relay function is executed. In this case, a message appears in the log / usr / sbin / opensips [5577]: ERROR: tm: w_t_relay: t_forward_nonack failed. How to correctly handle such cases in order to prevent such errors in the logs? -- Oleg Podguyko -------------- next part -------------- An HTML attachment was scrubbed... URL: From masked at vale.ski Sun Apr 5 13:54:29 2020 From: masked at vale.ski (Michael Vale) Date: Sun, 05 Apr 2020 23:54:29 +1000 Subject: [OpenSIPS-Users] using load balancer and lookup together In-Reply-To: References: Message-ID: <6342fd559df4c9765b17270d07bc04f11ad93b36.camel@vale.ski> Ok, to explain, Using your logic, To call say '555' will goto Voicemail, if I disable voicemail it will return a 404 instead of going to the load balancer. That's fine, if 555 is an extension, but if it's (for this example) a PSTN number (or a all-else catch all, like I'm trying to achieve) thats not OK because I get a 404 rather than it getting routed to the load balancer. If 555 is an extension/user the call will go through to the registered extension, but if it's not registered in the usrloc table, it goes to 404, instead of the load balancer. If I reverse the logic, It will goto the load balancer even if it's a registered extension, or Too Many Hops, depending on how I adjust the logic. I cannot seem to create a catch all for non-usrloc registered extension calls to goto the load balancer otherwise return a 404. I hope I explained it well enough. I will keep trying, Regards, Michael. On Sun, 2020-04-05 at 11:47 +0100, David Villasmil wrote: > Why are you trying to do all at once? > > Why not first do the lookup > > https://github.com/davidcsi/kamailio-private-public/blob/a81d7f777a8c5ee2dbb32311f7e6b5a3cf94bf32/kamailio.cfg#L771 > > > and then start load balancing? > > https://github.com/davidcsi/kamailio-private-public/blob/a81d7f777a8c5ee2dbb32311f7e6b5a3cf94bf32/kamailio.cfg#L1109 > > Do you have some special need to fulfill? > > David > On Sun, 5 Apr 2020 at 06:34, Michael Vale via Users < > users at lists.opensips.org> wrote: > > hi, > > > > > > > > perhaps this can be solved with a failure route and or a check > > status > > > > but i dont know and it would be nice if i could do it without it. > > > > > > > > no matter how i write the script, either a uac to uac call goes to > > the > > > > load balancer or the load balancer is stuck with a 404 reply from > > the > > > > script or uac to uac works but when one end is not registered it > > goes > > > > to the load balancer instead of getting a 404. > > > > > > > > i've tried failure routes and get the same problem. here is a > > snippet. > > > > > > > > if (!lb_start(1,"pstn")) && (!lookup("location","m",)) { > > > > lb_disable_dst(); > > > > #route(relay); > > > > #send_reply(404,"No user or gateway"); > > > > if (lb_start(1,"pstn")) { > > > > send_reply(500,"SIPSIPSIPS"); > > > > # t_relay(); > > > > exit; > > > > } > > > > # exit; > > > > } else if (lookup("location","m")) && > > > > (!lb_start(1,"pstn")) { > > > > lb_disable_dst(); > > > > route(relay); > > > > exit; > > > > } else if (lb_start(1,"pstn")) && > > > > (lookup("location","m")) { > > > > lb_disable_dst(); > > > > route(relay); > > > > exit; > > > > } else if (!lookup("location","m")) && > > > > (!lb_start(1,"pstn")) { > > > > send_reply(404,"Not Found"); > > > > exit; > > > > } else if (lb_start(1,"pstn")) && > > > > (!lookup("location","m")) { > > > > # #lb_disable_dst(); > > > > if (!lookup("location","m")) { > > > > route(relay); > > > > exit; > > > > } > > > > if (lookup("location","m")) { > > > > lb_disable_dst(); > > > > route(relay); > > > > exit; > > > > } > > > > } > > > > > > > > thanks in advance, > > > > > > > > michael. > > > > > > > > > > > > _______________________________________________ > > > > Users mailing list > > > > Users at lists.opensips.org > > > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > -- > Regards, > > David Villasmilemail: david.villasmil.work at gmail.com > phone: +34669448337 -------------- next part -------------- An HTML attachment was scrubbed... URL: From podguiko at mail.ru Sun Apr 5 17:58:28 2020 From: podguiko at mail.ru (=?UTF-8?B?T2xlZyBQb2RndXlrbw==?=) Date: Sun, 05 Apr 2020 20:58:28 +0300 Subject: [OpenSIPS-Users] =?utf-8?q?_Re=3A_t=5Fforward=5Fnonack_failed_=28?= =?utf-8?q?Ben_Newlin=29?= In-Reply-To: References: Message-ID: <1586109508.907336702@f327.i.mail.ru> Hello Ben,   Thank you for answer.   This is my RELAY route   route[RELAY] {     if (!t_relay())     {         sl_reply_error();     }     exit; }   I did the following experiment. Commented out the line #sl_reply_error(); Sent Invite and Cancel. Opensips did not send 500.   In fact, it is «sl_reply_error» that sends 500. it looks like:   SIP/2.0 500 No destination available (18/SL) It is clear now.   Following your advice I tried to use avp variable.   ## CANCEL processing     if (is_method("CANCEL"))     {         ## returns true if the current request is associated to a transaction         if (t_check_trans())         {             $avp(ci)=&ci             t_relay();         }         exit;     }     But when I try to use this variable( $avp(ci)) at the [resume] route of rest_client, it is     >Воскресенье, 5 апреля 2020, 16:55 +03:00 от users-request at lists.opensips.org: >  >Send Users mailing list submissions to >users at lists.opensips.org > >To subscribe or unsubscribe via the World Wide Web, visit >http://lists.opensips.org/cgi-bin/mailman/listinfo/users >or, via email, send a message with subject or body 'help' to >users-request at lists.opensips.org > >You can reach the person managing the list at >users-owner at lists.opensips.org > >When replying, please edit your Subject line so it is more specific >than "Re: Contents of Users digest..." > > >Today's Topics: > >   1. Re: t_forward_nonack failed (Ben Newlin) >   2. Re: using load balancer and lookup together (Michael Vale) > > >---------------------------------------------------------------------- > >Message: 1 >Date: Sun, 5 Apr 2020 13:11:34 +0000 >From: Ben Newlin < Ben.Newlin at genesys.com > >To: Oleg Podguyko < podguiko at mail.ru >, OpenSIPS users mailling list >< users at lists.opensips.org > >Subject: Re: [OpenSIPS-Users] t_forward_nonack failed >Message-ID: < 00BE023F-DA3A-45EE-988F-BA4891FB9E04 at genesys.com > >Content-Type: text/plain; charset="utf-8" > >Oleg, > >At first glance, it seems like t_was_cancelled [1] is exactly what you want. However, it can only be called from onreply_route or failure_route, which makes it unusable for this case. This has caused problems for me as well, as it makes it very hard to detect cancellation when working in asynchronous routes that don’t involve SIP messages. Similarly, t_cancel_branch can only be called from onreply_route, which also causes issues with not being able to manage branches with non-SIP async operations. It may be a good feature request to add the ability to use those functions in async resume routes, but for now they will not work. > >However, t_relay itself does not ever generate a 500 error back to the client as far as I know. It can only send a 477 response automatically. Are you sure you are not generating the 500 in your script when handling the error from t_relay? The documentation for the function [2] notes that a -3 response from t_relay indicates the request may have already been cancelled. In that case, you can just exit if you know the 487 has already been sent, or you can send a 487 reply yourself. To be honest, I’m not sure where the 487 is coming from in your case since I didn’t think OpenSIPS would automatically respond with a 487 for a cancelled transaction; I have always had to do that from the script myself. So in addition to sending the 500 reply yourself, you may have some code in your script which is also sending the 487. > >If none of that works, another option would be to set a flag or an avp in the transaction when processing the CANCEL. Then you can check the flag when you receive the HTTP response and if it is set just don’t call t_relay. > >[1] https://opensips.org/docs/modules/3.0.x/tm.html#func_t_was_cancelled >[2] https://opensips.org/docs/modules/3.0.x/tm.html#func_t_relay > >Ben Newlin > >From: Users < users-bounces at lists.opensips.org > on behalf of Oleg Podguyko via Users < users at lists.opensips.org > >Reply-To: Oleg Podguyko < podguiko at mail.ru >, OpenSIPS users mailling list < users at lists.opensips.org > >Date: Sunday, April 5, 2020 at 6:31 AM >To: users < users at lists.opensips.org > >Subject: [OpenSIPS-Users] t_forward_nonack failed > > >Opensips works like a proxy. Gets an Invite. In the process of processing it, opensips makes a request via http (rest_client module) and receives a response. >Adds the received information to Invite (as X-header) and sends through the dispatcher module to freeswitch. >Everything is working fine. >However, there is a scenario in which everything goes a little wrong. >Opensips receives an Invite, starts processing it, sends a request via http. At this time, Cancel arrives at Invite. The transaction is being destroyed. Opensips sends 200 Cancelling, and then 487. And here comes the response via HTTP, but since the transaction is no longer there, this call ends with an error 500 no route to destination when the t_relay function is executed. In this case, a message appears in the log >/ usr / sbin / opensips [5577]: ERROR: tm: w_t_relay: t_forward_nonack failed. >How to correctly handle such cases in order to prevent such errors in the logs? > >-- >Oleg Podguyko >-------------- next part -------------- >An HTML attachment was scrubbed... >URL: < http://lists.opensips.org/pipermail/users/attachments/20200405/8680c7ac/attachment-0001.html > > >------------------------------ > >Message: 2 >Date: Sun, 05 Apr 2020 23:54:29 +1000 >From: Michael Vale < masked at vale.ski > >To: David Villasmil < david.villasmil.work at gmail.com >, OpenSIPS users >mailling list < users at lists.opensips.org > >Subject: Re: [OpenSIPS-Users] using load balancer and lookup together >Message-ID: < 6342fd559df4c9765b17270d07bc04f11ad93b36.camel at vale.ski > >Content-Type: text/plain; charset="utf-8" > >Ok, to explain, >Using your logic, >To call say '555' will goto Voicemail, if I disable voicemail it will >return a 404 instead of going to the load balancer. >That's fine, if 555 is an extension, but if it's (for this example) a >PSTN number (or a all-else catch all, like I'm trying to achieve) thats >not OK because I get a 404 rather than it getting routed to the load >balancer. >If 555 is an extension/user the call will go through to the registered >extension, but if it's not registered in the usrloc table, it goes to >404, instead of the load balancer. >If I reverse the logic, It will goto the load balancer even if it's a >registered extension, or Too Many Hops, depending on how I adjust the >logic. >I cannot seem to create a catch all for non-usrloc registered extension >calls to goto the load balancer otherwise return a 404. >I hope I explained it well enough. I will keep trying, >Regards, >Michael. >On Sun, 2020-04-05 at 11:47 +0100, David Villasmil wrote: >> Why are you trying to do all at once? >> >> Why not first do the lookup >> >> https://github.com/davidcsi/kamailio-private-public/blob/a81d7f777a8c5ee2dbb32311f7e6b5a3cf94bf32/kamailio.cfg#L771 >> >> >> and then start load balancing? >> >> https://github.com/davidcsi/kamailio-private-public/blob/a81d7f777a8c5ee2dbb32311f7e6b5a3cf94bf32/kamailio.cfg#L1109 >> >> Do you have some special need to fulfill? >> >> David >> On Sun, 5 Apr 2020 at 06:34, Michael Vale via Users < >> users at lists.opensips.org > wrote: >> > hi, >> > >> > >> > >> > perhaps this can be solved with a failure route and or a check >> > status >> > >> > but i dont know and it would be nice if i could do it without it. >> > >> > >> > >> > no matter how i write the script, either a uac to uac call goes to >> > the >> > >> > load balancer or the load balancer is stuck with a 404 reply from >> > the >> > >> > script or uac to uac works but when one end is not registered it >> > goes >> > >> > to the load balancer instead of getting a 404. >> > >> > >> > >> > i've tried failure routes and get the same problem. here is a >> > snippet. >> > >> > >> > >> > if (!lb_start(1,"pstn")) && (!lookup("location","m",)) { >> > >> > lb_disable_dst(); >> > >> > #route(relay); >> > >> > #send_reply(404,"No user or gateway"); >> > >> > if (lb_start(1,"pstn")) { >> > >> > send_reply(500,"SIPSIPSIPS"); >> > >> > # t_relay(); >> > >> > exit; >> > >> > } >> > >> > # exit; >> > >> > } else if (lookup("location","m")) && >> > >> > (!lb_start(1,"pstn")) { >> > >> > lb_disable_dst(); >> > >> > route(relay); >> > >> > exit; >> > >> > } else if (lb_start(1,"pstn")) && >> > >> > (lookup("location","m")) { >> > >> > lb_disable_dst(); >> > >> > route(relay); >> > >> > exit; >> > >> > } else if (!lookup("location","m")) && >> > >> > (!lb_start(1,"pstn")) { >> > >> > send_reply(404,"Not Found"); >> > >> > exit; >> > >> > } else if (lb_start(1,"pstn")) && >> > >> > (!lookup("location","m")) { >> > >> > # #lb_disable_dst(); >> > >> > if (!lookup("location","m")) { >> > >> > route(relay); >> > >> > exit; >> > >> > } >> > >> > if (lookup("location","m")) { >> > >> > lb_disable_dst(); >> > >> > route(relay); >> > >> > exit; >> > >> > } >> > >> > } >> > >> > >> > >> > thanks in advance, >> > >> > >> > >> > michael. >> > >> > >> > >> > >> > >> > _______________________________________________ >> > >> > Users mailing list >> > >> > Users at lists.opensips.org >> > >> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> > >> -- >> Regards, >> >> David Villasmilemail: david.villasmil.work at gmail.com >> phone: +34669448337 >-------------- next part -------------- >An HTML attachment was scrubbed... >URL: < http://lists.opensips.org/pipermail/users/attachments/20200405/6ef5eec8/attachment.html > > >------------------------------ > >Subject: Digest Footer > >_______________________________________________ >Users mailing list >Users at lists.opensips.org >http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > >------------------------------ > >End of Users Digest, Vol 141, Issue 8 >*************************************     -- Олег Подгуйко   -------------- next part -------------- An HTML attachment was scrubbed... URL: From Ben.Newlin at genesys.com Sun Apr 5 19:08:28 2020 From: Ben.Newlin at genesys.com (Ben Newlin) Date: Sun, 5 Apr 2020 19:08:28 +0000 Subject: [OpenSIPS-Users] using load balancer and lookup together In-Reply-To: <6342fd559df4c9765b17270d07bc04f11ad93b36.camel@vale.ski> References: <6342fd559df4c9765b17270d07bc04f11ad93b36.camel@vale.ski> Message-ID: <398E61A5-CEE3-4034-8C19-8C29C101FA8B@genesys.com> That is what failure_routes are for. You can “catch” the 404 in a failure_route and choose somewhere else to attempt to send the call. You don’t have to just send back the response all the time. But what I think David was trying to ask is just why you are combining the lb_start and lookup calls always into a single if statement. This is very confusing and will end up with lb_start and/or lookup being called more than once in many cases. I don’t use the load_balancer module, but I think calling lb_start multiple times in the same call/transaction may skew the load balancing stats. In your code snippet, you are even calling both lb_start and lookup again within if legs where their return values have already been checked. Ben Newlin From: Users on behalf of Michael Vale via Users Reply-To: Michael Vale , OpenSIPS users mailling list Date: Sunday, April 5, 2020 at 9:57 AM To: David Villasmil , OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] using load balancer and lookup together Ok, to explain, Using your logic, To call say '555' will goto Voicemail, if I disable voicemail it will return a 404 instead of going to the load balancer. That's fine, if 555 is an extension, but if it's (for this example) a PSTN number (or a all-else catch all, like I'm trying to achieve) thats not OK because I get a 404 rather than it getting routed to the load balancer. If 555 is an extension/user the call will go through to the registered extension, but if it's not registered in the usrloc table, it goes to 404, instead of the load balancer. If I reverse the logic, It will goto the load balancer even if it's a registered extension, or Too Many Hops, depending on how I adjust the logic. I cannot seem to create a catch all for non-usrloc registered extension calls to goto the load balancer otherwise return a 404. I hope I explained it well enough. I will keep trying, Regards, Michael. On Sun, 2020-04-05 at 11:47 +0100, David Villasmil wrote: Why are you trying to do all at once? Why not first do the lookup https://github.com/davidcsi/kamailio-private-public/blob/a81d7f777a8c5ee2dbb32311f7e6b5a3cf94bf32/kamailio.cfg#L771 and then start load balancing? https://github.com/davidcsi/kamailio-private-public/blob/a81d7f777a8c5ee2dbb32311f7e6b5a3cf94bf32/kamailio.cfg#L1109 Do you have some special need to fulfill? David On Sun, 5 Apr 2020 at 06:34, Michael Vale via Users > wrote: hi, perhaps this can be solved with a failure route and or a check status but i dont know and it would be nice if i could do it without it. no matter how i write the script, either a uac to uac call goes to the load balancer or the load balancer is stuck with a 404 reply from the script or uac to uac works but when one end is not registered it goes to the load balancer instead of getting a 404. i've tried failure routes and get the same problem. here is a snippet. if (!lb_start(1,"pstn")) && (!lookup("location","m",)) { lb_disable_dst(); #route(relay); #send_reply(404,"No user or gateway"); if (lb_start(1,"pstn")) { send_reply(500,"SIPSIPSIPS"); # t_relay(); exit; } # exit; } else if (lookup("location","m")) && (!lb_start(1,"pstn")) { lb_disable_dst(); route(relay); exit; } else if (lb_start(1,"pstn")) && (lookup("location","m")) { lb_disable_dst(); route(relay); exit; } else if (!lookup("location","m")) && (!lb_start(1,"pstn")) { send_reply(404,"Not Found"); exit; } else if (lb_start(1,"pstn")) && (!lookup("location","m")) { # #lb_disable_dst(); if (!lookup("location","m")) { route(relay); exit; } if (lookup("location","m")) { lb_disable_dst(); route(relay); exit; } } thanks in advance, michael. _______________________________________________ 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 venefax at gmail.com Sun Apr 5 19:37:49 2020 From: venefax at gmail.com (Saint Michael) Date: Sun, 5 Apr 2020 15:37:49 -0400 Subject: [OpenSIPS-Users] ODBC unexplained error Message-ID: ERROR:db_unixodbc:db_unixodbc_extract_error: unixodbc:SQLDriverConnect=IM008:1:0:[unixODBC][ma-3.1.6]Dialog failed ... avpops:avpops_db_init: cannot initialize database connection for unixodbc://localhost/local but I can do "isql localhost", so odbc is working Any idea? -------------- next part -------------- An HTML attachment was scrubbed... URL: From masked at vale.ski Mon Apr 6 01:58:29 2020 From: masked at vale.ski (Michael Vale) Date: Mon, 06 Apr 2020 11:58:29 +1000 Subject: [OpenSIPS-Users] using load balancer and lookup together In-Reply-To: <398E61A5-CEE3-4034-8C19-8C29C101FA8B@genesys.com> References: <6342fd559df4c9765b17270d07bc04f11ad93b36.camel@vale.ski> <398E61A5-CEE3-4034-8C19-8C29C101FA8B@genesys.com> Message-ID: <5c8dad0271e41ba2429d26d4603e1919ef2fd685.camel@vale.ski> Thanks David and thanks Ben, I have implemented a failure route that works now. I will rectify the duplicate lb_start calls and I'm good. regards, Michael. On Sun, 2020-04-05 at 19:08 +0000, Ben Newlin wrote: > That is what failure_routes are for. You can “catch” the 404 in a > failure_route and choose somewhere else to attempt to send the call. > You don’t have to just send back the response all the time. > > But what I think David was trying to ask is just why you are > combining the lb_start and lookup calls always into a single if > statement. This is very confusing and will end up with lb_start > and/or lookup being called more than once in many > cases. I don’t use the load_balancer module, but I think calling > lb_start multiple times in the same call/transaction may skew the > load balancing stats. > > In your code snippet, you are even calling both lb_start and lookup > again within if legs where their return values have already been > checked. > > Ben Newlin > > > From: Users on behalf of Michael > Vale via Users > > Reply-To: Michael Vale , OpenSIPS users mailling > list > > Date: Sunday, April 5, 2020 at 9:57 AM > > To: David Villasmil , OpenSIPS users > mailling list > > Subject: Re: [OpenSIPS-Users] using load balancer and lookup together > > > > > > Ok, to explain, > > > > > > Using your logic, > > > > > > To call say '555' will goto Voicemail, if I disable voicemail it will > return a 404 instead of going to the load balancer. > > > > > > That's fine, if 555 is an extension, but if it's (for this example) a > PSTN number (or a all-else catch all, like I'm trying to achieve) > thats not OK because I get a 404 rather than it getting routed to the > load balancer. > > > > > > If 555 is an extension/user the call will go through to the > registered extension, but if it's not registered in the usrloc table, > it goes to 404, instead of the load balancer. > > > > > > If I reverse the logic, It will goto the load balancer even if it's a > registered extension, or Too Many Hops, depending on how I adjust the > logic. > > > > > > I cannot seem to create a catch all for non-usrloc registered > extension calls to goto the load balancer otherwise return a 404. > > > > > > I hope I explained it well enough. I will keep trying, > > > > > > Regards, > > > > > > Michael. > > > > > > On Sun, 2020-04-05 at 11:47 +0100, David Villasmil wrote: > > > > > Why are you trying to do all at once? > > > > > > > > > > > > Why not first do the lookup > > > > > > > > > > > > > > https://github.com/davidcsi/kamailio-private-public/blob/a81d7f777a8c5ee2dbb32311f7e6b5a3cf94bf32/kamailio.cfg#L771 > > > > > > > > > > > > > > > > > > > > and then start load balancing? > > > > > > > > > > > > > > https://github.com/davidcsi/kamailio-private-public/blob/a81d7f777a8c5ee2dbb32311f7e6b5a3cf94bf32/kamailio.cfg#L1109 > > > > > > > > > > > > > > > > Do you have some special need to fulfill? > > > > > > > > > > > > David > > > > > > > > > > > > > > On Sun, 5 Apr 2020 at 06:34, Michael Vale via Users < > > users at lists.opensips.org> wrote: > > > > > hi, > > > > > > > > > > > > perhaps this can be solved with a failure route and or a check > > > status > > > > > > but i dont know and it would be nice if i could do it without it. > > > > > > > > > > > > no matter how i write the script, either a uac to uac call goes > > > to the > > > > > > load balancer or the load balancer is stuck with a 404 reply from > > > the > > > > > > script or uac to uac works but when one end is not registered it > > > goes > > > > > > to the load balancer instead of getting a 404. > > > > > > > > > > > > i've tried failure routes and get the same problem. here is a > > > snippet. > > > > > > > > > > > > if (!lb_start(1,"pstn")) && (!lookup("location","m",)) { > > > > > > lb_disable_dst(); > > > > > > #route(relay); > > > > > > #send_reply(404,"No user or gateway"); > > > > > > if (lb_start(1,"pstn")) { > > > > > > send_reply(500,"SIPSIPSIPS"); > > > > > > # t_relay(); > > > > > > exit; > > > > > > } > > > > > > # exit; > > > > > > } else if (lookup("location","m")) && > > > > > > (!lb_start(1,"pstn")) { > > > > > > lb_disable_dst(); > > > > > > route(relay); > > > > > > exit; > > > > > > } else if (lb_start(1,"pstn")) && > > > > > > (lookup("location","m")) { > > > > > > lb_disable_dst(); > > > > > > route(relay); > > > > > > exit; > > > > > > } else if (!lookup("location","m")) && > > > > > > (!lb_start(1,"pstn")) { > > > > > > send_reply(404,"Not Found"); > > > > > > exit; > > > > > > } else if (lb_start(1,"pstn")) && > > > > > > (!lookup("location","m")) { > > > > > > # #lb_disable_dst(); > > > > > > if (!lookup("location","m")) { > > > > > > route(relay); > > > > > > exit; > > > > > > } > > > > > > if (lookup("location","m")) { > > > > > > lb_disable_dst(); > > > > > > route(relay); > > > > > > exit; > > > > > > } > > > > > > } > > > > > > > > > > > > thanks in advance, > > > > > > > > > > > > michael. > > > > > > > > > > > > > > > > > > _______________________________________________ > > > > > > 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 y.kirsanov at gmail.com Mon Apr 6 04:04:08 2020 From: y.kirsanov at gmail.com (Yury Kirsanov) Date: Mon, 6 Apr 2020 14:04:08 +1000 Subject: [OpenSIPS-Users] Issue with 'To' tag and t_reply In-Reply-To: References: Message-ID: Hi Ben, The problem we're facing now is that according to RFC3261 dialog is established after 183 Session Progress with a To tag, so Server A can't continue to receive out-of-dialog SIP messages. In this case we're unable to send a OpenSIPS-generated message with different To tag which occurs in my situation. Is there any way to resolve this situation? It looks to me that behaviour of Server A is correct as OpenSIPS acts as a proxy and passes messages from Server B and then suddenly injects a SIP message originated by itself. Looks like there has to be two 'legs' of the call, one between Server A and OpenSIPS and another one between OpenSIPS and multiple servers it tries to reach in order to establish the call, but in this case OpenSIPS can't act as a pure proxy. Please advise? Thanks!. P.S. For some reason I'm not receiving your responses in my mailbox? пт, 27 мар. 2020 г. в 01:56, Yury Kirsanov : > Thanks a lot for your explanation, Ben! I thought that there can be an > issue with Server A not accepting my new SIP response, it looks like > they're doing matching only based on SIP To tag and completely ignoring any > Call-ID or DID matching as well as From tag matching, in my case From tag > is always the same. > > One more question, how do you think, can there be anything related with > topology hiding I'm using? I really doubt that but just in case...As far as > I understand my issue is not because I'm using topology hiding, but because > OpenSIPS first passes To tag from remote server and then generates one by > itself when I'm using a 't_reply' and Server A is just not accepting such > behaviour, trying to match any SIP responses to To tag passed in 183 > Session Progress. I tried to change topology_hiding() to loose_route() and > nothing changed in my chain except for Server A now being able to see all > RRs and Vias inside my network. > > Regards, > Yury. > > пт, 27 мар. 2020 г. в 01:37, Yury Kirsanov : > >> But the question is still here - how can I send a different t_reply code >> from failure_route? And then stop processing any further SIP messages? >> >> пт, 27 мар. 2020 г. в 01:23, Yury Kirsanov : >> >>> The problem is that I need to go through a list of SIP servers, analyze >>> response of each of them and if it's an error like 4XX, 5XX or 6XX I need >>> to send appropriate response to originating server. Let's say I'm not only >>> adding a Reason field but upon receipt of 404 Not Found I want to respond >>> with 480 Temporarily Unavailable with Reason: Q.850;Cause=41 for example? >>> But Server B first replied with 183 Session Progress playing back a message >>> 'Sorry, you need to top up your account' and then replied with SIP 402 >>> Payment Required. I had to proxy 183 Session Progress back to Server A so >>> its SIP client could hear that message and then I'd like to signal 480 >>> Temporarily Unavailable - but I can't as OpenSIPS is using completely >>> different To tag. >>> >>> I can't do this in onreply_route as I'm going through a list of SIP >>> servers (upstreams or downstreams), so it definitely needs to be done from >>> failure route, as far as I understand, and yes, I'm matching against 4XX, >>> 5XX and 6XX codes and I need to reply with 480 Temporarily Unavailable in >>> most cases so Server A would have a possibility to do failover to any other >>> server in that case. I don't want to just proxy 4XX, 5XX or 6XX response to >>> it. >>> >>> I've figured out why I have two 404 responses in my original call log - >>> I was using sl_send_reply instead of t_reply and it was using original To >>> tag but only on second attempt. >>> >>> Regards, >>> Yury. >>> >>> пт, 27 мар. 2020 г. в 01:02, Yury Kirsanov : >>> >>>> Hi, >>>> As per my original email: >>>> 1. I was doing exactly as you suggested, in failure_route I'm using >>>> t_reply("404","Not Found") and it comes out with a wrong To: tag. >>>> 2. I don't need to proxy response from server B, I need to analyze its >>>> response and send a response to server A according to my needs. >>>> >>>> Currently it seems that t_reply is not using same To tag if 183 Session >>>> Progress has been proxied, which is strange as I have dialog running. >>>> >>>> Regards, >>>> Yury. >>>> >>>> чт, 26 мар. 2020 г. в 19:13, Yury Kirsanov : >>>> >>>>> Hi, >>>>> I'm using an OpenSIPS as a proxy between two servers. First one is >>>>> sending SIP INVITE to OpenSIPS, then OpenSIPS forwards request to second >>>>> server. I'm creating a dialog on initial INVITE. Second server then replies >>>>> with SIP 183 Session Progress, plays back a message and then responds with >>>>> 4XX code, for example SIP 404 Not Found (indicating that number dialed is >>>>> disconnected). In OpenSIPS I'm receiving that reply and in failure_route >>>>> I'd like to change that code to a bit different SIP 404, so I'm using >>>>> following code: >>>>> >>>>> append_to_reply("Reason: Q.850;cause=1"); >>>>> t_reply("404","Not Found"); >>>>> exit; >>>>> >>>>> But in this case I can see that OpenSIPS generates additional branch >>>>> (??? not sure here) with different "To" tag and pushes it out and then >>>>> forwards original reply SIP packet even though I have an exit statement in >>>>> my failure_route. I tried to do sl_send_reply and behavior is pretty much >>>>> the same. Can someone let me know what I may be doing wrong? I need correct >>>>> "To" tag to be used (based on 183 Session Progress message from server B >>>>> and passed to server A previously) and second 404 shouldn't be forwarded >>>>> out. >>>>> >>>>> Here's an example of a call with my explanations >>>>> >>>>> Initial invite from server A, no 'to tag' as expected: >>>>> >>>>> INVITE sip:XXXXXXXXX at B.B.B.B SIP/2.0 >>>>> Max-Forwards: 67 >>>>> To: "XXXXXXXXX" >>>>> Call-ID: 469A5568-E092-4038-B1B8-13AC9B9571CA >>>>> Via: SIP/2.0/UDP A.A.A.A:5060;rport;branch=z9hG4bK773616538 >>>>> From: "YYYYYYYYY" ;tag=117583367 >>>>> CSeq: 1741310 INVITE >>>>> User-Agent: User Agent >>>>> Contact: >>>>> Allow: ACK, INVITE, BYE, CANCEL, REGISTER, REFER, OPTIONS, INFO, >>>>> SUBSCRIBE, NOTIFY >>>>> Date: Thu, 26 Mar 2020 07:54:55 GMT >>>>> Content-Type: application/sdp >>>>> Content-Length: 250 >>>>> >>>>> v=0 >>>>> o=dcom 1585209295 1585209295 IN IP4 A.A.A.A >>>>> s=SIP Call >>>>> c=IN IP4 A.A.A.A >>>>> t=0 0 >>>>> m=audio 15340 RTP/AVP 8 0 18 101 >>>>> a=rtpmap:8 PCMA/8000 >>>>> a=rtpmap:0 PCMU/8000 >>>>> a=rtpmap:18 G729/8000 >>>>> a=fmtp:18 annexb=no >>>>> a=rtpmap:101 telephone-event/8000 >>>>> >>>>> Response from OpenSIPS: >>>>> >>>>> SIP/2.0 100 Giving a try >>>>> To: "XXXXXXXXX" >>>>> Call-ID: 469A5568-E092-4038-B1B8-13AC9B9571CA >>>>> Via: SIP/2.0/UDP >>>>> A.A.A.A:5060;received=A.A.A.A;rport=5060;branch=z9hG4bK773616538 >>>>> From: "YYYYYYYYY" ;tag=117583367 >>>>> CSeq: 1741310 INVITE >>>>> Server: Server Signature >>>>> Content-Length: 0 >>>>> >>>>> OpenSIPS has forwarded packet to Server B and Server B responded with >>>>> 183 and assigned a 'To' tag: >>>>> >>>>> SIP/2.0 183 Session Progress >>>>> Via: SIP/2.0/UDP >>>>> A.A.A.A:5060;received=A.A.A.A;rport=5060;branch=z9hG4bK773616538 >>>>> Call-ID: 469A5568-E092-4038-B1B8-13AC9B9571CA >>>>> From: "YYYYYYYYY" ;tag=117583367 >>>>> To: "XXXXXXXXX" ; >>>>> *tag=0b49dc32-2c4b-413e-a349-c781a23d53b9* >>>>> CSeq: 1741310 INVITE >>>>> Server: PBX >>>>> Contact: >>>>> Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, >>>>> BYE, CANCEL, UPDATE, PRACK, REFER >>>>> Content-Type: application/sdp >>>>> Content-Length: 354 >>>>> >>>>> v=0 >>>>> o=- 1585209295 1585209297 IN IP4 B.B.B.B >>>>> s=Asterisk >>>>> c=IN IP4 B.B.B.B >>>>> t=0 0 >>>>> a=rtpengine:673f999268ae >>>>> m=audio 32386 RTP/AVP 0 8 18 101 >>>>> a=maxptime:150 >>>>> a=rtpmap:0 PCMU/8000 >>>>> a=rtpmap:8 PCMA/8000 >>>>> a=rtpmap:18 G729/8000 >>>>> a=rtpmap:101 telephone-event/8000 >>>>> a=fmtp:18 annexb=no >>>>> a=fmtp:101 0-16 >>>>> a=sendrecv >>>>> a=rtcp:32387 >>>>> a=ptime:20 >>>>> >>>>> Server B responds with SIP 404 after playing back message that number >>>>> is disconnected and I'm trying to reply to server A with custom Reason >>>>> message. To_tag is completely different from the To tag that has been >>>>> passed to server A after initial 183!!! >>>>> >>>>> SIP/2.0 404 Not Found >>>>> To: "XXXXXXXXX" ; >>>>> *tag=a976.21514595b467be41a9b712a6b0b621d9* >>>>> Call-ID: 469A5568-E092-4038-B1B8-13AC9B9571CA >>>>> Via: SIP/2.0/UDP >>>>> A.A.A.A:5060;received=A.A.A.A;rport=5060;branch=z9hG4bK773616538 >>>>> From: "YYYYYYYYY" ;tag=117583367 >>>>> CSeq: 1741310 INVITE >>>>> Reason: Q.850;cause=1;text="Number is disconnected" >>>>> Server: Server Signature >>>>> Content-Length: 0 >>>>> >>>>> Of course, server A just ignores this message as it can't match 'To' >>>>> tag to its transaction. Now, for some reason, OpenSIPS forwards original >>>>> reply from Server B to Server A with the same 'To' tag as in 183 Session >>>>> Progress: >>>>> >>>>> SIP/2.0 404 Not Found >>>>> Via: SIP/2.0/UDP >>>>> A.A.A.A:5060;received=A.A.A.A;rport=5060;branch=z9hG4bK773616538 >>>>> Call-ID: 469A5568-E092-4038-B1B8-13AC9B9571CA >>>>> From: "YYYYYYYYY" ;tag=117583367 >>>>> To: "XXXXXXXXX" ; >>>>> *tag=0b49dc32-2c4b-413e-a349-c781a23d53b9* >>>>> CSeq: 1741310 INVITE >>>>> Server: PBX >>>>> Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, >>>>> BYE, CANCEL, UPDATE, PRACK, REFER >>>>> Reason: Q.850;cause=1 >>>>> Content-Length: 0 >>>>> >>>>> And at this point Server A can match this reply and responds with an >>>>> ACK: >>>>> >>>>> ACK sip:XXXXXXXXX at B.B.B.B SIP/2.0 >>>>> Via: SIP/2.0/UDP A.A.A.A:5060;rport;branch=z9hG4bK773616538 >>>>> From: "YYYYYYYYY" ;tag=117583367 >>>>> To: "XXXXXXXXX" ; >>>>> *tag=0b49dc32-2c4b-413e-a349-c781a23d53b9* >>>>> Call-ID: 469A5568-E092-4038-B1B8-13AC9B9571CA >>>>> CSeq: 1741310 ACK >>>>> Max-Forwards: 67 >>>>> Contact: >>>>> User-Agent: User Agent >>>>> Content-Length: 0 >>>>> >>>>> I think that t_reply is creating a new transaction instead of using >>>>> existing one, but I'm not sure why and how to fix this? >>>>> >>>>> Thanks! >>>>> >>>>> Best regards, >>>>> Yury. >>>>> >>>> -------------- next part -------------- An HTML attachment was scrubbed... URL: From farmorg at gmail.com Mon Apr 6 11:08:13 2020 From: farmorg at gmail.com (Mark Farmer) Date: Mon, 6 Apr 2020 12:08:13 +0100 Subject: [OpenSIPS-Users] Access to listen/advertised IP Addresses In-Reply-To: References: Message-ID: Many thanks for the reply. $Ri is certainly useful when the request comes from a non-natted interface. Thanks for pointing that out :) Is there a way to reference the advertised IP address defined in the listen statement? listen=udp:xxx.xxx.xxx.xxx:5060 as xxx.xxx.xxx.xxx:5060 Thanks Mark. On Thu, 2 Apr 2020 at 17:32, Sharad Kumar via Users < users at lists.opensips.org> wrote: > Hi Mark, > > If your initial goal is to get the interface IP where request is received > then you can try these variables. > > *$Ri* - reference to IP address of the interface where the request has > been received > > *$Rp* - reference to the port where the message was received > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -- Mark Farmer farmorg at gmail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From david.villasmil.work at gmail.com Mon Apr 6 12:42:58 2020 From: david.villasmil.work at gmail.com (David Villasmil) Date: Mon, 6 Apr 2020 13:42:58 +0100 Subject: [OpenSIPS-Users] Access to listen/advertised IP Addresses In-Reply-To: References: Message-ID: Right here: https://www.kamailio.org/wiki/cookbooks/5.2.x/pseudovariables#rai_-_received_advertised_ip_address Regards, David Villasmil email: david.villasmil.work at gmail.com phone: +34669448337 On Mon, Apr 6, 2020 at 12:08 PM Mark Farmer wrote: > Many thanks for the reply. > > $Ri is certainly useful when the request comes from a non-natted > interface. Thanks for pointing that out :) > > Is there a way to reference the advertised IP address defined in the > listen statement? > > listen=udp:xxx.xxx.xxx.xxx:5060 as xxx.xxx.xxx.xxx:5060 > > Thanks > Mark. > > > On Thu, 2 Apr 2020 at 17:32, Sharad Kumar via Users < > users at lists.opensips.org> wrote: > >> Hi Mark, >> >> If your initial goal is to get the interface IP where request is >> received then you can try these variables. >> >> *$Ri* - reference to IP address of the interface where the request has >> been received >> >> *$Rp* - reference to the port where the message was received >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> > > > -- > Mark Farmer > farmorg at gmail.com > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -------------- next part -------------- An HTML attachment was scrubbed... URL: From farmorg at gmail.com Mon Apr 6 13:04:04 2020 From: farmorg at gmail.com (Mark Farmer) Date: Mon, 6 Apr 2020 14:04:04 +0100 Subject: [OpenSIPS-Users] Access to listen/advertised IP Addresses In-Reply-To: References: Message-ID: Thanks David. But I see no reference to the same variable in OpenSIPS. https://www.opensips.org/Documentation/Script-CoreVar-2-4 Am I missing something? On Mon, 6 Apr 2020 at 13:45, David Villasmil wrote: > Right here: > > > https://www.kamailio.org/wiki/cookbooks/5.2.x/pseudovariables#rai_-_received_advertised_ip_address > > Regards, > > David Villasmil > email: david.villasmil.work at gmail.com > phone: +34669448337 > > > On Mon, Apr 6, 2020 at 12:08 PM Mark Farmer wrote: > >> Many thanks for the reply. >> >> $Ri is certainly useful when the request comes from a non-natted >> interface. Thanks for pointing that out :) >> >> Is there a way to reference the advertised IP address defined in the >> listen statement? >> >> listen=udp:xxx.xxx.xxx.xxx:5060 as xxx.xxx.xxx.xxx:5060 >> >> Thanks >> Mark. >> >> >> On Thu, 2 Apr 2020 at 17:32, Sharad Kumar via Users < >> users at lists.opensips.org> wrote: >> >>> Hi Mark, >>> >>> If your initial goal is to get the interface IP where request is >>> received then you can try these variables. >>> >>> *$Ri* - reference to IP address of the interface where the request has >>> been received >>> >>> *$Rp* - reference to the port where the message was received >>> _______________________________________________ >>> Users mailing list >>> Users at lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >> >> >> -- >> Mark Farmer >> farmorg at gmail.com >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -- Mark Farmer farmorg at gmail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From david.villasmil.work at gmail.com Mon Apr 6 14:49:36 2020 From: david.villasmil.work at gmail.com (David Villasmil) Date: Mon, 6 Apr 2020 15:49:36 +0100 Subject: [OpenSIPS-Users] Access to listen/advertised IP Addresses In-Reply-To: References: Message-ID: No, you’re right. It’s not in the core variables and I can’t find it either. Which makes me think it’s either not exposed or somewhere in a module (it’s not in proto_udp) I will research a little to try and find it.. On Mon, 6 Apr 2020 at 14:04, Mark Farmer wrote: > Thanks David. But I see no reference to the same variable in OpenSIPS. > > https://www.opensips.org/Documentation/Script-CoreVar-2-4 > > Am I missing something? > > > On Mon, 6 Apr 2020 at 13:45, David Villasmil < > david.villasmil.work at gmail.com> wrote: > >> Right here: >> >> >> https://www.kamailio.org/wiki/cookbooks/5.2.x/pseudovariables#rai_-_received_advertised_ip_address >> >> Regards, >> >> David Villasmil >> email: david.villasmil.work at gmail.com >> phone: +34669448337 >> >> >> On Mon, Apr 6, 2020 at 12:08 PM Mark Farmer wrote: >> >>> Many thanks for the reply. >>> >>> $Ri is certainly useful when the request comes from a non-natted >>> interface. Thanks for pointing that out :) >>> >>> Is there a way to reference the advertised IP address defined in the >>> listen statement? >>> >>> listen=udp:xxx.xxx.xxx.xxx:5060 as xxx.xxx.xxx.xxx:5060 >>> >>> Thanks >>> Mark. >>> >>> >>> On Thu, 2 Apr 2020 at 17:32, Sharad Kumar via Users < >>> users at lists.opensips.org> wrote: >>> >>>> Hi Mark, >>>> >>>> If your initial goal is to get the interface IP where request is >>>> received then you can try these variables. >>>> >>>> *$Ri* - reference to IP address of the interface where the request has >>>> been received >>>> >>>> *$Rp* - reference to the port where the message was received >>>> _______________________________________________ >>>> Users mailing list >>>> Users at lists.opensips.org >>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>> >>> >>> >>> -- >>> Mark Farmer >>> farmorg at gmail.com >>> _______________________________________________ >>> Users mailing list >>> Users at lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> > > > -- > Mark Farmer > farmorg at gmail.com > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -- Regards, David Villasmil email: david.villasmil.work at gmail.com phone: +34669448337 -------------- next part -------------- An HTML attachment was scrubbed... URL: From johan at democon.be Mon Apr 6 15:00:44 2020 From: johan at democon.be (Johan De Clercq) Date: Mon, 6 Apr 2020 15:00:44 +0000 Subject: [OpenSIPS-Users] Access to listen/advertised IP Addresses In-Reply-To: References: , Message-ID: It,s not exposed I think. I can’t find it back either Outlook voor iOS downloaden ________________________________ Van: Users namens David Villasmil Verzonden: Monday, April 6, 2020 4:49:36 PM Aan: OpenSIPS users mailling list Onderwerp: Re: [OpenSIPS-Users] Access to listen/advertised IP Addresses No, you’re right. It’s not in the core variables and I can’t find it either. Which makes me think it’s either not exposed or somewhere in a module (it’s not in proto_udp) I will research a little to try and find it.. On Mon, 6 Apr 2020 at 14:04, Mark Farmer > wrote: Thanks David. But I see no reference to the same variable in OpenSIPS. https://www.opensips.org/Documentation/Script-CoreVar-2-4 Am I missing something? On Mon, 6 Apr 2020 at 13:45, David Villasmil > wrote: Right here: https://www.kamailio.org/wiki/cookbooks/5.2.x/pseudovariables#rai_-_received_advertised_ip_address Regards, David Villasmil email: david.villasmil.work at gmail.com phone: +34669448337 On Mon, Apr 6, 2020 at 12:08 PM Mark Farmer > wrote: Many thanks for the reply. $Ri is certainly useful when the request comes from a non-natted interface. Thanks for pointing that out :) Is there a way to reference the advertised IP address defined in the listen statement? listen=udp:xxx.xxx.xxx.xxx:5060 as xxx.xxx.xxx.xxx:5060 Thanks Mark. On Thu, 2 Apr 2020 at 17:32, Sharad Kumar via Users > wrote: Hi Mark, If your initial goal is to get the interface IP where request is received then you can try these variables. $Ri - reference to IP address of the interface where the request has been received $Rp - reference to the port where the message was received _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Mark Farmer farmorg at gmail.com _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Mark Farmer farmorg at gmail.com _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- Regards, David Villasmil email: david.villasmil.work at gmail.com phone: +34669448337 -------------- next part -------------- An HTML attachment was scrubbed... URL: From david.villasmil.work at gmail.com Mon Apr 6 15:12:23 2020 From: david.villasmil.work at gmail.com (David Villasmil) Date: Mon, 6 Apr 2020 15:12:23 +0000 Subject: [OpenSIPS-Users] Access to listen/advertised IP Addresses In-Reply-To: References: Message-ID: i only see $rd which is the domain to which the sip message was sent, it "should" have the advertised ip, or de actual domain, in which case if you need the actual ip, it is useless Regards, David Villasmil email: david.villasmil.work at gmail.com phone: +34669448337 On Mon, Apr 6, 2020 at 4:00 PM Johan De Clercq wrote: > It,s not exposed I think. I can’t find it back either > > Outlook voor iOS downloaden > ------------------------------ > *Van:* Users namens David Villasmil < > david.villasmil.work at gmail.com> > *Verzonden:* Monday, April 6, 2020 4:49:36 PM > *Aan:* OpenSIPS users mailling list > *Onderwerp:* Re: [OpenSIPS-Users] Access to listen/advertised IP Addresses > > No, you’re right. It’s not in the core variables and I can’t find it > either. Which makes me think it’s either not exposed or somewhere in a > module (it’s not in proto_udp) > > I will research a little to try and find it.. > > On Mon, 6 Apr 2020 at 14:04, Mark Farmer wrote: > > Thanks David. But I see no reference to the same variable in OpenSIPS. > > https://www.opensips.org/Documentation/Script-CoreVar-2-4 > > Am I missing something? > > > On Mon, 6 Apr 2020 at 13:45, David Villasmil < > david.villasmil.work at gmail.com> wrote: > > Right here: > > > https://www.kamailio.org/wiki/cookbooks/5.2.x/pseudovariables#rai_-_received_advertised_ip_address > > Regards, > > David Villasmil > email: david.villasmil.work at gmail.com > phone: +34669448337 > > > On Mon, Apr 6, 2020 at 12:08 PM Mark Farmer wrote: > > Many thanks for the reply. > > $Ri is certainly useful when the request comes from a non-natted > interface. Thanks for pointing that out :) > > Is there a way to reference the advertised IP address defined in the > listen statement? > > listen=udp:xxx.xxx.xxx.xxx:5060 as xxx.xxx.xxx.xxx:5060 > > Thanks > Mark. > > > On Thu, 2 Apr 2020 at 17:32, Sharad Kumar via Users < > users at lists.opensips.org> wrote: > > Hi Mark, > > If your initial goal is to get the interface IP where request is received > then you can try these variables. > > *$Ri* - reference to IP address of the interface where the request has > been received > > *$Rp* - reference to the port where the message was received > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > -- > Mark Farmer > farmorg at gmail.com > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > -- > Mark Farmer > farmorg at gmail.com > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > -- > 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 Mon Apr 6 15:27:07 2020 From: david.villasmil.work at gmail.com (David Villasmil) Date: Mon, 6 Apr 2020 15:27:07 +0000 Subject: [OpenSIPS-Users] Access to listen/advertised IP Addresses In-Reply-To: References: Message-ID: this is where the advertised address is set https://github.com/OpenSIPS/opensips/blob/628a126fe3523e800440855f10f9841d6a2c39eb/cfg.y#L2373 Regards, David Villasmil email: david.villasmil.work at gmail.com phone: +34669448337 On Mon, Apr 6, 2020 at 4:12 PM David Villasmil < david.villasmil.work at gmail.com> wrote: > i only see $rd which is the domain to which the sip message was sent, it > "should" have the advertised ip, or de actual domain, in which case if you > need the actual ip, it is useless > Regards, > > David Villasmil > email: david.villasmil.work at gmail.com > phone: +34669448337 > > > On Mon, Apr 6, 2020 at 4:00 PM Johan De Clercq wrote: > >> It,s not exposed I think. I can’t find it back either >> >> Outlook voor iOS downloaden >> ------------------------------ >> *Van:* Users namens David Villasmil < >> david.villasmil.work at gmail.com> >> *Verzonden:* Monday, April 6, 2020 4:49:36 PM >> *Aan:* OpenSIPS users mailling list >> *Onderwerp:* Re: [OpenSIPS-Users] Access to listen/advertised IP >> Addresses >> >> No, you’re right. It’s not in the core variables and I can’t find it >> either. Which makes me think it’s either not exposed or somewhere in a >> module (it’s not in proto_udp) >> >> I will research a little to try and find it.. >> >> On Mon, 6 Apr 2020 at 14:04, Mark Farmer wrote: >> >> Thanks David. But I see no reference to the same variable in OpenSIPS. >> >> https://www.opensips.org/Documentation/Script-CoreVar-2-4 >> >> Am I missing something? >> >> >> On Mon, 6 Apr 2020 at 13:45, David Villasmil < >> david.villasmil.work at gmail.com> wrote: >> >> Right here: >> >> >> https://www.kamailio.org/wiki/cookbooks/5.2.x/pseudovariables#rai_-_received_advertised_ip_address >> >> Regards, >> >> David Villasmil >> email: david.villasmil.work at gmail.com >> phone: +34669448337 >> >> >> On Mon, Apr 6, 2020 at 12:08 PM Mark Farmer wrote: >> >> Many thanks for the reply. >> >> $Ri is certainly useful when the request comes from a non-natted >> interface. Thanks for pointing that out :) >> >> Is there a way to reference the advertised IP address defined in the >> listen statement? >> >> listen=udp:xxx.xxx.xxx.xxx:5060 as xxx.xxx.xxx.xxx:5060 >> >> Thanks >> Mark. >> >> >> On Thu, 2 Apr 2020 at 17:32, Sharad Kumar via Users < >> users at lists.opensips.org> wrote: >> >> Hi Mark, >> >> If your initial goal is to get the interface IP where request is >> received then you can try these variables. >> >> *$Ri* - reference to IP address of the interface where the request has >> been received >> >> *$Rp* - reference to the port where the message was received >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> >> >> -- >> Mark Farmer >> farmorg at gmail.com >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> >> >> -- >> Mark Farmer >> farmorg at gmail.com >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> -- >> 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 johan at democon.be Mon Apr 6 15:56:03 2020 From: johan at democon.be (johan) Date: Mon, 6 Apr 2020 17:56:03 +0200 Subject: [OpenSIPS-Users] t_check_trans Message-ID: <1d6ef1dc-d82e-3187-fe11-6a8347fb54e3@democon.be> Is there a specific reason why t_check_trans can't be called in a reply route ? To me this seems handy to see if a 200 OK is a retransmission. BR, From alex at canyan.io Tue Apr 7 09:31:42 2020 From: alex at canyan.io (Aleksandar Sosic) Date: Tue, 7 Apr 2020 11:31:42 +0200 Subject: [OpenSIPS-Users] failed to load module siptrace.so Message-ID: Hi Guys, I'm installing Opensips v3.0 via `deb http://apt.opensips.org buster 3.0-releases` with: ``` apt-get install -y opensips opensips-json-module opensips-restclient-module opensips-http-modules ``` When specifying in the conf `loadmodule "siptrace.so"` upon running opensips I get this error: ``` Apr 7 09:21:49 [333] CRITICAL:core:yyerror: parse error in /etc/opensips/opensips.cfg:38:13-14: failed to load module siptrace.so ``` The file indeed is not present in `/usr/lib/x86_64-linux-gnu/opensips/modules/`. I do have `sipcapture.so` and `proto_hep.so` although but no siptrace. What am I missing here? I'm trying to send all to a HEP agent like I do in kamailio with: ``` loadmodule "siptrace.so" modparam("siptrace", "trace_on", 1) modparam("siptrace", "trace_to_database", 0) modparam("siptrace", "hep_mode_on", 1) modparam("siptrace", "hep_version", 3) modparam("siptrace", "hep_capture_id", 1) request_route { sip_trace("hep-agent.local", "$ci"); ... } ``` Any ideas or examples of how to do this with Opensips v3? Thanks, -- Alex From david.villasmil.work at gmail.com Tue Apr 7 09:39:08 2020 From: david.villasmil.work at gmail.com (David Villasmil) Date: Tue, 7 Apr 2020 09:39:08 +0000 Subject: [OpenSIPS-Users] failed to load module siptrace.so In-Reply-To: References: Message-ID: did you compile sipcapture? Regards, David Villasmil email: david.villasmil.work at gmail.com phone: +34669448337 On Tue, Apr 7, 2020 at 10:32 AM Aleksandar Sosic wrote: > Hi Guys, > > I'm installing Opensips v3.0 via `deb http://apt.opensips.org buster > 3.0-releases` with: > ``` > apt-get install -y opensips opensips-json-module > opensips-restclient-module opensips-http-modules > ``` > > When specifying in the conf `loadmodule "siptrace.so"` upon running > opensips I get this error: > ``` > Apr 7 09:21:49 [333] CRITICAL:core:yyerror: parse error in > /etc/opensips/opensips.cfg:38:13-14: failed to load module siptrace.so > ``` > > The file indeed is not present in > `/usr/lib/x86_64-linux-gnu/opensips/modules/`. > I do have `sipcapture.so` and `proto_hep.so` although but no siptrace. > What am I missing here? > > I'm trying to send all to a HEP agent like I do in kamailio with: > ``` > loadmodule "siptrace.so" > > modparam("siptrace", "trace_on", 1) > modparam("siptrace", "trace_to_database", 0) > modparam("siptrace", "hep_mode_on", 1) > modparam("siptrace", "hep_version", 3) > modparam("siptrace", "hep_capture_id", 1) > > request_route { > sip_trace("hep-agent.local", "$ci"); > ... > } > ``` > > Any ideas or examples of how to do this with Opensips v3? > > Thanks, > -- > Alex > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -------------- next part -------------- An HTML attachment was scrubbed... URL: From david.villasmil.work at gmail.com Tue Apr 7 09:41:01 2020 From: david.villasmil.work at gmail.com (David Villasmil) Date: Tue, 7 Apr 2020 09:41:01 +0000 Subject: [OpenSIPS-Users] failed to load module siptrace.so In-Reply-To: References: Message-ID: or install? Regards, David Villasmil email: david.villasmil.work at gmail.com phone: +34669448337 On Tue, Apr 7, 2020 at 10:39 AM David Villasmil < david.villasmil.work at gmail.com> wrote: > did you compile sipcapture? > > Regards, > > David Villasmil > email: david.villasmil.work at gmail.com > phone: +34669448337 > > > On Tue, Apr 7, 2020 at 10:32 AM Aleksandar Sosic wrote: > >> Hi Guys, >> >> I'm installing Opensips v3.0 via `deb http://apt.opensips.org buster >> 3.0-releases` with: >> ``` >> apt-get install -y opensips opensips-json-module >> opensips-restclient-module opensips-http-modules >> ``` >> >> When specifying in the conf `loadmodule "siptrace.so"` upon running >> opensips I get this error: >> ``` >> Apr 7 09:21:49 [333] CRITICAL:core:yyerror: parse error in >> /etc/opensips/opensips.cfg:38:13-14: failed to load module siptrace.so >> ``` >> >> The file indeed is not present in >> `/usr/lib/x86_64-linux-gnu/opensips/modules/`. >> I do have `sipcapture.so` and `proto_hep.so` although but no siptrace. >> What am I missing here? >> >> I'm trying to send all to a HEP agent like I do in kamailio with: >> ``` >> loadmodule "siptrace.so" >> >> modparam("siptrace", "trace_on", 1) >> modparam("siptrace", "trace_to_database", 0) >> modparam("siptrace", "hep_mode_on", 1) >> modparam("siptrace", "hep_version", 3) >> modparam("siptrace", "hep_capture_id", 1) >> >> request_route { >> sip_trace("hep-agent.local", "$ci"); >> ... >> } >> ``` >> >> Any ideas or examples of how to do this with Opensips v3? >> >> Thanks, >> -- >> Alex >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Tue Apr 7 10:13:31 2020 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Tue, 7 Apr 2020 13:13:31 +0300 Subject: [OpenSIPS-Users] [BLOG] Real-Time Rating and Cost Based Routing in OpenSIPS 3.1 Message-ID: <81cf4783-41ea-def7-14d5-3ddf7cd111ef@opensips.org> While there are numerous external rating and billing engines available in the wild, having a quick and easy way of putting a price for a call, without relying on external applications, can be a valuable asset to have. https://blog.opensips.org/2020/04/07/real-time-rating-and-cost-based-routing-in-opensips-3-1/ Thank you Vlad Paiu for the valuable contribution and post ! Best Regards, -- Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com From david.villasmil.work at gmail.com Tue Apr 7 10:17:13 2020 From: david.villasmil.work at gmail.com (David Villasmil) Date: Tue, 7 Apr 2020 10:17:13 +0000 Subject: [OpenSIPS-Users] failed to load module siptrace.so In-Reply-To: References: Message-ID: I just made a fresh install on a docker and indeed siptrace is not there... going to compile it Regards, David Villasmil email: david.villasmil.work at gmail.com phone: +34669448337 On Tue, Apr 7, 2020 at 10:41 AM David Villasmil < david.villasmil.work at gmail.com> wrote: > or install? > > Regards, > > David Villasmil > email: david.villasmil.work at gmail.com > phone: +34669448337 > > > On Tue, Apr 7, 2020 at 10:39 AM David Villasmil < > david.villasmil.work at gmail.com> wrote: > >> did you compile sipcapture? >> >> Regards, >> >> David Villasmil >> email: david.villasmil.work at gmail.com >> phone: +34669448337 >> >> >> On Tue, Apr 7, 2020 at 10:32 AM Aleksandar Sosic wrote: >> >>> Hi Guys, >>> >>> I'm installing Opensips v3.0 via `deb http://apt.opensips.org buster >>> 3.0-releases` with: >>> ``` >>> apt-get install -y opensips opensips-json-module >>> opensips-restclient-module opensips-http-modules >>> ``` >>> >>> When specifying in the conf `loadmodule "siptrace.so"` upon running >>> opensips I get this error: >>> ``` >>> Apr 7 09:21:49 [333] CRITICAL:core:yyerror: parse error in >>> /etc/opensips/opensips.cfg:38:13-14: failed to load module siptrace.so >>> ``` >>> >>> The file indeed is not present in >>> `/usr/lib/x86_64-linux-gnu/opensips/modules/`. >>> I do have `sipcapture.so` and `proto_hep.so` although but no siptrace. >>> What am I missing here? >>> >>> I'm trying to send all to a HEP agent like I do in kamailio with: >>> ``` >>> loadmodule "siptrace.so" >>> >>> modparam("siptrace", "trace_on", 1) >>> modparam("siptrace", "trace_to_database", 0) >>> modparam("siptrace", "hep_mode_on", 1) >>> modparam("siptrace", "hep_version", 3) >>> modparam("siptrace", "hep_capture_id", 1) >>> >>> request_route { >>> sip_trace("hep-agent.local", "$ci"); >>> ... >>> } >>> ``` >>> >>> Any ideas or examples of how to do this with Opensips v3? >>> >>> Thanks, >>> -- >>> Alex >>> >>> _______________________________________________ >>> Users mailing list >>> Users at lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >> -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Tue Apr 7 10:25:48 2020 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Tue, 7 Apr 2020 13:25:48 +0300 Subject: [OpenSIPS-Users] failed to load module siptrace.so In-Reply-To: References: Message-ID: <08ddcf0a-8a25-98c7-9bb5-d8021ef2c392@opensips.org> Hi Aleksandar, Starting 3.0, the siptrace module was named "tracer" as it's not sip-centric anymore, so see https://opensips.org/html/docs/modules/3.0.x/tracer.html Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com On 4/7/20 12:31 PM, Aleksandar Sosic wrote: > Hi Guys, > > I'm installing Opensips v3.0 via `deb http://apt.opensips.org buster > 3.0-releases` with: > ``` > apt-get install -y opensips opensips-json-module > opensips-restclient-module opensips-http-modules > ``` > > When specifying in the conf `loadmodule "siptrace.so"` upon running > opensips I get this error: > ``` > Apr 7 09:21:49 [333] CRITICAL:core:yyerror: parse error in > /etc/opensips/opensips.cfg:38:13-14: failed to load module siptrace.so > ``` > > The file indeed is not present in `/usr/lib/x86_64-linux-gnu/opensips/modules/`. > I do have `sipcapture.so` and `proto_hep.so` although but no siptrace. > What am I missing here? > > I'm trying to send all to a HEP agent like I do in kamailio with: > ``` > loadmodule "siptrace.so" > > modparam("siptrace", "trace_on", 1) > modparam("siptrace", "trace_to_database", 0) > modparam("siptrace", "hep_mode_on", 1) > modparam("siptrace", "hep_version", 3) > modparam("siptrace", "hep_capture_id", 1) > > request_route { > sip_trace("hep-agent.local", "$ci"); > ... > } > ``` > > Any ideas or examples of how to do this with Opensips v3? > > Thanks, > -- > Alex > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users From bogdan at opensips.org Tue Apr 7 10:34:29 2020 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Tue, 7 Apr 2020 13:34:29 +0300 Subject: [OpenSIPS-Users] t_check_trans In-Reply-To: <1d6ef1dc-d82e-3187-fe11-6a8347fb54e3@democon.be> References: <1d6ef1dc-d82e-3187-fe11-6a8347fb54e3@democon.be> Message-ID: The t_check_trans() is performing a matching of the current request against the known transaction. Such matching, for relies, doesn't make too much of a sense as:     1) if you use an onreply_route[], it means your reply already matched the transaction     2) for replies, there is no special handling for retransmissions, they are handled in the same way as the first reply. (unlikely for the requests) If you want to "catch" the 200 OK retransmission, simply use a bflag - set it on first 200OK and test it for the next 200 OKs :) Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com On 4/6/20 6:56 PM, johan wrote: > Is there a specific reason why t_check_trans can't be called in a > reply route ? > > To me this seems handy to see if a 200 OK is a retransmission. > > > BR, > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users From bogdan at opensips.org Tue Apr 7 10:42:18 2020 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Tue, 7 Apr 2020 13:42:18 +0300 Subject: [OpenSIPS-Users] Access to listen/advertised IP Addresses In-Reply-To: References: Message-ID: Hi guys, Maybe adding a new core variable like $in_socket.XXXX, to give access to various fields, like $in_socket.ip, $in_socket.port, $in_socket.advertised_ip, etc. This will replace the $Ri and $Rp And we can also add $out_socket, that will similarly replace the $fs (forced socket) Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com OpenSIPS Summit, Amsterdam, May 2020 https://www.opensips.org/events/Summit-2020Amsterdam/ On 4/6/20 6:00 PM, Johan De Clercq wrote: > It,s not exposed I think. I can’t find it back either > > Outlook voor iOS downloaden > ------------------------------------------------------------------------ > *Van:* Users namens David Villasmil > > *Verzonden:* Monday, April 6, 2020 4:49:36 PM > *Aan:* OpenSIPS users mailling list > *Onderwerp:* Re: [OpenSIPS-Users] Access to listen/advertised IP > Addresses > No, you’re right. It’s not in the core variables and I can’t find it > either. Which makes me think it’s either not exposed or somewhere in a > module (it’s not in proto_udp) > > I will research a little to try and find it.. > > On Mon, 6 Apr 2020 at 14:04, Mark Farmer > wrote: > > Thanks David. But I see no reference to the same variable in OpenSIPS. > > https://www.opensips.org/Documentation/Script-CoreVar-2-4 > > Am I missing something? > > > On Mon, 6 Apr 2020 at 13:45, David Villasmil > > wrote: > > Right here: > > https://www.kamailio.org/wiki/cookbooks/5.2.x/pseudovariables#rai_-_received_advertised_ip_address > > Regards, > > David Villasmil > email: david.villasmil.work at gmail.com > > phone: +34669448337 > > > On Mon, Apr 6, 2020 at 12:08 PM Mark Farmer > wrote: > > Many thanks for the reply. > > $Ri is certainly useful when the request comes from a > non-natted interface. Thanks for pointing that out :) > > Is there a way to reference the advertised IP address > defined in the listen statement? > > listen=udp:xxx.xxx.xxx.xxx:5060 as xxx.xxx.xxx.xxx:5060 > > Thanks > Mark. > > > On Thu, 2 Apr 2020 at 17:32, Sharad Kumar via Users > > wrote: > > Hi Mark, > >  If your initial goal is to get the interface IP where > request is received then you can try these variables. > > *$Ri* - reference to IP address of the interface where > the request has been received > > *$Rp* - reference to the port where the message was > received > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > -- > Mark Farmer > farmorg at gmail.com > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > -- > Mark Farmer > farmorg at gmail.com > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > -- > Regards, > > David Villasmil > email: david.villasmil.work at gmail.com > > phone: +34669448337 > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From farmorg at gmail.com Tue Apr 7 11:10:57 2020 From: farmorg at gmail.com (Mark Farmer) Date: Tue, 7 Apr 2020 12:10:57 +0100 Subject: [OpenSIPS-Users] Access to listen/advertised IP Addresses In-Reply-To: References: Message-ID: Hi Bogdan The root of my issue is that I need 2 variables containing the IP's of my 2 interfaces (mhomed=yes) but the advertised address of the NAT'd DMZ interface while keeping changes per server to a bare minimum to ease deployment. I actually solved my issue by using include_file and using cfgutils to set 2 script variables. So now all deployment changes are confined to a much simpler/smaller file. However, the proposed changes would make things even nicer. Would cfgutils be able to accept those variables as parameters to the 'varset' function? Regards Mark. On Tue, 7 Apr 2020 at 11:44, Bogdan-Andrei Iancu wrote: > Hi guys, > > Maybe adding a new core variable like $in_socket.XXXX, to give access to > various fields, like $in_socket.ip, $in_socket.port, $in_socket.advertised_ip, > etc. This will replace the $Ri and $Rp > > And we can also add $out_socket, that will similarly replace the $fs > (forced socket) > > Regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > OpenSIPS Summit, Amsterdam, May 2020 > https://www.opensips.org/events/Summit-2020Amsterdam/ > > On 4/6/20 6:00 PM, Johan De Clercq wrote: > > It,s not exposed I think. I can’t find it back either > > Outlook voor iOS downloaden > ------------------------------ > *Van:* Users > namens David Villasmil > > *Verzonden:* Monday, April 6, 2020 4:49:36 PM > *Aan:* OpenSIPS users mailling list > > *Onderwerp:* Re: [OpenSIPS-Users] Access to listen/advertised IP Addresses > > No, you’re right. It’s not in the core variables and I can’t find it > either. Which makes me think it’s either not exposed or somewhere in a > module (it’s not in proto_udp) > > I will research a little to try and find it.. > > On Mon, 6 Apr 2020 at 14:04, Mark Farmer wrote: > > Thanks David. But I see no reference to the same variable in OpenSIPS. > > https://www.opensips.org/Documentation/Script-CoreVar-2-4 > > Am I missing something? > > > On Mon, 6 Apr 2020 at 13:45, David Villasmil < > david.villasmil.work at gmail.com> wrote: > > Right here: > > > https://www.kamailio.org/wiki/cookbooks/5.2.x/pseudovariables#rai_-_received_advertised_ip_address > > Regards, > > David Villasmil > email: david.villasmil.work at gmail.com > phone: +34669448337 > > > On Mon, Apr 6, 2020 at 12:08 PM Mark Farmer wrote: > > Many thanks for the reply. > > $Ri is certainly useful when the request comes from a non-natted > interface. Thanks for pointing that out :) > > Is there a way to reference the advertised IP address defined in the > listen statement? > > listen=udp:xxx.xxx.xxx.xxx:5060 as xxx.xxx.xxx.xxx:5060 > > Thanks > Mark. > > > On Thu, 2 Apr 2020 at 17:32, Sharad Kumar via Users < > users at lists.opensips.org> wrote: > > Hi Mark, > > If your initial goal is to get the interface IP where request is received > then you can try these variables. > > *$Ri* - reference to IP address of the interface where the request has > been received > > *$Rp* - reference to the port where the message was received > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > -- > Mark Farmer > farmorg at gmail.com > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > -- > Mark Farmer > farmorg at gmail.com > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > -- > Regards, > > David Villasmil > email: david.villasmil.work at gmail.com > phone: +34669448337 > > _______________________________________________ > Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -- Mark Farmer farmorg at gmail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From alex at canyan.io Tue Apr 7 11:18:40 2020 From: alex at canyan.io (Aleksandar Sosic) Date: Tue, 7 Apr 2020 13:18:40 +0200 Subject: [OpenSIPS-Users] failed to load module siptrace.so In-Reply-To: <08ddcf0a-8a25-98c7-9bb5-d8021ef2c392@opensips.org> References: <08ddcf0a-8a25-98c7-9bb5-d8021ef2c392@opensips.org> Message-ID: Ok, because I was looking at this: https://opensips.org/html/docs/modules/3.0.x/siptrace.html so I expected it to be available. Should this page then be removed? Thanks, Alex On Tue, Apr 7, 2020 at 12:25 PM Bogdan-Andrei Iancu wrote: > > Hi Aleksandar, > > Starting 3.0, the siptrace module was named "tracer" as it's not > sip-centric anymore, so see > https://opensips.org/html/docs/modules/3.0.x/tracer.html > > Regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > > On 4/7/20 12:31 PM, Aleksandar Sosic wrote: > > Hi Guys, > > > > I'm installing Opensips v3.0 via `deb http://apt.opensips.org buster > > 3.0-releases` with: > > ``` > > apt-get install -y opensips opensips-json-module > > opensips-restclient-module opensips-http-modules > > ``` > > > > When specifying in the conf `loadmodule "siptrace.so"` upon running > > opensips I get this error: > > ``` > > Apr 7 09:21:49 [333] CRITICAL:core:yyerror: parse error in > > /etc/opensips/opensips.cfg:38:13-14: failed to load module siptrace.so > > ``` > > > > The file indeed is not present in `/usr/lib/x86_64-linux-gnu/opensips/modules/`. > > I do have `sipcapture.so` and `proto_hep.so` although but no siptrace. > > What am I missing here? > > > > I'm trying to send all to a HEP agent like I do in kamailio with: > > ``` > > loadmodule "siptrace.so" > > > > modparam("siptrace", "trace_on", 1) > > modparam("siptrace", "trace_to_database", 0) > > modparam("siptrace", "hep_mode_on", 1) > > modparam("siptrace", "hep_version", 3) > > modparam("siptrace", "hep_capture_id", 1) > > > > request_route { > > sip_trace("hep-agent.local", "$ci"); > > ... > > } > > ``` > > > > Any ideas or examples of how to do this with Opensips v3? > > > > Thanks, > > -- > > Alex > > > > _______________________________________________ > > Users mailing list > > Users at lists.opensips.org > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > From bogdan at opensips.org Tue Apr 7 11:33:40 2020 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Tue, 7 Apr 2020 14:33:40 +0300 Subject: [OpenSIPS-Users] failed to load module siptrace.so In-Reply-To: References: <08ddcf0a-8a25-98c7-9bb5-d8021ef2c392@opensips.org> Message-ID: <63a37505-c08d-8095-b2af-fa70ba6e11cf@opensips.org> yeah, sorry, it was a left over page from the 2.4 to 3.0 migration of the docs. I will rebuild the full module docs, to be sure there are no left-overs. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com OpenSIPS Summit, Amsterdam, May 2020 https://www.opensips.org/events/Summit-2020Amsterdam/ On 4/7/20 2:18 PM, Aleksandar Sosic wrote: > Ok, because I was looking at this: > https://opensips.org/html/docs/modules/3.0.x/siptrace.html > > so I expected it to be available. > Should this page then be removed? > > Thanks, > Alex > > On Tue, Apr 7, 2020 at 12:25 PM Bogdan-Andrei Iancu wrote: >> Hi Aleksandar, >> >> Starting 3.0, the siptrace module was named "tracer" as it's not >> sip-centric anymore, so see >> https://opensips.org/html/docs/modules/3.0.x/tracer.html >> >> Regards, >> >> Bogdan-Andrei Iancu >> >> OpenSIPS Founder and Developer >> https://www.opensips-solutions.com >> >> On 4/7/20 12:31 PM, Aleksandar Sosic wrote: >>> Hi Guys, >>> >>> I'm installing Opensips v3.0 via `deb http://apt.opensips.org buster >>> 3.0-releases` with: >>> ``` >>> apt-get install -y opensips opensips-json-module >>> opensips-restclient-module opensips-http-modules >>> ``` >>> >>> When specifying in the conf `loadmodule "siptrace.so"` upon running >>> opensips I get this error: >>> ``` >>> Apr 7 09:21:49 [333] CRITICAL:core:yyerror: parse error in >>> /etc/opensips/opensips.cfg:38:13-14: failed to load module siptrace.so >>> ``` >>> >>> The file indeed is not present in `/usr/lib/x86_64-linux-gnu/opensips/modules/`. >>> I do have `sipcapture.so` and `proto_hep.so` although but no siptrace. >>> What am I missing here? >>> >>> I'm trying to send all to a HEP agent like I do in kamailio with: >>> ``` >>> loadmodule "siptrace.so" >>> >>> modparam("siptrace", "trace_on", 1) >>> modparam("siptrace", "trace_to_database", 0) >>> modparam("siptrace", "hep_mode_on", 1) >>> modparam("siptrace", "hep_version", 3) >>> modparam("siptrace", "hep_capture_id", 1) >>> >>> request_route { >>> sip_trace("hep-agent.local", "$ci"); >>> ... >>> } >>> ``` >>> >>> Any ideas or examples of how to do this with Opensips v3? >>> >>> Thanks, >>> -- >>> Alex >>> >>> _______________________________________________ >>> Users mailing list >>> Users at lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users From johan at democon.be Tue Apr 7 11:43:34 2020 From: johan at democon.be (Johan De Clercq) Date: Tue, 7 Apr 2020 11:43:34 +0000 Subject: [OpenSIPS-Users] t_check_trans In-Reply-To: References: <1d6ef1dc-d82e-3187-fe11-6a8347fb54e3@democon.be>, Message-ID: But where should I then check? In the reply-route? Outlook voor iOS downloaden ________________________________ Van: Bogdan-Andrei Iancu Verzonden: Tuesday, April 7, 2020 12:34:29 PM Aan: OpenSIPS users mailling list ; johan Onderwerp: Re: [OpenSIPS-Users] t_check_trans The t_check_trans() is performing a matching of the current request against the known transaction. Such matching, for relies, doesn't make too much of a sense as: 1) if you use an onreply_route[], it means your reply already matched the transaction 2) for replies, there is no special handling for retransmissions, they are handled in the same way as the first reply. (unlikely for the requests) If you want to "catch" the 200 OK retransmission, simply use a bflag - set it on first 200OK and test it for the next 200 OKs :) Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com On 4/6/20 6:56 PM, johan wrote: > Is there a specific reason why t_check_trans can't be called in a > reply route ? > > To me this seems handy to see if a 200 OK is a retransmission. > > > BR, > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From jeff at ugnd.org Tue Apr 7 12:57:23 2020 From: jeff at ugnd.org (Jeff Pyle) Date: Tue, 7 Apr 2020 08:57:23 -0400 Subject: [OpenSIPS-Users] drouting gateway status with route_to_gw() Message-ID: Hello, On v2.4.7 I'm using route_to_gw() to load routing for a particular gateway. It works well. It also works when the gateway is in an "Inactive" status, which surprises me. I would expect the function to return with an error code in that scenario, or at least have the option to achieve that behavior. I don't see any mention in the documentation of a way to manually check the status of a gateway from the script (only from MI). Is there? Perhaps the gateway status is taken into consideration only with do_routing()? If that's the case, I'm wondering if I can emulate route_to_gw() with do_routing() while specifying a gw_whitelist of only the gateway I'm interested in routing to. I thought I'd ask the experts before I get crazy with it. - Jeff -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Tue Apr 7 13:06:05 2020 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Tue, 7 Apr 2020 16:06:05 +0300 Subject: [OpenSIPS-Users] t_check_trans In-Reply-To: References: <1d6ef1dc-d82e-3187-fe11-6a8347fb54e3@democon.be> Message-ID: <3def19dc-39a6-f65e-e351-d525732adc24@opensips.org> Yup :) Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com OpenSIPS Summit, Amsterdam, May 2020 https://www.opensips.org/events/Summit-2020Amsterdam/ On 4/7/20 2:43 PM, Johan De Clercq wrote: > But where should I then check? In the reply-route? > > Outlook voor iOS downloaden > ------------------------------------------------------------------------ > *Van:* Bogdan-Andrei Iancu > *Verzonden:* Tuesday, April 7, 2020 12:34:29 PM > *Aan:* OpenSIPS users mailling list ; johan > > *Onderwerp:* Re: [OpenSIPS-Users] t_check_trans > The t_check_trans() is performing a matching of the current request > against the known transaction. Such matching, for relies, doesn't make > too much of a sense as: > >      1) if you use an onreply_route[], it means your reply already > matched the transaction > >      2) for replies, there is no special handling for retransmissions, > they are handled in the same way as the first reply. (unlikely for the > requests) > > If you want to "catch" the 200 OK retransmission, simply use a bflag - > set it on first 200OK and test it for the next 200 OKs :) > > Regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > > > On 4/6/20 6:56 PM, johan wrote: > > Is there a specific reason why t_check_trans can't be called in a > > reply route ? > > > > To me this seems handy to see if a 200 OK is a retransmission. > > > > > > BR, > > > > > > _______________________________________________ > > Users mailing list > > Users at lists.opensips.org > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Tue Apr 7 13:40:02 2020 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Tue, 7 Apr 2020 16:40:02 +0300 Subject: [OpenSIPS-Users] Access to listen/advertised IP Addresses In-Reply-To: References: Message-ID: <5eb6a6ca-83aa-c59e-2f2a-2c4f216a4118@opensips.org> Hi Mark, ingenious solution :) In regards to the proposed solution, I do not understand the question about varset (cfgutils), as there is no relation between the script vars and these new $socket vars. Maybe I'm missing something from your question. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com OpenSIPS Summit, Amsterdam, May 2020 https://www.opensips.org/events/Summit-2020Amsterdam/ On 4/7/20 2:10 PM, Mark Farmer wrote: > Hi Bogdan > > The root of my issue is that I need 2 variables containing the IP's of > my 2 interfaces (mhomed=yes) but the advertised address of the NAT'd > DMZ interface while keeping changes per server to a bare minimum to > ease deployment. > > I actually solved my issue by using include_file and using cfgutils to > set 2 script variables. So now all deployment changes are confined to > a much simpler/smaller file. > > However, the proposed changes would make things even nicer. Would > cfgutils be able to accept those variables as parameters to the > 'varset' function? > > Regards > Mark. > > > > On Tue, 7 Apr 2020 at 11:44, Bogdan-Andrei Iancu > wrote: > > Hi guys, > > Maybe adding a new core variable like $in_socket.XXXX, to give > access to various fields, like $in_socket.ip, $in_socket.port, > $in_socket.advertised_ip, etc. This will replace the $Ri and $Rp > > And we can also add $out_socket, that will similarly replace the > $fs (forced socket) > > Regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > OpenSIPS Summit, Amsterdam, May 2020 > https://www.opensips.org/events/Summit-2020Amsterdam/ > > On 4/6/20 6:00 PM, Johan De Clercq wrote: >> It,s not exposed I think. I can’t find it back either >> >> Outlook voor iOS downloaden >> ------------------------------------------------------------------------ >> *Van:* Users >> namens David Villasmil >> >> >> *Verzonden:* Monday, April 6, 2020 4:49:36 PM >> *Aan:* OpenSIPS users mailling list >> >> *Onderwerp:* Re: [OpenSIPS-Users] Access to listen/advertised IP >> Addresses >> No, you’re right. It’s not in the core variables and I can’t find >> it either. Which makes me think it’s either not exposed or >> somewhere in a module (it’s not in proto_udp) >> >> I will research a little to try and find it.. >> >> On Mon, 6 Apr 2020 at 14:04, Mark Farmer > > wrote: >> >> Thanks David. But I see no reference to the same variable in >> OpenSIPS. >> >> https://www.opensips.org/Documentation/Script-CoreVar-2-4 >> >> Am I missing something? >> >> >> On Mon, 6 Apr 2020 at 13:45, David Villasmil >> > > wrote: >> >> Right here: >> >> https://www.kamailio.org/wiki/cookbooks/5.2.x/pseudovariables#rai_-_received_advertised_ip_address >> >> Regards, >> >> David Villasmil >> email: david.villasmil.work at gmail.com >> >> phone: +34669448337 >> >> >> On Mon, Apr 6, 2020 at 12:08 PM Mark Farmer >> > wrote: >> >> Many thanks for the reply. >> >> $Ri is certainly useful when the request comes from a >> non-natted interface. Thanks for pointing that out :) >> >> Is there a way to reference the advertised IP address >> defined in the listen statement? >> >> listen=udp:xxx.xxx.xxx.xxx:5060 as xxx.xxx.xxx.xxx:5060 >> >> Thanks >> Mark. >> >> >> On Thu, 2 Apr 2020 at 17:32, Sharad Kumar via Users >> > > wrote: >> >> Hi Mark, >> >>  If your initial goal is to get the interface IP >> where request is received then you can try these >> variables. >> >> *$Ri* - reference to IP address of the interface >> where the request has been received >> >> *$Rp* - reference to the port where the message >> was received >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> >> >> -- >> Mark Farmer >> farmorg at gmail.com >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> >> >> -- >> Mark Farmer >> farmorg at gmail.com >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> -- >> 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 > > > > -- > Mark Farmer > farmorg at gmail.com > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Tue Apr 7 13:47:39 2020 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Tue, 7 Apr 2020 16:47:39 +0300 Subject: [OpenSIPS-Users] drouting gateway status with route_to_gw() In-Reply-To: References: Message-ID: <1f89599b-a259-6270-5829-f7b9ce1ee7b8@opensips.org> Hi Jeff, Yeah, the route_to_gw() gives you raw, low level access to the routing, so the whole soring and filtering mechanisms are skipped. This is why the status is not checked, but it makes sense, as time as you can specify a list of GWs. Could you open a bug report on this please ? About fetching the status of a GW from script, no there is no such mechanism for now. Normally it should be needed (if we fix the above issue, right?) Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com OpenSIPS Summit, Amsterdam, May 2020 https://www.opensips.org/events/Summit-2020Amsterdam/ On 4/7/20 3:57 PM, Jeff Pyle wrote: > Hello, > > On v2.4.7 I'm using route_to_gw() to load routing for a particular > gateway.  It works well.  It also works when the gateway is in an > "Inactive" status, which surprises me.  I would expect the function to > return with an error code in that scenario, or at least have the > option to achieve that behavior. > > I don't see any mention in the documentation of a way to manually > check the status of a gateway from the script (only from MI).  Is there? > > Perhaps the gateway status is taken into consideration only with > do_routing()?  If that's the case, I'm wondering if I can emulate > route_to_gw() with do_routing() while specifying a gw_whitelist of > only the gateway I'm interested in routing to. > > I thought I'd ask the experts before I get crazy with it. > > > - Jeff > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Tue Apr 7 14:07:19 2020 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Tue, 7 Apr 2020 17:07:19 +0300 Subject: [OpenSIPS-Users] problem in call forwarding scenario In-Reply-To: References: Message-ID: <6e5f5e1a-258c-b347-f4bd-305e69593a40@opensips.org> Hi Kamlesh, Different TO-tag in provisional replies is perfectly legal from the SIP RFC3261 and the originator must accept that. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com OpenSIPS Summit, Amsterdam, May 2020 https://www.opensips.org/events/Summit-2020Amsterdam/ On 4/4/20 9:10 PM, Kamlesh . wrote: > Dear All, > > version: opensips 2.4.6 (x86_64/linux) > > flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, > F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT > > ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, > MAX_URI_SIZE 1024, BUF_SIZE 65535 > > poll method support: poll, epoll, sigio_rt, select. > > git revision: edef893c5 > > main.c compiled on 23:26:34 Dec 14 2019 with gcc 4.8.5 > > I have a problem with call forwarding on busy. The issue arises when I > get an INCOMING call from the gateway and the extension(callee) > responds with busy and callee is configured as an external number as > forwarding on busy so the forwarded call sends to the same gateway and > receives the 183. So in that case the gateway has two provisional > responses one is 180 from callee and 183 from the gateway. It is > obvious that both the responses have different to tag. So the second > response is not accepted by the gateway. Any solution for the issue. > Below is the call flow. > Regards, > Kamlesh > > Disclaimer : > > This e-mail and any file transmitted with it are for exclusive use of > the intended recipient(s) > and may contain confidential and privileged information. If you are > not the intended recipient, > please contact the sender by replying this e-mail and destroy all > copies and original message. > Any unauthorized review,use, disclosure, dissemination, forwarding, > printing and copying of this > email or any action taken in reliance of this e-mail is strictly > prohibited and may be unlawful. > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From farmorg at gmail.com Tue Apr 7 15:56:32 2020 From: farmorg at gmail.com (Mark Farmer) Date: Tue, 7 Apr 2020 16:56:32 +0100 Subject: [OpenSIPS-Users] Access to listen/advertised IP Addresses In-Reply-To: <5eb6a6ca-83aa-c59e-2f2a-2c4f216a4118@opensips.org> References: <5eb6a6ca-83aa-c59e-2f2a-2c4f216a4118@opensips.org> Message-ID: I was thinking something like: modparam("cfgutils", "varset", "extip=s:$in_socket.advertised_ip") On Tue, 7 Apr 2020 at 14:40, Bogdan-Andrei Iancu wrote: > Hi Mark, > > ingenious solution :) > > In regards to the proposed solution, I do not understand the question > about varset (cfgutils), as there is no relation between the script vars > and these new $socket vars. Maybe I'm missing something from your question. > > Regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > OpenSIPS Summit, Amsterdam, May 2020 > https://www.opensips.org/events/Summit-2020Amsterdam/ > > On 4/7/20 2:10 PM, Mark Farmer wrote: > > Hi Bogdan > > The root of my issue is that I need 2 variables containing the IP's of my > 2 interfaces (mhomed=yes) but the advertised address of the NAT'd DMZ > interface while keeping changes per server to a bare minimum to ease > deployment. > > I actually solved my issue by using include_file and using cfgutils to set > 2 script variables. So now all deployment changes are confined to a much > simpler/smaller file. > > However, the proposed changes would make things even nicer. Would > cfgutils be able to accept those variables as parameters to the 'varset' > function? > > Regards > Mark. > > > > On Tue, 7 Apr 2020 at 11:44, Bogdan-Andrei Iancu > wrote: > >> Hi guys, >> >> Maybe adding a new core variable like $in_socket.XXXX, to give access to >> various fields, like $in_socket.ip, $in_socket.port, $in_socket.advertised_ip, >> etc. This will replace the $Ri and $Rp >> >> And we can also add $out_socket, that will similarly replace the $fs >> (forced socket) >> >> Regards, >> >> Bogdan-Andrei Iancu >> >> OpenSIPS Founder and Developer >> https://www.opensips-solutions.com >> OpenSIPS Summit, Amsterdam, May 2020 >> https://www.opensips.org/events/Summit-2020Amsterdam/ >> >> On 4/6/20 6:00 PM, Johan De Clercq wrote: >> >> It,s not exposed I think. I can’t find it back either >> >> Outlook voor iOS downloaden >> ------------------------------ >> *Van:* Users >> namens David Villasmil >> >> *Verzonden:* Monday, April 6, 2020 4:49:36 PM >> *Aan:* OpenSIPS users mailling list >> >> *Onderwerp:* Re: [OpenSIPS-Users] Access to listen/advertised IP >> Addresses >> >> No, you’re right. It’s not in the core variables and I can’t find it >> either. Which makes me think it’s either not exposed or somewhere in a >> module (it’s not in proto_udp) >> >> I will research a little to try and find it.. >> >> On Mon, 6 Apr 2020 at 14:04, Mark Farmer wrote: >> >> Thanks David. But I see no reference to the same variable in OpenSIPS. >> >> https://www.opensips.org/Documentation/Script-CoreVar-2-4 >> >> Am I missing something? >> >> >> On Mon, 6 Apr 2020 at 13:45, David Villasmil < >> david.villasmil.work at gmail.com> wrote: >> >> Right here: >> >> >> https://www.kamailio.org/wiki/cookbooks/5.2.x/pseudovariables#rai_-_received_advertised_ip_address >> >> Regards, >> >> David Villasmil >> email: david.villasmil.work at gmail.com >> phone: +34669448337 >> >> >> On Mon, Apr 6, 2020 at 12:08 PM Mark Farmer wrote: >> >> Many thanks for the reply. >> >> $Ri is certainly useful when the request comes from a non-natted >> interface. Thanks for pointing that out :) >> >> Is there a way to reference the advertised IP address defined in the >> listen statement? >> >> listen=udp:xxx.xxx.xxx.xxx:5060 as xxx.xxx.xxx.xxx:5060 >> >> Thanks >> Mark. >> >> >> On Thu, 2 Apr 2020 at 17:32, Sharad Kumar via Users < >> users at lists.opensips.org> wrote: >> >> Hi Mark, >> >> If your initial goal is to get the interface IP where request is >> received then you can try these variables. >> >> *$Ri* - reference to IP address of the interface where the request has >> been received >> >> *$Rp* - reference to the port where the message was received >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> >> >> -- >> Mark Farmer >> farmorg at gmail.com >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> >> >> -- >> Mark Farmer >> farmorg at gmail.com >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> -- >> Regards, >> >> David Villasmil >> email: david.villasmil.work at gmail.com >> phone: +34669448337 >> >> _______________________________________________ >> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> > > > -- > Mark Farmer > farmorg at gmail.com > > _______________________________________________ > Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > -- Mark Farmer farmorg at gmail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Tue Apr 7 16:29:51 2020 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Tue, 7 Apr 2020 19:29:51 +0300 Subject: [OpenSIPS-Users] Access to listen/advertised IP Addresses In-Reply-To: References: <5eb6a6ca-83aa-c59e-2f2a-2c4f216a4118@opensips.org> Message-ID: <5957bcba-f283-50e4-19a7-4b4d43f8b987@opensips.org> No need, just use in script, where ever you need $socket_in(advertised_ip) and it will be evaluated for the current socket (used for receiving the request) Regardsm Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com OpenSIPS Summit, Amsterdam, May 2020 https://www.opensips.org/events/Summit-2020Amsterdam/ On 4/7/20 6:56 PM, Mark Farmer wrote: > I was thinking something like: > > modparam("cfgutils", "varset", "extip=s:$in_socket.advertised_ip") > > > On Tue, 7 Apr 2020 at 14:40, Bogdan-Andrei Iancu > wrote: > > Hi Mark, > > ingenious solution :) > > In regards to the proposed solution, I do not understand the > question about varset (cfgutils), as there is no relation between > the script vars and these new $socket vars. Maybe I'm missing > something from your question. > > Regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > OpenSIPS Summit, Amsterdam, May 2020 > https://www.opensips.org/events/Summit-2020Amsterdam/ > > On 4/7/20 2:10 PM, Mark Farmer wrote: >> Hi Bogdan >> >> The root of my issue is that I need 2 variables containing the >> IP's of my 2 interfaces (mhomed=yes) but the advertised address >> of the NAT'd DMZ interface while keeping changes per server to a >> bare minimum to ease deployment. >> >> I actually solved my issue by using include_file and using >> cfgutils to set 2 script variables. So now all deployment changes >> are confined to a much simpler/smaller file. >> >> However, the proposed changes would make things even nicer. Would >> cfgutils be able to accept those variables as parameters to the >> 'varset' function? >> >> Regards >> Mark. >> >> >> >> On Tue, 7 Apr 2020 at 11:44, Bogdan-Andrei Iancu >> > wrote: >> >> Hi guys, >> >> Maybe adding a new core variable like $in_socket.XXXX, to >> give access to various fields, like $in_socket.ip, >> $in_socket.port, $in_socket.advertised_ip, etc. This will >> replace the $Ri and $Rp >> >> And we can also add $out_socket, that will similarly replace >> the $fs (forced socket) >> >> Regards, >> >> Bogdan-Andrei Iancu >> >> OpenSIPS Founder and Developer >> https://www.opensips-solutions.com >> OpenSIPS Summit, Amsterdam, May 2020 >> https://www.opensips.org/events/Summit-2020Amsterdam/ >> >> On 4/6/20 6:00 PM, Johan De Clercq wrote: >>> It,s not exposed I think. I can’t find it back either >>> >>> Outlook voor iOS downloaden >>> ------------------------------------------------------------------------ >>> *Van:* Users >>> namens David >>> Villasmil >>> >>> *Verzonden:* Monday, April 6, 2020 4:49:36 PM >>> *Aan:* OpenSIPS users mailling list >>> >>> *Onderwerp:* Re: [OpenSIPS-Users] Access to >>> listen/advertised IP Addresses >>> No, you’re right. It’s not in the core variables and I can’t >>> find it either. Which makes me think it’s either not exposed >>> or somewhere in a module (it’s not in proto_udp) >>> >>> I will research a little to try and find it.. >>> >>> On Mon, 6 Apr 2020 at 14:04, Mark Farmer >> > wrote: >>> >>> Thanks David. But I see no reference to the same >>> variable in OpenSIPS. >>> >>> https://www.opensips.org/Documentation/Script-CoreVar-2-4 >>> >>> Am I missing something? >>> >>> >>> On Mon, 6 Apr 2020 at 13:45, David Villasmil >>> >> > wrote: >>> >>> Right here: >>> >>> https://www.kamailio.org/wiki/cookbooks/5.2.x/pseudovariables#rai_-_received_advertised_ip_address >>> >>> Regards, >>> >>> David Villasmil >>> email: david.villasmil.work at gmail.com >>> >>> phone: +34669448337 >>> >>> >>> On Mon, Apr 6, 2020 at 12:08 PM Mark Farmer >>> > wrote: >>> >>> Many thanks for the reply. >>> >>> $Ri is certainly useful when the request comes >>> from a non-natted interface. Thanks for pointing >>> that out :) >>> >>> Is there a way to reference the advertised IP >>> address defined in the listen statement? >>> >>> listen=udp:xxx.xxx.xxx.xxx:5060 as >>> xxx.xxx.xxx.xxx:5060 >>> >>> Thanks >>> Mark. >>> >>> >>> On Thu, 2 Apr 2020 at 17:32, Sharad Kumar via >>> Users >> > wrote: >>> >>> Hi Mark, >>> >>>  If your initial goal is to get the >>> interface IP where request is received then >>> you can try these variables. >>> >>> *$Ri* - reference to IP address of the >>> interface where the request has been received >>> >>> *$Rp* - reference to the port where the >>> message was received >>> >>> _______________________________________________ >>> Users mailing list >>> Users at lists.opensips.org >>> >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >>> >>> >>> -- >>> Mark Farmer >>> farmorg at gmail.com >>> _______________________________________________ >>> Users mailing list >>> Users at lists.opensips.org >>> >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >>> _______________________________________________ >>> Users mailing list >>> Users at lists.opensips.org >>> >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >>> >>> >>> -- >>> Mark Farmer >>> farmorg at gmail.com >>> _______________________________________________ >>> Users mailing list >>> Users at lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >>> -- >>> 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 >> >> >> >> -- >> Mark Farmer >> farmorg at gmail.com >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > -- > Mark Farmer > farmorg at gmail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From xaled at web.de Wed Apr 8 09:32:24 2020 From: xaled at web.de (xaled) Date: Wed, 8 Apr 2020 11:32:24 +0200 Subject: [OpenSIPS-Users] Access to parameters from generic headers Message-ID: <039401d60d88$9cf00fa0$d6d02ee0$@web.de> Hi, I'm having problems with accessing values of header parameters. I tried to use regex to extract cause value from Reason header, but it does not work properly. I'm not an regex expert so the regex could wrong. It would be good to have a generic mechanism for this type of values extraction from header parameters. Reason: Q.850;cause=1;text="Unallocated (unassigned) number", SIP;cause=500;text="Server internal error" $var(cause_reg) = "/(.*)cause=(.*);(.*)/\2/i"; $var(cause) = $(hdr(Reason){re.subst,$var(cause_reg)}); I added a feature request some time ago for this issue: https://github.com/OpenSIPS/opensips/issues/1289 From pasan_5 at yahoo.com Wed Apr 8 09:35:41 2020 From: pasan_5 at yahoo.com (Pasan Meemaduma) Date: Wed, 8 Apr 2020 09:35:41 +0000 (UTC) Subject: [OpenSIPS-Users] tls error References: <383031527.2439829.1586338541743.ref@mail.yahoo.com> Message-ID: <383031527.2439829.1586338541743@mail.yahoo.com> Hi Guys. Hope everyone is safe and be safe. I'm running into an issue with using tls in opensips. I'm trying to have two connections from asterisk servers and onlyone server connection is accepted at a time, Both asterisk servers are using the same wild card cert for their tls connections. I'm getting the below error Apr  8 09:22:46 ip-172-31-36-39 opensips[2846]: Apr  8 09:22:46 [2863] ERROR:proto_tls:_tls_read: SYSCALL error -> (11) Apr  8 09:22:46 ip-172-31-36-39 opensips[2846]: Apr  8 09:22:46 [2863] ERROR:proto_tls:_tls_read: TLS connection to x.x.x.x:60550 read failed Apr  8 09:22:46 ip-172-31-36-39 opensips[2846]: Apr  8 09:22:46 [2863] ERROR:proto_tls:_tls_read: TLS read error: 5 Apr  8 09:22:46 ip-172-31-36-39 opensips[2846]: Apr  8 09:22:46 [2863] ERROR:proto_tls:tls_print_errstack: TLS errstack: error:0200100D:system library:fopen:Permission denied Apr  8 09:22:46 ip-172-31-36-39 opensips[2846]: Apr  8 09:22:46 [2863] ERROR:proto_tls:tls_print_errstack: TLS errstack: error:20074002:BIO routines:file_ctrl:system lib Apr  8 09:22:46 ip-172-31-36-39 opensips[2846]: Apr  8 09:22:46 [2863] ERROR:proto_tls:tls_print_errstack: TLS errstack: error:0B06F002:x509 certificate routines:X509_load_cert_file:system lib Apr  8 09:22:46 ip-172-31-36-39 opensips[2846]: Apr  8 09:22:46 [2863] ERROR:proto_tls:tls_read_req: failed to read asterisk 1  (tls) ---> opensipsasterisk 2  (tls) ---> I'm using below opensips opensips -V version: opensips 2.4.6 (x86_64/linux) flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535 poll method support: poll, epoll, sigio_rt, select. main.c compiled on  with gcc 6.3.0 I tried to update to latest 2.4.7 and then opensips processes get stuck in a loop consuming all CPUs when tls module loaded with exact config which running on 2.4.6. Any hint clue would be helpful opensips config is as below, ####### Global Parameters ######### log_level=5 log_stderror=yes log_facility=LOG_LOCAL0 #udp_workers=1 #tcp_workers=1 tcp_connect_timeout=900 auto_aliases=no alias=tls:x.cloud:5061 alias=udp:172.31.36.39:5060 listen=tls:172.31.36.39:5061 listen=udp:172.31.36.39:5060   # CUSTOMIZE ME advertised_address=x.x.x.x ####### Modules Section ######## #set module path mpath="/usr/lib/x86_64-linux-gnu/opensips/modules/" loadmodule "tls_mgm.so" loadmodule "proto_tls.so" #loadmodule "proto_hep.so" loadmodule "uri.so" loadmodule "drouting.so" loadmodule "db_mysql.so" #### SIGNALING module loadmodule "signaling.so" loadmodule "textops.so" #### StateLess module loadmodule "sl.so" loadmodule "avpops.so" #### Transaction Module loadmodule "tm.so" modparam("tm", "fr_timeout", 30) modparam("tm", "fr_inv_timeout", 60) modparam("tm", "restart_fr_on_each_reply", 0) modparam("tm", "onreply_avp_mode", 1) modparam("tm", "via1_matching", 0) modparam("tm", "ruri_matching", 0) modparam("tm", "T1_timer", 1000) #### Record Route Module loadmodule "rr.so" #modparam("rr", "append_fromtag", 1) #### MAX ForWarD module loadmodule "maxfwd.so" loadmodule "nathelper.so" #### SIP MSG OPerationS module loadmodule "sipmsgops.so" #### FIFO Management Interface loadmodule "mi_fifo.so" modparam("mi_fifo", "fifo_name", "/tmp/opensips_fifo") modparam("mi_fifo", "fifo_mode", 0666) loadmodule "permissions.so" modparam("permissions", "db_url","mysql://opensips:xx at localhost/opensips") loadmodule "proto_udp.so" # RULE of THUMB make sure certs can be read by opensips user # otherwise Its a nightmare to debug :( modparam("tls_mgm", "certificate", "/etc/opensips/tls/default.crt") modparam("tls_mgm", "private_key","/etc/opensips/tls/default.key") modparam("tls_mgm", "ca_list", "/etc/opensips/tls/ca-default.crt") modparam("tls_mgm", "ca_dir", "/etc/ssl/certs/") modparam("tls_mgm","verify_cert", "1") modparam("tls_mgm","require_cert", "1") modparam("tls_mgm", "server_domain", "dom1=172.31.36.39:5061") modparam("tls_mgm","verify_cert", "[dom1]1") modparam("tls_mgm","require_cert", "[dom1]1") modparam("tls_mgm","tls_method", "[dom1]TLSv1_2") modparam("tls_mgm","certificate", "[dom1]/etc/tls/x.cloud/x.cloud.crt") modparam("tls_mgm","private_key", "[dom1]/etc/tls/x.cloud/x.cloud.key") modparam("tls_mgm", "ca_list", "[dom1]/etc/tls/x.cloud/x.cloud-ca.crt") modparam("tls_mgm", "ca_dir", "[dom1]/etc/ssl/certs/") modparam("tls_mgm", "tls_handshake_timeout", 900) modparam("proto_tls", "tls_max_msg_chunks", 1024) modparam("drouting", "db_url","mysql://opensips:x at localhost/opensips") modparam("drouting", "probing_from", "sip:pinger at x.x.x.x") modparam("avpops","db_url","mysql://opensips:x at localhost/opensips") ####### Routing Logic ######## # main request routing logic route{     force_rport();     if (!mf_process_maxfwd_header("10")) {         sl_send_reply("483","Too Many Hops");         exit;     }     if(is_method("OPTIONS")) {           xlog("L_INFO", "[MS TEAMS] OPTIONS In\n");           sl_send_reply("200", "OK");           exit;     }     # absorb retransmissions, but do not create transaction     t_check_trans();     if (has_totag()) {         # sequential request within a dialog should         # take the path determined by record-routing         if(is_method("INVITE|BYE") && check_source_address("0")) {             xlog("In dialog Method=$rm, RURI=$ruri, SI=$si ,DU=$du\n");             t_relay();         }         if ( !loose_route() ) {             # we do record-routing for all our traffic, so we should not             # receive any sequential requests without Route hdr.             sl_send_reply("404", "Not here");             exit;         }         # route it out to whatever destination was set by loose_route()         # in $du (destination URI).         route(relay);         exit;     }     # CANCEL processing     if (is_method("CANCEL")) {         if (t_check_trans())             t_relay();         exit;     }         # record routing     if (is_method("INVITE") && ! has_totag() && ! check_source_address("0")) {         xlog("Incoming call to MS: RURI=$ruri, SI=$si, M=$rm\n");         if(!avp_db_query("SELECT msteams_domain FROM vpabx_routing WHERE phone_number='$(rU{s.escape.common})'", "$avp(teamsdomain)")){             sl_send_reply("404", "User Not Found");                         exit;         }         $var(rrhdr) = $avp(teamsdomain) + ":5061;transport=tls";         strip(1);         do_routing("1");         prefix("+");                 record_route_preset("$var(rrhdr)", "172.31.36.39:5060");                 add_rr_param(";r2=on");         route(relay);                 } else if (is_method("INVITE") && ! has_totag()) {         record_route();         xlog("Incoming call from MS: RURI=$ruri, SI=$si, M=$rm\n");         if(!avp_db_query("SELECT vpabx_domain FROM vpabx_routing WHERE phone_number='$(fU{s.escape.common})'","$avp(ddomain)")){             sl_send_reply("404", "User Not Found");             exit;         }         else {             $rd = $avp(ddomain);             route(relay);         }     }     if (!is_myself("$rd")) {         append_hf("P-hint: outbound\r\n");                  route(relay);     }     # requests for my domain          if (is_method("PUBLISH|SUBSCRIBE")) {         sl_send_reply("503", "Service Unavailable");         exit;     }     if ($rU==NULL) {         # request with no Username in RURI         sl_send_reply("484", "Address Incomplete");         exit;     }     # when routing via usrloc, log the missed calls also     #route(relay); } route[relay] {     # for INVITEs enable some additional helper routes     if (is_method("INVITE") && !has_totag() ) {         t_newtran();         t_on_reply("handle_nat");     }     xlog("Method=$rm, RURI=$ruri, SI=$si ,DU=$du\n");     if (!t_relay()) {         sl_send_reply("500", "Internal Error");     }     exit; } branch_route[per_branch_ops] {     xlog("new branch at $ru\n"); } onreply_route[handle_nat] {     xlog("incoming reply: RR=$rr, RS=$rs, SI=$si\n"); } failure_route[missed_call] {     if (t_was_cancelled()) {         exit;     } } local_route {   $var(dst) = "pstnhub.microsoft.com";   if (is_method("OPTIONS") && ($(ru{s.index, $var(dst)}) != NULL))     append_hf("Contact: \r\n"); }   Thank you Pasan Distinguishing What && How ! 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URL: From farmorg at gmail.com Wed Apr 8 09:41:37 2020 From: farmorg at gmail.com (Mark Farmer) Date: Wed, 8 Apr 2020 10:41:37 +0100 Subject: [OpenSIPS-Users] Access to listen/advertised IP Addresses In-Reply-To: <5957bcba-f283-50e4-19a7-4b4d43f8b987@opensips.org> References: <5eb6a6ca-83aa-c59e-2f2a-2c4f216a4118@opensips.org> <5957bcba-f283-50e4-19a7-4b4d43f8b987@opensips.org> Message-ID: Of course :) On Tue, 7 Apr 2020 at 17:29, Bogdan-Andrei Iancu wrote: > No need, just use in script, where ever you need $socket_in(advertised_ip) > and it will be evaluated for the current socket (used for receiving the > request) > > Regardsm > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > OpenSIPS Summit, Amsterdam, May 2020 > https://www.opensips.org/events/Summit-2020Amsterdam/ > > On 4/7/20 6:56 PM, Mark Farmer wrote: > > I was thinking something like: > > modparam("cfgutils", "varset", "extip=s:$in_socket.advertised_ip") > > > On Tue, 7 Apr 2020 at 14:40, Bogdan-Andrei Iancu > wrote: > >> Hi Mark, >> >> ingenious solution :) >> >> In regards to the proposed solution, I do not understand the question >> about varset (cfgutils), as there is no relation between the script vars >> and these new $socket vars. Maybe I'm missing something from your question. >> >> Regards, >> >> Bogdan-Andrei Iancu >> >> OpenSIPS Founder and Developer >> https://www.opensips-solutions.com >> OpenSIPS Summit, Amsterdam, May 2020 >> https://www.opensips.org/events/Summit-2020Amsterdam/ >> >> On 4/7/20 2:10 PM, Mark Farmer wrote: >> >> Hi Bogdan >> >> The root of my issue is that I need 2 variables containing the IP's of my >> 2 interfaces (mhomed=yes) but the advertised address of the NAT'd DMZ >> interface while keeping changes per server to a bare minimum to ease >> deployment. >> >> I actually solved my issue by using include_file and using cfgutils to >> set 2 script variables. So now all deployment changes are confined to a >> much simpler/smaller file. >> >> However, the proposed changes would make things even nicer. Would >> cfgutils be able to accept those variables as parameters to the 'varset' >> function? >> >> Regards >> Mark. >> >> >> >> On Tue, 7 Apr 2020 at 11:44, Bogdan-Andrei Iancu >> wrote: >> >>> Hi guys, >>> >>> Maybe adding a new core variable like $in_socket.XXXX, to give access to >>> various fields, like $in_socket.ip, $in_socket.port, $in_socket.advertised_ip, >>> etc. This will replace the $Ri and $Rp >>> >>> And we can also add $out_socket, that will similarly replace the $fs >>> (forced socket) >>> >>> Regards, >>> >>> Bogdan-Andrei Iancu >>> >>> OpenSIPS Founder and Developer >>> https://www.opensips-solutions.com >>> OpenSIPS Summit, Amsterdam, May 2020 >>> https://www.opensips.org/events/Summit-2020Amsterdam/ >>> >>> On 4/6/20 6:00 PM, Johan De Clercq wrote: >>> >>> It,s not exposed I think. I can’t find it back either >>> >>> Outlook voor iOS downloaden >>> ------------------------------ >>> *Van:* Users >>> namens David Villasmil >>> >>> *Verzonden:* Monday, April 6, 2020 4:49:36 PM >>> *Aan:* OpenSIPS users mailling list >>> >>> *Onderwerp:* Re: [OpenSIPS-Users] Access to listen/advertised IP >>> Addresses >>> >>> No, you’re right. It’s not in the core variables and I can’t find it >>> either. Which makes me think it’s either not exposed or somewhere in a >>> module (it’s not in proto_udp) >>> >>> I will research a little to try and find it.. >>> >>> On Mon, 6 Apr 2020 at 14:04, Mark Farmer wrote: >>> >>> Thanks David. But I see no reference to the same variable in OpenSIPS. >>> >>> https://www.opensips.org/Documentation/Script-CoreVar-2-4 >>> >>> Am I missing something? >>> >>> >>> On Mon, 6 Apr 2020 at 13:45, David Villasmil < >>> david.villasmil.work at gmail.com> wrote: >>> >>> Right here: >>> >>> >>> https://www.kamailio.org/wiki/cookbooks/5.2.x/pseudovariables#rai_-_received_advertised_ip_address >>> >>> Regards, >>> >>> David Villasmil >>> email: david.villasmil.work at gmail.com >>> phone: +34669448337 >>> >>> >>> On Mon, Apr 6, 2020 at 12:08 PM Mark Farmer wrote: >>> >>> Many thanks for the reply. >>> >>> $Ri is certainly useful when the request comes from a non-natted >>> interface. Thanks for pointing that out :) >>> >>> Is there a way to reference the advertised IP address defined in the >>> listen statement? >>> >>> listen=udp:xxx.xxx.xxx.xxx:5060 as xxx.xxx.xxx.xxx:5060 >>> >>> Thanks >>> Mark. >>> >>> >>> On Thu, 2 Apr 2020 at 17:32, Sharad Kumar via Users < >>> users at lists.opensips.org> wrote: >>> >>> Hi Mark, >>> >>> If your initial goal is to get the interface IP where request is >>> received then you can try these variables. >>> >>> *$Ri* - reference to IP address of the interface where the request has >>> been received >>> >>> *$Rp* - reference to the port where the message was received >>> _______________________________________________ >>> Users mailing list >>> Users at lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >>> >>> >>> -- >>> Mark Farmer >>> farmorg at gmail.com >>> _______________________________________________ >>> Users mailing list >>> Users at lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >>> _______________________________________________ >>> Users mailing list >>> Users at lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >>> >>> >>> -- >>> Mark Farmer >>> farmorg at gmail.com >>> _______________________________________________ >>> Users mailing list >>> Users at lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >>> -- >>> Regards, >>> >>> David Villasmil >>> email: david.villasmil.work at gmail.com >>> phone: +34669448337 >>> >>> _______________________________________________ >>> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >>> >>> _______________________________________________ >>> Users mailing list >>> Users at lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >> >> >> -- >> Mark Farmer >> farmorg at gmail.com >> >> _______________________________________________ >> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> >> > > -- > Mark Farmer > farmorg at gmail.com > > > -- Mark Farmer farmorg at gmail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From farmorg at gmail.com Wed Apr 8 09:49:00 2020 From: farmorg at gmail.com (Mark Farmer) Date: Wed, 8 Apr 2020 10:49:00 +0100 Subject: [OpenSIPS-Users] Modules not inheriting db_default_url Message-ID: Hi everyone I have noticed that some modules do not inherit db_default_url. In my case: dialplan.so permissions.so rtpengine.so I also noticed that the documentation for permissions.so actually says that this is the case but dialplan & rtpengine do not. Is it possible for these/all modules to inherit db_default_url or is there a reason for this? OpenSIPS 2.4.7 Best regards Mark. -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Wed Apr 8 10:19:43 2020 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Wed, 8 Apr 2020 13:19:43 +0300 Subject: [OpenSIPS-Users] Access to parameters from generic headers In-Reply-To: <039401d60d88$9cf00fa0$d6d02ee0$@web.de> References: <039401d60d88$9cf00fa0$d6d02ee0$@web.de> Message-ID: <4a904efa-64e6-7464-1c2d-37f542145241@opensips.org> Hi, Try to use the param transformation https://opensips.org/Documentation/Script-Tran-3-0#toc60 Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com OpenSIPS Summit, Amsterdam, May 2020 https://www.opensips.org/events/Summit-2020Amsterdam/ On 4/8/20 12:32 PM, xaled wrote: > Hi, > > I'm having problems with accessing values of header parameters. I tried to use regex to extract cause value from Reason header, but it does not work properly. I'm not an regex expert so the regex could wrong. It would be good to have a generic mechanism for this type of values extraction from header parameters. > > Reason: Q.850;cause=1;text="Unallocated (unassigned) number", SIP;cause=500;text="Server internal error" > > $var(cause_reg) = "/(.*)cause=(.*);(.*)/\2/i"; > $var(cause) = $(hdr(Reason){re.subst,$var(cause_reg)}); > > I added a feature request some time ago for this issue: > https://github.com/OpenSIPS/opensips/issues/1289 > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users From liviu at opensips.org Wed Apr 8 10:47:37 2020 From: liviu at opensips.org (Liviu Chircu) Date: Wed, 8 Apr 2020 13:47:37 +0300 Subject: [OpenSIPS-Users] Modules not inheriting db_default_url In-Reply-To: References: Message-ID: <0e2386b3-5a29-b2c1-40c9-b5ada9c9b1e5@opensips.org> On 08.04.2020 12:49, Mark Farmer wrote: > Is it possible for these/all modules to inherit db_default_url or is > there a reason for this? Hi, Mark! Logically speaking, I think it makes sense for all those 3 modules you listed to inherit from db_default_url [1].  Also, this should be the case for any module which requires a db_url by default (e.g. dialog, usrloc, etc.).  Most likely, we have to take the modules one-by-one and correct this issue for any of them which still have it. [1]: https://www.opensips.org/Documentation/Script-CoreParameters-3-1#toc14 -- Liviu Chircu www.twitter.com/liviuchircu | www.opensips-solutions.com OpenSIPS Summit, Amsterdam, May 2020 www.opensips.org/events From farmorg at gmail.com Wed Apr 8 11:15:15 2020 From: farmorg at gmail.com (Mark Farmer) Date: Wed, 8 Apr 2020 12:15:15 +0100 Subject: [OpenSIPS-Users] Modules not inheriting db_default_url In-Reply-To: <0e2386b3-5a29-b2c1-40c9-b5ada9c9b1e5@opensips.org> References: <0e2386b3-5a29-b2c1-40c9-b5ada9c9b1e5@opensips.org> Message-ID: Hi Liviu That sounds good to me. Do you need me to do anything in order to get this implemented? Mark. On Wed, 8 Apr 2020 at 11:49, Liviu Chircu wrote: > On 08.04.2020 12:49, Mark Farmer wrote: > > Is it possible for these/all modules to inherit db_default_url or is > > there a reason for this? > > Hi, Mark! > > Logically speaking, I think it makes sense for all those 3 modules you > listed to inherit from db_default_url [1]. Also, this should be the > case for any module which requires a db_url by default (e.g. dialog, > usrloc, etc.). Most likely, we have to take the modules one-by-one and > correct this issue for any of them which still have it. > > [1]: > https://www.opensips.org/Documentation/Script-CoreParameters-3-1#toc14 > > -- > Liviu Chircu > www.twitter.com/liviuchircu | www.opensips-solutions.com > > OpenSIPS Summit, Amsterdam, May 2020 > www.opensips.org/events > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -- Mark Farmer farmorg at gmail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Wed Apr 8 11:20:10 2020 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Wed, 8 Apr 2020 14:20:10 +0300 Subject: [OpenSIPS-Users] Modules not inheriting db_default_url In-Reply-To: <0e2386b3-5a29-b2c1-40c9-b5ada9c9b1e5@opensips.org> References: <0e2386b3-5a29-b2c1-40c9-b5ada9c9b1e5@opensips.org> Message-ID: <2032d2ac-4abe-1597-2b8f-450861296ec4@opensips.org> Liviu, not all the modules exposing a db_url should inherit the db_default_url. For some modules the db_url is optional and acts as a switch for the module (if db backend should be used or not). Forcing the default db_url may break this behavior. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com OpenSIPS Summit, Amsterdam, May 2020 https://www.opensips.org/events/Summit-2020Amsterdam/ On 4/8/20 1:47 PM, Liviu Chircu wrote: > On 08.04.2020 12:49, Mark Farmer wrote: >> Is it possible for these/all modules to inherit db_default_url or is >> there a reason for this? > > Hi, Mark! > > Logically speaking, I think it makes sense for all those 3 modules you > listed to inherit from db_default_url [1].  Also, this should be the > case for any module which requires a db_url by default (e.g. dialog, > usrloc, etc.).  Most likely, we have to take the modules one-by-one > and correct this issue for any of them which still have it. > > [1]: > https://www.opensips.org/Documentation/Script-CoreParameters-3-1#toc14 > From liviu at opensips.org Wed Apr 8 11:25:07 2020 From: liviu at opensips.org (Liviu Chircu) Date: Wed, 8 Apr 2020 14:25:07 +0300 Subject: [OpenSIPS-Users] Modules not inheriting db_default_url In-Reply-To: References: <0e2386b3-5a29-b2c1-40c9-b5ada9c9b1e5@opensips.org> Message-ID: On 08.04.2020 14:15, Mark Farmer wrote: > Do you need me to do anything in order to get this implemented? Well, if you want, you could go through the modules and see if there are any other ones which require this, apart from dialplan/permissions/rtpengine (which I agree with: they should all inherit the default DB URL).  Next, I suggest you re-use the init_db_url() macro [1] to easily incorporate the default DB URL logic, just like dialog does it [2], for example. PS: I will be happy to review the PR if you're willing to do it! :) Best regards, [1]: https://github.com/OpenSIPS/opensips/blob/master/db/db.h#L471 [2]: https://github.com/OpenSIPS/opensips/blob/master/modules/dialog/dialog.c#L661 -- Liviu Chircu www.twitter.com/liviuchircu | www.opensips-solutions.com OpenSIPS Summit, Amsterdam, May 2020 www.opensips.org/events From liviu at opensips.org Wed Apr 8 11:25:57 2020 From: liviu at opensips.org (Liviu Chircu) Date: Wed, 8 Apr 2020 14:25:57 +0300 Subject: [OpenSIPS-Users] Modules not inheriting db_default_url In-Reply-To: <2032d2ac-4abe-1597-2b8f-450861296ec4@opensips.org> References: <0e2386b3-5a29-b2c1-40c9-b5ada9c9b1e5@opensips.org> <2032d2ac-4abe-1597-2b8f-450861296ec4@opensips.org> Message-ID: On 08.04.2020 14:20, Bogdan-Andrei Iancu wrote: > not all the modules exposing a db_url should inherit the db_default_url Totally agree!  However, giving it some thought, I couldn't find such an example, mostly because all such modules (e.g. dialog, usrloc) already have a "MODE" switch which forces that behavior.  So inheriting the default URL string is fully backwards-compatible. Bottom line: if we do this extension in a smart way, I think that we will end up with a 100% better experience (no downsides) -- Liviu Chircu www.twitter.com/liviuchircu | www.opensips-solutions.com OpenSIPS Summit, Amsterdam, May 2020 www.opensips.org/events From bogdan at opensips.org Wed Apr 8 11:30:31 2020 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Wed, 8 Apr 2020 14:30:31 +0300 Subject: [OpenSIPS-Users] Modules not inheriting db_default_url In-Reply-To: References: <0e2386b3-5a29-b2c1-40c9-b5ada9c9b1e5@opensips.org> <2032d2ac-4abe-1597-2b8f-450861296ec4@opensips.org> Message-ID: <677b150b-b685-fce9-2139-839f22b0bc84@opensips.org> I recalled this issue from the times when the default db_url was added. Maybe things have changed in the mean while, but I considered worthy to be mentioned. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com OpenSIPS Summit, Amsterdam, May 2020 https://www.opensips.org/events/Summit-2020Amsterdam/ On 4/8/20 2:25 PM, Liviu Chircu wrote: > On 08.04.2020 14:20, Bogdan-Andrei Iancu wrote: >> not all the modules exposing a db_url should inherit the db_default_url > > Totally agree!  However, giving it some thought, I couldn't find such > an example, mostly because all such modules (e.g. dialog, usrloc) > already have a "MODE" switch which forces that behavior. So inheriting > the default URL string is fully backwards-compatible. > > Bottom line: if we do this extension in a smart way, I think that we > will end up with a 100% better experience (no downsides) > From xaled at web.de Wed Apr 8 12:10:25 2020 From: xaled at web.de (xaled) Date: Wed, 8 Apr 2020 14:10:25 +0200 Subject: [OpenSIPS-Users] Access to parameters from generic headers In-Reply-To: <4a904efa-64e6-7464-1c2d-37f542145241@opensips.org> References: <039401d60d88$9cf00fa0$d6d02ee0$@web.de> <4a904efa-64e6-7464-1c2d-37f542145241@opensips.org> Message-ID: <03b201d60d9e$affb3c80$0ff1b580$@web.de> Hi Bogdan, works perfectly, dunno why I did not find it earlier. Thanks! -----Original Message----- From: Bogdan-Andrei Iancu [mailto:bogdan at opensips.org] Sent: Wednesday, April 08, 2020 12:20 PM To: OpenSIPS users mailling list ; xaled Subject: Re: [OpenSIPS-Users] Access to parameters from generic headers Hi, Try to use the param transformation https://opensips.org/Documentation/Script-Tran-3-0#toc60 Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com OpenSIPS Summit, Amsterdam, May 2020 https://www.opensips.org/events/Summit-2020Amsterdam/ On 4/8/20 12:32 PM, xaled wrote: > Hi, > > I'm having problems with accessing values of header parameters. I tried to use regex to extract cause value from Reason header, but it does not work properly. I'm not an regex expert so the regex could wrong. It would be good to have a generic mechanism for this type of values extraction from header parameters. > > Reason: Q.850;cause=1;text="Unallocated (unassigned) number", SIP;cause=500;text="Server internal error" > > $var(cause_reg) = "/(.*)cause=(.*);(.*)/\2/i"; > $var(cause) = $(hdr(Reason){re.subst,$var(cause_reg)}); > > I added a feature request some time ago for this issue: > https://github.com/OpenSIPS/opensips/issues/1289 > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users From mrsanvicente at gmail.com Thu Apr 9 05:11:00 2020 From: mrsanvicente at gmail.com (Mario San Vicente) Date: Thu, 9 Apr 2020 00:11:00 -0500 Subject: [OpenSIPS-Users] PRACK support / configuration Message-ID: Hello Everyone, I have been trying to enable PRACK support, and on the docs it says it is supported on module b2b_entitier modules/b2b_entities/dlg.c /* PRACK handling *//* if the provisional reply contains a - Require: 100rel header -> send PRACK */ It also says that B2BA top hiding is a predefined service that works without any scenario definition and for any type of dialog. So I am traying to enable top hiding to get the PRACK working. just adding thhe line: b2b_init_request("top hiding"); But when i add this line, the call flow stops working. Any advice on how to configure this feature? NOTE: In my working configuration i have a dialog created, just to let you know if that is supported at the same time. top hiding and dialog. Thank you -- Mario San Vicente Cheers! -------------- next part -------------- An HTML attachment was scrubbed... URL: From razvan at opensips.org Thu Apr 9 08:06:36 2020 From: razvan at opensips.org (=?UTF-8?Q?R=c4=83zvan_Crainea?=) Date: Thu, 9 Apr 2020 11:06:36 +0300 Subject: [OpenSIPS-Users] Fwd: Opensips 3.0.2 - Cannot install with db_mysql module enabled In-Reply-To: References: Message-ID: <4a5f8a5f-c328-ad0a-3399-af6fcf1d8e40@opensips.org> Hi, Gordon! Could you post somewhere the content of Makefile.conf? Best regards, Răzvan Crainea OpenSIPS Core Developer http://www.opensips-solutions.com On 3/19/20 3:18 AM, Gordon Yeong wrote: > > hi, guys, > > I am trying to test out the blacklist functionality on my sandboxes > (Centos 7). > > > Replication steps: > =============== > >  I have the sources for opensips 3.0.2 and in "make menuconfig", I have: > 1) enabled "db_mysql" by selecting it > > > 2) specified my install path as /opt/bin (don't ask why). > > > I then hit the "Save Changes". I get the message 'You have enabled the > 'db_mysql' module, so please install ' development libraries of > mysql-client , typically libmysqlclient-dev'*'.* > > ** > > > >   On my Centos 7 machine, I have already isntalled the mysql client dev > modules. > >  ⚡ root at localhost  ~/opensips-3.0.2  yum install mysql-client > Loaded plugins: fastestmirror, langpacks > Loading mirror speeds from cached hostfile >  * base: centos.melbourneitmirror.net > >  * epel: fedora.melbourneitmirror.net > >  * extras: centos.melbourneitmirror.net > >  * updates: centos.melbourneitmirror.net > > Package MariaDB-client-10.4.12-1.el7.centos.x86_64 already installed > and latest version > Nothing to do > > > > ISSUE: > ======== > >  The mysql client dev modules have already been installed and yet when > I hit 'Compile and Install Opensips", there's so many issues primarily > around around shm. > > > > I have attached a text version of the output > (failed_compilation_install_log.txt) when I tried to compile and install > in this thread. > > Can anyone please shed some light on this? > I have also tried to run the following as a measure to try and not use > "make menuconfig" to no avail. > > " make prefix=/opt/opensips all &&  make prefix=/opt/opensips/ > install".  The compilation output remains the same as what I have > encountered by using "make menuconfig". > > > Please assist. > >  Thank you > > > Gordon > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > From anexiole at gmail.com Thu Apr 9 08:23:22 2020 From: anexiole at gmail.com (Gordon Yeong) Date: Thu, 9 Apr 2020 18:23:22 +1000 Subject: [OpenSIPS-Users] Fwd: Opensips 3.0.2 - Cannot install with db_mysql module enabled In-Reply-To: <4a5f8a5f-c328-ad0a-3399-af6fcf1d8e40@opensips.org> References: <4a5f8a5f-c328-ad0a-3399-af6fcf1d8e40@opensips.org> Message-ID: Hello there. I had to reinstall the vm and the fault was no longer produceable. I suspect it was a bad install of the centos7 vm Thank you and happy easter. Stay safe :) On Thu, 9 Apr 2020, 6:08 pm Răzvan Crainea, wrote: > Hi, Gordon! > > Could you post somewhere the content of Makefile.conf? > > Best regards, > > Răzvan Crainea > OpenSIPS Core Developer > http://www.opensips-solutions.com > > On 3/19/20 3:18 AM, Gordon Yeong wrote: > > > > hi, guys, > > > > I am trying to test out the blacklist functionality on my sandboxes > > (Centos 7). > > > > > > Replication steps: > > =============== > > > > I have the sources for opensips 3.0.2 and in "make menuconfig", I have: > > 1) enabled "db_mysql" by selecting it > > > > > > 2) specified my install path as /opt/bin (don't ask why). > > > > > > I then hit the "Save Changes". I get the message 'You have enabled the > > 'db_mysql' module, so please install ' development libraries of > > mysql-client , typically libmysqlclient-dev'*'.* > > > > ** > > > > > > > > On my Centos 7 machine, I have already isntalled the mysql client dev > > modules. > > > > ⚡ root at localhost  ~/opensips-3.0.2  yum install mysql-client > > Loaded plugins: fastestmirror, langpacks > > Loading mirror speeds from cached hostfile > > * base: centos.melbourneitmirror.net > > > > * epel: fedora.melbourneitmirror.net > > > > * extras: centos.melbourneitmirror.net > > > > * updates: centos.melbourneitmirror.net > > > > Package MariaDB-client-10.4.12-1.el7.centos.x86_64 already installed > > and latest version > > Nothing to do > > > > > > > > ISSUE: > > ======== > > > > The mysql client dev modules have already been installed and yet when > > I hit 'Compile and Install Opensips", there's so many issues primarily > > around around shm. > > > > > > > > I have attached a text version of the output > > (failed_compilation_install_log.txt) when I tried to compile and install > > in this thread. > > > > Can anyone please shed some light on this? > > I have also tried to run the following as a measure to try and not use > > "make menuconfig" to no avail. > > > > " make prefix=/opt/opensips all && make prefix=/opt/opensips/ > > install". The compilation output remains the same as what I have > > encountered by using "make menuconfig". > > > > > > Please assist. > > > > Thank you > > > > > > Gordon > > > > > > > > _______________________________________________ > > 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 masked at vale.ski Thu Apr 9 08:43:41 2020 From: masked at vale.ski (Michael Vale) Date: Thu, 09 Apr 2020 18:43:41 +1000 Subject: [OpenSIPS-Users] callfwd avp with opensips-cli Message-ID: hi. is it possible to set the avp for call forwarding with opensips-cli and if so, how? regards, michael. From donat.zenichev at gmail.com Thu Apr 9 12:03:15 2020 From: donat.zenichev at gmail.com (Donat Zenichev) Date: Thu, 9 Apr 2020 15:03:15 +0300 Subject: [OpenSIPS-Users] A behavior of Clusterer module during networking issues Message-ID: Hi there! I have question and it's almost theoretic. The question relates to Clusterer module and its behavior. Lets imagine we have a regular Active/Stand-by setup. And both instances are sharing the same sharing tag, for e.g.: "vip/1" This cluster also has some automatic tool kit that handles: - shared IP migration - shared tag activation One day a networking failure happens on Active (Master) side, and Stand-by side decides to allocate shared IP address on itself, and also Stand-by machine activates shared tag (sets it to active). For a while, Stand-by side acts as Active machine and everything works great. But, since that failure with a regular Master, was networking related issue. And, operation system keeps on working on regular Master, then a regular Master also supposes itself as a real Active node, and it thinks that it's Stand-by side that is unreachable and not-working, meanwhile it's not true. What happens with a sharing tag, when a Master side comes back into play? (no more networking issues) Master keeps on working (all the time) with already present active "vip/1" shared tag. So when Master returns into work, and meets Stand-by side, they both have active shared tag. Will Clusterer module solve this contradiction on its own? And if so, to which side the precedence is given? The other way around could be to manually re-activate all services, when all the cluster resumes into normal working process (all nodes are present). Thus this gives us a warranty that shared tag will only be activated on one of the sides. Any answer is appreciated. -- Best regards, Donat Zenichev -------------- next part -------------- An HTML attachment was scrubbed... URL: From m.shirazi at gmail.com Thu Apr 9 16:47:35 2020 From: m.shirazi at gmail.com (Mehdi Shirazi) Date: Thu, 9 Apr 2020 21:17:35 +0430 Subject: [OpenSIPS-Users] [BLOG] Real-Time Rating and Cost Based Routing in OpenSIPS 3.1 Message-ID: Hi Thanks for new features. Is it possible(or planed) to use this module for basic prepaid billing ? Regards M.Shirazi >While there are numerous external rating and billing engines available >in the wild, having a quick and easy way of putting a price for a call, >without relying on external applications, can be a valuable asset to have. >https://blog.opensips.org/2020/04/07/real-time-rating-and-cost-based-routing-in-opensips-3-1/ >Thank you Vlad Paiu for the valuable contribution and post ! >Best Regards, >Bogdan-Andrei Iancu -------------- next part -------------- An HTML attachment was scrubbed... URL: From alex at canyan.io Fri Apr 10 04:37:00 2020 From: alex at canyan.io (Aleksandar Sosic) Date: Fri, 10 Apr 2020 06:37:00 +0200 Subject: [OpenSIPS-Users] failed to load module siptrace.so In-Reply-To: <63a37505-c08d-8095-b2af-fa70ba6e11cf@opensips.org> References: <08ddcf0a-8a25-98c7-9bb5-d8021ef2c392@opensips.org> <63a37505-c08d-8095-b2af-fa70ba6e11cf@opensips.org> Message-ID: Ok thanks guys, seems to work now. On Tue, Apr 7, 2020 at 1:33 PM Bogdan-Andrei Iancu wrote: > yeah, sorry, it was a left over page from the 2.4 to 3.0 migration of > the docs. > > I will rebuild the full module docs, to be sure there are no left-overs. > > Regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > OpenSIPS Summit, Amsterdam, May 2020 > https://www.opensips.org/events/Summit-2020Amsterdam/ > > On 4/7/20 2:18 PM, Aleksandar Sosic wrote: > > Ok, because I was looking at this: > > https://opensips.org/html/docs/modules/3.0.x/siptrace.html > > > > so I expected it to be available. > > Should this page then be removed? > > > > Thanks, > > Alex > > > > On Tue, Apr 7, 2020 at 12:25 PM Bogdan-Andrei Iancu > wrote: > >> Hi Aleksandar, > >> > >> Starting 3.0, the siptrace module was named "tracer" as it's not > >> sip-centric anymore, so see > >> https://opensips.org/html/docs/modules/3.0.x/tracer.html > >> > >> Regards, > >> > >> Bogdan-Andrei Iancu > >> > >> OpenSIPS Founder and Developer > >> https://www.opensips-solutions.com > >> > >> On 4/7/20 12:31 PM, Aleksandar Sosic wrote: > >>> Hi Guys, > >>> > >>> I'm installing Opensips v3.0 via `deb http://apt.opensips.org buster > >>> 3.0-releases` with: > >>> ``` > >>> apt-get install -y opensips opensips-json-module > >>> opensips-restclient-module opensips-http-modules > >>> ``` > >>> > >>> When specifying in the conf `loadmodule "siptrace.so"` upon running > >>> opensips I get this error: > >>> ``` > >>> Apr 7 09:21:49 [333] CRITICAL:core:yyerror: parse error in > >>> /etc/opensips/opensips.cfg:38:13-14: failed to load module siptrace.so > >>> ``` > >>> > >>> The file indeed is not present in > `/usr/lib/x86_64-linux-gnu/opensips/modules/`. > >>> I do have `sipcapture.so` and `proto_hep.so` although but no siptrace. > >>> What am I missing here? > >>> > >>> I'm trying to send all to a HEP agent like I do in kamailio with: > >>> ``` > >>> loadmodule "siptrace.so" > >>> > >>> modparam("siptrace", "trace_on", 1) > >>> modparam("siptrace", "trace_to_database", 0) > >>> modparam("siptrace", "hep_mode_on", 1) > >>> modparam("siptrace", "hep_version", 3) > >>> modparam("siptrace", "hep_capture_id", 1) > >>> > >>> request_route { > >>> sip_trace("hep-agent.local", "$ci"); > >>> ... > >>> } > >>> ``` > >>> > >>> Any ideas or examples of how to do this with Opensips v3? > >>> > >>> Thanks, > >>> -- > >>> Alex > >>> > >>> _______________________________________________ > >>> Users mailing list > >>> Users at lists.opensips.org > >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From alexandru.tripon at itsyscom.com Fri Apr 10 09:28:57 2020 From: alexandru.tripon at itsyscom.com (Alexandru Tripon) Date: Fri, 10 Apr 2020 12:28:57 +0300 Subject: [OpenSIPS-Users] Stir_shaken signature length Message-ID: Hi, I tried to populate the Identity header with the stir_shaken module. The header is populated but when I try to verify the signature using an external tool it fails because of the length. I have the folowing Identity generated by Opensips: ` eyJhbGciOiJFUzI1NiIsInBwdCI6InNoYWtlbiIsInR5cCI6InBhc3Nwb3J0IiwieDV1IjoiL2hvbWUvdHJpYWwvTHVjcnUvQ29kZS9zdGlyU2hha2VuL215cHVia2V5LnBlbSJ9.eyJhdHRlc3QiOiJBIiwiZGVzdCI6eyJ0biI6WyIxMDAyIl19LCJpYXQiOjE1ODY1MDMxODcsIm9yaWciOnsidG4iOiIxMDAxIn0sIm9yaWdpZCI6IjEyMzQ1NiJ9.MEYCIQCjIx6w8IeilqHq0jbc6uwIB9v1RDmecoep0gRJJC4EmQIhANH1MO9jwRtqH6jgFH12XqROFv-nUroEgzsRAaMJtAsR;info=\u003c/home/trial/Lucru/Code/stirShaken/mypubkey.pem\u003e;ppt=\"shaken\" ` the lenght of encoded signature(in base64) is 96 and in the decoded one is 72. In the RFC for ES256 algorithm( https://tools.ietf.org/html/rfc7518#section-3.4) the length of the decoded signature is 64. Am I missing something here? Thanks, Alexandru Tripon -------------- next part -------------- An HTML attachment was scrubbed... URL: From vladpaiu at opensips.org Fri Apr 10 10:00:57 2020 From: vladpaiu at opensips.org (Vlad Paiu) Date: Fri, 10 Apr 2020 13:00:57 +0300 Subject: [OpenSIPS-Users] [BLOG] Real-Time Rating and Cost Based Routing in OpenSIPS 3.1 In-Reply-To: References: Message-ID: <7259f651-4f66-802f-1b05-d7a79f914b3c@opensips.org> Hello, Currently the module can only be used for a basic postpaid type of rating. Regards, Vlad On 09.04.2020 19:47, Mehdi Shirazi wrote: > Hi > Thanks for new features. > Is it possible(or planed) to use this module for basic prepaid billing ? > > Regards > M.Shirazi > > >While there are numerous external rating and billing engines available > >in the wild, having a quick and easy way of putting a price for a call, > >without relying on external applications, can be a valuable asset to have. > > >https://blog.opensips.org/2020/04/07/real-time-rating-and-cost-based-routing-in-opensips-3-1/ > > >Thank you Vlad Paiu for the valuable contribution and post ! > > >Best Regards, > > >Bogdan-Andrei Iancu > > > > _______________________________________________ > 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 jeff at ugnd.org Fri Apr 10 19:59:40 2020 From: jeff at ugnd.org (Jeff Pyle) Date: Fri, 10 Apr 2020 15:59:40 -0400 Subject: [OpenSIPS-Users] drouting probe_mode in active/passive cluster Message-ID: Hello, On v2.4.7 I have a two-node cluster configured as active/standby with keepalived managing which side has the IP. I want to use drouting probe_mode=2, but I don't see how to prevent the passive side from trying to ping the gateways even when it doesn't have the IP. And, of course, it fails: /usr/sbin/opensips[2212445]: ERROR:core:get_out_socket: connect failed: Network is unreachable /usr/sbin/opensips[2212445]: ERROR:core:get_out_socket: no socket found /usr/sbin/opensips[2212445]: ERROR:tm:t_uac: no corresponding socket for af 2 /usr/sbin/opensips[2212445]: ERROR:drouting:dr_prob_handler: unable to execute dialog, disabling destination... If I have status_replication_cluster configured, then the passive side effectively poisons the active side by indicating the gateway is down. How can I tell the passive side not to send pings? - Jeff -------------- next part -------------- An HTML attachment was scrubbed... URL: From liviu at opensips.org Sat Apr 11 11:11:23 2020 From: liviu at opensips.org (Liviu Chircu) Date: Sat, 11 Apr 2020 14:11:23 +0300 Subject: [OpenSIPS-Users] A behavior of Clusterer module during networking issues In-Reply-To: References: Message-ID: <522532ff-c612-d09f-9877-163b9316977a@opensips.org> On 09.04.2020 15:03, Donat Zenichev wrote: > I have question and it's almost theoretic. > The question relates to Clusterer module and its behavior. > > Will Clusterer module solve this contradiction on its own? > And if so, to which side the precedence is given? > > The other way around could be to manually re-activate all services, > when all the cluster resumes into normal working process (all nodes > are present). > Thus this gives us a warranty that shared tag will only be activated > on one of the sides. Hi, Donat! A very good question and one that we had to answer ourselves when we came up with the current design.  To begin with, in your scenario, for all OpenSIPS 2.4+ clusterer versions, after the link between the nodes comes back online, you will have the following: * node A: ACTIVE state (holds the VIP), sharing tag state: ACTIVE (1) * node B: BACKUP state, sharing tag state: ACTIVE (1) The main reason behind this inconsistent state is that we did not provide an MI command to force a sharing tag to BACKUP (0), which could be triggered on node B's transition from ACTIVE -> BACKUP once the link is restored, so recovering from this state will not work automatically - you have to provide handling for this scenario as well (see last paragraph). Reasoning behind this design ---------------------------- Ultimately, our priority was not to get into solving consensus problems, Paxos algorithms, etc.  What we wanted is a robust active/backup solution which you could flip back and forth with ease, thus achieving both High-Availability and easy maintenance.  By not providing a "set_sharing_tag vip 0" command, we _avoid_ the situation where, due to a developer error, both tags end up being BACKUP (0)!!  In such a scenario: there will be no more CDRs and you will be able to run infinite CPS/CC through that instance, since all call profile counters are equal to 0.  None of the instances take responsibility for any call running through them, so a lot of data will be lost. On the flip side, in a scenario where both tags are bugged in the ACTIVE (1) state, you would have: duplicated CDRs (along with maybe some DB error logs due to conflicting unique keys) and possibly extra-counted calls, leading to a reduction of the maximally supported CC/CPS.  Assume that the platform wasn't even at 50% of the max limits to begin with, and the latter has 0 impact on the live system.  Thinking about this, this didn't sound that bad at all to us: no data loss, at the expense of a few error logs and possibly some added call limits. So you can see that we went for a design which minimizes any errors that the developers can make, and protects the overall system.  The platform will work decently, regardless of network conditions or how the tag-managing MI commands are sent or abused. How to automatically recover from the ACTIVE/ACTIVE sharing tag state --------------------------------------------------------------------- Given that the "clusterer_shtag_set_active" [1] MI command issued to a node will force all other nodes to transition from ACTIVE -> BACKUP, you could enhance your system with a logic that sends this command to the opposite node any time a node's VIP performs the ACTIVE -> BACKUP transition.  This should fix the original problem, where both tags end up in the ACTIVE state, due to the link between nodes being temporarily down, without any of the OS'es necessarily being down. PS: we haven't implemented the above ^ ourselves yet, but it should work in theory :) let me know if it works for you if you do decide to plug this rare issue for your setup! Best regards, [1]: https://opensips.org/docs/modules/3.1.x/clusterer#mi_clusterer_shtag_set_active -- Liviu Chircu www.twitter.com/liviuchircu | www.opensips-solutions.com OpenSIPS Summit, Amsterdam, May 2020 www.opensips.org/events From kamlesh at worldphone.in Sat Apr 11 12:51:04 2020 From: kamlesh at worldphone.in (Kamlesh .) Date: Sat, 11 Apr 2020 18:21:04 +0530 Subject: [OpenSIPS-Users] Topology Hiding and NAT Message-ID: Hello Team, I have a problem with TOPOLOGY HIDING and NAT issues while using forwarding on busy/no-answer. I fix the NATed contact and then create dialog with topology hiding. The UPSTREAM (callee to caller) BYE relays to private IP instead of public. All the previous requests relay after fixing the contact. This only happens when a call is forwarded on BUSY/NO ANSWER Relay the packets to opensips itself, We just change the request number and add some required headers to detect the forwarded request. So the all dialog related things are the same for both the request one is initial request and forwarded request. Anyone can help me to short out this issue. I’m ready to provide other information if any to solve the issue. Regards, Kamlesh -- Disclaimer : This e-mail and any file transmitted with it are for exclusive use of the intended recipient(s) and may contain confidential and privileged information. If you are not the intended recipient,  please contact the sender by replying this e-mail and destroy all copies and original message. Any unauthorized review,use, disclosure, dissemination, forwarding, printing and copying of this email or any action taken in reliance of this e-mail is strictly prohibited and may be unlawful. -------------- next part -------------- An HTML attachment was scrubbed... URL: From alexei.vasilyev at gmail.com Sat Apr 11 13:28:55 2020 From: alexei.vasilyev at gmail.com (Alexey Vasilyev) Date: Sat, 11 Apr 2020 06:28:55 -0700 (MST) Subject: [OpenSIPS-Users] drouting probe_mode in active/passive cluster In-Reply-To: References: Message-ID: <1586611735171-0.post@n2.nabble.com> Hi Jeff. I made one solution for 2.4. You can cherry-pick https://github.com/OpenSIPS/opensips/commit/05ca54a37d82c605e2cd6d10e5a62fb4f7c35b78 And may be this: https://github.com/OpenSIPS/opensips/commit/94a3ede1e276984a91f93f6ece832d174b071ab8 There is documentation in commits. ----- --- Alexey Vasilyev -- Sent from: http://opensips-open-sip-server.1449251.n2.nabble.com/OpenSIPS-Users-f1449235.html From donat.zenichev at gmail.com Sat Apr 11 16:42:09 2020 From: donat.zenichev at gmail.com (Donat Zenichev) Date: Sat, 11 Apr 2020 19:42:09 +0300 Subject: [OpenSIPS-Users] A behavior of Clusterer module during networking issues In-Reply-To: <522532ff-c612-d09f-9877-163b9316977a@opensips.org> References: <522532ff-c612-d09f-9877-163b9316977a@opensips.org> Message-ID: Hello Liviu! And first of all thank you for your detailed explanation. Now I completely understand the approach was taken when developing this feature of Clusterer. And this looks logical to me. It gives less chances to make a human error. What I did for now, is a clustering super-structure (that works apart OpenSIPS), and here what happens when both nodes see each other again (when recovering from networking issues): Shared IP remains working on the Master side, and one-shot systemd service activates "clusterer_shtag_set_active" on the Master right away. Thus suppressing a Stand-by node to apply backup state. For now this schema works perfectly. Might be I will come up with more robust solution later.. In case this happens, I will share my experience. Have a nice day! On Sat, Apr 11, 2020 at 2:13 PM Liviu Chircu wrote: > On 09.04.2020 15:03, Donat Zenichev wrote: > > I have question and it's almost theoretic. > > The question relates to Clusterer module and its behavior. > > > > Will Clusterer module solve this contradiction on its own? > > And if so, to which side the precedence is given? > > > > The other way around could be to manually re-activate all services, > > when all the cluster resumes into normal working process (all nodes > > are present). > > Thus this gives us a warranty that shared tag will only be activated > > on one of the sides. > > Hi, Donat! > > A very good question and one that we had to answer ourselves when we > came up with the current design. To begin with, in your scenario, for > all OpenSIPS 2.4+ clusterer versions, after the link between the nodes > comes back online, you will have the following: > > * node A: ACTIVE state (holds the VIP), sharing tag state: ACTIVE (1) > * node B: BACKUP state, sharing tag state: ACTIVE (1) > > The main reason behind this inconsistent state is that we did not > provide an MI command to force a sharing tag to BACKUP (0), which could > be triggered on node B's transition from ACTIVE -> BACKUP once the link > is restored, so recovering from this state will not work automatically - > you have to provide handling for this scenario as well (see last > paragraph). > > Reasoning behind this design > ---------------------------- > > Ultimately, our priority was not to get into solving consensus problems, > Paxos algorithms, etc. What we wanted is a robust active/backup > solution which you could flip back and forth with ease, thus achieving > both High-Availability and easy maintenance. By not providing a > "set_sharing_tag vip 0" command, we _avoid_ the situation where, due to > a developer error, both tags end up being BACKUP (0)!! In such a > scenario: there will be no more CDRs and you will be able to run > infinite CPS/CC through that instance, since all call profile counters > are equal to 0. None of the instances take responsibility for any call > running through them, so a lot of data will be lost. > > On the flip side, in a scenario where both tags are bugged in the ACTIVE > (1) state, you would have: duplicated CDRs (along with maybe some DB > error logs due to conflicting unique keys) and possibly extra-counted > calls, leading to a reduction of the maximally supported CC/CPS. Assume > that the platform wasn't even at 50% of the max limits to begin with, > and the latter has 0 impact on the live system. Thinking about this, > this didn't sound that bad at all to us: no data loss, at the expense of > a few error logs and possibly some added call limits. > > So you can see that we went for a design which minimizes any errors that > the developers can make, and protects the overall system. The platform > will work decently, regardless of network conditions or how the > tag-managing MI commands are sent or abused. > > How to automatically recover from the ACTIVE/ACTIVE sharing tag state > --------------------------------------------------------------------- > > Given that the "clusterer_shtag_set_active" [1] MI command issued to a > node will force all other nodes to transition from ACTIVE -> BACKUP, you > could enhance your system with a logic that sends this command to the > opposite node any time a node's VIP performs the ACTIVE -> BACKUP > transition. This should fix the original problem, where both tags end > up in the ACTIVE state, due to the link between nodes being temporarily > down, without any of the OS'es necessarily being down. > > PS: we haven't implemented the above ^ ourselves yet, but it should work > in theory :) let me know if it works for you if you do decide to plug > this rare issue for your setup! > > Best regards, > > [1]: > > https://opensips.org/docs/modules/3.1.x/clusterer#mi_clusterer_shtag_set_active > > -- > Liviu Chircu > www.twitter.com/liviuchircu | www.opensips-solutions.com > > OpenSIPS Summit, Amsterdam, May 2020 > www.opensips.org/events > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -- Best regards, Donat Zenichev -------------- next part -------------- An HTML attachment was scrubbed... URL: From jeff at ugnd.org Sun Apr 12 00:37:17 2020 From: jeff at ugnd.org (Jeff Pyle) Date: Sat, 11 Apr 2020 20:37:17 -0400 Subject: [OpenSIPS-Users] drouting probe_mode in active/passive cluster In-Reply-To: <1586611735171-0.post@n2.nabble.com> References: <1586611735171-0.post@n2.nabble.com> Message-ID: Hi Alexey, I see the "dr_enable_probing" MI command; that's great! Is there an equivalent modparam I can add to default to 0, so that it will only probe when we know it has the IP (when enabled by MI)? It will be some time until I'm able to cherry-pick and test. The two proxies are currently running from the published APT repo and it will take a moment to convert them to source or construct a local repo. - Jeff On Sat, Apr 11, 2020 at 9:29 AM Alexey Vasilyev wrote: > Hi Jeff. > > I made one solution for 2.4. You can cherry-pick > > https://github.com/OpenSIPS/opensips/commit/05ca54a37d82c605e2cd6d10e5a62fb4f7c35b78 > > And may be this: > > https://github.com/OpenSIPS/opensips/commit/94a3ede1e276984a91f93f6ece832d174b071ab8 > > There is documentation in commits. > > > > ----- > --- > Alexey Vasilyev > -- > Sent from: > http://opensips-open-sip-server.1449251.n2.nabble.com/OpenSIPS-Users-f1449235.html > > _______________________________________________ > 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 stvicent at hotmail.com Sun Apr 12 19:36:59 2020 From: stvicent at hotmail.com (Mario San Vicente) Date: Sun, 12 Apr 2020 19:36:59 +0000 Subject: [OpenSIPS-Users] PRACK and reinvite_ping_interval Message-ID: Estimated team, I have been working in a configuration that supports a PRACK (using B2b topology hiding) and reinvite_ping_interval ( using create_dialog("rR") ). I was able to make it work, but not together. It seems that B2B topology hiding and create dialog can not work together.. Any idea how to have both working. Thank you in advance. Vicent -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Mon Apr 13 14:37:44 2020 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Mon, 13 Apr 2020 17:37:44 +0300 Subject: [OpenSIPS-Users] PRACK support / configuration In-Reply-To: References: Message-ID: <6de4650b-99c6-bda0-fdfd-0a1a99da0376@opensips.org> Hi Mario, First of all, you need to pick one b2b or dialog - you cannot use both in the same time for the same call. Secondly, please see the B2B tutorial (https://www.opensips.org/Documentation/Tutorials-B2BUA) - once you started the B2B session (by the b2b_init script function), the control over the call is passed to the b2b module, so no script returning. The b2b module will take care of your call from that point further, there is nothing for you to do. Best Regard, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com OpenSIPS Summit, Amsterdam, May 2020 https://www.opensips.org/events/Summit-2020Amsterdam/ On 4/9/20 8:11 AM, Mario San Vicente wrote: > Hello Everyone, > > I have been trying to enable PRACK support, and on the docs it says it > is supported on module b2b_entitier > > modules/b2b_entities/dlg.c > > /* PRACK handling // > // if the provisional reply contains a > > * Require: 100rel header -> send PRACK */ > > It also says that B2BA top hiding is a predefined service that works > without any scenario definition and for any type of dialog. > > So I am traying to enable top hiding to get the PRACK working. > just adding thhe line: > > b2b_init_request("top hiding"); > > But when i add this line, the call flow stops working. > > Any advice on how to configure this feature? > > NOTE: In my working configuration i have a dialog created, just to let > you know if that is supported at the same time.  top hiding and dialog. > > Thank you > > -- > Mario San Vicente > Cheers! > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Mon Apr 13 14:45:09 2020 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Mon, 13 Apr 2020 17:45:09 +0300 Subject: [OpenSIPS-Users] Topology Hiding and NAT In-Reply-To: References: Message-ID: <0effb9dc-f01e-7131-4ff7-140c54c83945@opensips.org> Hi Kamlesh, I would suspect you do something wrong in the failure_route, when handling the FWD cases. Maybe another change over the contact hdr in the INVITE ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com OpenSIPS Summit, Amsterdam, May 2020 https://www.opensips.org/events/Summit-2020Amsterdam/ On 4/11/20 3:51 PM, Kamlesh . wrote: > Hello Team, > > I have a problem with TOPOLOGY HIDING and NAT issues while using > forwarding on busy/no-answer. I fix the NATed contact and then create > dialog with topology hiding. The UPSTREAM (callee to caller) BYE > relays to private IP instead of public. All the previous requests > relay after fixing the contact. This only happens when a call is > forwarded on BUSY/NO ANSWER Relay the packetsto opensips itself, We > just change the request number and add some required headers to detect > the forwarded request. So the all dialog related things are the same > for both the request one is initial request and forwarded request. > Anyone can help me to short out this issue. I’m ready to provide other > information if any to solve the issue. > > Regards, > > Kamlesh > > > > Disclaimer : > > This e-mail and any file transmitted with it are for exclusive use of > the intended recipient(s) > and may contain confidential and privileged information. If you are > not the intended recipient, > please contact the sender by replying this e-mail and destroy all > copies and original message. > Any unauthorized review,use, disclosure, dissemination, forwarding, > printing and copying of this > email or any action taken in reliance of this e-mail is strictly > prohibited and may be unlawful. > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Mon Apr 13 14:51:55 2020 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Mon, 13 Apr 2020 17:51:55 +0300 Subject: [OpenSIPS-Users] drouting probe_mode in active/passive cluster In-Reply-To: References: <1586611735171-0.post@n2.nabble.com> Message-ID: Hi guys, actually 3.0 provides a better approach to the problem, by using the sharing tags provided by the clustering layer [1]. Only the OpenSIPS instance having the tag active will perform the pinging/probing and broadcast the status changes into the cluster. Unfortunately 2.4 does not have such a mechanism, the clustering support is more rudimentary there. [1] https://opensips.org/html/docs/modules/3.0.x/drouting.html#param_cluster_sharing_tag Regards Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com OpenSIPS Summit, Amsterdam, May 2020 https://www.opensips.org/events/Summit-2020Amsterdam/ On 4/12/20 3:37 AM, Jeff Pyle wrote: > Hi Alexey, > > I see the "dr_enable_probing" MI command; that's great!  Is there an > equivalent modparam I can add to default to 0, so that it will only > probe when we know it has the IP (when enabled by MI)? > > It will be some time until I'm able to cherry-pick and test.  The two > proxies are currently running from the published APT repo and it will > take a moment to convert them to source or construct a local repo. > > > - Jeff > > On Sat, Apr 11, 2020 at 9:29 AM Alexey Vasilyev > > wrote: > > Hi Jeff. > > I made one solution for 2.4. You can cherry-pick > https://github.com/OpenSIPS/opensips/commit/05ca54a37d82c605e2cd6d10e5a62fb4f7c35b78 > > And may be this: > https://github.com/OpenSIPS/opensips/commit/94a3ede1e276984a91f93f6ece832d174b071ab8 > > There is documentation in commits. > > > > ----- > --- > Alexey Vasilyev > -- > Sent from: > http://opensips-open-sip-server.1449251.n2.nabble.com/OpenSIPS-Users-f1449235.html > > _______________________________________________ > 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 vladp at opensips.org Mon Apr 13 15:01:14 2020 From: vladp at opensips.org (Vlad Patrascu) Date: Mon, 13 Apr 2020 18:01:14 +0300 Subject: [OpenSIPS-Users] Stir_shaken signature length In-Reply-To: References: Message-ID: <1d9c0e31-4708-ead6-e6ea-5152cba4e22e@opensips.org> Hi Alexandru, OpenSIPS is using the signature in DER encoded format (as it is directly generated by openssl) but indeed it is not the proper format as per RFC 7518. Thanks for the report, I am working on a fix. Regards, Vlad Patrascu On 10.04.2020 12:28, Alexandru Tripon wrote: > Hi, > > I tried to populate the Identity header with the stir_shaken module. > The header is populated but when I try to verify the signature using > an external tool it fails because of the length. > I have the folowing Identity generated by Opensips: > ` > eyJhbGciOiJFUzI1NiIsInBwdCI6InNoYWtlbiIsInR5cCI6InBhc3Nwb3J0IiwieDV1IjoiL2hvbWUvdHJpYWwvTHVjcnUvQ29kZS9zdGlyU2hha2VuL215cHVia2V5LnBlbSJ9.eyJhdHRlc3QiOiJBIiwiZGVzdCI6eyJ0biI6WyIxMDAyIl19LCJpYXQiOjE1ODY1MDMxODcsIm9yaWciOnsidG4iOiIxMDAxIn0sIm9yaWdpZCI6IjEyMzQ1NiJ9.MEYCIQCjIx6w8IeilqHq0jbc6uwIB9v1RDmecoep0gRJJC4EmQIhANH1MO9jwRtqH6jgFH12XqROFv-nUroEgzsRAaMJtAsR;info=\u003c/home/trial/Lucru/Code/stirShaken/mypubkey.pem\u003e;ppt=\"shaken\" > ` > the lenght of encoded signature(in base64) is 96 and in the decoded > one is 72. > In the RFC for ES256 > algorithm(https://tools.ietf.org/html/rfc7518#section-3.4) the length > of the decoded signature is 64. > Am I missing something here? > > Thanks, > Alexandru Tripon > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Mon Apr 13 15:10:21 2020 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Mon, 13 Apr 2020 18:10:21 +0300 Subject: [OpenSIPS-Users] PRACK and reinvite_ping_interval In-Reply-To: References: Message-ID: Hi Mario, yes, you cannot use both. The dialog module is expecting the call to be handled in proxy mode, while the B2B module expect the call to handled in ....back2back mode. So logically speaking you cannot have them both working together. Maybe you can open a feature request on OpenSIPS tracker for adding PRACK support in the dialog module. Best regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com OpenSIPS Summit, Amsterdam, May 2020 https://www.opensips.org/events/Summit-2020Amsterdam/ On 4/12/20 10:36 PM, Mario San Vicente wrote: > Estimated team, > > I have been working in a configuration that supports a PRACK (using > B2b topology hiding) and reinvite_ping_interval ( > using create_dialog("rR") ). > > I was able to make it work, but not together.  It seems that B2B > topology hiding and create dialog can not work together.. Any idea how > to have both working. > > Thank you in advance. > Vicent > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Mon Apr 13 15:11:04 2020 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Mon, 13 Apr 2020 18:11:04 +0300 Subject: [OpenSIPS-Users] callfwd avp with opensips-cli In-Reply-To: References: Message-ID: <10f821c5-73f7-41d9-04fb-50c5ebf0f95b@opensips.org> Hi Michael, No, it is not possible. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com OpenSIPS Summit, Amsterdam, May 2020 https://www.opensips.org/events/Summit-2020Amsterdam/ On 4/9/20 11:43 AM, Michael Vale via Users wrote: > hi. > > is it possible to set the avp for call forwarding with opensips-cli and > if so, how? > > regards, > > michael. > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users From alexei.vasilyev at gmail.com Mon Apr 13 15:11:14 2020 From: alexei.vasilyev at gmail.com (Alexey Vasilyev) Date: Mon, 13 Apr 2020 08:11:14 -0700 (MST) Subject: [OpenSIPS-Users] drouting probe_mode in active/passive cluster In-Reply-To: References: <1586611735171-0.post@n2.nabble.com> Message-ID: <1586790674754-0.post@n2.nabble.com> Hi Bogdan, Yes, of course, in 3.0 you have implemented full support for the sharing tags. But when these commits were made, the sharing tags were only in process of discussion. So it's absolutely temporarily solution only for 2.4 branch. And for Jeff, there is no modparam. Just call "dr_enable_probing 0" after starting opensips. ----- --- Alexey Vasilyev -- Sent from: http://opensips-open-sip-server.1449251.n2.nabble.com/OpenSIPS-Users-f1449235.html From venefax at gmail.com Mon Apr 13 15:13:12 2020 From: venefax at gmail.com (Saint Michael) Date: Mon, 13 Apr 2020 11:13:12 -0400 Subject: [OpenSIPS-Users] Stir_shaken signature length In-Reply-To: <1d9c0e31-4708-ead6-e6ea-5152cba4e22e@opensips.org> References: <1d9c0e31-4708-ead6-e6ea-5152cba4e22e@opensips.org> Message-ID: I am trying to do the same. The question I need to ask here is: how do you generate the signature from the certificate, the caller ID and the destination number? I have the API working in staging mode, but now I need to really sign a call and send it forward with Opensips 2.4.7 Federico On Mon, Apr 13, 2020 at 11:03 AM Vlad Patrascu wrote: > Hi Alexandru, > > OpenSIPS is using the signature in DER encoded format (as it is directly > generated by openssl) but indeed it is not the proper format as per RFC > 7518. Thanks for the report, I am working on a fix. > > Regards, > > Vlad Patrascu > On 10.04.2020 12:28, Alexandru Tripon wrote: > > Hi, > > I tried to populate the Identity header with the stir_shaken module. > The header is populated but when I try to verify the signature using an > external tool it fails because of the length. > I have the folowing Identity generated by Opensips: > ` > > eyJhbGciOiJFUzI1NiIsInBwdCI6InNoYWtlbiIsInR5cCI6InBhc3Nwb3J0IiwieDV1IjoiL2hvbWUvdHJpYWwvTHVjcnUvQ29kZS9zdGlyU2hha2VuL215cHVia2V5LnBlbSJ9.eyJhdHRlc3QiOiJBIiwiZGVzdCI6eyJ0biI6WyIxMDAyIl19LCJpYXQiOjE1ODY1MDMxODcsIm9yaWciOnsidG4iOiIxMDAxIn0sIm9yaWdpZCI6IjEyMzQ1NiJ9.MEYCIQCjIx6w8IeilqHq0jbc6uwIB9v1RDmecoep0gRJJC4EmQIhANH1MO9jwRtqH6jgFH12XqROFv-nUroEgzsRAaMJtAsR;info=\u003c/home/trial/Lucru/Code/stirShaken/mypubkey.pem\u003e;ppt=\"shaken\" > ` > the lenght of encoded signature(in base64) is 96 and in the decoded one is > 72. > In the RFC for ES256 algorithm( > https://tools.ietf.org/html/rfc7518#section-3.4) the length of the > decoded signature is 64. > Am I missing something here? > > Thanks, > Alexandru Tripon > > _______________________________________________ > Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Mon Apr 13 15:15:10 2020 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Mon, 13 Apr 2020 18:15:10 +0300 Subject: [OpenSIPS-Users] drouting probe_mode in active/passive cluster In-Reply-To: <1586790674754-0.post@n2.nabble.com> References: <1586611735171-0.post@n2.nabble.com> <1586790674754-0.post@n2.nabble.com> Message-ID: You are 100% right Alexey, I was just to lure Jeff to go for 3.0 ;) Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com OpenSIPS Summit, Amsterdam, May 2020 https://www.opensips.org/events/Summit-2020Amsterdam/ On 4/13/20 6:11 PM, Alexey Vasilyev wrote: > Hi Bogdan, > > Yes, of course, in 3.0 you have implemented full support for the sharing > tags. But when these commits were made, the sharing tags were only in > process of discussion. So it's absolutely temporarily solution only for 2.4 > branch. > > And for Jeff, there is no modparam. Just call "dr_enable_probing 0" after > starting opensips. > > > > ----- > --- > Alexey Vasilyev > -- > Sent from: http://opensips-open-sip-server.1449251.n2.nabble.com/OpenSIPS-Users-f1449235.html > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users From vladp at opensips.org Mon Apr 13 15:32:31 2020 From: vladp at opensips.org (Vlad Patrascu) Date: Mon, 13 Apr 2020 18:32:31 +0300 Subject: [OpenSIPS-Users] Stir_shaken signature length In-Reply-To: References: <1d9c0e31-4708-ead6-e6ea-5152cba4e22e@opensips.org> Message-ID: <0f6097c5-245e-1f33-b1c3-ab3228195708@opensips.org> Hi Frederico, I'm not really sure I understand your question of "how" to generate the signature. Are you refering to how the scripting should look like or something else ? But anyway, it is not possible with OpenSIPS 2.4.7 as the stir_shaken module is available starting with OpenSIPS 3.1. Regards, Vlad Patrascu On 13.04.2020 18:13, Saint Michael wrote: > I am trying to do the same. The question I need to ask here is: how do > you generate the signature from the certificate, the caller ID and the > destination number? > I have the API working in staging mode, but now I need to really sign > a call and send it forward with Opensips 2.4.7 > > Federico > > On Mon, Apr 13, 2020 at 11:03 AM Vlad Patrascu > wrote: > > Hi Alexandru, > > OpenSIPS is using the signature in DER encoded format (as it is > directly generated by openssl) but indeed it is not the proper > format as per RFC 7518. Thanks for the report, I am working on a fix. > > Regards, > > Vlad Patrascu > > On 10.04.2020 12:28, Alexandru Tripon wrote: >> Hi, >> >> I tried to populate the Identity header with the stir_shaken module. >> The header is populated but when I try to verify the signature >> using an external tool it fails because of the length. >> I have the folowing Identity generated by Opensips: >> ` >> eyJhbGciOiJFUzI1NiIsInBwdCI6InNoYWtlbiIsInR5cCI6InBhc3Nwb3J0IiwieDV1IjoiL2hvbWUvdHJpYWwvTHVjcnUvQ29kZS9zdGlyU2hha2VuL215cHVia2V5LnBlbSJ9.eyJhdHRlc3QiOiJBIiwiZGVzdCI6eyJ0biI6WyIxMDAyIl19LCJpYXQiOjE1ODY1MDMxODcsIm9yaWciOnsidG4iOiIxMDAxIn0sIm9yaWdpZCI6IjEyMzQ1NiJ9.MEYCIQCjIx6w8IeilqHq0jbc6uwIB9v1RDmecoep0gRJJC4EmQIhANH1MO9jwRtqH6jgFH12XqROFv-nUroEgzsRAaMJtAsR;info=\u003c/home/trial/Lucru/Code/stirShaken/mypubkey.pem\u003e;ppt=\"shaken\" >> ` >> the lenght of encoded signature(in base64) is 96 and in the >> decoded one is 72. >> In the RFC for ES256 >> algorithm(https://tools.ietf.org/html/rfc7518#section-3.4) the >> length of the decoded signature is 64. >> Am I missing something here? >> >> Thanks, >> Alexandru Tripon >> >> _______________________________________________ >> 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 venefax at gmail.com Mon Apr 13 15:58:15 2020 From: venefax at gmail.com (Saint Michael) Date: Mon, 13 Apr 2020 11:58:15 -0400 Subject: [OpenSIPS-Users] Stir_shaken signature length In-Reply-To: <0f6097c5-245e-1f33-b1c3-ab3228195708@opensips.org> References: <1d9c0e31-4708-ead6-e6ea-5152cba4e22e@opensips.org> <0f6097c5-245e-1f33-b1c3-ab3228195708@opensips.org> Message-ID: I see, so I need to update my Opensips to 3.1, and then how does it work? the module grabs my certificate and generates the signature? Is there a command line tool that can do that meanwhile? We can always add the signature like any other header. Can somebody paste a sample code here so I my try? On Mon, Apr 13, 2020 at 11:34 AM Vlad Patrascu wrote: > Hi Frederico, > > I'm not really sure I understand your question of "how" to generate the > signature. Are you refering to how the scripting should look like or > something else ? But anyway, it is not possible with OpenSIPS 2.4.7 as the > stir_shaken module is available starting with OpenSIPS 3.1. > > Regards, > > Vlad Patrascu > On 13.04.2020 18:13, Saint Michael wrote: > > I am trying to do the same. The question I need to ask here is: how do you > generate the signature from the certificate, the caller ID and the > destination number? > I have the API working in staging mode, but now I need to really sign a > call and send it forward with Opensips 2.4.7 > > Federico > > On Mon, Apr 13, 2020 at 11:03 AM Vlad Patrascu wrote: > >> Hi Alexandru, >> >> OpenSIPS is using the signature in DER encoded format (as it is directly >> generated by openssl) but indeed it is not the proper format as per RFC >> 7518. Thanks for the report, I am working on a fix. >> >> Regards, >> >> Vlad Patrascu >> On 10.04.2020 12:28, Alexandru Tripon wrote: >> >> Hi, >> >> I tried to populate the Identity header with the stir_shaken module. >> The header is populated but when I try to verify the signature using an >> external tool it fails because of the length. >> I have the folowing Identity generated by Opensips: >> ` >> >> eyJhbGciOiJFUzI1NiIsInBwdCI6InNoYWtlbiIsInR5cCI6InBhc3Nwb3J0IiwieDV1IjoiL2hvbWUvdHJpYWwvTHVjcnUvQ29kZS9zdGlyU2hha2VuL215cHVia2V5LnBlbSJ9.eyJhdHRlc3QiOiJBIiwiZGVzdCI6eyJ0biI6WyIxMDAyIl19LCJpYXQiOjE1ODY1MDMxODcsIm9yaWciOnsidG4iOiIxMDAxIn0sIm9yaWdpZCI6IjEyMzQ1NiJ9.MEYCIQCjIx6w8IeilqHq0jbc6uwIB9v1RDmecoep0gRJJC4EmQIhANH1MO9jwRtqH6jgFH12XqROFv-nUroEgzsRAaMJtAsR;info=\u003c/home/trial/Lucru/Code/stirShaken/mypubkey.pem\u003e;ppt=\"shaken\" >> ` >> the lenght of encoded signature(in base64) is 96 and in the decoded one >> is 72. >> In the RFC for ES256 algorithm( >> https://tools.ietf.org/html/rfc7518#section-3.4) the length of the >> decoded signature is 64. >> Am I missing something here? >> >> Thanks, >> Alexandru Tripon >> >> _______________________________________________ >> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> > > _______________________________________________ > Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Mon Apr 13 16:37:39 2020 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Mon, 13 Apr 2020 19:37:39 +0300 Subject: [OpenSIPS-Users] Stir_shaken signature length In-Reply-To: References: <1d9c0e31-4708-ead6-e6ea-5152cba4e22e@opensips.org> <0f6097c5-245e-1f33-b1c3-ab3228195708@opensips.org> Message-ID: Maybe you should first take a look at https://blog.opensips.org/2020/01/23/shaken-not-stirred/ and https://opensips.org/docs/modules/3.1.x/stir_shaken.html Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com OpenSIPS Summit, Amsterdam, May 2020 https://www.opensips.org/events/Summit-2020Amsterdam/ On 4/13/20 6:58 PM, Saint Michael wrote: > I see, so I need to update my Opensips to 3.1, and then how does it > work? the module grabs my certificate and generates the signature? > Is there a command line tool that can do that meanwhile? We can always > add the signature like any other header. > Can somebody paste a sample code here so I my try? > > > > > On Mon, Apr 13, 2020 at 11:34 AM Vlad Patrascu > wrote: > > Hi Frederico, > > I'm not really sure I understand your question of "how" to > generate the signature. Are you refering to how the scripting > should look like or something else ? But anyway, it is not > possible with OpenSIPS 2.4.7 as the stir_shaken module is > available starting with OpenSIPS 3.1. > > Regards, > > Vlad Patrascu > > On 13.04.2020 18:13, Saint Michael wrote: >> I am trying to do the same. The question I need to ask here is: >> how do you generate the signature from the certificate, the >> caller ID and the destination number? >> I have the API working in staging mode, but now I need to really >> sign a call and send it forward with Opensips 2.4.7 >> >> Federico >> >> On Mon, Apr 13, 2020 at 11:03 AM Vlad Patrascu >> > wrote: >> >> Hi Alexandru, >> >> OpenSIPS is using the signature in DER encoded format (as it >> is directly generated by openssl) but indeed it is not the >> proper format as per RFC 7518. Thanks for the report, I am >> working on a fix. >> >> Regards, >> >> Vlad Patrascu >> >> On 10.04.2020 12:28, Alexandru Tripon wrote: >>> Hi, >>> >>> I tried to populate the Identity header with the stir_shaken >>> module. >>> The header is populated but when I try to verify the >>> signature using an external tool it fails because of the length. >>> I have the folowing Identity generated by Opensips: >>> ` >>> eyJhbGciOiJFUzI1NiIsInBwdCI6InNoYWtlbiIsInR5cCI6InBhc3Nwb3J0IiwieDV1IjoiL2hvbWUvdHJpYWwvTHVjcnUvQ29kZS9zdGlyU2hha2VuL215cHVia2V5LnBlbSJ9.eyJhdHRlc3QiOiJBIiwiZGVzdCI6eyJ0biI6WyIxMDAyIl19LCJpYXQiOjE1ODY1MDMxODcsIm9yaWciOnsidG4iOiIxMDAxIn0sIm9yaWdpZCI6IjEyMzQ1NiJ9.MEYCIQCjIx6w8IeilqHq0jbc6uwIB9v1RDmecoep0gRJJC4EmQIhANH1MO9jwRtqH6jgFH12XqROFv-nUroEgzsRAaMJtAsR;info=\u003c/home/trial/Lucru/Code/stirShaken/mypubkey.pem\u003e;ppt=\"shaken\" >>> ` >>> the lenght of encoded signature(in base64) is 96 and in the >>> decoded one is 72. >>> In the RFC for ES256 >>> algorithm(https://tools.ietf.org/html/rfc7518#section-3.4) >>> the length of the decoded signature is 64. >>> Am I missing something here? >>> >>> Thanks, >>> Alexandru Tripon >>> >>> _______________________________________________ >>> Users mailing list >>> Users at lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From calvin.ellison at voxox.com Mon Apr 13 16:47:55 2020 From: calvin.ellison at voxox.com (Calvin Ellison) Date: Mon, 13 Apr 2020 09:47:55 -0700 Subject: [OpenSIPS-Users] Stir_shaken signature length In-Reply-To: References: <1d9c0e31-4708-ead6-e6ea-5152cba4e22e@opensips.org> Message-ID: On Mon, Apr 13, 2020 at 8:13 AM Saint Michael wrote: > I am trying to do the same. The question I need to ask here is: how do you generate the signature from the certificate, the caller ID and the destination number? > I have the API working in staging mode, but now I need to really sign a call and send it forward with Opensips 2.4.7 For 2.4.x you could try using the exec module with a CLI tool, or an external HTTP API. A C Library with CLI and HTTP server was posted to the VoceOps mailing list in January: https://puck.nether.net/pipermail/voiceops/2020-January/008264.html https://github.com/asipto/secsipidx From brett at nemeroff.com Mon Apr 13 17:16:22 2020 From: brett at nemeroff.com (Brett Nemeroff) Date: Mon, 13 Apr 2020 12:16:22 -0500 Subject: [OpenSIPS-Users] Stir_shaken signature length In-Reply-To: References: <1d9c0e31-4708-ead6-e6ea-5152cba4e22e@opensips.org> Message-ID: While this may work, I'd caution you against using the exec module for anything performant. It is a heavy beast. Or rather, your spawned shell is. Beware. On Mon, Apr 13, 2020 at 11:50 AM Calvin Ellison wrote: > On Mon, Apr 13, 2020 at 8:13 AM Saint Michael wrote: > > I am trying to do the same. The question I need to ask here is: how do > you generate the signature from the certificate, the caller ID and the > destination number? > > I have the API working in staging mode, but now I need to really sign a > call and send it forward with Opensips 2.4.7 > > For 2.4.x you could try using the exec module with a CLI tool, or an > external HTTP API. A C Library with CLI and HTTP server was posted to > the VoceOps mailing list in January: > > https://puck.nether.net/pipermail/voiceops/2020-January/008264.html > > https://github.com/asipto/secsipidx > > _______________________________________________ > 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 Mon Apr 13 17:59:23 2020 From: venefax at gmail.com (Saint Michael) Date: Mon, 13 Apr 2020 13:59:23 -0400 Subject: [OpenSIPS-Users] Stir_shaken signature length In-Reply-To: References: <1d9c0e31-4708-ead6-e6ea-5152cba4e22e@opensips.org> Message-ID: Hi Brett. I am going with the 3.1 version. Exec is unusable in production. On Mon, Apr 13, 2020 at 1:19 PM Brett Nemeroff wrote: > While this may work, I'd caution you against using the exec module for > anything performant. It is a heavy beast. Or rather, your spawned shell is. > Beware. > > On Mon, Apr 13, 2020 at 11:50 AM Calvin Ellison > wrote: > >> On Mon, Apr 13, 2020 at 8:13 AM Saint Michael wrote: >> > I am trying to do the same. The question I need to ask here is: how do >> you generate the signature from the certificate, the caller ID and the >> destination number? >> > I have the API working in staging mode, but now I need to really sign a >> call and send it forward with Opensips 2.4.7 >> >> For 2.4.x you could try using the exec module with a CLI tool, or an >> external HTTP API. A C Library with CLI and HTTP server was posted to >> the VoceOps mailing list in January: >> >> https://puck.nether.net/pipermail/voiceops/2020-January/008264.html >> >> https://github.com/asipto/secsipidx >> >> _______________________________________________ >> 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 stvicent at hotmail.com Mon Apr 13 20:21:10 2020 From: stvicent at hotmail.com (Mario San Vicente) Date: Mon, 13 Apr 2020 20:21:10 +0000 Subject: [OpenSIPS-Users] PRACK support / configuration In-Reply-To: <6de4650b-99c6-bda0-fdfd-0a1a99da0376@opensips.org> References: <6de4650b-99c6-bda0-fdfd-0a1a99da0376@opensips.org> Message-ID: Thank you Bogdan Mario sv El 13 abr 2020, a la(s) 9:40, Bogdan-Andrei Iancu escribió:  Hi Mario, First of all, you need to pick one b2b or dialog - you cannot use both in the same time for the same call. Secondly, please see the B2B tutorial (https://www.opensips.org/Documentation/Tutorials-B2BUA) - once you started the B2B session (by the b2b_init script function), the control over the call is passed to the b2b module, so no script returning. The b2b module will take care of your call from that point further, there is nothing for you to do. Best Regard, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com OpenSIPS Summit, Amsterdam, May 2020 https://www.opensips.org/events/Summit-2020Amsterdam/ On 4/9/20 8:11 AM, Mario San Vicente wrote: Hello Everyone, I have been trying to enable PRACK support, and on the docs it says it is supported on module b2b_entitier modules/b2b_entities/dlg.c /* PRACK handling / / if the provisional reply contains a * Require: 100rel header -> send PRACK */ It also says that B2BA top hiding is a predefined service that works without any scenario definition and for any type of dialog. So I am traying to enable top hiding to get the PRACK working. just adding thhe line: b2b_init_request("top hiding"); But when i add this line, the call flow stops working. Any advice on how to configure this feature? NOTE: In my working configuration i have a dialog created, just to let you know if that is supported at the same time. top hiding and dialog. Thank you -- Mario San Vicente Cheers! _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From stvicent at hotmail.com Tue Apr 14 01:19:08 2020 From: stvicent at hotmail.com (Mario San Vicente) Date: Tue, 14 Apr 2020 01:19:08 +0000 Subject: [OpenSIPS-Users] PRACK and reinvite_ping_interval In-Reply-To: References: , Message-ID: Thank you Bogdan, Ok i will open the feature request. Mario sv El 13 abr 2020, a la(s) 10:12, Bogdan-Andrei Iancu escribió:  Hi Mario, yes, you cannot use both. The dialog module is expecting the call to be handled in proxy mode, while the B2B module expect the call to handled in ....back2back mode. So logically speaking you cannot have them both working together. Maybe you can open a feature request on OpenSIPS tracker for adding PRACK support in the dialog module. Best regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com OpenSIPS Summit, Amsterdam, May 2020 https://www.opensips.org/events/Summit-2020Amsterdam/ On 4/12/20 10:36 PM, Mario San Vicente wrote: Estimated team, I have been working in a configuration that supports a PRACK (using B2b topology hiding) and reinvite_ping_interval ( using create_dialog("rR") ). I was able to make it work, but not together. It seems that B2B topology hiding and create dialog can not work together.. Any idea how to have both working. Thank you in advance. Vicent _______________________________________________ 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 Apr 14 07:18:42 2020 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Tue, 14 Apr 2020 10:18:42 +0300 Subject: [OpenSIPS-Users] PRACK and reinvite_ping_interval In-Reply-To: References: Message-ID: Got it https://github.com/OpenSIPS/opensips/issues/2076 Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com OpenSIPS Summit, https://www.opensips.org/events/Summit-2020Amsterdam/ On 4/14/20 4:19 AM, Mario San Vicente wrote: > Thank you Bogdan, > > Ok i will open the feature request. > > Mario sv > >> El 13 abr 2020, a la(s) 10:12, Bogdan-Andrei Iancu >> escribió: >> >>  Hi Mario, >> >> yes, you cannot use both. The dialog module is expecting the call to >> be handled in proxy mode, while the B2B module expect the call to >> handled in ....back2back mode. So logically speaking you cannot have >> them both working together. >> >> Maybe you can open a feature request on OpenSIPS tracker for adding >> PRACK support in the dialog module. >> >> Best regards, >> Bogdan-Andrei Iancu >> >> OpenSIPS Founder and Developer >> https://www.opensips-solutions.com >> OpenSIPS Summit, Amsterdam, May 2020 >> https://www.opensips.org/events/Summit-2020Amsterdam/ >> >> On 4/12/20 10:36 PM, Mario San Vicente wrote: >>> Estimated team, >>> >>> I have been working in a configuration that supports a PRACK (using >>> B2b topology hiding) and reinvite_ping_interval ( >>> using create_dialog("rR") ). >>> >>> I was able to make it work, but not together.  It seems that B2B >>> topology hiding and create dialog can not work together.. Any idea >>> how to have both working. >>> >>> Thank you in advance. >>> Vicent >>> >>> _______________________________________________ >>> 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 babak.freeswitch at gmail.com Tue Apr 14 08:16:40 2020 From: babak.freeswitch at gmail.com (Babak Yakhchali) Date: Tue, 14 Apr 2020 12:46:40 +0430 Subject: [OpenSIPS-Users] Forking call to more than one AOR Message-ID: Hi I'm trying to implement this scenario: each user has two AORs. one for mobile and one for a userId (random string) at any time user may be registered on one, both or none of these AORs when a call comes to one of these AORs the other one must be called too (if there is a registered contact for it) reading the documents I think I need to use lookup("location","r"). but I don't know how should I add branches before calling it for the two AORs thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: From miha at softnet.si Tue Apr 14 08:57:02 2020 From: miha at softnet.si (Miha) Date: Tue, 14 Apr 2020 10:57:02 +0200 Subject: [OpenSIPS-Users] Alias domain / dns srv Message-ID: <66e399c9-fda5-ba7a-b163-d959d5149df0@softnet.si> Hello we have dns srv record for failover. In dns srv we have two record. So, one version of our devices does not support dns srv records. Is it possible to register device directly to one A record which is wirtten in DNS SRV record and then use ALIAS in opensips to right domain? DNS SRV. sip.test.com  (proxy1.test.com, proxy2.test.com) Devices that do not support will register to proxy1.test.com (opensips will have alias which will point to sip.test.com)? thank you miha -------------- next part -------------- An HTML attachment was scrubbed... URL: From babak.freeswitch at gmail.com Tue Apr 14 13:25:24 2020 From: babak.freeswitch at gmail.com (Babak Yakhchali) Date: Tue, 14 Apr 2020 17:55:24 +0430 Subject: [OpenSIPS-Users] Forking call to more than one AOR In-Reply-To: References: Message-ID: For anyone interested: I used append_branch() two times to create 2 branches. then using $(branch(uri)[1]) = $avp(other_aor); I set the second branch to go for other AOR, now calling lookup() like this: lookup("location","r") will create all needed branches based on the 2 already created branches and removes them from destination set. Till now invites are generated for different contacts, BUT there is a problem with clients registered with second contact, because To header is based on the first AOR in ruri and they will not accept the invite message, to solve this I used a branch route and changed the To header like this: branch_route[per_branch_ops] { if($tU != $rU){ uac_replace_to("","$avp(other_aor)"); } } On Tue, Apr 14, 2020 at 12:46 PM Babak Yakhchali wrote: > Hi > I'm trying to implement this scenario: > each user has two AORs. one for mobile and one for a userId (random string) > at any time user may be registered on one, both or none of these AORs > when a call comes to one of these AORs the other one must be called too > (if there is a registered contact for it) > > reading the documents I think I need to use lookup("location","r"). but I > don't know how should I add branches before calling it for the two AORs > > thanks > -------------- next part -------------- An HTML attachment was scrubbed... URL: From podguiko at mail.ru Tue Apr 14 13:29:01 2020 From: podguiko at mail.ru (=?UTF-8?B?T2xlZyBQb2RndXlrbw==?=) Date: Tue, 14 Apr 2020 16:29:01 +0300 Subject: [OpenSIPS-Users] =?utf-8?q?_choice_of_AS_architecture?= Message-ID: <1586870941.864026772@f395.i.mail.ru> Hi Bogdan and all your team!   We have been working with opensips for quite some time and we like it. We keep the load on the order of 300-400 CAPS, and for example, the SIP-I module is perhaps the only solution that allows you to fully work with ISUP. Our programmers prepared several PRs based on the results of work with SIP-I and SCTP modules. We are faced with the task of creating a classic bundle of cscf + AS server, in the future architecture opensips will play the role of cscf, and as AS we look at the module SEAS + external server. We are all moving towards IMS.   What do you think, how justified can such a choice be? Perhaps the opensips team already has its own new vision for this? What would you choose as AS?     -- Oleg Podguyko -------------- next part -------------- An HTML attachment was scrubbed... URL: From john.quick at smartvox.co.uk Tue Apr 14 15:24:34 2020 From: john.quick at smartvox.co.uk (John Quick) Date: Tue, 14 Apr 2020 16:24:34 +0100 Subject: [OpenSIPS-Users] Problem with add_rr_param() in v2.4.7 Message-ID: <001b01d61270$cd7d8200$68788600$@smartvox.co.uk> I've tried running the same opensips.cfg script in v2.4.6 and then 2.4.7 When it is changed to 2.4.7, the function add_rr_param() does nothing. When run under 2.4.6 it updates the Record-Route header as you would expect. John Quick Smartvox Limited Web: www.smartvox.co.uk From alain.bieuzent at free.fr Tue Apr 14 15:50:31 2020 From: alain.bieuzent at free.fr (Alain Bieuzent) Date: Tue, 14 Apr 2020 17:50:31 +0200 Subject: [OpenSIPS-Users] Forking call to more than one AOR In-Reply-To: References: Message-ID: <927C0970-FC87-4821-B498-2A46B35562D7@free.fr> thanks for sharing Alain De : Users au nom de Babak Yakhchali Répondre à : OpenSIPS users mailling list Date : mardi 14 avril 2020 à 15:27 À : Objet : Re: [OpenSIPS-Users] Forking call to more than one AOR For anyone interested: I used append_branch() two times to create 2 branches. then using $(branch(uri)[1]) = $avp(other_aor); I set the second branch to go for other AOR, now calling lookup() like this: lookup("location","r") will create all needed branches based on the 2 already created branches and removes them from destination set. Till now invites are generated for different contacts, BUT there is a problem with clients registered with second contact, because To header is based on the first AOR in ruri and they will not accept the invite message, to solve this I used a branch route and changed the To header like this: branch_route[per_branch_ops] { if($tU != $rU){ uac_replace_to("","$avp(other_aor)"); } } On Tue, Apr 14, 2020 at 12:46 PM Babak Yakhchali wrote: Hi I'm trying to implement this scenario: each user has two AORs. one for mobile and one for a userId (random string) at any time user may be registered on one, both or none of these AORs when a call comes to one of these AORs the other one must be called too (if there is a registered contact for it) reading the documents I think I need to use lookup("location","r"). but I don't know how should I add branches before calling it for the two AORs thanks _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From sanderson at ArnoldMagnetics.com Tue Apr 14 16:41:10 2020 From: sanderson at ArnoldMagnetics.com (Samuel Anderson) Date: Tue, 14 Apr 2020 16:41:10 +0000 Subject: [OpenSIPS-Users] Local Route Socket Message-ID: Hi All, I know you're unable to do routing or signaling in the local route, so I'm hoping someone can help me think of a workaround. My OpenSIPS proxy has 2 IP addresses on the LAN and 1 on the WAN. When a SIP trunk sends an invite to my proxy, it forwards that invite out a specific LAN IP address, so Asterisk recognizes it as from the PSTN. When an external UAC sends a registration request to the proxy, it's forwarded to Asterisk from the other LAN IP address, so Asterisk recognizes it as a UAC to authenticate. I use the mid_registrar module in OpenSIPS to save the location of the UAC. The problem is if the UAC becomes unresponsive and the max_pings_lost threshold of the nathelper module is reached, OpenSIPS sends a registration with an expiration of 0 to Asterisk. However, it's sent with the wrong source IP address, and Asterisk does not recognize it as an AOR to update. I've tried modifying the local_route to change the $fs variable, but this does not work. local_route { if (is_method("REGISTER")) { #Asterisk is expecting Registrations from UAC phones to come from 10.x.x.x $fs = "udp:10.x.x.x:5060"; } } Thank you, -Sam This message (including any attachments) is intended only for the use of the individual or entity to which it is addressed and may contain information that is non-public, proprietary, privileged, confidential, and exempt from disclosure under applicable law or may constitute as attorney work product. If you are not the intended recipient, you are hereby notified that any use, dissemination, distribution, or copying of this communication is strictly prohibited. If you have received this communication in error, notify us immediately by telephone and (i) destroy this message if a facsimile or (ii) delete this message immediately if this is an electronic communication. Thank you. -------------- next part -------------- An HTML attachment was scrubbed... URL: From vladp at opensips.org Tue Apr 14 20:26:57 2020 From: vladp at opensips.org (Vlad Patrascu) Date: Tue, 14 Apr 2020 23:26:57 +0300 Subject: [OpenSIPS-Users] Stir_shaken signature length In-Reply-To: References: Message-ID: <64e46d29-29a7-2277-5637-addfe5b6ecf8@opensips.org> Hi, Please check the fix on the latest master branch on github and let me know if everything is ok. Regards, Vlad Patrascu On 10.04.2020 12:28, Alexandru Tripon wrote: > Hi, > > I tried to populate the Identity header with the stir_shaken module. > The header is populated but when I try to verify the signature using > an external tool it fails because of the length. > I have the folowing Identity generated by Opensips: > ` > eyJhbGciOiJFUzI1NiIsInBwdCI6InNoYWtlbiIsInR5cCI6InBhc3Nwb3J0IiwieDV1IjoiL2hvbWUvdHJpYWwvTHVjcnUvQ29kZS9zdGlyU2hha2VuL215cHVia2V5LnBlbSJ9.eyJhdHRlc3QiOiJBIiwiZGVzdCI6eyJ0biI6WyIxMDAyIl19LCJpYXQiOjE1ODY1MDMxODcsIm9yaWciOnsidG4iOiIxMDAxIn0sIm9yaWdpZCI6IjEyMzQ1NiJ9.MEYCIQCjIx6w8IeilqHq0jbc6uwIB9v1RDmecoep0gRJJC4EmQIhANH1MO9jwRtqH6jgFH12XqROFv-nUroEgzsRAaMJtAsR;info=\u003c/home/trial/Lucru/Code/stirShaken/mypubkey.pem\u003e;ppt=\"shaken\" > ` > the lenght of encoded signature(in base64) is 96 and in the decoded > one is 72. > In the RFC for ES256 > algorithm(https://tools.ietf.org/html/rfc7518#section-3.4) the length > of the decoded signature is 64. > Am I missing something here? > > Thanks, > Alexandru Tripon > > _______________________________________________ > 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 Apr 14 21:40:28 2020 From: venefax at gmail.com (Saint Michael) Date: Tue, 14 Apr 2020 17:40:28 -0400 Subject: [OpenSIPS-Users] memory In-Reply-To: <64e46d29-29a7-2277-5637-addfe5b6ecf8@opensips.org> References: <64e46d29-29a7-2277-5637-addfe5b6ecf8@opensips.org> Message-ID: > > I see hundreds of messages like this how I raise the memory? My box has unlimited memory, but opensips does not use it. I changed the parameters inside the file opensips.service, but it did not changed the memory allocated. Apr 14 21:06:13 [117509] ERROR:core:fm_malloc: not enough free shm memory (17320 bytes left, need 6720), please increase the "-m" command line parameter! Apr 14 21:06:13 [117509] ERROR:tm:new_t: out of mem Apr 14 21:06:13 [117509] ERROR:tm:t_newtran: new_t failed Apr 14 21:06:13 [117449] CRITICAL:dialog:w_set_dlg_profile: BUG - setting profile from script, but no dialog found Apr 14 21:06:13 [117518] Vendor IP 216.158.87.250 has a channel limit of 15000 - currently 666 channels Apr 14 21:06:13 [117449] CRITICAL:dialog:w_set_dlg_profile: BUG - setting profile from script, but no dialog found Apr 14 21:06:13 [117483] CRITICAL:dialog:w_set_dlg_profile: BUG - setting profile from script, but no dialog found Apr 14 21:06:13 [117483] Vendor IP 216.158.87.250 has a channel limit of 15000 - currently 666 channels Apr 14 21:06:13 [117507] ERROR:core:fm_malloc: not enough free shm memory (36376 bytes left, need 6720), please increase the "-m" command line parameter! Apr 14 21:06:13 [117507] ERROR:tm:new_t: out of mem -------------- next part -------------- An HTML attachment was scrubbed... URL: From venefax at gmail.com Tue Apr 14 21:59:45 2020 From: venefax at gmail.com (Saint Michael) Date: Tue, 14 Apr 2020 17:59:45 -0400 Subject: [OpenSIPS-Users] Messages Message-ID: I see thousands of the messages below. If they are not important, how do I hide them ARNING:dialog:dlg_onroute: tight matching failed for ACK with callid='5063212c34ac27e23d25ea143ca8bf4f at 64.140.166.100:5060'/52, ftag='as480e6ada'/10, ttag='8e2e-a7407c352db2ab880c67b8cc744fbbfe'/37 and direction=1 Apr 14 21:56:06 brian opensips[22702]: Apr 14 21:56:06 [22787] WARNING:dialog:dlg_onroute: dialog identification elements are callid='5063212c34ac27e23d25ea143ca8bf4f at 64.140.166.100:5060'/52, caller tag='as480e6ada'/10, callee tag='KSQ5DeD148vBD'/13 -------------- next part -------------- An HTML attachment was scrubbed... URL: From david.villasmil.work at gmail.com Tue Apr 14 22:57:19 2020 From: david.villasmil.work at gmail.com (David Villasmil) Date: Tue, 14 Apr 2020 23:57:19 +0100 Subject: [OpenSIPS-Users] Messages In-Reply-To: References: Message-ID: Maybe lower the debug? On Tue, 14 Apr 2020 at 23:00, Saint Michael wrote: > I see thousands of the messages below. If they are not important, how do I > hide them > ARNING:dialog:dlg_onroute: tight matching failed for ACK with > callid='5063212c34ac27e23d25ea143ca8bf4f at 64.140.166.100:5060'/52, > ftag='as480e6ada'/10, ttag='8e2e-a7407c352db2ab880c67b8cc744fbbfe'/37 and > direction=1 > Apr 14 21:56:06 brian opensips[22702]: Apr 14 21:56:06 [22787] > WARNING:dialog:dlg_onroute: dialog identification elements are > callid='5063212c34ac27e23d25ea143ca8bf4f at 64.140.166.100:5060'/52, caller > tag='as480e6ada'/10, callee tag='KSQ5DeD148vBD'/13 > _______________________________________________ > 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 venefax at gmail.com Tue Apr 14 23:09:02 2020 From: venefax at gmail.com (Saint Michael) Date: Tue, 14 Apr 2020 19:09:02 -0400 Subject: [OpenSIPS-Users] Messages In-Reply-To: References: Message-ID: what is the lowest debug level that is safe to use? On Tue, Apr 14, 2020 at 6:59 PM David Villasmil < david.villasmil.work at gmail.com> wrote: > Maybe lower the debug? > > On Tue, 14 Apr 2020 at 23:00, Saint Michael wrote: > >> I see thousands of the messages below. If they are not important, how do >> I hide them >> ARNING:dialog:dlg_onroute: tight matching failed for ACK with >> callid='5063212c34ac27e23d25ea143ca8bf4f at 64.140.166.100:5060'/52, >> ftag='as480e6ada'/10, ttag='8e2e-a7407c352db2ab880c67b8cc744fbbfe'/37 and >> direction=1 >> Apr 14 21:56:06 brian opensips[22702]: Apr 14 21:56:06 [22787] >> WARNING:dialog:dlg_onroute: dialog identification elements are >> callid='5063212c34ac27e23d25ea143ca8bf4f at 64.140.166.100:5060'/52, caller >> tag='as480e6ada'/10, callee tag='KSQ5DeD148vBD'/13 >> _______________________________________________ >> 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 david.villasmil.work at gmail.com Tue Apr 14 23:54:31 2020 From: david.villasmil.work at gmail.com (David Villasmil) Date: Wed, 15 Apr 2020 00:54:31 +0100 Subject: [OpenSIPS-Users] Messages In-Reply-To: References: Message-ID: Have you tried with 1 or 0? Regards, David Villasmil email: david.villasmil.work at gmail.com phone: +34669448337 On Wed, Apr 15, 2020 at 12:09 AM Saint Michael wrote: > what is the lowest debug level that is safe to use? > > > On Tue, Apr 14, 2020 at 6:59 PM David Villasmil < > david.villasmil.work at gmail.com> wrote: > >> Maybe lower the debug? >> >> On Tue, 14 Apr 2020 at 23:00, Saint Michael wrote: >> >>> I see thousands of the messages below. If they are not important, how do >>> I hide them >>> ARNING:dialog:dlg_onroute: tight matching failed for ACK with >>> callid='5063212c34ac27e23d25ea143ca8bf4f at 64.140.166.100:5060'/52, >>> ftag='as480e6ada'/10, ttag='8e2e-a7407c352db2ab880c67b8cc744fbbfe'/37 and >>> direction=1 >>> Apr 14 21:56:06 brian opensips[22702]: Apr 14 21:56:06 [22787] >>> WARNING:dialog:dlg_onroute: dialog identification elements are >>> callid='5063212c34ac27e23d25ea143ca8bf4f at 64.140.166.100:5060'/52, >>> caller tag='as480e6ada'/10, callee tag='KSQ5DeD148vBD'/13 >>> _______________________________________________ >>> 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 venefax at gmail.com Wed Apr 15 00:05:04 2020 From: venefax at gmail.com (Saint Michael) Date: Tue, 14 Apr 2020 20:05:04 -0400 Subject: [OpenSIPS-Users] Messages In-Reply-To: References: Message-ID: Let me try 0. On Tue, Apr 14, 2020, 7:56 PM David Villasmil < david.villasmil.work at gmail.com> wrote: > Have you tried with 1 or 0? > Regards, > > David Villasmil > email: david.villasmil.work at gmail.com > phone: +34669448337 > > > On Wed, Apr 15, 2020 at 12:09 AM Saint Michael wrote: > >> what is the lowest debug level that is safe to use? >> >> >> On Tue, Apr 14, 2020 at 6:59 PM David Villasmil < >> david.villasmil.work at gmail.com> wrote: >> >>> Maybe lower the debug? >>> >>> On Tue, 14 Apr 2020 at 23:00, Saint Michael wrote: >>> >>>> I see thousands of the messages below. If they are not important, how >>>> do I hide them >>>> ARNING:dialog:dlg_onroute: tight matching failed for ACK with >>>> callid='5063212c34ac27e23d25ea143ca8bf4f at 64.140.166.100:5060'/52, >>>> ftag='as480e6ada'/10, ttag='8e2e-a7407c352db2ab880c67b8cc744fbbfe'/37 and >>>> direction=1 >>>> Apr 14 21:56:06 brian opensips[22702]: Apr 14 21:56:06 [22787] >>>> WARNING:dialog:dlg_onroute: dialog identification elements are >>>> callid='5063212c34ac27e23d25ea143ca8bf4f at 64.140.166.100:5060'/52, >>>> caller tag='as480e6ada'/10, callee tag='KSQ5DeD148vBD'/13 >>>> _______________________________________________ >>>> 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 >> > _______________________________________________ > 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 alexandru.tripon at itsyscom.com Wed Apr 15 07:29:35 2020 From: alexandru.tripon at itsyscom.com (Alexandru Tripon) Date: Wed, 15 Apr 2020 10:29:35 +0300 Subject: [OpenSIPS-Users] Stir_shaken signature length In-Reply-To: <64e46d29-29a7-2277-5637-addfe5b6ecf8@opensips.org> References: <64e46d29-29a7-2277-5637-addfe5b6ecf8@opensips.org> Message-ID: Hi Vlad, It works as intended on the latest master. Thanks for your quick response, Alexandru Tripon On Tue, Apr 14, 2020 at 11:28 PM Vlad Patrascu wrote: > Hi, > > Please check the fix on the latest master branch on github and let me know > if everything is ok. > > Regards, > > Vlad Patrascu > On 10.04.2020 12:28, Alexandru Tripon wrote: > > Hi, > > I tried to populate the Identity header with the stir_shaken module. > The header is populated but when I try to verify the signature using an > external tool it fails because of the length. > I have the folowing Identity generated by Opensips: > ` > > eyJhbGciOiJFUzI1NiIsInBwdCI6InNoYWtlbiIsInR5cCI6InBhc3Nwb3J0IiwieDV1IjoiL2hvbWUvdHJpYWwvTHVjcnUvQ29kZS9zdGlyU2hha2VuL215cHVia2V5LnBlbSJ9.eyJhdHRlc3QiOiJBIiwiZGVzdCI6eyJ0biI6WyIxMDAyIl19LCJpYXQiOjE1ODY1MDMxODcsIm9yaWciOnsidG4iOiIxMDAxIn0sIm9yaWdpZCI6IjEyMzQ1NiJ9.MEYCIQCjIx6w8IeilqHq0jbc6uwIB9v1RDmecoep0gRJJC4EmQIhANH1MO9jwRtqH6jgFH12XqROFv-nUroEgzsRAaMJtAsR;info=\u003c/home/trial/Lucru/Code/stirShaken/mypubkey.pem\u003e;ppt=\"shaken\" > ` > the lenght of encoded signature(in base64) is 96 and in the decoded one is > 72. > In the RFC for ES256 algorithm( > https://tools.ietf.org/html/rfc7518#section-3.4) the length of the > decoded signature is 64. > Am I missing something here? > > Thanks, > Alexandru Tripon > > _______________________________________________ > Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Wed Apr 15 11:56:46 2020 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Wed, 15 Apr 2020 14:56:46 +0300 Subject: [OpenSIPS-Users] memory In-Reply-To: References: <64e46d29-29a7-2277-5637-addfe5b6ecf8@opensips.org> Message-ID: <4aecacc1-56cd-3500-a328-8649576a816d@opensips.org> Hi, See https://opensips.org/Documentation/TroubleShooting-OutOfMem Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com On 4/15/20 12:40 AM, Saint Michael wrote: > > I see hundreds of messages like this > > how I raise the memory? My box has unlimited memory, but opensips does > not use it. > I changed the parameters inside the file opensips.service, but it > did not changed the memory allocated. > > Apr 14 21:06:13 [117509] ERROR:core:fm_malloc: not enough free shm > memory (17320 bytes left, need 6720), please increase the "-m" command > line parameter! > Apr 14 21:06:13 [117509] ERROR:tm:new_t: out of mem > Apr 14 21:06:13 [117509] ERROR:tm:t_newtran: new_t failed > Apr 14 21:06:13 [117449] CRITICAL:dialog:w_set_dlg_profile: BUG - > setting profile from script, but no dialog found > Apr 14 21:06:13 [117518] Vendor IP 216.158.87.250 has a channel limit > of 15000 - currently 666 channels > Apr 14 21:06:13 [117449] CRITICAL:dialog:w_set_dlg_profile: BUG - > setting profile from script, but no dialog found > Apr 14 21:06:13 [117483] CRITICAL:dialog:w_set_dlg_profile: BUG - > setting profile from script, but no dialog found > Apr 14 21:06:13 [117483] Vendor IP 216.158.87.250 has a channel limit > of 15000 - currently 666 channels > Apr 14 21:06:13 [117507] ERROR:core:fm_malloc: not enough free shm > memory (36376 bytes left, need 6720), please increase the "-m" command > line parameter! > Apr 14 21:06:13 [117507] ERROR:tm:new_t: out of mem > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Wed Apr 15 17:31:27 2020 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Wed, 15 Apr 2020 20:31:27 +0300 Subject: [OpenSIPS-Users] Alias domain / dns srv In-Reply-To: <66e399c9-fda5-ba7a-b163-d959d5149df0@softnet.si> References: <66e399c9-fda5-ba7a-b163-d959d5149df0@softnet.si> Message-ID: <3e873bf7-2082-4a95-fffc-9277c3f0c414@opensips.org> Hi Miha, You are mixing the SIP domains with the SIP server location. The SIP domains have nothing to do with SRV, while for SIP server location you can use it. The idea is to set as SIP user user at sip.test.com (and 'sip.test.com' is the SIP domain all the time). If the domain does not support SRV, it will do an A lookup on sip.test.com, and you can point it , as IP, to proxy1.test.com. If the domain supports SRV....you know the drill . But in both cases the SIP domain in SIP messages will be 'sip.test.com' Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com On 4/14/20 11:57 AM, Miha via Users wrote: > Hello > > we have dns srv record for failover. In dns srv we have two record. > So, one version of our devices does not support dns srv records. Is it > possible to register device directly to one A record which is wirtten > in DNS SRV record and then use ALIAS in opensips to right domain? > > DNS SRV. > > sip.test.com  (proxy1.test.com, proxy2.test.com) > Devices that do not support will register to proxy1.test.com (opensips > will have alias which will point to sip.test.com)? > > > > > thank you > miha > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Wed Apr 15 17:37:07 2020 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Wed, 15 Apr 2020 20:37:07 +0300 Subject: [OpenSIPS-Users] Problem with add_rr_param() in v2.4.7 In-Reply-To: <001b01d61270$cd7d8200$68788600$@smartvox.co.uk> References: <001b01d61270$cd7d8200$68788600$@smartvox.co.uk> Message-ID: Hi John, So you say you experience a regression between 2.4.6 and 2.4.7.... Any particularities in terms of how you do the record_routing() and add_rr_param(), like sequence, other msg changes or signaling ? Any simple way to reproduce the issue? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com On 4/14/20 6:24 PM, John Quick wrote: > I've tried running the same opensips.cfg script in v2.4.6 and then 2.4.7 > When it is changed to 2.4.7, the function add_rr_param() does nothing. > When run under 2.4.6 it updates the Record-Route header as you would expect. > > John Quick > Smartvox Limited > Web: www.smartvox.co.uk > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users From bogdan at opensips.org Wed Apr 15 17:40:41 2020 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Wed, 15 Apr 2020 20:40:41 +0300 Subject: [OpenSIPS-Users] Forking call to more than one AOR In-Reply-To: References: Message-ID: <23797e3f-6e1a-9ea5-018c-5c4f6ab87d45@opensips.org> Hi Babak, Yes, the 'r' flag in lookup() is the magic one here, but you do not have to use append_branch() 2 times before the lookup. Just set the first AOR into RURI and the second one have it appended as branch - this is all you need to do before the lookup(). Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com On 4/14/20 4:25 PM, Babak Yakhchali wrote: > For anyone interested: > I used append_branch() two times to create 2 branches. then using >     $(branch(uri)[1]) = $avp(other_aor); > I set the second branch to go for other AOR, now calling lookup() like > this: >     lookup("location","r") > will create all needed branches based on the 2 already created > branches and removes them from destination set. Till now invites are > generated for different contacts, BUT there is a problem with clients > registered with second contact, because To header is based on the > first AOR in ruri and they will not accept the invite message, to > solve this I used a branch route and changed the To header like this: > > branch_route[per_branch_ops] { >    if($tU != $rU){ >         uac_replace_to("","$avp(other_aor)"); >    } > } > > > On Tue, Apr 14, 2020 at 12:46 PM Babak Yakhchali > > wrote: > > Hi > I'm trying to implement this scenario: > each user has two AORs. one for mobile and one for a userId > (random string) > at any time user may be registered on one, both or none of these AORs > when a call comes to one of these AORs the other one must be > called too (if there is a registered contact for it) > > reading the documents I think I need to use > lookup("location","r"). but I don't know how should I add branches > before calling it for the two AORs > > thanks > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Wed Apr 15 17:46:31 2020 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Wed, 15 Apr 2020 20:46:31 +0300 Subject: [OpenSIPS-Users] Local Route Socket In-Reply-To: References: Message-ID: <384a3314-6bfa-209e-5e92-3a388bb48862@opensips.org> Hi Samuel, Any change of the send_socket in local route should be taken into consideration - be sure, by placing some xlog, that the execution gets to the $fs assignment. Also check the logs, maybe the socket you are setting there is not matching any opensips listener, so ignored. Best regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com On 4/14/20 7:41 PM, Samuel Anderson wrote: > > Hi All, > > I know you’re unable to do routing or signaling in the local route, so > I’m hoping someone can help me think of a workaround. > > My OpenSIPS proxy has 2 IP addresses on the LAN and 1 on the WAN. When > a SIP trunk sends an invite to my proxy, it forwards that invite out a > specific LAN IP address, so Asterisk recognizes it as from the PSTN. > When an external UAC sends a registration request to the proxy, it’s > forwarded to Asterisk from the other LAN IP address, so Asterisk > recognizes it as a UAC to authenticate. I use the mid_registrar module > in OpenSIPS to save the location of the UAC. > > The problem is if the UAC becomes unresponsive and the max_pings_lost > threshold of the nathelper module is reached, OpenSIPS sends a > registration with an expiration of 0 to Asterisk. However, it’s sent > with the wrong source IP address, and Asterisk does not recognize it > as an AOR to update. > > I’ve tried modifying the local_route to change the $fs variable, but > this does not work. > > local_route { > >                 if (is_method("REGISTER")) { > >                                 #Asterisk is expecting Registrations > from UAC phones to come from 10.x.x.x > >                                 $fs = "udp:10.x.x.x:5060"; > >                 } > > } > > Thank you, > > -Sam > > This message (including any attachments) is intended only for the use > of the individual or entity to which it is addressed and may contain > information that is non-public, proprietary, privileged, confidential, > and exempt from disclosure under applicable law or may constitute as > attorney work product. If you are not the intended recipient, you are > hereby notified that any use, dissemination, distribution, or copying > of this communication is strictly prohibited. If you have received > this communication in error, notify us immediately by telephone and > (i) destroy this message if a facsimile or (ii) delete this message > immediately if this is an electronic communication. 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 Wed Apr 15 17:52:33 2020 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Wed, 15 Apr 2020 20:52:33 +0300 Subject: [OpenSIPS-Users] Messages In-Reply-To: References: Message-ID: <80472d11-fb01-8636-7ddd-11f246bec393@opensips.org> Hi Michael, I wouldn't say the messages are not important, they can reveal broken signaling. As per the message, the To-tag of the ACK does not match what the dialog learned from 200 OK. I would advise you to double check that via a SIP capture of the traffic. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com On 4/15/20 12:59 AM, Saint Michael wrote: > I see thousands of the messages below. If they are not important, how > do I hide them > ARNING:dialog:dlg_onroute: tight matching failed for ACK with > callid='5063212c34ac27e23d25ea143ca8bf4f at 64.140.166.100:5060'/52, > ftag='as480e6ada'/10, ttag='8e2e-a7407c352db2ab880c67b8cc744fbbfe'/37 > and direction=1 > Apr 14 21:56:06 brian opensips[22702]: Apr 14 21:56:06 [22787] > WARNING:dialog:dlg_onroute: dialog identification elements are > callid='5063212c34ac27e23d25ea143ca8bf4f at 64.140.166.100:5060'/52, > caller tag='as480e6ada'/10, callee tag='KSQ5DeD148vBD'/13 > > _______________________________________________ > 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 tito at xsvoce.com Wed Apr 15 19:00:15 2020 From: tito at xsvoce.com (Tito Cumpen) Date: Wed, 15 Apr 2020 12:00:15 -0700 Subject: [OpenSIPS-Users] choice of AS architecture In-Reply-To: <1586870941.864026772@f395.i.mail.ru> References: <1586870941.864026772@f395.i.mail.ru> Message-ID: Oleg, I tried using the Seas module a while back and found out the support for it was has dropped so I moved on to using restcomm sip servlets as an AS. I recommend you look into it. On Tue, Apr 14, 2020 at 6:30 AM Oleg Podguyko via Users < users at lists.opensips.org> wrote: > Hi Bogdan and all your team! > > We have been working with opensips for quite some time and we like it. We > keep the load on the order of 300-400 CAPS, and for example, the SIP-I > module is perhaps the only solution that allows you to fully work with > ISUP. Our programmers prepared several PRs based on the results of work > with SIP-I and SCTP modules. > We are faced with the task of creating a classic bundle of cscf + AS > server, in the future architecture opensips will play the role of cscf, and > as AS we look at the module SEAS + external server. > > We are all moving towards IMS. > > What do you think, how justified can such a choice be? Perhaps the > opensips team already has its own new vision for this? > What would you choose as AS? > > > -- > Oleg Podguyko > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -------------- next part -------------- An HTML attachment was scrubbed... URL: From john.quick at smartvox.co.uk Wed Apr 15 19:19:43 2020 From: john.quick at smartvox.co.uk (John Quick) Date: Wed, 15 Apr 2020 20:19:43 +0100 Subject: [OpenSIPS-Users] Problem with add_rr_param() in v2.4.7 In-Reply-To: References: <001b01d61270$cd7d8200$68788600$@smartvox.co.uk> Message-ID: <001801d6135a$d1db1a70$75914f50$@smartvox.co.uk> Hi Bogdan, The only things that are unusual are: 1) transport protocol conversion between TLS and UDP so it requires double RR's. 2) it must use FQDN instead of IP address in the Record Route headers for the TLS interface The script adds two RR headers like this: record_route_preset(":5061;transport=tls", ":5060"); add_rr_param(";r2=on"); There is also one more call to add_rr_param() after those two lines, but none of the parameters is added. For calls going in the other direction: record_route_preset(":5060", ":5061;transport=tls"); add_rr_param(";r2=on"); When testing with v2.4.7, the listen statements were like this: auto_aliases=no alias=udp::5060 alias=tls::5061 listen=udp::5060 listen=tls::5061 One idea I had is that it might work to use record_route() instead of record_route_preset() provided I changed the second listen statement to this: listen=tls::5061 AS :5061 ...but I haven't tested to see if the fault only happens with record_route_preset() and not with record_route(). John Quick Smartvox Limited -----Original Message----- From: Bogdan-Andrei Iancu Sent: 15 April 2020 18:37 To: john.quick at smartvox.co.uk; OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] Problem with add_rr_param() in v2.4.7 Hi John, So you say you experience a regression between 2.4.6 and 2.4.7.... Any particularities in terms of how you do the record_routing() and add_rr_param(), like sequence, other msg changes or signaling ? Any simple way to reproduce the issue? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com On 4/14/20 6:24 PM, John Quick wrote: > I've tried running the same opensips.cfg script in v2.4.6 and then > 2.4.7 When it is changed to 2.4.7, the function add_rr_param() does nothing. > When run under 2.4.6 it updates the Record-Route header as you would expect. > > John Quick > Smartvox Limited > Web: www.smartvox.co.uk > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users From miha at softnet.si Thu Apr 16 05:55:26 2020 From: miha at softnet.si (Miha) Date: Thu, 16 Apr 2020 07:55:26 +0200 Subject: [OpenSIPS-Users] Alias domain / dns srv In-Reply-To: <3e873bf7-2082-4a95-fffc-9277c3f0c414@opensips.org> References: <66e399c9-fda5-ba7a-b163-d959d5149df0@softnet.si> <3e873bf7-2082-4a95-fffc-9277c3f0c414@opensips.org> Message-ID: Hello Bogdan not mixing, just maybe wrong discribing :) That is what i did. Thank you for your help and explenation! br miha Bogdan-Andrei Iancu je 4/15/2020 ob 7:31 PM napisal: > Hi Miha, > > You are mixing the SIP domains with the SIP server location. The SIP > domains have nothing to do with SRV, while for SIP server location you > can use it. > > The idea is to set as SIP user user at sip.test.com (and 'sip.test.com' > is the SIP domain all the time). > > If the domain does not support SRV, it will do an A lookup on > sip.test.com, and you can point it , as IP, to proxy1.test.com. > > If the domain supports SRV....you know the drill . > > But in both cases the SIP domain in SIP messages will be 'sip.test.com' > > Regards, > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > > On 4/14/20 11:57 AM, Miha via Users wrote: >> Hello >> >> we have dns srv record for failover. In dns srv we have two record. >> So, one version of our devices does not support dns srv records. Is >> it possible to register device directly to one A record which is >> wirtten in DNS SRV record and then use ALIAS in opensips to right domain? >> >> DNS SRV. >> >> sip.test.com  (proxy1.test.com, proxy2.test.com) >> Devices that do not support will register to proxy1.test.com >> (opensips will have alias which will point to sip.test.com)? >> >> >> >> >> thank you >> miha >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Thu Apr 16 11:00:45 2020 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Thu, 16 Apr 2020 14:00:45 +0300 Subject: [OpenSIPS-Users] choice of AS architecture In-Reply-To: <1586870941.864026772@f395.i.mail.ru> References: <1586870941.864026772@f395.i.mail.ru> Message-ID: Hey Oleg, What SIP features/capabilities do you want to have on the AS component? IMHO this is the point to start - based on your own requirements on what the AS must do, you can do a proper selection. And yes, as Tito said, seas is dead fish in the water for a long time. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com On 4/14/20 4:29 PM, Oleg Podguyko via Users wrote: > Hi Bogdan and all your team! > We have been working with opensips for quite some time and we like it. > We keep the load on the order of 300-400 CAPS, and for example, the > SIP-I module is perhaps the only solution that allows you to fully > work with ISUP. Our programmers prepared several PRs based on the > results of work with SIP-I and SCTP modules. > We are faced with the task of creating a classic bundle of cscf + AS > server, in the future architecture opensips will play the role of > cscf, and as AS we look at the module SEAS + external server. > > We are all moving towards IMS. > What do you think, how justified can such a choice be? Perhaps the > opensips team already has its own new vision for this? > What would you choose as AS? > -- > Oleg Podguyko > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From farmorg at gmail.com Thu Apr 16 15:14:27 2020 From: farmorg at gmail.com (Mark Farmer) Date: Thu, 16 Apr 2020 16:14:27 +0100 Subject: [OpenSIPS-Users] Compare $fU to drouting prefix Message-ID: Hi everyone I am looking for a way to compare $fU in INVITE to the matching drouting() prefix of another group and retrieve the rule_attrs from that rule. At the moment I am thinking I'll have to run a custom DB query so I have 2 questions: 1. Is there a better way to do this? 2. If not, what is the best way to run custom DB queries? I have been reading through the drouting() documentation but that hasn't helped. OpenSIPS 2.4.7 Many thanks! Mark. -------------- next part -------------- An HTML attachment was scrubbed... URL: From gmaruzz at gmail.com Thu Apr 16 15:36:08 2020 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Thu, 16 Apr 2020 17:36:08 +0200 Subject: [OpenSIPS-Users] Announcing SaraPhone, SIP WebRTC Open Source business phone Message-ID: My fellow VoIPers, I am pleased to announce the early availability of: SaraPhone ------------------ SaraPhone is a bare bone SIP WebRTC voice phone, complete with most features real companies want to use in real world: HotDesking, Redial, BLFs, MWI, DND, PhoneBook, Hold, Transfer, Mute, Attended Transfer, Notifications, running on all Browsers both on Desktop and SmartPhone. SaraPhone is fully integrated with FusionPBX, the full-featured domain based multi-tenant PBX and voice switch for FreeSwitch. Based on SIP.js, SaraPhone works with all WebRTC compliant SIP proxies, gateways, and servers (Asterisk, OpenSIPS, Kamailio, etc). Initial author is Giovanni Maruzzelli, and SaraPhone gets its name from Giovanni's wife, Sara Hosseini. In addition to providing all of the usual DeskPhone functionality, SaraPhone got: - Desktop Notification for Incoming Calls - Live MWI update - Real Time BLFs status update - BLF click to call - Caller Name and Number Display - Call Error Cause Display - AutoAnswer - Network Disconnect Reload - Show and Set Caller-ID (incoming-outbound) You an find it in GitHub ( https://github.com/gmaruzz/saraphone ). Anyone interested can play with it :). Have fun, giovanni -- Sincerely, Giovanni Maruzzelli OpenTelecom.IT cell: +39 347 266 56 18 -------------- next part -------------- An HTML attachment was scrubbed... URL: From Ben.Newlin at genesys.com Sun Apr 5 19:15:05 2020 From: Ben.Newlin at genesys.com (Ben Newlin) Date: Sun, 5 Apr 2020 19:15:05 +0000 Subject: [OpenSIPS-Users] t_forward_nonack failed (Ben Newlin) In-Reply-To: <1586109508.907336702@f327.i.mail.ru> References: <1586109508.907336702@f327.i.mail.ru> Message-ID: I am not positive whether flags or AVPs set within the CANCEL transaction would be visible back in the INVITE transaction. It was just a thought. You might try a flag instead. But also in your code snippet you do have a typo. It should be: $avp(ci)=$ci; If flags and avps don’t work you may have to use something like a local cache to store the CANCEL data. I can’t think of any other solutions. It does seem OpenSIPS should provide the ability to detect this by allowing t_was_cancelled (and possibly t_cancel_branch) outside of just onreply_route. Ben Newlin From: Users on behalf of Oleg Podguyko via Users Reply-To: Oleg Podguyko , OpenSIPS users mailling list Date: Sunday, April 5, 2020 at 2:01 PM To: "users at lists.opensips.org" Subject: [OpenSIPS-Users] Re: t_forward_nonack failed (Ben Newlin) Hello Ben, Thank you for answer. This is my RELAY route route[RELAY] { if (!t_relay()) { sl_reply_error(); } exit; } I did the following experiment. Commented out the line #sl_reply_error(); Sent Invite and Cancel. Opensips did not send 500. In fact, it is «sl_reply_error» that sends 500. it looks like: SIP/2.0 500 No destination available (18/SL) It is clear now. Following your advice I tried to use avp variable. ## CANCEL processing if (is_method("CANCEL")) { ## returns true if the current request is associated to a transaction if (t_check_trans()) { $avp(ci)=&ci t_relay(); } exit; } But when I try to use this variable( $avp(ci)) at the [resume] route of rest_client, it is Воскресенье, 5 апреля 2020, 16:55 +03:00 от users-request at lists.opensips.org: Send Users mailing list submissions to users at lists.opensips.org To subscribe or unsubscribe via the World Wide Web, visit http://lists.opensips.org/cgi-bin/mailman/listinfo/users or, via email, send a message with subject or body 'help' to users-request at lists.opensips.org You can reach the person managing the list at users-owner at lists.opensips.org When replying, please edit your Subject line so it is more specific than "Re: Contents of Users digest..." Today's Topics: 1. Re: t_forward_nonack failed (Ben Newlin) 2. Re: using load balancer and lookup together (Michael Vale) ---------------------------------------------------------------------- Message: 1 Date: Sun, 5 Apr 2020 13:11:34 +0000 From: Ben Newlin > To: Oleg Podguyko >, OpenSIPS users mailling list > Subject: Re: [OpenSIPS-Users] t_forward_nonack failed Message-ID: <00BE023F-DA3A-45EE-988F-BA4891FB9E04 at genesys.com> Content-Type: text/plain; charset="utf-8" Oleg, At first glance, it seems like t_was_cancelled [1] is exactly what you want. However, it can only be called from onreply_route or failure_route, which makes it unusable for this case. This has caused problems for me as well, as it makes it very hard to detect cancellation when working in asynchronous routes that don’t involve SIP messages. Similarly, t_cancel_branch can only be called from onreply_route, which also causes issues with not being able to manage branches with non-SIP async operations. It may be a good feature request to add the ability to use those functions in async resume routes, but for now they will not work. However, t_relay itself does not ever generate a 500 error back to the client as far as I know. It can only send a 477 response automatically. Are you sure you are not generating the 500 in your script when handling the error from t_relay? The documentation for the function [2] notes that a -3 response from t_relay indicates the request may have already been cancelled. In that case, you can just exit if you know the 487 has already been sent, or you can send a 487 reply yourself. To be honest, I’m not sure where the 487 is coming from in your case since I didn’t think OpenSIPS would automatically respond with a 487 for a cancelled transaction; I have always had to do that from the script myself. So in addition to sending the 500 reply yourself, you may have some code in your script which is also sending the 487. If none of that works, another option would be to set a flag or an avp in the transaction when processing the CANCEL. Then you can check the flag when you receive the HTTP response and if it is set just don’t call t_relay. [1] https://opensips.org/docs/modules/3.0.x/tm.html#func_t_was_cancelled [2] https://opensips.org/docs/modules/3.0.x/tm.html#func_t_relay Ben Newlin From: Users > on behalf of Oleg Podguyko via Users > Reply-To: Oleg Podguyko >, OpenSIPS users mailling list > Date: Sunday, April 5, 2020 at 6:31 AM To: users > Subject: [OpenSIPS-Users] t_forward_nonack failed Opensips works like a proxy. Gets an Invite. In the process of processing it, opensips makes a request via http (rest_client module) and receives a response. Adds the received information to Invite (as X-header) and sends through the dispatcher module to freeswitch. Everything is working fine. However, there is a scenario in which everything goes a little wrong. Opensips receives an Invite, starts processing it, sends a request via http. At this time, Cancel arrives at Invite. The transaction is being destroyed. Opensips sends 200 Cancelling, and then 487. And here comes the response via HTTP, but since the transaction is no longer there, this call ends with an error 500 no route to destination when the t_relay function is executed. In this case, a message appears in the log / usr / sbin / opensips [5577]: ERROR: tm: w_t_relay: t_forward_nonack failed. How to correctly handle such cases in order to prevent such errors in the logs? -- Oleg Podguyko -------------- next part -------------- An HTML attachment was scrubbed... URL: > ------------------------------ Message: 2 Date: Sun, 05 Apr 2020 23:54:29 +1000 From: Michael Vale > To: David Villasmil >, OpenSIPS users mailling list > Subject: Re: [OpenSIPS-Users] using load balancer and lookup together Message-ID: <6342fd559df4c9765b17270d07bc04f11ad93b36.camel at vale.ski> Content-Type: text/plain; charset="utf-8" Ok, to explain, Using your logic, To call say '555' will goto Voicemail, if I disable voicemail it will return a 404 instead of going to the load balancer. That's fine, if 555 is an extension, but if it's (for this example) a PSTN number (or a all-else catch all, like I'm trying to achieve) thats not OK because I get a 404 rather than it getting routed to the load balancer. If 555 is an extension/user the call will go through to the registered extension, but if it's not registered in the usrloc table, it goes to 404, instead of the load balancer. If I reverse the logic, It will goto the load balancer even if it's a registered extension, or Too Many Hops, depending on how I adjust the logic. I cannot seem to create a catch all for non-usrloc registered extension calls to goto the load balancer otherwise return a 404. I hope I explained it well enough. I will keep trying, Regards, Michael. On Sun, 2020-04-05 at 11:47 +0100, David Villasmil wrote: > Why are you trying to do all at once? > > Why not first do the lookup > > https://github.com/davidcsi/kamailio-private-public/blob/a81d7f777a8c5ee2dbb32311f7e6b5a3cf94bf32/kamailio.cfg#L771 > > > and then start load balancing? > > https://github.com/davidcsi/kamailio-private-public/blob/a81d7f777a8c5ee2dbb32311f7e6b5a3cf94bf32/kamailio.cfg#L1109 > > Do you have some special need to fulfill? > > David > On Sun, 5 Apr 2020 at 06:34, Michael Vale via Users < > users at lists.opensips.org> wrote: > > hi, > > > > > > > > perhaps this can be solved with a failure route and or a check > > status > > > > but i dont know and it would be nice if i could do it without it. > > > > > > > > no matter how i write the script, either a uac to uac call goes to > > the > > > > load balancer or the load balancer is stuck with a 404 reply from > > the > > > > script or uac to uac works but when one end is not registered it > > goes > > > > to the load balancer instead of getting a 404. > > > > > > > > i've tried failure routes and get the same problem. here is a > > snippet. > > > > > > > > if (!lb_start(1,"pstn")) && (!lookup("location","m",)) { > > > > lb_disable_dst(); > > > > #route(relay); > > > > #send_reply(404,"No user or gateway"); > > > > if (lb_start(1,"pstn")) { > > > > send_reply(500,"SIPSIPSIPS"); > > > > # t_relay(); > > > > exit; > > > > } > > > > # exit; > > > > } else if (lookup("location","m")) && > > > > (!lb_start(1,"pstn")) { > > > > lb_disable_dst(); > > > > route(relay); > > > > exit; > > > > } else if (lb_start(1,"pstn")) && > > > > (lookup("location","m")) { > > > > lb_disable_dst(); > > > > route(relay); > > > > exit; > > > > } else if (!lookup("location","m")) && > > > > (!lb_start(1,"pstn")) { > > > > send_reply(404,"Not Found"); > > > > exit; > > > > } else if (lb_start(1,"pstn")) && > > > > (!lookup("location","m")) { > > > > # #lb_disable_dst(); > > > > if (!lookup("location","m")) { > > > > route(relay); > > > > exit; > > > > } > > > > if (lookup("location","m")) { > > > > lb_disable_dst(); > > > > route(relay); > > > > exit; > > > > } > > > > } > > > > > > > > thanks in advance, > > > > > > > > michael. > > > > > > > > > > > > _______________________________________________ > > > > Users mailing list > > > > Users at lists.opensips.org > > > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > -- > Regards, > > David Villasmilemail: david.villasmil.work at gmail.com > phone: +34669448337 -------------- next part -------------- An HTML attachment was scrubbed... URL: > ------------------------------ Subject: Digest Footer _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ------------------------------ End of Users Digest, Vol 141, Issue 8 ************************************* -- Олег Подгуйко -------------- next part -------------- An HTML attachment was scrubbed... URL: From podguiko at mail.ru Sun Apr 5 20:12:25 2020 From: podguiko at mail.ru (=?UTF-8?B?T2xlZyBQb2RndXlrbw==?=) Date: Sun, 05 Apr 2020 23:12:25 +0300 Subject: [OpenSIPS-Users] =?utf-8?q?t=5Fforward=5Fnonack_failed_=28Ben_New?= =?utf-8?q?lin=29?= In-Reply-To: References: <1586109508.907336702@f327.i.mail.ru> Message-ID: <1586117545.50299149@f422.i.mail.ru> I tried to use flags, it doesn’t work too. You a right,  the way  of ability to detect this by allowing «t_was_cancelled» at the any «routes» is a best way. I will open a issue.   Thanks again for your reply!   >Воскресенье, 5 апреля 2020, 22:15 +03:00 от Ben Newlin : >  >I am not positive whether flags or AVPs set within the CANCEL transaction would be visible back in the INVITE transaction. It was just a thought. You might try a flag instead. But also in your code snippet you do have a typo. It should be: >  >$avp(ci)=$ci; >  >If flags and avps don’t work you may have to use something like a local cache to store the CANCEL data. I can’t think of any other solutions. It does seem OpenSIPS should provide the ability to detect this by allowing t_was_cancelled (and possibly t_cancel_branch) outside of just onreply_route. >  >Ben Newlin >  >From: Users < users-bounces at lists.opensips.org > on behalf of Oleg Podguyko via Users < users at lists.opensips.org > >Reply-To: Oleg Podguyko < podguiko at mail.ru >, OpenSIPS users mailling list < users at lists.opensips.org > >Date: Sunday, April 5, 2020 at 2:01 PM >To: "users at lists.opensips.org" < users at lists.opensips.org > >Subject: [OpenSIPS-Users] Re: t_forward_nonack failed (Ben Newlin) >  > >Hello Ben, >  >Thank you for answer. >  >This is my RELAY route >  >route[RELAY] { >    if (!t_relay()) >    { >        sl_reply_error(); >    } >    exit; >} >  >I did the following experiment. Commented out the line >#sl_reply_error(); > >Sent Invite and Cancel. Opensips did not send 500. >  >In fact, it is «sl_reply_error» that sends 500. >it looks like: >  >SIP/2.0 500 No destination available (18/SL) >It is clear now. >  >Following your advice I tried to use avp variable. >  >## CANCEL processing >    if (is_method("CANCEL")) >    { >        ## returns true if the current request is associated to a transaction >        if (t_check_trans()) >        { >            $avp(ci)=&ci >            t_relay(); >        } >        exit; >    } >  >  >But when I try to use this variable( $avp(ci)) at the [resume] route of rest_client, it is >  >  >>Воскресенье, 5 апреля 2020, 16:55 +03:00 от users-request at lists.opensips.org: >>  >>Send Users mailing list submissions to >>users at lists.opensips.org >> >>To subscribe or unsubscribe via the World Wide Web, visit >>http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>or, via email, send a message with subject or body 'help' to >>users-request at lists.opensips.org >> >>You can reach the person managing the list at >>users-owner at lists.opensips.org >> >>When replying, please edit your Subject line so it is more specific >>than "Re: Contents of Users digest..." >> >> >>Today's Topics: >> >>   1. Re: t_forward_nonack failed (Ben Newlin) >>   2. Re: using load balancer and lookup together (Michael Vale) >> >> >>---------------------------------------------------------------------- >> >>Message: 1 >>Date: Sun, 5 Apr 2020 13:11:34 +0000 >>From: Ben Newlin < Ben.Newlin at genesys.com > >>To: Oleg Podguyko < podguiko at mail.ru >, OpenSIPS users mailling list >>< users at lists.opensips.org > >>Subject: Re: [OpenSIPS-Users] t_forward_nonack failed >>Message-ID: < 00BE023F-DA3A-45EE-988F-BA4891FB9E04 at genesys.com > >>Content-Type: text/plain; charset="utf-8" >> >>Oleg, >> >>At first glance, it seems like t_was_cancelled [1] is exactly what you want. However, it can only be called from onreply_route or failure_route, which makes it unusable for this case. This has caused problems for me as well, as it makes it very hard to detect cancellation when working in asynchronous routes that don’t involve SIP messages. Similarly, t_cancel_branch can only be called from onreply_route, which also causes issues with not being able to manage branches with non-SIP async operations. It may be a good feature request to add the ability to use those functions in async resume routes, but for now they will not work. >> >>However, t_relay itself does not ever generate a 500 error back to the client as far as I know. It can only send a 477 response automatically. Are you sure you are not generating the 500 in your script when handling the error from t_relay? The documentation for the function [2] notes that a -3 response from t_relay indicates the request may have already been cancelled. In that case, you can just exit if you know the 487 has already been sent, or you can send a 487 reply yourself. To be honest, I’m not sure where the 487 is coming from in your case since I didn’t think OpenSIPS would automatically respond with a 487 for a cancelled transaction; I have always had to do that from the script myself. So in addition to sending the 500 reply yourself, you may have some code in your script which is also sending the 487. >> >>If none of that works, another option would be to set a flag or an avp in the transaction when processing the CANCEL. Then you can check the flag when you receive the HTTP response and if it is set just don’t call t_relay. >> >>[1] https://opensips.org/docs/modules/3.0.x/tm.html#func_t_was_cancelled >>[2] https://opensips.org/docs/modules/3.0.x/tm.html#func_t_relay >> >>Ben Newlin >> >>From: Users < users-bounces at lists.opensips.org > on behalf of Oleg Podguyko via Users < users at lists.opensips.org > >>Reply-To: Oleg Podguyko < podguiko at mail.ru >, OpenSIPS users mailling list < users at lists.opensips.org > >>Date: Sunday, April 5, 2020 at 6:31 AM >>To: users < users at lists.opensips.org > >>Subject: [OpenSIPS-Users] t_forward_nonack failed >> >> >>Opensips works like a proxy. Gets an Invite. In the process of processing it, opensips makes a request via http (rest_client module) and receives a response. >>Adds the received information to Invite (as X-header) and sends through the dispatcher module to freeswitch. >>Everything is working fine. >>However, there is a scenario in which everything goes a little wrong. >>Opensips receives an Invite, starts processing it, sends a request via http. At this time, Cancel arrives at Invite. The transaction is being destroyed. Opensips sends 200 Cancelling, and then 487. And here comes the response via HTTP, but since the transaction is no longer there, this call ends with an error 500 no route to destination when the t_relay function is executed. In this case, a message appears in the log >>/ usr / sbin / opensips [5577]: ERROR: tm: w_t_relay: t_forward_nonack failed. >>How to correctly handle such cases in order to prevent such errors in the logs? >> >>-- >>Oleg Podguyko >>-------------- next part -------------- >>An HTML attachment was scrubbed... >>URL: < http://lists.opensips.org/pipermail/users/attachments/20200405/8680c7ac/attachment-0001.html > >> >>------------------------------ >> >>Message: 2 >>Date: Sun, 05 Apr 2020 23:54:29 +1000 >>From: Michael Vale < masked at vale.ski > >>To: David Villasmil < david.villasmil.work at gmail.com >, OpenSIPS users >>mailling list < users at lists.opensips.org > >>Subject: Re: [OpenSIPS-Users] using load balancer and lookup together >>Message-ID: < 6342fd559df4c9765b17270d07bc04f11ad93b36.camel at vale.ski > >>Content-Type: text/plain; charset="utf-8" >> >>Ok, to explain, >>Using your logic, >>To call say '555' will goto Voicemail, if I disable voicemail it will >>return a 404 instead of going to the load balancer. >>That's fine, if 555 is an extension, but if it's (for this example) a >>PSTN number (or a all-else catch all, like I'm trying to achieve) thats >>not OK because I get a 404 rather than it getting routed to the load >>balancer. >>If 555 is an extension/user the call will go through to the registered >>extension, but if it's not registered in the usrloc table, it goes to >>404, instead of the load balancer. >>If I reverse the logic, It will goto the load balancer even if it's a >>registered extension, or Too Many Hops, depending on how I adjust the >>logic. >>I cannot seem to create a catch all for non-usrloc registered extension >>calls to goto the load balancer otherwise return a 404. >>I hope I explained it well enough. I will keep trying, >>Regards, >>Michael. >>On Sun, 2020-04-05 at 11:47 +0100, David Villasmil wrote: >>> Why are you trying to do all at once? >>> >>> Why not first do the lookup >>> >>> https://github.com/davidcsi/kamailio-private-public/blob/a81d7f777a8c5ee2dbb32311f7e6b5a3cf94bf32/kamailio.cfg#L771 >>> >>> >>> and then start load balancing? >>> >>> https://github.com/davidcsi/kamailio-private-public/blob/a81d7f777a8c5ee2dbb32311f7e6b5a3cf94bf32/kamailio.cfg#L1109 >>> >>> Do you have some special need to fulfill? >>> >>> David >>> On Sun, 5 Apr 2020 at 06:34, Michael Vale via Users < >>> users at lists.opensips.org > wrote: >>> > hi, >>> > >>> > >>> > >>> > perhaps this can be solved with a failure route and or a check >>> > status >>> > >>> > but i dont know and it would be nice if i could do it without it. >>> > >>> > >>> > >>> > no matter how i write the script, either a uac to uac call goes to >>> > the >>> > >>> > load balancer or the load balancer is stuck with a 404 reply from >>> > the >>> > >>> > script or uac to uac works but when one end is not registered it >>> > goes >>> > >>> > to the load balancer instead of getting a 404. >>> > >>> > >>> > >>> > i've tried failure routes and get the same problem. here is a >>> > snippet. >>> > >>> > >>> > >>> > if (!lb_start(1,"pstn")) && (!lookup("location","m",)) { >>> > >>> > lb_disable_dst(); >>> > >>> > #route(relay); >>> > >>> > #send_reply(404,"No user or gateway"); >>> > >>> > if (lb_start(1,"pstn")) { >>> > >>> > send_reply(500,"SIPSIPSIPS"); >>> > >>> > # t_relay(); >>> > >>> > exit; >>> > >>> > } >>> > >>> > # exit; >>> > >>> > } else if (lookup("location","m")) && >>> > >>> > (!lb_start(1,"pstn")) { >>> > >>> > lb_disable_dst(); >>> > >>> > route(relay); >>> > >>> > exit; >>> > >>> > } else if (lb_start(1,"pstn")) && >>> > >>> > (lookup("location","m")) { >>> > >>> > lb_disable_dst(); >>> > >>> > route(relay); >>> > >>> > exit; >>> > >>> > } else if (!lookup("location","m")) && >>> > >>> > (!lb_start(1,"pstn")) { >>> > >>> > send_reply(404,"Not Found"); >>> > >>> > exit; >>> > >>> > } else if (lb_start(1,"pstn")) && >>> > >>> > (!lookup("location","m")) { >>> > >>> > # #lb_disable_dst(); >>> > >>> > if (!lookup("location","m")) { >>> > >>> > route(relay); >>> > >>> > exit; >>> > >>> > } >>> > >>> > if (lookup("location","m")) { >>> > >>> > lb_disable_dst(); >>> > >>> > route(relay); >>> > >>> > exit; >>> > >>> > } >>> > >>> > } >>> > >>> > >>> > >>> > thanks in advance, >>> > >>> > >>> > >>> > michael. >>> > >>> > >>> > >>> > >>> > >>> > _______________________________________________ >>> > >>> > Users mailing list >>> > >>> > Users at lists.opensips.org >>> > >>> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> > >>> -- >>> Regards, >>> >>> David Villasmilemail: david.villasmil.work at gmail.com >>> phone: +34669448337 >>-------------- next part -------------- >>An HTML attachment was scrubbed... >>URL: < http://lists.opensips.org/pipermail/users/attachments/20200405/6ef5eec8/attachment.html > >> >>------------------------------ >> >>Subject: Digest Footer >> >>_______________________________________________ >>Users mailing list >>Users at lists.opensips.org >>http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> >>------------------------------ >> >>End of Users Digest, Vol 141, Issue 8 >>************************************* >  >  >-- >Олег Подгуйко >      -- Олег Подгуйко   -------------- next part -------------- An HTML attachment was scrubbed... URL: From Ben.Newlin at genesys.com Mon Apr 6 13:24:10 2020 From: Ben.Newlin at genesys.com (Ben Newlin) Date: Mon, 6 Apr 2020 13:24:10 +0000 Subject: [OpenSIPS-Users] Issue with 'To' tag and t_reply In-Reply-To: References: Message-ID: <28968715-5132-409E-BBF6-200462569AC8@genesys.com> Unfortunately I don’t think I can provide any further assistance. As far as I know Server A is still operating out of spec. The receipt of the 183 with To tag does establish an *early* dialog, but nowhere does that mandate that the server must stop receiving responses from any other dialog. Servers must be tolerant of downstream forking of SIP calls by proxies (like OpenSIPS) and until the final response is received they must be able to receive multiple provisional responses potentially with different To tags. As I said before, on the OpenSIPS side there really is no way to solve this well. If OpenSIPS forks a call and receives multiple provisional responses from different downstream servers, how would it choose one to send back upstream? How could it know that is the one that will eventually answer? It can’t, so it sends them upstream as is and the UAC must handle it. When it needs to send a message it is operating per the RFC which recommends creating your own tag rather than randomly choosing a downstream tag which may or may not be the final responder. I suppose some enhancements could be made to make it work in your specific case, where you do not have multiple downstream legs, but personally I don’t feel it would be worth the added complication. But that would be up to the OpenSIPS maintainers. The only thing I can think that you could do to workaround the issue with Server A is to drop all provisional responses and only provide the final response, because you can’t guarantee until the final response what the final To tag will be. Also, I will say that we do failover just like this in our OpenSIPS. We relay provisional responses as they come and we relay final responses, sometimes coming from different downstream servers due to failover. And sometimes we send the response ourselves. We’ve never had an issue with an upstream server complaining about the tag changing. Ben Newlin From: Users on behalf of Yury Kirsanov Reply-To: OpenSIPS users mailling list Date: Monday, April 6, 2020 at 12:05 AM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] Issue with 'To' tag and t_reply Hi Ben, The problem we're facing now is that according to RFC3261 dialog is established after 183 Session Progress with a To tag, so Server A can't continue to receive out-of-dialog SIP messages. In this case we're unable to send a OpenSIPS-generated message with different To tag which occurs in my situation. Is there any way to resolve this situation? It looks to me that behaviour of Server A is correct as OpenSIPS acts as a proxy and passes messages from Server B and then suddenly injects a SIP message originated by itself. Looks like there has to be two 'legs' of the call, one between Server A and OpenSIPS and another one between OpenSIPS and multiple servers it tries to reach in order to establish the call, but in this case OpenSIPS can't act as a pure proxy. Please advise? Thanks!. P.S. For some reason I'm not receiving your responses in my mailbox? пт, 27 мар. 2020 г. в 01:56, Yury Kirsanov >: Thanks a lot for your explanation, Ben! I thought that there can be an issue with Server A not accepting my new SIP response, it looks like they're doing matching only based on SIP To tag and completely ignoring any Call-ID or DID matching as well as From tag matching, in my case From tag is always the same. One more question, how do you think, can there be anything related with topology hiding I'm using? I really doubt that but just in case...As far as I understand my issue is not because I'm using topology hiding, but because OpenSIPS first passes To tag from remote server and then generates one by itself when I'm using a 't_reply' and Server A is just not accepting such behaviour, trying to match any SIP responses to To tag passed in 183 Session Progress. I tried to change topology_hiding() to loose_route() and nothing changed in my chain except for Server A now being able to see all RRs and Vias inside my network. Regards, Yury. пт, 27 мар. 2020 г. в 01:37, Yury Kirsanov >: But the question is still here - how can I send a different t_reply code from failure_route? And then stop processing any further SIP messages? пт, 27 мар. 2020 г. в 01:23, Yury Kirsanov >: The problem is that I need to go through a list of SIP servers, analyze response of each of them and if it's an error like 4XX, 5XX or 6XX I need to send appropriate response to originating server. Let's say I'm not only adding a Reason field but upon receipt of 404 Not Found I want to respond with 480 Temporarily Unavailable with Reason: Q.850;Cause=41 for example? But Server B first replied with 183 Session Progress playing back a message 'Sorry, you need to top up your account' and then replied with SIP 402 Payment Required. I had to proxy 183 Session Progress back to Server A so its SIP client could hear that message and then I'd like to signal 480 Temporarily Unavailable - but I can't as OpenSIPS is using completely different To tag. I can't do this in onreply_route as I'm going through a list of SIP servers (upstreams or downstreams), so it definitely needs to be done from failure route, as far as I understand, and yes, I'm matching against 4XX, 5XX and 6XX codes and I need to reply with 480 Temporarily Unavailable in most cases so Server A would have a possibility to do failover to any other server in that case. I don't want to just proxy 4XX, 5XX or 6XX response to it. I've figured out why I have two 404 responses in my original call log - I was using sl_send_reply instead of t_reply and it was using original To tag but only on second attempt. Regards, Yury. пт, 27 мар. 2020 г. в 01:02, Yury Kirsanov >: Hi, As per my original email: 1. I was doing exactly as you suggested, in failure_route I'm using t_reply("404","Not Found") and it comes out with a wrong To: tag. 2. I don't need to proxy response from server B, I need to analyze its response and send a response to server A according to my needs. Currently it seems that t_reply is not using same To tag if 183 Session Progress has been proxied, which is strange as I have dialog running. Regards, Yury. чт, 26 мар. 2020 г. в 19:13, Yury Kirsanov >: Hi, I'm using an OpenSIPS as a proxy between two servers. First one is sending SIP INVITE to OpenSIPS, then OpenSIPS forwards request to second server. I'm creating a dialog on initial INVITE. Second server then replies with SIP 183 Session Progress, plays back a message and then responds with 4XX code, for example SIP 404 Not Found (indicating that number dialed is disconnected). In OpenSIPS I'm receiving that reply and in failure_route I'd like to change that code to a bit different SIP 404, so I'm using following code: append_to_reply("Reason: Q.850;cause=1"); t_reply("404","Not Found"); exit; But in this case I can see that OpenSIPS generates additional branch (??? not sure here) with different "To" tag and pushes it out and then forwards original reply SIP packet even though I have an exit statement in my failure_route. I tried to do sl_send_reply and behavior is pretty much the same. Can someone let me know what I may be doing wrong? I need correct "To" tag to be used (based on 183 Session Progress message from server B and passed to server A previously) and second 404 shouldn't be forwarded out. Here's an example of a call with my explanations Initial invite from server A, no 'to tag' as expected: INVITE sip:XXXXXXXXX at B.B.B.B SIP/2.0 Max-Forwards: 67 To: "XXXXXXXXX" Call-ID: 469A5568-E092-4038-B1B8-13AC9B9571CA Via: SIP/2.0/UDP A.A.A.A:5060;rport;branch=z9hG4bK773616538 From: "YYYYYYYYY" ;tag=117583367 CSeq: 1741310 INVITE User-Agent: User Agent Contact: Allow: ACK, INVITE, BYE, CANCEL, REGISTER, REFER, OPTIONS, INFO, SUBSCRIBE, NOTIFY Date: Thu, 26 Mar 2020 07:54:55 GMT Content-Type: application/sdp Content-Length: 250 v=0 o=dcom 1585209295 1585209295 IN IP4 A.A.A.A s=SIP Call c=IN IP4 A.A.A.A t=0 0 m=audio 15340 RTP/AVP 8 0 18 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:101 telephone-event/8000 Response from OpenSIPS: SIP/2.0 100 Giving a try To: "XXXXXXXXX" Call-ID: 469A5568-E092-4038-B1B8-13AC9B9571CA Via: SIP/2.0/UDP A.A.A.A:5060;received=A.A.A.A;rport=5060;branch=z9hG4bK773616538 From: "YYYYYYYYY" ;tag=117583367 CSeq: 1741310 INVITE Server: Server Signature Content-Length: 0 OpenSIPS has forwarded packet to Server B and Server B responded with 183 and assigned a 'To' tag: SIP/2.0 183 Session Progress Via: SIP/2.0/UDP A.A.A.A:5060;received=A.A.A.A;rport=5060;branch=z9hG4bK773616538 Call-ID: 469A5568-E092-4038-B1B8-13AC9B9571CA From: "YYYYYYYYY" ;tag=117583367 To: "XXXXXXXXX" ;tag=0b49dc32-2c4b-413e-a349-c781a23d53b9 CSeq: 1741310 INVITE Server: PBX Contact: Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REFER Content-Type: application/sdp Content-Length: 354 v=0 o=- 1585209295 1585209297 IN IP4 B.B.B.B s=Asterisk c=IN IP4 B.B.B.B t=0 0 a=rtpengine:673f999268ae m=audio 32386 RTP/AVP 0 8 18 101 a=maxptime:150 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:18 annexb=no a=fmtp:101 0-16 a=sendrecv a=rtcp:32387 a=ptime:20 Server B responds with SIP 404 after playing back message that number is disconnected and I'm trying to reply to server A with custom Reason message. To_tag is completely different from the To tag that has been passed to server A after initial 183!!! SIP/2.0 404 Not Found To: "XXXXXXXXX" ;tag=a976.21514595b467be41a9b712a6b0b621d9 Call-ID: 469A5568-E092-4038-B1B8-13AC9B9571CA Via: SIP/2.0/UDP A.A.A.A:5060;received=A.A.A.A;rport=5060;branch=z9hG4bK773616538 From: "YYYYYYYYY" ;tag=117583367 CSeq: 1741310 INVITE Reason: Q.850;cause=1;text="Number is disconnected" Server: Server Signature Content-Length: 0 Of course, server A just ignores this message as it can't match 'To' tag to its transaction. Now, for some reason, OpenSIPS forwards original reply from Server B to Server A with the same 'To' tag as in 183 Session Progress: SIP/2.0 404 Not Found Via: SIP/2.0/UDP A.A.A.A:5060;received=A.A.A.A;rport=5060;branch=z9hG4bK773616538 Call-ID: 469A5568-E092-4038-B1B8-13AC9B9571CA From: "YYYYYYYYY" ;tag=117583367 To: "XXXXXXXXX" ;tag=0b49dc32-2c4b-413e-a349-c781a23d53b9 CSeq: 1741310 INVITE Server: PBX Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REFER Reason: Q.850;cause=1 Content-Length: 0 And at this point Server A can match this reply and responds with an ACK: ACK sip:XXXXXXXXX at B.B.B.B SIP/2.0 Via: SIP/2.0/UDP A.A.A.A:5060;rport;branch=z9hG4bK773616538 From: "YYYYYYYYY" ;tag=117583367 To: "XXXXXXXXX" ;tag=0b49dc32-2c4b-413e-a349-c781a23d53b9 Call-ID: 469A5568-E092-4038-B1B8-13AC9B9571CA CSeq: 1741310 ACK Max-Forwards: 67 Contact: User-Agent: User Agent Content-Length: 0 I think that t_reply is creating a new transaction instead of using existing one, but I'm not sure why and how to fix this? Thanks! Best regards, Yury. -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Thu Apr 16 15:55:02 2020 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Thu, 16 Apr 2020 18:55:02 +0300 Subject: [OpenSIPS-Users] Compare $fU to drouting prefix In-Reply-To: References: Message-ID: <61a53c5b-3d70-4054-a837-3b9d70f135fd@opensips.org> Hi Mark, What kind of matching you want to do between $fU and the dr prefixes ? You want to do the same as what drouting() does with $rU ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com On 4/16/20 6:14 PM, Mark Farmer wrote: > Hi everyone > > I am looking for a way to compare $fU in INVITE to the matching > drouting() prefix of another group and retrieve the rule_attrs from > that rule. > > At the moment I am thinking I'll have to run a custom DB query so I > have 2 questions: > > 1. Is there a better way to do this? > 2. If not, what is the best way to run custom DB queries? > > I have been reading through the drouting() documentation but that > hasn't helped. > > OpenSIPS 2.4.7 > > Many thanks! > Mark. > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From liviu at opensips.org Thu Apr 16 16:06:13 2020 From: liviu at opensips.org (Liviu Chircu) Date: Thu, 16 Apr 2020 19:06:13 +0300 Subject: [OpenSIPS-Users] Announcing SaraPhone, SIP WebRTC Open Source business phone In-Reply-To: References: Message-ID: On 16.04.2020 18:36, Giovanni Maruzzelli wrote: > Initial author is Giovanni Maruzzelli, and SaraPhone gets its name > from Giovanni's wife, Sara Hosseini. Hi, Giovanni! This is amazing!  With SIPML5 slowly falling behind more and more, SaraPhone is a premium addition to the open-source WebRTC softphone niche. As the proud owner of star #1 on the GitHub project, I truly wish it will turn into 1 thousand, followed by 20 thousand (but not more, because you are already a busy man). Thank you!! -- Liviu Chircu www.twitter.com/liviuchircu | www.opensips-solutions.com OpenSIPS Summit, Amsterdam, May 2020 www.opensips.org/events From gmaruzz at gmail.com Thu Apr 16 16:14:46 2020 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Thu, 16 Apr 2020 18:14:46 +0200 Subject: [OpenSIPS-Users] Announcing SaraPhone, SIP WebRTC Open Source business phone In-Reply-To: References: Message-ID: On Thu, Apr 16, 2020 at 6:09 PM Liviu Chircu wrote: > On 16.04.2020 18:36, Giovanni Maruzzelli wrote: > > Initial author is Giovanni Maruzzelli, and SaraPhone gets its name > > from Giovanni's wife, Sara Hosseini. > > Hi, Giovanni! > > This is amazing! With SIPML5 slowly falling behind more and more, > SaraPhone is a premium addition to the open-source WebRTC softphone niche. > > As the proud owner of star #1 on the GitHub project, I truly wish it > will turn into 1 thousand, followed by 20 thousand (but not more, > because you are already a busy man). > > I've got license #1 for MySQL, some years ago I get a nice print to be framed from Monty and Axel, and then they sold to Oracle. I swear, if Oracle buy, I buy you a dinner :D ! Thanks a lot, Liviu!!! > Thank you!! > > -- > Liviu Chircu > www.twitter.com/liviuchircu | www.opensips-solutions.com > > OpenSIPS Summit, Amsterdam, May 2020 > www.opensips.org/events > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -- Sincerely, Giovanni Maruzzelli OpenTelecom.IT cell: +39 347 266 56 18 -------------- next part -------------- An HTML attachment was scrubbed... URL: From farmorg at gmail.com Fri Apr 17 08:13:30 2020 From: farmorg at gmail.com (Mark Farmer) Date: Fri, 17 Apr 2020 09:13:30 +0100 Subject: [OpenSIPS-Users] Compare $fU to drouting prefix In-Reply-To: <61a53c5b-3d70-4054-a837-3b9d70f135fd@opensips.org> References: <61a53c5b-3d70-4054-a837-3b9d70f135fd@opensips.org> Message-ID: Hi Bogdan, I will try to explain better. In rule_attrs I have a customer identifier which is used by acc to add the identifier into the CDR database. This works fine for calls from PSTN which are routed to another SIP gateway but calls from that gateway routed to PSTN can come from multiple customers and there is no way to identify which. So I'd like to match the incoming $fU to the rule that would match $rU in the from PSTN scenario in order to retrieve the rule_attrs (the customer identifier) from that rule. Does that make sense? Many thanks and regards Mark. On Thu, 16 Apr 2020 at 16:55, Bogdan-Andrei Iancu wrote: > Hi Mark, > > What kind of matching you want to do between $fU and the dr prefixes ? You > want to do the same as what drouting() does with $rU ? > > Regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > > On 4/16/20 6:14 PM, Mark Farmer wrote: > > Hi everyone > > I am looking for a way to compare $fU in INVITE to the matching drouting() > prefix of another group and retrieve the rule_attrs from that rule. > > At the moment I am thinking I'll have to run a custom DB query so I have 2 > questions: > > 1. Is there a better way to do this? > 2. If not, what is the best way to run custom DB queries? > > I have been reading through the drouting() documentation but that hasn't > helped. > > OpenSIPS 2.4.7 > > Many thanks! > Mark. > > > _______________________________________________ > Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > -- Mark Farmer farmorg at gmail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Fri Apr 17 08:15:33 2020 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Fri, 17 Apr 2020 11:15:33 +0300 Subject: [OpenSIPS-Users] Compare $fU to drouting prefix In-Reply-To: References: <61a53c5b-3d70-4054-a837-3b9d70f135fd@opensips.org> Message-ID: Hey Mark, It is not nice, but you can do: $var(tmp) = $rU; $rU = $fU do_routing(); $rU = $var(tmp); Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com On 4/17/20 11:13 AM, Mark Farmer wrote: > Hi Bogdan, I will try to explain better. > > In rule_attrs I have a customer identifier which is used by acc to add > the identifier into the CDR database. > This works fine for calls from PSTN which are routed to another SIP > gateway but calls from that gateway routed to PSTN can come from > multiple customers and there is no way to identify which. So I'd like > to match the incoming $fU to the rule that would match $rU in the from > PSTN scenario in order to retrieve the rule_attrs (the customer > identifier) from that rule. > > Does that make sense? > > Many thanks and regards > Mark. > > > > > On Thu, 16 Apr 2020 at 16:55, Bogdan-Andrei Iancu > wrote: > > Hi Mark, > > What kind of matching you want to do between $fU and the dr > prefixes ? You want to do the same as what drouting() does with $rU ? > > Regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > > On 4/16/20 6:14 PM, Mark Farmer wrote: >> Hi everyone >> >> I am looking for a way to compare $fU in INVITE to the matching >> drouting() prefix of another group and retrieve the rule_attrs >> from that rule. >> >> At the moment I am thinking I'll have to run a custom DB query so >> I have 2 questions: >> >> 1. Is there a better way to do this? >> 2. If not, what is the best way to run custom DB queries? >> >> I have been reading through the drouting() documentation but that >> hasn't helped. >> >> OpenSIPS 2.4.7 >> >> Many thanks! >> Mark. >> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > -- > Mark Farmer > farmorg at gmail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From farmorg at gmail.com Fri Apr 17 10:40:28 2020 From: farmorg at gmail.com (Mark Farmer) Date: Fri, 17 Apr 2020 11:40:28 +0100 Subject: [OpenSIPS-Users] Compare $fU to drouting prefix In-Reply-To: References: <61a53c5b-3d70-4054-a837-3b9d70f135fd@opensips.org> Message-ID: Thanks Bogdan, that's mostly working now. My issue now is with passing that identifier into acc_extra() as a variable which does not seem to be working. Using xlog() I can see that the variable is populated right before calling acc_extra() ... if (isflagset(PBX_PSTN) || isflagset(TPTY_PSTN)) { xlog("CUSTOM_LOG: Customer ID = $var(custID)"); $acc_extra(customer_id) = $var(custID); ... do_accounting("db","cdr"); } Does acc_extra() not accept variables as input? Thanks again! Mark. On Fri, 17 Apr 2020 at 09:15, Bogdan-Andrei Iancu wrote: > Hey Mark, > > It is not nice, but you can do: > > $var(tmp) = $rU; > $rU = $fU > do_routing(); > $rU = $var(tmp); > > Regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > > On 4/17/20 11:13 AM, Mark Farmer wrote: > > Hi Bogdan, I will try to explain better. > > In rule_attrs I have a customer identifier which is used by acc to add the > identifier into the CDR database. > This works fine for calls from PSTN which are routed to another SIP > gateway but calls from that gateway routed to PSTN can come from multiple > customers and there is no way to identify which. So I'd like to match the > incoming $fU to the rule that would match $rU in the from PSTN scenario in > order to retrieve the rule_attrs (the customer identifier) from that rule. > > Does that make sense? > > Many thanks and regards > Mark. > > > > > On Thu, 16 Apr 2020 at 16:55, Bogdan-Andrei Iancu > wrote: > >> Hi Mark, >> >> What kind of matching you want to do between $fU and the dr prefixes ? >> You want to do the same as what drouting() does with $rU ? >> >> Regards, >> >> Bogdan-Andrei Iancu >> >> OpenSIPS Founder and Developer >> https://www.opensips-solutions.com >> >> On 4/16/20 6:14 PM, Mark Farmer wrote: >> >> Hi everyone >> >> I am looking for a way to compare $fU in INVITE to the matching >> drouting() prefix of another group and retrieve the rule_attrs from that >> rule. >> >> At the moment I am thinking I'll have to run a custom DB query so I have >> 2 questions: >> >> 1. Is there a better way to do this? >> 2. If not, what is the best way to run custom DB queries? >> >> I have been reading through the drouting() documentation but that hasn't >> helped. >> >> OpenSIPS 2.4.7 >> >> Many thanks! >> Mark. >> >> >> _______________________________________________ >> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> >> > > -- > Mark Farmer > farmorg at gmail.com > > > -- Mark Farmer farmorg at gmail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From johan at democon.be Fri Apr 17 11:11:49 2020 From: johan at democon.be (johan) Date: Fri, 17 Apr 2020 13:11:49 +0200 Subject: [OpenSIPS-Users] Compare $fU to drouting prefix In-Reply-To: References: <61a53c5b-3d70-4054-a837-3b9d70f135fd@opensips.org> Message-ID: <13aad111-b785-a5ad-cc4b-bf2f428a9602@democon.be> you need to call accounting first. then you fill up your acc_extra stuff. accounting is only done on bye. so it might be that you need to call 1.8.4. |acc_db_request(comment, table)| wkr, Johan. On 17/04/2020 12:40, Mark Farmer wrote: > Thanks Bogdan, that's mostly working now. > > My issue now is with passing that identifier into acc_extra() as a > variable which does not seem to be working. > Using xlog() I can see that the variable is populated right before > calling acc_extra() > > ... > if (isflagset(PBX_PSTN) || isflagset(TPTY_PSTN)) { >                 xlog("CUSTOM_LOG: Customer ID = $var(custID)"); >                 $acc_extra(customer_id) = $var(custID); > ... > do_accounting("db","cdr"); > } > > Does acc_extra() not accept variables as input? > > Thanks again! > Mark. > > > > > On Fri, 17 Apr 2020 at 09:15, Bogdan-Andrei Iancu > wrote: > > Hey Mark, > > It is not nice, but you can do: > > $var(tmp) = $rU; > $rU = $fU > do_routing(); > $rU = $var(tmp); > > Regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > > On 4/17/20 11:13 AM, Mark Farmer wrote: >> Hi Bogdan, I will try to explain better. >> >> In rule_attrs I have a customer identifier which is used by acc >> to add the identifier into the CDR database. >> This works fine for calls from PSTN which are routed to another >> SIP gateway but calls from that gateway routed to PSTN can come >> from multiple customers and there is no way to identify which. So >> I'd like to match the incoming $fU to the rule that would match >> $rU in the from PSTN scenario in order to retrieve the rule_attrs >> (the customer identifier) from that rule. >> >> Does that make sense? >> >> Many thanks and regards >> Mark. >> >> >> >> >> On Thu, 16 Apr 2020 at 16:55, Bogdan-Andrei Iancu >> > wrote: >> >> Hi Mark, >> >> What kind of matching you want to do between $fU and the dr >> prefixes ? You want to do the same as what drouting() does >> with $rU ? >> >> Regards, >> >> Bogdan-Andrei Iancu >> >> OpenSIPS Founder and Developer >> https://www.opensips-solutions.com >> >> On 4/16/20 6:14 PM, Mark Farmer wrote: >>> Hi everyone >>> >>> I am looking for a way to compare $fU in INVITE to the >>> matching drouting() prefix of another group and retrieve the >>> rule_attrs from that rule. >>> >>> At the moment I am thinking I'll have to run a custom DB >>> query so I have 2 questions: >>> >>> 1. Is there a better way to do this? >>> 2. If not, what is the best way to run custom DB queries? >>> >>> I have been reading through the drouting() documentation but >>> that hasn't helped. >>> >>> OpenSIPS 2.4.7 >>> >>> Many thanks! >>> Mark. >>> >>> >>> _______________________________________________ >>> Users mailing list >>> Users at lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> >> >> -- >> Mark Farmer >> farmorg at gmail.com > > > > -- > Mark Farmer > farmorg at gmail.com > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From johan at democon.be Fri Apr 17 11:12:29 2020 From: johan at democon.be (johan) Date: Fri, 17 Apr 2020 13:12:29 +0200 Subject: [OpenSIPS-Users] Compare $fU to drouting prefix In-Reply-To: References: <61a53c5b-3d70-4054-a837-3b9d70f135fd@opensips.org> Message-ID: <8bef710d-1f7c-46c7-9aea-4c8271ac4671@democon.be> and if doesn't accept a var then copy the var to an avp. On 17/04/2020 12:40, Mark Farmer wrote: > Thanks Bogdan, that's mostly working now. > > My issue now is with passing that identifier into acc_extra() as a > variable which does not seem to be working. > Using xlog() I can see that the variable is populated right before > calling acc_extra() > > ... > if (isflagset(PBX_PSTN) || isflagset(TPTY_PSTN)) { >                 xlog("CUSTOM_LOG: Customer ID = $var(custID)"); >                 $acc_extra(customer_id) = $var(custID); > ... > do_accounting("db","cdr"); > } > > Does acc_extra() not accept variables as input? > > Thanks again! > Mark. > > > > > On Fri, 17 Apr 2020 at 09:15, Bogdan-Andrei Iancu > wrote: > > Hey Mark, > > It is not nice, but you can do: > > $var(tmp) = $rU; > $rU = $fU > do_routing(); > $rU = $var(tmp); > > Regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > > On 4/17/20 11:13 AM, Mark Farmer wrote: >> Hi Bogdan, I will try to explain better. >> >> In rule_attrs I have a customer identifier which is used by acc >> to add the identifier into the CDR database. >> This works fine for calls from PSTN which are routed to another >> SIP gateway but calls from that gateway routed to PSTN can come >> from multiple customers and there is no way to identify which. So >> I'd like to match the incoming $fU to the rule that would match >> $rU in the from PSTN scenario in order to retrieve the rule_attrs >> (the customer identifier) from that rule. >> >> Does that make sense? >> >> Many thanks and regards >> Mark. >> >> >> >> >> On Thu, 16 Apr 2020 at 16:55, Bogdan-Andrei Iancu >> > wrote: >> >> Hi Mark, >> >> What kind of matching you want to do between $fU and the dr >> prefixes ? You want to do the same as what drouting() does >> with $rU ? >> >> Regards, >> >> Bogdan-Andrei Iancu >> >> OpenSIPS Founder and Developer >> https://www.opensips-solutions.com >> >> On 4/16/20 6:14 PM, Mark Farmer wrote: >>> Hi everyone >>> >>> I am looking for a way to compare $fU in INVITE to the >>> matching drouting() prefix of another group and retrieve the >>> rule_attrs from that rule. >>> >>> At the moment I am thinking I'll have to run a custom DB >>> query so I have 2 questions: >>> >>> 1. Is there a better way to do this? >>> 2. If not, what is the best way to run custom DB queries? >>> >>> I have been reading through the drouting() documentation but >>> that hasn't helped. >>> >>> OpenSIPS 2.4.7 >>> >>> Many thanks! >>> Mark. >>> >>> >>> _______________________________________________ >>> Users mailing list >>> Users at lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> >> >> -- >> Mark Farmer >> farmorg at gmail.com > > > > -- > Mark Farmer > farmorg at gmail.com > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Fri Apr 17 12:50:08 2020 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Fri, 17 Apr 2020 15:50:08 +0300 Subject: [OpenSIPS-Users] Compare $fU to drouting prefix In-Reply-To: References: <61a53c5b-3d70-4054-a837-3b9d70f135fd@opensips.org> Message-ID: <9fc8685d-dffe-852f-5c85-0fddd5de3485@opensips.org> Mark, You can populate the $acc_extra() from whatever other variable or string operations. Most probably your issue is in other place, in regards to the acc logic. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com On 4/17/20 1:40 PM, Mark Farmer wrote: > Thanks Bogdan, that's mostly working now. > > My issue now is with passing that identifier into acc_extra() as a > variable which does not seem to be working. > Using xlog() I can see that the variable is populated right before > calling acc_extra() > > ... > if (isflagset(PBX_PSTN) || isflagset(TPTY_PSTN)) { >                 xlog("CUSTOM_LOG: Customer ID = $var(custID)"); >                 $acc_extra(customer_id) = $var(custID); > ... > do_accounting("db","cdr"); > } > > Does acc_extra() not accept variables as input? > > Thanks again! > Mark. > > > > > On Fri, 17 Apr 2020 at 09:15, Bogdan-Andrei Iancu > wrote: > > Hey Mark, > > It is not nice, but you can do: > > $var(tmp) = $rU; > $rU = $fU > do_routing(); > $rU = $var(tmp); > > Regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > > On 4/17/20 11:13 AM, Mark Farmer wrote: >> Hi Bogdan, I will try to explain better. >> >> In rule_attrs I have a customer identifier which is used by acc >> to add the identifier into the CDR database. >> This works fine for calls from PSTN which are routed to another >> SIP gateway but calls from that gateway routed to PSTN can come >> from multiple customers and there is no way to identify which. So >> I'd like to match the incoming $fU to the rule that would match >> $rU in the from PSTN scenario in order to retrieve the rule_attrs >> (the customer identifier) from that rule. >> >> Does that make sense? >> >> Many thanks and regards >> Mark. >> >> >> >> >> On Thu, 16 Apr 2020 at 16:55, Bogdan-Andrei Iancu >> > wrote: >> >> Hi Mark, >> >> What kind of matching you want to do between $fU and the dr >> prefixes ? You want to do the same as what drouting() does >> with $rU ? >> >> Regards, >> >> Bogdan-Andrei Iancu >> >> OpenSIPS Founder and Developer >> https://www.opensips-solutions.com >> >> On 4/16/20 6:14 PM, Mark Farmer wrote: >>> Hi everyone >>> >>> I am looking for a way to compare $fU in INVITE to the >>> matching drouting() prefix of another group and retrieve the >>> rule_attrs from that rule. >>> >>> At the moment I am thinking I'll have to run a custom DB >>> query so I have 2 questions: >>> >>> 1. Is there a better way to do this? >>> 2. If not, what is the best way to run custom DB queries? >>> >>> I have been reading through the drouting() documentation but >>> that hasn't helped. >>> >>> OpenSIPS 2.4.7 >>> >>> Many thanks! >>> Mark. >>> >>> >>> _______________________________________________ >>> Users mailing list >>> Users at lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> >> >> -- >> Mark Farmer >> farmorg at gmail.com > > > > -- > Mark Farmer > farmorg at gmail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From farmorg at gmail.com Fri Apr 17 13:08:24 2020 From: farmorg at gmail.com (Mark Farmer) Date: Fri, 17 Apr 2020 14:08:24 +0100 Subject: [OpenSIPS-Users] Compare $fU to drouting prefix In-Reply-To: <9fc8685d-dffe-852f-5c85-0fddd5de3485@opensips.org> References: <61a53c5b-3d70-4054-a837-3b9d70f135fd@opensips.org> <9fc8685d-dffe-852f-5c85-0fddd5de3485@opensips.org> Message-ID: Thats what I thought :) This is getting quite odd now. I am adding 2 extra fields, Call_Flow & customer_id. The odd thing is that Call_Flow is working perfectly in all cases. customer_id works fine for the latter 2 scenarios but in the first scenario the customer_id data is never added to the database. I have this configured for the acc module: modparam("acc", "extra_fields", "db: from_usr; to_usr; customer_id; Call_Flow") The variable is set earlier on: ... do_routing("3",,,"$var(custID)"); ... And I am doing all of the accounting in a dedicated route: route[ACCEXTRA] { do_accounting("db","cdr"); xlog("CUSTOM_LOG: Adding extra accounting: from_usr: $fU customer_id: $var(rule_attrs) $var(custID)"); *# variable is visible here* $acc_extra(from_usr) = $fU; $acc_extra(to_usr) = $tU; if (isflagset(PBX_PSTN) || isflagset(TPTY_PSTN)) { xlog("CUSTOM_LOG: Customer ID = $var(custID)"); *# variable is visible here* $acc_extra(Call_Flow) = "Outbound"; $acc_extra(customer_id) = $var(custID); xlog("CUSTOM_LOG: $$acc_extra(customer_id) = $acc_extra(customer_id)"); *# variable is visible here* } else if (isflagset(PSTN_TPTY) || isflagset(PSTN_PBX)) { $acc_extra(Call_Flow) = "Inbound"; $acc_extra(customer_id) = $var(rule_attrs); } else $acc_extra(Call_Flow) = "Internal"; $acc_extra(customer_id) = $var(rule_attrs); } On Fri, 17 Apr 2020 at 13:50, Bogdan-Andrei Iancu wrote: > Mark, > > You can populate the $acc_extra() from whatever other variable or string > operations. Most probably your issue is in other place, in regards to the > acc logic. > > Regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > > On 4/17/20 1:40 PM, Mark Farmer wrote: > > Thanks Bogdan, that's mostly working now. > > My issue now is with passing that identifier into acc_extra() as a > variable which does not seem to be working. > Using xlog() I can see that the variable is populated right before calling > acc_extra() > > ... > if (isflagset(PBX_PSTN) || isflagset(TPTY_PSTN)) { > xlog("CUSTOM_LOG: Customer ID = $var(custID)"); > $acc_extra(customer_id) = $var(custID); > ... > do_accounting("db","cdr"); > } > > Does acc_extra() not accept variables as input? > > Thanks again! > Mark. > > > > > On Fri, 17 Apr 2020 at 09:15, Bogdan-Andrei Iancu > wrote: > >> Hey Mark, >> >> It is not nice, but you can do: >> >> $var(tmp) = $rU; >> $rU = $fU >> do_routing(); >> $rU = $var(tmp); >> >> Regards, >> >> Bogdan-Andrei Iancu >> >> OpenSIPS Founder and Developer >> https://www.opensips-solutions.com >> >> On 4/17/20 11:13 AM, Mark Farmer wrote: >> >> Hi Bogdan, I will try to explain better. >> >> In rule_attrs I have a customer identifier which is used by acc to add >> the identifier into the CDR database. >> This works fine for calls from PSTN which are routed to another SIP >> gateway but calls from that gateway routed to PSTN can come from multiple >> customers and there is no way to identify which. So I'd like to match the >> incoming $fU to the rule that would match $rU in the from PSTN scenario in >> order to retrieve the rule_attrs (the customer identifier) from that rule. >> >> Does that make sense? >> >> Many thanks and regards >> Mark. >> >> >> >> >> On Thu, 16 Apr 2020 at 16:55, Bogdan-Andrei Iancu >> wrote: >> >>> Hi Mark, >>> >>> What kind of matching you want to do between $fU and the dr prefixes ? >>> You want to do the same as what drouting() does with $rU ? >>> >>> Regards, >>> >>> Bogdan-Andrei Iancu >>> >>> OpenSIPS Founder and Developer >>> https://www.opensips-solutions.com >>> >>> On 4/16/20 6:14 PM, Mark Farmer wrote: >>> >>> Hi everyone >>> >>> I am looking for a way to compare $fU in INVITE to the matching >>> drouting() prefix of another group and retrieve the rule_attrs from that >>> rule. >>> >>> At the moment I am thinking I'll have to run a custom DB query so I have >>> 2 questions: >>> >>> 1. Is there a better way to do this? >>> 2. If not, what is the best way to run custom DB queries? >>> >>> I have been reading through the drouting() documentation but that hasn't >>> helped. >>> >>> OpenSIPS 2.4.7 >>> >>> Many thanks! >>> Mark. >>> >>> >>> _______________________________________________ >>> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >>> >>> >> >> -- >> Mark Farmer >> farmorg at gmail.com >> >> >> > > -- > Mark Farmer > farmorg at gmail.com > > > -- Mark Farmer farmorg at gmail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From farmorg at gmail.com Fri Apr 17 14:58:46 2020 From: farmorg at gmail.com (Mark Farmer) Date: Fri, 17 Apr 2020 15:58:46 +0100 Subject: [OpenSIPS-Users] Compare $fU to drouting prefix In-Reply-To: References: <61a53c5b-3d70-4054-a837-3b9d70f135fd@opensips.org> <9fc8685d-dffe-852f-5c85-0fddd5de3485@opensips.org> Message-ID: OK, fixed it. Turned out to be this breaking it by overwriting $acc_extra(customer_id) with a blank value. ... else $acc_extra(Call_Flow) = "Internal"; $acc_extra(customer_id) = $var(rule_attrs); ... Changed it to: ... else if (isflagset(TPTY_PBX) || isflagset(PBX_TPTY)) { $acc_extra(Call_Flow) = "Internal"; $acc_extra(customer_id) = $var(rule_attrs); ... And all works nicely :) Thanks for the help! Mark. On Fri, 17 Apr 2020 at 14:08, Mark Farmer wrote: > Thats what I thought :) > > This is getting quite odd now. > I am adding 2 extra fields, Call_Flow & customer_id. The odd thing is that > Call_Flow is working perfectly in all cases. > customer_id works fine for the latter 2 scenarios but in the first > scenario the customer_id data is never added to the database. > > I have this configured for the acc module: > modparam("acc", "extra_fields", "db: from_usr; to_usr; customer_id; > Call_Flow") > > The variable is set earlier on: > ... > do_routing("3",,,"$var(custID)"); > ... > > And I am doing all of the accounting in a dedicated route: > > route[ACCEXTRA] { > do_accounting("db","cdr"); > xlog("CUSTOM_LOG: Adding extra accounting: from_usr: $fU > customer_id: $var(rule_attrs) $var(custID)"); *# variable is visible here* > $acc_extra(from_usr) = $fU; > $acc_extra(to_usr) = $tU; > if (isflagset(PBX_PSTN) || isflagset(TPTY_PSTN)) { > xlog("CUSTOM_LOG: Customer ID = $var(custID)"); *# > variable is visible here* > $acc_extra(Call_Flow) = "Outbound"; > $acc_extra(customer_id) = $var(custID); > xlog("CUSTOM_LOG: $$acc_extra(customer_id) = > $acc_extra(customer_id)"); *# variable is visible here* > } else if (isflagset(PSTN_TPTY) || isflagset(PSTN_PBX)) { > $acc_extra(Call_Flow) = "Inbound"; > $acc_extra(customer_id) = $var(rule_attrs); > } else $acc_extra(Call_Flow) = "Internal"; > $acc_extra(customer_id) = $var(rule_attrs); > } > > > On Fri, 17 Apr 2020 at 13:50, Bogdan-Andrei Iancu > wrote: > >> Mark, >> >> You can populate the $acc_extra() from whatever other variable or string >> operations. Most probably your issue is in other place, in regards to the >> acc logic. >> >> Regards, >> >> Bogdan-Andrei Iancu >> >> OpenSIPS Founder and Developer >> https://www.opensips-solutions.com >> >> On 4/17/20 1:40 PM, Mark Farmer wrote: >> >> Thanks Bogdan, that's mostly working now. >> >> My issue now is with passing that identifier into acc_extra() as a >> variable which does not seem to be working. >> Using xlog() I can see that the variable is populated right before >> calling acc_extra() >> >> ... >> if (isflagset(PBX_PSTN) || isflagset(TPTY_PSTN)) { >> xlog("CUSTOM_LOG: Customer ID = $var(custID)"); >> $acc_extra(customer_id) = $var(custID); >> ... >> do_accounting("db","cdr"); >> } >> >> Does acc_extra() not accept variables as input? >> >> Thanks again! >> Mark. >> >> >> >> >> On Fri, 17 Apr 2020 at 09:15, Bogdan-Andrei Iancu >> wrote: >> >>> Hey Mark, >>> >>> It is not nice, but you can do: >>> >>> $var(tmp) = $rU; >>> $rU = $fU >>> do_routing(); >>> $rU = $var(tmp); >>> >>> Regards, >>> >>> Bogdan-Andrei Iancu >>> >>> OpenSIPS Founder and Developer >>> https://www.opensips-solutions.com >>> >>> On 4/17/20 11:13 AM, Mark Farmer wrote: >>> >>> Hi Bogdan, I will try to explain better. >>> >>> In rule_attrs I have a customer identifier which is used by acc to add >>> the identifier into the CDR database. >>> This works fine for calls from PSTN which are routed to another SIP >>> gateway but calls from that gateway routed to PSTN can come from multiple >>> customers and there is no way to identify which. So I'd like to match the >>> incoming $fU to the rule that would match $rU in the from PSTN scenario in >>> order to retrieve the rule_attrs (the customer identifier) from that rule. >>> >>> Does that make sense? >>> >>> Many thanks and regards >>> Mark. >>> >>> >>> >>> >>> On Thu, 16 Apr 2020 at 16:55, Bogdan-Andrei Iancu >>> wrote: >>> >>>> Hi Mark, >>>> >>>> What kind of matching you want to do between $fU and the dr prefixes ? >>>> You want to do the same as what drouting() does with $rU ? >>>> >>>> Regards, >>>> >>>> Bogdan-Andrei Iancu >>>> >>>> OpenSIPS Founder and Developer >>>> https://www.opensips-solutions.com >>>> >>>> On 4/16/20 6:14 PM, Mark Farmer wrote: >>>> >>>> Hi everyone >>>> >>>> I am looking for a way to compare $fU in INVITE to the matching >>>> drouting() prefix of another group and retrieve the rule_attrs from that >>>> rule. >>>> >>>> At the moment I am thinking I'll have to run a custom DB query so I >>>> have 2 questions: >>>> >>>> 1. Is there a better way to do this? >>>> 2. If not, what is the best way to run custom DB queries? >>>> >>>> I have been reading through the drouting() documentation but that >>>> hasn't helped. >>>> >>>> OpenSIPS 2.4.7 >>>> >>>> Many thanks! >>>> Mark. >>>> >>>> >>>> _______________________________________________ >>>> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>> >>>> >>>> >>> >>> -- >>> Mark Farmer >>> farmorg at gmail.com >>> >>> >>> >> >> -- >> Mark Farmer >> farmorg at gmail.com >> >> >> > > -- > Mark Farmer > farmorg at gmail.com > -- Mark Farmer farmorg at gmail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From podguiko at mail.ru Fri Apr 17 19:43:18 2020 From: podguiko at mail.ru (=?UTF-8?B?T2xlZyBQb2RndXlrbw==?=) Date: Fri, 17 Apr 2020 22:43:18 +0300 Subject: [OpenSIPS-Users] =?utf-8?q?choice_of_AS_architecture?= In-Reply-To: References: <1586870941.864026772@f395.i.mail.ru> Message-ID: <1587152598.815633507@f301.i.mail.ru> We looked at the seas for a reason. It seemed to us that was a good idea is when the SIP protocol is decoded only once and further work is already underway with this data. This approach, it seems to us, allows us to significantly increase the processing speed and avoid unnecessary encoding of such a message back to SIP for transmission to AS.   In our current scheme, part of the logic runs on the http server. Which we work with via http_rest_client. We send part of the SIP headers to the http server, and to do this, we encode it all in json, then decode the response from the http server, and encode it in SIP. the HTTP server essentially performs the role of AS. Everything works, but the load will grow. And we would like to have a light tool for working with SIP messages, without overhead such as SIP to JSON, JSON to SIP and so on. Our opensips are already processing 400-500 CAPS. We periodically have problems (segfault). We even opened a ticket about this.   We are currently searching on the cscf + AS architecture. And we are trying to use the experience we have already gained We think that there should be an idea of seas or maybe the possibility of using any other programming languages for some logic except lUA, for example python or something else to build AS next to opensips.  It would be great also to have a mechanism where you can make changes to configs without restarting all opensips. >Четверг, 16 апреля 2020, 14:00 +03:00 от Bogdan-Andrei Iancu : >  >Hey Oleg, > >What SIP features/capabilities do you want to have on the AS component? IMHO this is the point to start - based on your own requirements on what the AS must do, you can do a proper selection. > >And yes, as Tito said, seas is dead fish in the water for a long time. > >Regards, >Bogdan-Andrei Iancu > >OpenSIPS Founder and Developer > https://www.opensips-solutions.com > > >On 4/14/20 4:29 PM, Oleg Podguyko via Users wrote: >>Hi Bogdan and all your team! >>  >>We have been working with opensips for quite some time and we like it. We keep the load on the order of 300-400 CAPS, and for example, the SIP-I module is perhaps the only solution that allows you to fully work with ISUP. Our programmers prepared several PRs based on the results of work with SIP-I and SCTP modules. >>We are faced with the task of creating a classic bundle of cscf + AS server, in the future architecture opensips will play the role of cscf, and as AS we look at the module SEAS + external server. >> >>We are all moving towards IMS. >>  >>What do you think, how justified can such a choice be? Perhaps the opensips team already has its own new vision for this? >>What would you choose as AS? >>  >>  >>-- >>Oleg Podguyko   >>  >>_______________________________________________ >>Users mailing list >>Users at lists.opensips.org >>http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>     -- Олег Подгуйко   -------------- next part -------------- An HTML attachment was scrubbed... URL: From mark at markmiranda.com Fri Apr 17 19:53:50 2020 From: mark at markmiranda.com (Mark Miranda) Date: Fri, 17 Apr 2020 14:53:50 -0500 Subject: [OpenSIPS-Users] OpenSIPS as Teams SBC RTP->SRTP Question Message-ID: <524FBEF1-0042-48B4-A97C-6C4E119A4A38@markmiranda.com> I have configured OpenSIPS to act as an SBC for MS Teams. My voip provider (voip.ms) supports TLS, so my outgoing calls from MS Teams work great. On the incoming side though, voip.ms does not support TLS 1.2 so I’m just using TCP or UDP unsecured. Of course when the call goes out to the MS Teams user, Teams replies with 488 Not Acceptable Here and I assume this is because it is trying to use RTP and not SRTP. I see that there is an rtpengine module for OpenSIPS but I could not find a good example of how this could be used to encrypt the RTP packets as it passes to teams? Anyone have a working example of this? Or even better, something already working with Teams? Mark From tito at xsvoce.com Fri Apr 17 20:43:10 2020 From: tito at xsvoce.com (Tito Cumpen) Date: Fri, 17 Apr 2020 13:43:10 -0700 Subject: [OpenSIPS-Users] Tls using t_relay Message-ID: Hello, I am attempting to use t_relay("tls:domain:port") but I am not having much success with it on 2.4. opensips sends a syn to the peer then gets an syn ack and sends a rst . The logs claim that the send failed but it never opened the socket entirely and did not send the client hello at all. Are there any other configs to be considered when attempting this ? -------------- next part -------------- An HTML attachment was scrubbed... URL: From volga629 at networklab.ca Sat Apr 18 01:38:33 2020 From: volga629 at networklab.ca (volga629) Date: Fri, 17 Apr 2020 22:38:33 -0300 Subject: [OpenSIPS-Users] ms teams ACK Message-ID: <94a0f640-5400-9124-0a98-96556fa000b6@skillsearch.ca> An HTML attachment was scrubbed... URL: From callum.guy at x-on.co.uk Sat Apr 18 07:51:39 2020 From: callum.guy at x-on.co.uk (Callum Guy) Date: Sat, 18 Apr 2020 08:51:39 +0100 Subject: [OpenSIPS-Users] Compare $fU to drouting prefix In-Reply-To: References: <61a53c5b-3d70-4054-a837-3b9d70f135fd@opensips.org> <9fc8685d-dffe-852f-5c85-0fddd5de3485@opensips.org> Message-ID: That'll do it.. On Fri, 17 Apr 2020 at 16:00, Mark Farmer wrote: > OK, fixed it. > > Turned out to be this breaking it by overwriting $acc_extra(customer_id) > with a blank value. > > ... > else $acc_extra(Call_Flow) = "Internal"; > $acc_extra(customer_id) = $var(rule_attrs); > ... > > Changed it to: > > ... > else if (isflagset(TPTY_PBX) || isflagset(PBX_TPTY)) { > $acc_extra(Call_Flow) = "Internal"; > $acc_extra(customer_id) = $var(rule_attrs); > ... > > And all works nicely :) > > Thanks for the help! > Mark. > > > On Fri, 17 Apr 2020 at 14:08, Mark Farmer wrote: > >> Thats what I thought :) >> >> This is getting quite odd now. >> I am adding 2 extra fields, Call_Flow & customer_id. The odd thing is >> that Call_Flow is working perfectly in all cases. >> customer_id works fine for the latter 2 scenarios but in the first >> scenario the customer_id data is never added to the database. >> >> I have this configured for the acc module: >> modparam("acc", "extra_fields", "db: from_usr; to_usr; customer_id; >> Call_Flow") >> >> The variable is set earlier on: >> ... >> do_routing("3",,,"$var(custID)"); >> ... >> >> And I am doing all of the accounting in a dedicated route: >> >> route[ACCEXTRA] { >> do_accounting("db","cdr"); >> xlog("CUSTOM_LOG: Adding extra accounting: from_usr: $fU >> customer_id: $var(rule_attrs) $var(custID)"); *# variable is >> visible here* >> $acc_extra(from_usr) = $fU; >> $acc_extra(to_usr) = $tU; >> if (isflagset(PBX_PSTN) || isflagset(TPTY_PSTN)) { >> xlog("CUSTOM_LOG: Customer ID = $var(custID)"); *# >> variable is visible here* >> $acc_extra(Call_Flow) = "Outbound"; >> $acc_extra(customer_id) = $var(custID); >> xlog("CUSTOM_LOG: $$acc_extra(customer_id) = >> $acc_extra(customer_id)"); *# variable is visible here* >> } else if (isflagset(PSTN_TPTY) || isflagset(PSTN_PBX)) { >> $acc_extra(Call_Flow) = "Inbound"; >> $acc_extra(customer_id) = $var(rule_attrs); >> } else $acc_extra(Call_Flow) = "Internal"; >> $acc_extra(customer_id) = $var(rule_attrs); >> } >> >> >> On Fri, 17 Apr 2020 at 13:50, Bogdan-Andrei Iancu >> wrote: >> >>> Mark, >>> >>> You can populate the $acc_extra() from whatever other variable or string >>> operations. Most probably your issue is in other place, in regards to the >>> acc logic. >>> >>> Regards, >>> >>> Bogdan-Andrei Iancu >>> >>> OpenSIPS Founder and Developer >>> https://www.opensips-solutions.com >>> >>> On 4/17/20 1:40 PM, Mark Farmer wrote: >>> >>> Thanks Bogdan, that's mostly working now. >>> >>> My issue now is with passing that identifier into acc_extra() as a >>> variable which does not seem to be working. >>> Using xlog() I can see that the variable is populated right before >>> calling acc_extra() >>> >>> ... >>> if (isflagset(PBX_PSTN) || isflagset(TPTY_PSTN)) { >>> xlog("CUSTOM_LOG: Customer ID = $var(custID)"); >>> $acc_extra(customer_id) = $var(custID); >>> ... >>> do_accounting("db","cdr"); >>> } >>> >>> Does acc_extra() not accept variables as input? >>> >>> Thanks again! >>> Mark. >>> >>> >>> >>> >>> On Fri, 17 Apr 2020 at 09:15, Bogdan-Andrei Iancu >>> wrote: >>> >>>> Hey Mark, >>>> >>>> It is not nice, but you can do: >>>> >>>> $var(tmp) = $rU; >>>> $rU = $fU >>>> do_routing(); >>>> $rU = $var(tmp); >>>> >>>> Regards, >>>> >>>> Bogdan-Andrei Iancu >>>> >>>> OpenSIPS Founder and Developer >>>> https://www.opensips-solutions.com >>>> >>>> On 4/17/20 11:13 AM, Mark Farmer wrote: >>>> >>>> Hi Bogdan, I will try to explain better. >>>> >>>> In rule_attrs I have a customer identifier which is used by acc to add >>>> the identifier into the CDR database. >>>> This works fine for calls from PSTN which are routed to another SIP >>>> gateway but calls from that gateway routed to PSTN can come from multiple >>>> customers and there is no way to identify which. So I'd like to match the >>>> incoming $fU to the rule that would match $rU in the from PSTN scenario in >>>> order to retrieve the rule_attrs (the customer identifier) from that rule. >>>> >>>> Does that make sense? >>>> >>>> Many thanks and regards >>>> Mark. >>>> >>>> >>>> >>>> >>>> On Thu, 16 Apr 2020 at 16:55, Bogdan-Andrei Iancu >>>> wrote: >>>> >>>>> Hi Mark, >>>>> >>>>> What kind of matching you want to do between $fU and the dr prefixes ? >>>>> You want to do the same as what drouting() does with $rU ? >>>>> >>>>> Regards, >>>>> >>>>> Bogdan-Andrei Iancu >>>>> >>>>> OpenSIPS Founder and Developer >>>>> https://www.opensips-solutions.com >>>>> >>>>> On 4/16/20 6:14 PM, Mark Farmer wrote: >>>>> >>>>> Hi everyone >>>>> >>>>> I am looking for a way to compare $fU in INVITE to the matching >>>>> drouting() prefix of another group and retrieve the rule_attrs from that >>>>> rule. >>>>> >>>>> At the moment I am thinking I'll have to run a custom DB query so I >>>>> have 2 questions: >>>>> >>>>> 1. Is there a better way to do this? >>>>> 2. If not, what is the best way to run custom DB queries? >>>>> >>>>> I have been reading through the drouting() documentation but that >>>>> hasn't helped. >>>>> >>>>> OpenSIPS 2.4.7 >>>>> >>>>> Many thanks! >>>>> Mark. >>>>> >>>>> >>>>> _______________________________________________ >>>>> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>>> >>>>> >>>>> >>>> >>>> -- >>>> Mark Farmer >>>> farmorg at gmail.com >>>> >>>> >>>> >>> >>> -- >>> Mark Farmer >>> farmorg at gmail.com >>> >>> >>> >> >> -- >> Mark Farmer >> farmorg at gmail.com >> > > > -- > Mark Farmer > farmorg at gmail.com > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -- *0333 332 0000  |  x-on.co.uk   |   **      **  |  Coronavirus * THE ITSPA AWARDS 2020 AND Best ITSP - Mid Market, Best Software and Best Vertical Solution are trade marks of the Internet Telephony Services Providers' Association, used under licence. 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URL: From johan at democon.be Sat Apr 18 08:54:58 2020 From: johan at democon.be (Johan De Clercq) Date: Sat, 18 Apr 2020 08:54:58 +0000 Subject: [OpenSIPS-Users] ms teams ACK In-Reply-To: <94a0f640-5400-9124-0a98-96556fa000b6@skillsearch.ca> References: <94a0f640-5400-9124-0a98-96556fa000b6@skillsearch.ca> Message-ID: Please find the necessary manips in this doc. https://www.oracle.com/webfolder/technetwork/acmepacket/Microsoft/SBC-MSFTTeams-NON-MB.pdf https://blog.opensips.org/2019/09/16/opensips-as-ms-teams-sbc/ Outlook voor iOS downloaden ________________________________ Van: Users namens volga629 via Users Verzonden: Saturday, April 18, 2020 3:38:33 AM Aan: OpenSIPS users mailling list Onderwerp: [OpenSIPS-Users] ms teams ACK Hello Everyone, Is possible rewrite ACK contact header in dialog ? My guess it expecting FQDN. MS Teams disconnect the call after 20 sec REASON: Q.850;cause=18;text="bc427610-edae-47b9-9daa-7ea74d40dcc7;Call Controller timed out while waiting for acknowledgement." In this case ACK come from asterisk box. Opensips Log: /usr/sbin/opensips[3321]: [IN-DIALOG] [ACK] Contact header ~> [ volga629 -------------- next part -------------- An HTML attachment was scrubbed... URL: From john.quick at smartvox.co.uk Sat Apr 18 14:26:19 2020 From: john.quick at smartvox.co.uk (John Quick) Date: Sat, 18 Apr 2020 15:26:19 +0100 Subject: [OpenSIPS-Users] OpenSIPS as Teams SBC RTP->SRTP Question Message-ID: <000401d6158d$543156e0$fc9404a0$@smartvox.co.uk> I have written a couple of articles which, between them, should help you with this question. The first article looks at WebRTC <--> SIP using rtpengine: https://kb.smartvox.co.uk/opensips/webrtc-using-opensips-and-rtpengine/ The other one discusses how you configure OpenSIPS 2.2.x for TLS: https://kb.smartvox.co.uk/opensips/using-tls-in-opensips-v2-2-x/ ..in the third paragraph it mentions about SRTP-RTP transcoding with a cross reference to the first article and with a note about how to adjust the parameters sent to rtpengine so they will work for SRTP (SAVP) instead of WebRTC (SAVPF). So together it should provide you with the examples and explanations you seek. I have been working on setting up a Teams SBC over the last 2 weeks. I have it working with calls in both directions and it is using rtpengine exactly as described in my two articles. Have you read the blog/knowledgebase article on the opensips web site about Teams SBC? https://blog.opensips.org/2019/09/16/opensips-as-ms-teams-sbc/ John Quick Smartvox Limited Web: www.smartvox.co.uk >> I see that there is an rtpengine module for OpenSIPS but I could not find a good example of how this could be used to encrypt the RTP packets as it passes to teams? >> >> Anyone have a working example of this? Or even better, something already working with Teams? From volga629 at networklab.ca Sat Apr 18 19:25:37 2020 From: volga629 at networklab.ca (volga629) Date: Sat, 18 Apr 2020 16:25:37 -0300 Subject: [OpenSIPS-Users] ms teams ACK In-Reply-To: References: <94a0f640-5400-9124-0a98-96556fa000b6@skillsearch.ca> Message-ID: <6f80948f-fa0a-696f-eb11-2a46e2f49591@skillsearch.ca> An HTML attachment was scrubbed... URL: From osas at voipembedded.com Sat Apr 18 19:33:06 2020 From: osas at voipembedded.com (Ovidiu Sas) Date: Sat, 18 Apr 2020 15:33:06 -0400 Subject: [OpenSIPS-Users] ms teams ACK In-Reply-To: <6f80948f-fa0a-696f-eb11-2a46e2f49591@skillsearch.ca> References: <94a0f640-5400-9124-0a98-96556fa000b6@skillsearch.ca> <6f80948f-fa0a-696f-eb11-2a46e2f49591@skillsearch.ca> Message-ID: You don't need to mess with the Contact header (unless you are connecting NATed endpoints with MS servers). You need to populate proper Record-Route headers in the initial INVITE as explained int the blog: https://blog.opensips.org/2019/09/16/opensips-as-ms-teams-sbc/ -ovidiu On Sat, Apr 18, 2020 at 3:26 PM volga629 via Users wrote: > > Thank you for reply, > > I fixed Contact header in ACK, but Microsoft still unhappy call drops after 20 sec. > > ACK debug > > https://paste.centos.org/view/21b816d1 > > volga629 > > On 4/18/20 5:54 AM, Johan De Clercq wrote: > > Please find the necessary manips in this doc. > > https://www.oracle.com/webfolder/technetwork/acmepacket/Microsoft/SBC-MSFTTeams-NON-MB.pdf > > https://blog.opensips.org/2019/09/16/opensips-as-ms-teams-sbc/ > > > Outlook voor iOS downloaden > ________________________________ > Van: Users namens volga629 via Users > Verzonden: Saturday, April 18, 2020 3:38:33 AM > Aan: OpenSIPS users mailling list > Onderwerp: [OpenSIPS-Users] ms teams ACK > > > Hello Everyone, > > Is possible rewrite ACK contact header in dialog ? > > My guess it expecting FQDN. > > MS Teams disconnect the call after 20 sec > > REASON: Q.850;cause=18;text="bc427610-edae-47b9-9daa-7ea74d40dcc7;Call Controller timed out while waiting for acknowledgement." > > In this case ACK come from asterisk box. > > Opensips Log: > > /usr/sbin/opensips[3321]: [IN-DIALOG] [ACK] Contact header ~> [ > > volga629 > > _______________________________________________ > 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 volga629 at networklab.ca Sat Apr 18 19:43:57 2020 From: volga629 at networklab.ca (volga629) Date: Sat, 18 Apr 2020 16:43:57 -0300 Subject: [OpenSIPS-Users] ms teams ACK In-Reply-To: References: <94a0f640-5400-9124-0a98-96556fa000b6@skillsearch.ca> <6f80948f-fa0a-696f-eb11-2a46e2f49591@skillsearch.ca> Message-ID: <0c16126f-931a-d8a6-5aaa-07a67a8727d1@skillsearch.ca> An HTML attachment was scrubbed... URL: From alexei.vasilyev at gmail.com Sat Apr 18 20:13:54 2020 From: alexei.vasilyev at gmail.com (Alexey Vasilyev) Date: Sat, 18 Apr 2020 13:13:54 -0700 (MST) Subject: [OpenSIPS-Users] ms teams ACK In-Reply-To: <0c16126f-931a-d8a6-5aaa-07a67a8727d1@skillsearch.ca> References: <94a0f640-5400-9124-0a98-96556fa000b6@skillsearch.ca> <6f80948f-fa0a-696f-eb11-2a46e2f49591@skillsearch.ca> <0c16126f-931a-d8a6-5aaa-07a67a8727d1@skillsearch.ca> Message-ID: <1587240834447-0.post@n2.nabble.com> Hi volga629, There were nothing special for ACK. You don't need to change To/From/Contact. All the necessary steps were in the article https://blog.opensips.org/2019/09/16/opensips-as-ms-teams-sbc/ and for most people it still works. So I'm not sure, that MS changed anything, because all the hardware SBCs should change behaviour, so they need new firmware. SBC vendors should inform customers to update etc. So this is not so simple process. And it definitely make no sense for anybody. And in the test lab for the article I've used absolutely the same architecture with asterisk, the only difference was RTPEngine to transcode SRTP-RTP. And within test lab I've tested not only calls, but transfers worked fine too. ----- --- Alexey Vasilyev -- Sent from: http://opensips-open-sip-server.1449251.n2.nabble.com/OpenSIPS-Users-f1449235.html From volga629 at networklab.ca Sat Apr 18 21:01:19 2020 From: volga629 at networklab.ca (volga629) Date: Sat, 18 Apr 2020 18:01:19 -0300 Subject: [OpenSIPS-Users] ms teams ACK In-Reply-To: <1587240834447-0.post@n2.nabble.com> References: <94a0f640-5400-9124-0a98-96556fa000b6@skillsearch.ca> <6f80948f-fa0a-696f-eb11-2a46e2f49591@skillsearch.ca> <0c16126f-931a-d8a6-5aaa-07a67a8727d1@skillsearch.ca> <1587240834447-0.post@n2.nabble.com> Message-ID: <4e26f956-8513-3863-8ced-67bcac216861@skillsearch.ca> An HTML attachment was scrubbed... URL: From johan at democon.be Sun Apr 19 06:04:51 2020 From: johan at democon.be (Johan De Clercq) Date: Sun, 19 Apr 2020 06:04:51 +0000 Subject: [OpenSIPS-Users] ms teams ACK In-Reply-To: <4e26f956-8513-3863-8ced-67bcac216861@skillsearch.ca> References: <94a0f640-5400-9124-0a98-96556fa000b6@skillsearch.ca> <6f80948f-fa0a-696f-eb11-2a46e2f49591@skillsearch.ca> <0c16126f-931a-d8a6-5aaa-07a67a8727d1@skillsearch.ca> <1587240834447-0.post@n2.nabble.com>, <4e26f956-8513-3863-8ced-67bcac216861@skillsearch.ca> Message-ID: Can’t it be a NAT problem? The IP address where the bye is coming from doesn’t seem a pstnhub to me. Outlook voor iOS downloaden ________________________________ Van: Users namens volga629 via Users Verzonden: Saturday, April 18, 2020 11:01:19 PM Aan: OpenSIPS users mailling list ; Alexey Vasilyev Onderwerp: Re: [OpenSIPS-Users] ms teams ACK Hello Alexey, Thank you on reply, I undone all changes regard headers changes and MS Teams send BYE directly to asterisk. No Route header present. But INVITE ACK 183 180 all travel with proper routing information. 2020/04/18 17:54:28.599711 190.109.70.77:5060 -> 190.109.68.250:5060 BYE sip:11988582770 at 190.109.68.250:5060 SIP/2.0 FROM: ;tag=4d7fb0763c224e39a13a03c669c4b387 TO: ;tag=as41e97ff5 CSEQ: 3 BYE CALL-ID: 2e6c1a8d2383a4752403e94512ced077 at 190.109.70.77 MAX-FORWARDS: 69 Via: SIP/2.0/UDP 190.109.70.77:5060;branch=z9hG4bK050e.e400e373.0;i=66c9c603 VIA: SIP/2.0/TLS 52.114.14.70:5061;rport=8208;received=52.114.14.70;branch=z9hG4bK9594cd7 REASON: Q.850;cause=18;text="fcb37a2a-4bc4-49b6-a5e3-aabddc8f7a22;Call Controller timed out while waiting for acknowledgement." CONTACT: CONTENT-LENGTH: 0 USER-AGENT: Microsoft.PSTNHub.SIPProxy v.2020.4.13.7 i.ASSE.3 ALLOW: INVITE,ACK,OPTIONS,CANCEL,BYE,NOTIFY volga629 On 4/18/20 5:13 PM, Alexey Vasilyev wrote: Hi volga629, There were nothing special for ACK. You don't need to change To/From/Contact. All the necessary steps were in the article https://blog.opensips.org/2019/09/16/opensips-as-ms-teams-sbc/ and for most people it still works. So I'm not sure, that MS changed anything, because all the hardware SBCs should change behaviour, so they need new firmware. SBC vendors should inform customers to update etc. So this is not so simple process. And it definitely make no sense for anybody. And in the test lab for the article I've used absolutely the same architecture with asterisk, the only difference was RTPEngine to transcode SRTP-RTP. And within test lab I've tested not only calls, but transfers worked fine too. ----- --- Alexey Vasilyev -- Sent from: http://opensips-open-sip-server.1449251.n2.nabble.com/OpenSIPS-Users-f1449235.html _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From volga629 at networklab.ca Sun Apr 19 12:40:02 2020 From: volga629 at networklab.ca (volga629) Date: Sun, 19 Apr 2020 09:40:02 -0300 Subject: [OpenSIPS-Users] ms teams ACK In-Reply-To: References: <94a0f640-5400-9124-0a98-96556fa000b6@skillsearch.ca> <6f80948f-fa0a-696f-eb11-2a46e2f49591@skillsearch.ca> <0c16126f-931a-d8a6-5aaa-07a67a8727d1@skillsearch.ca> <1587240834447-0.post@n2.nabble.com> <4e26f956-8513-3863-8ced-67bcac216861@skillsearch.ca> Message-ID: <76b1f17e-21c7-8ccb-d5c9-abe54701d564@skillsearch.ca> An HTML attachment was scrubbed... URL: From nnikeshala at yahoo.com Sun Apr 19 20:50:41 2020 From: nnikeshala at yahoo.com (Nayani Nikeshala) Date: Sun, 19 Apr 2020 20:50:41 +0000 (UTC) Subject: [OpenSIPS-Users] OpenSIPS installation with PostgresSQL In-Reply-To: <129541852.2099756.1587328682918@mail.yahoo.com> References: <129541852.2099756.1587328682918.ref@mail.yahoo.com> <129541852.2099756.1587328682918@mail.yahoo.com> Message-ID: <1985829642.2102360.1587329441034@mail.yahoo.com> Hello, I'm trying to install  opensips-2.4.7 with psql -8.4.20 on a Centos-6 platform. I can connect to psql DB as below. [root at SIPserver ~]# psql -h localhost -U opensips -d opensipspsql (8.4.20)Type "help" for help. opensips=>  When I try to start OpenSIP as below, it gives me the following error.  [root at SIPserver ~]# opensipsctl start INFO: Starting OpenSIPS : ERROR: PID file /var/run/opensips.pid does not exist -- OpenSIPS start failed[root at SIPserver ~]# /var/log/opensips.log shows the below error.  Apr 20 01:31:20 SIPserver opensips: ERROR:core:sr_load_module: could not open module : /usr/local//lib/opensips/modules/db_postgres.so: undefined symbol: PQconnectdbParamsApr 20 01:31:20 SIPserver opensips: ERROR:core:load_module: failed to load moduleApr 20 01:31:20 SIPserver opensips: CRITICAL:core:yyerror: parse error in config file /usr/local//etc/opensips/opensips.cfg, line 89, column 13-14: failed to load module db_postgres.so#012 I noted that PQconnectdbParams is available from psql-9 onwards and could not figure out, what does this error mean.  Could someone assist me please.  Thanks! -------------- next part -------------- An HTML attachment was scrubbed... URL: From john.quick at smartvox.co.uk Mon Apr 20 08:51:01 2020 From: john.quick at smartvox.co.uk (John Quick) Date: Mon, 20 Apr 2020 09:51:01 +0100 Subject: [OpenSIPS-Users] ms teams ACK Message-ID: <000601d616f0$d165cc40$743164c0$@smartvox.co.uk> Johan, Check that you're not calling fix_nated_contact() in the onreply_route that handles the 200 OK from Teams Proxy. You must not fix the Contact in the 200 OK because then the ACK from your end will be mis-routed. John Quick Smartvox Limited Web: www.smartvox.co.uk From johan at democon.be Mon Apr 20 08:52:20 2020 From: johan at democon.be (Johan De Clercq) Date: Mon, 20 Apr 2020 08:52:20 +0000 Subject: [OpenSIPS-Users] ms teams ACK In-Reply-To: <000601d616f0$d165cc40$743164c0$@smartvox.co.uk> References: <000601d616f0$d165cc40$743164c0$@smartvox.co.uk> Message-ID: Thanks for the response John, but it is for Volga. Outlook voor iOS downloaden ________________________________ Van: John Quick Verzonden: Monday, April 20, 2020 10:51:01 AM Aan: users at lists.opensips.org ; 'Johan De Clercq' Onderwerp: Re: [OpenSIPS-Users] ms teams ACK Johan, Check that you're not calling fix_nated_contact() in the onreply_route that handles the 200 OK from Teams Proxy. You must not fix the Contact in the 200 OK because then the ACK from your end will be mis-routed. John Quick Smartvox Limited Web: www.smartvox.co.uk -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Mon Apr 20 14:32:08 2020 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Mon, 20 Apr 2020 17:32:08 +0300 Subject: [OpenSIPS-Users] Tls using t_relay In-Reply-To: References: Message-ID: <39b60bcc-945b-f659-ae7f-37f82327c63c@opensips.org> Hi Tito, You say OpenSIPS is sending a RESET without being any data exchanged on the connection? What are the logs for the failed send ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com On 4/17/20 11:43 PM, Tito Cumpen wrote: > Hello, > > I am attempting to use t_relay("tls:domain:port") but I am not having > much success with it on 2.4. opensips sends a syn  to the peer then > gets an syn ack and sends a rst . The logs claim that the send failed > but it never opened the socket entirely and did not send the client > hello at all. > > Are there any other configs to be considered when attempting this ? > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Mon Apr 20 14:33:58 2020 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Mon, 20 Apr 2020 17:33:58 +0300 Subject: [OpenSIPS-Users] OpenSIPS installation with PostgresSQL In-Reply-To: <1985829642.2102360.1587329441034@mail.yahoo.com> References: <129541852.2099756.1587328682918.ref@mail.yahoo.com> <129541852.2099756.1587328682918@mail.yahoo.com> <1985829642.2102360.1587329441034@mail.yahoo.com> Message-ID: <7812bb25-eef4-0b34-04f5-a4f5c5ba589f@opensips.org> Hi, Is OpenSIPS compiled by you or installed from packages (if yes, what is the repo url)? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com On 4/19/20 11:50 PM, Nayani Nikeshala via Users wrote: > Hello, > > I'm trying to install opensips-2.4.7 with psql -8.4.20 on a Centos-6 > platform. I can connect to psql DB as below. > > [root at SIPserver ~]# psql -h localhost -U opensips -d opensips > psql (8.4.20) > Type "help" for help. > > opensips=> > > When I try to start OpenSIP as below, it gives me the following error. > > [root at SIPserver ~]# opensipsctl start > > INFO: Starting OpenSIPS : > > ERROR: PID file /var/run/opensips.pid does not exist -- OpenSIPS start > failed > [root at SIPserver ~]# > > > /var/log/opensips.log shows the below error. > > Apr 20 01:31:20 SIPserver opensips: ERROR:core:sr_load_module: could > not open module : > /usr/local//lib/opensips/modules/db_postgres.so: undefined symbol: > PQconnectdbParams > Apr 20 01:31:20 SIPserver opensips: ERROR:core:load_module: failed to > load module > Apr 20 01:31:20 SIPserver opensips: CRITICAL:core:yyerror: parse error > in config file /usr/local//etc/opensips/opensips.cfg, line 89, column > 13-14: failed to load module db_postgres.so#012 > > I noted that PQconnectdbParams is available from psql-9 onwards and > could not figure out, what does this error mean. > > Could someone assist me please. > > Thanks! > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From nnikeshala at yahoo.com Mon Apr 20 16:24:04 2020 From: nnikeshala at yahoo.com (Nayani Nikeshala) Date: Mon, 20 Apr 2020 16:24:04 +0000 (UTC) Subject: [OpenSIPS-Users] OpenSIPS installation with PostgresSQL In-Reply-To: <7812bb25-eef4-0b34-04f5-a4f5c5ba589f@opensips.org> References: <129541852.2099756.1587328682918.ref@mail.yahoo.com> <129541852.2099756.1587328682918@mail.yahoo.com> <1985829642.2102360.1587329441034@mail.yahoo.com> <7812bb25-eef4-0b34-04f5-a4f5c5ba589f@opensips.org> Message-ID: <1950080153.2366807.1587399844986@mail.yahoo.com> Hello, OpenSIPS was compiled by me and I followed the OpenSIPS Installation Webinar tutorial.  Thanks! On Monday, April 20, 2020, 08:04:10 PM GMT+5:30, Bogdan-Andrei Iancu wrote: Hi, Is OpenSIPS compiled by you or installed from packages (if yes, what is the repo url)? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com On 4/19/20 11:50 PM, Nayani Nikeshala via Users wrote: Hello, I'm trying to install  opensips-2.4.7 with psql -8.4.20 on a Centos-6 platform. I can connect to psql DB as below. [root at SIPserver ~]# psql -h localhost -U opensips -d opensips psql (8.4.20) Type "help" for help. opensips=>  When I try to start OpenSIP as below, it gives me the following error.  [root at SIPserver ~]# opensipsctl start INFO: Starting OpenSIPS : ERROR: PID file /var/run/opensips.pid does not exist -- OpenSIPS start failed [root at SIPserver ~]# /var/log/opensips.log shows the below error.  Apr 20 01:31:20 SIPserver opensips: ERROR:core:sr_load_module: could not open module : /usr/local//lib/opensips/modules/db_postgres.so: undefined symbol: PQconnectdbParams Apr 20 01:31:20 SIPserver opensips: ERROR:core:load_module: failed to load module Apr 20 01:31:20 SIPserver opensips: CRITICAL:core:yyerror: parse error in config file /usr/local//etc/opensips/opensips.cfg, line 89, column 13-14: failed to load module db_postgres.so#012 I noted that PQconnectdbParams is available from psql-9 onwards and could not figure out, what does this error mean.  Could someone assist me please.  Thanks! _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From sobomax at sippysoft.com Mon Apr 20 18:59:47 2020 From: sobomax at sippysoft.com (Maxim Sobolev) Date: Mon, 20 Apr 2020 11:59:47 -0700 Subject: [OpenSIPS-Users] Join us for SIP Chronicles Live #1, featuring Giovanni Maruzzelli Message-ID: Dear Real-Time Friends and Colleagues! As many of you we have been totally devastated that we will have no chance to see you in the next few months to come. :-/ Some people in the community believe it might be years. I don’t necessarily agree with that opinion myself. Over the course of the last few years our team had a great time extending live coverage for some of those events that have been affected, got some experience and equipment. Instead of just waiting for the virus to clear, we decided to organize a series of bi-weekly live casts with some of the speakers that we have hoped to see at those events presenting their latest developments live and then answering questions from the audience. So without further ado, let me introduce our first guest Giovanni Maruzzelli, who is going to introduce his newest project SaraPhone ( https://github.com/gmaruzz/saraphone). Join us this Saturday, April 25th 4:30pm UTC and get a chance to ask Giovanni a question about his project live: https://youtu.be/mF9elIcVGE8 Or if you miss that opportunity, you can always watch the recording later on Sippy Labs channel on YouTube and email Giovanni your question at < gmaruzz at gmail.com>. SaraPhone is a bare bone SIP WebRTC phone, complete with most features real companies want to use in real world: HotDesking, Redial, BLFs, MWI, DND, PhoneBook, Hold, Transfer, Mute, Attended Transfer, Notifications, running on all Browsers both on Desktop and SmartPhone. SaraPhone is fully integrated with FusionPBX, the full-featured domain based multi-tenant PBX and voice switch for FreeSwitch. Based on SIP.js, SaraPhone works with all WebRTC compliant SIP proxies, gateways, and servers (Asterisk, OpenSIPS, Kamailio, Janus, etc). Initial author is Giovanni Maruzzelli, and SaraPhone gets its name from Giovanni's wife, Sara Hosseini. In addition to providing all of the usual DeskPhone functionality, SaraPhone got: - Desktop Notification for Incoming Calls - Live MWI update - Real Time BLFs status update - BLF click to call - Caller Name and Number Display - Call Error Cause Display - AutoAnswer - Network Disconnect Reload - Show and Set Caller-ID (incoming-outbound) Stay healthy, optimistic and productive! Also share, like and subscribe. See you soon!!! Regards, Max -------------- next part -------------- An HTML attachment was scrubbed... URL: From tito at xsvoce.com Mon Apr 20 20:49:29 2020 From: tito at xsvoce.com (Tito Cumpen) Date: Mon, 20 Apr 2020 13:49:29 -0700 Subject: [OpenSIPS-Users] Tls using t_relay In-Reply-To: <39b60bcc-945b-f659-ae7f-37f82327c63c@opensips.org> References: <39b60bcc-945b-f659-ae7f-37f82327c63c@opensips.org> Message-ID: Yes, It closes the socket right after the syn+ack. I can provide the trace if necessary.Here is an output of the logs Apr 20 20:41:53 sip1 /sbin/opensips[5106]: ERROR:proto_tls:proto_tls_send: connect failed Apr 20 20:41:53 sip1 /sbin/opensips[5106]: ERROR:tm:msg_send: send() to 192.xx.xx.xxx:5081 for proto tls/3 failed Apr 20 20:41:53 sip1 /sbin/opensips[5106]: ERROR:tm:t_forward_nonack: sending request failed Apr 20 20:41:53 sip1 /sbin/opensips[5106]: WARNING:rabbitmq:amqp_check_status: [ID1] socket error: Connection reset by peer(104) Apr 20 20:41:53 sip1 /sbin/opensips[5106]: ERROR:rabbitmq:rmq_error: closing channel: a socket error occurred Apr 20 20:41:53 sip1 /sbin/opensips[5106]: ERROR:rabbitmq:rmq_error: closing connection: a socket error occurred Apr 20 20:41:53 sip1 /sbin/opensips[5106]: should be removing pw Apr 20 20:41:53 sip1 /sbin/opensips[5106]: new branch at sip:support.test at test.org within logic per branch route Apr 20 20:41:53 sip1 /sbin/opensips[5106]: ERROR:core:tcp_connect_blocking_timeout: poll error: flags 28 - 4 8 16 32 Apr 20 20:41:53 sip1 /sbin/opensips[5106]: ERROR:core:tcp_connect_blocking_timeout: failed to retrieve SO_ERROR [server=127.0.0.1:443] (111) Connection refused Apr 20 20:41:53 sip1 /sbin/opensips[5106]: ERROR:proto_tls:tls_sync_connect: tcp_blocking_connect failed Apr 20 20:41:53 sip1 /sbin/opensips[5106]: ERROR:proto_tls:proto_tls_send: connect failed Apr 20 20:41:53 sip1 /sbin/opensips[5106]: ERROR:tm:msg_send: send() to 127.0.0.1:443 for proto tls/3 failed Apr 20 20:41:53 sip1 /sbin/opensips[5106]: ERROR:tm:t_forward_nonack: sending request failed Apr 20 20:41:53 sip1 /sbin/opensips[5106]: ERROR:core:tcp_connect_blocking_timeout: poll error: flags 28 - 4 8 16 32 Apr 20 20:41:53 sip1 /sbin/opensips[5106]: ERROR:core:tcp_connect_blocking_timeout: failed to retrieve SO_ERROR [server=127.0.0.1:443] (111) Connection refused Apr 20 20:41:53 sip1 /sbin/opensips[5106]: ERROR:proto_tls:tls_sync_connect: tcp_blocking_connect failed Apr 20 20:41:53 sip1 /sbin/opensips[5106]: ERROR:proto_tls:proto_tls_send: connect failed Apr 20 20:41:53 sip1 /sbin/opensips[5106]: ERROR:tm:msg_send: send() to 127.0.0.1:443 for proto tls/3 failed Apr 20 20:41:53 sip1 /sbin/opensips[5106]: ERROR:tm:t_forward_nonack: sending request failed Apr 20 20:41:53 sip1 /sbin/opensips[5106]: ERROR:core:tcpconn_async_connect: poll error: flags 1c Apr 20 20:41:53 sip1 /sbin/opensips[5106]: ERROR:core:tcpconn_async_connect: failed to retrieve SO_ERROR [server= 127.0.0.1:5060] (111) Connection refused Apr 20 20:41:53 sip1 /sbin/opensips[5106]: ERROR:core:proto_tcp_send: async TCP connect failed Apr 20 20:41:53 sip1 /sbin/opensips[5106]: ERROR:tm:msg_send: send() to 127.0.0.1:5060 for proto tcp/2 failed Apr 20 20:41:53 sip1 /sbin/opensips[5106]: ERROR:tm:t_forward_nonack: sending request failed Apr 20 20:41:53 sip1 /sbin/opensips[5106]: ERROR:core:tcpconn_async_connect: poll error: flags 1c Apr 20 20:41:53 sip1 /sbin/opensips[5106]: ERROR:core:tcpconn_async_connect: failed to retrieve SO_ERROR [server= 127.0.0.1:5060] (111) Connection refused Apr 20 20:41:53 sip1 /sbin/opensips[5106]: ERROR:core:proto_tcp_send: async TCP connect failed Apr 20 20:41:53 sip1 /sbin/opensips[5106]: ERROR:tm:msg_send: send() to 127.0.0.1:5060 for proto tcp/2 failed Apr 20 20:41:53 sip1 /sbin/opensips[5106]: ERROR:tm:t_forward_nonack: sending request failed Apr 20 20:41:53 sip1 /sbin/opensips[5106]: ERROR:core:tcpconn_async_connect: poll error: flags 1c Apr 20 20:41:53 sip1 /sbin/opensips[5106]: ERROR:core:tcpconn_async_connect: failed to retrieve SO_ERROR [server= 127.0.0.1:5223] (111) Connection refused Apr 20 20:41:53 sip1 /sbin/opensips[5106]: ERROR:core:proto_tcp_send: async TCP connect failed Apr 20 20:41:53 sip1 /sbin/opensips[5106]: ERROR:tm:msg_send: send() to 127.0.0.1:5223 for proto tcp/2 failed Apr 20 20:41:53 sip1 /sbin/opensips[5106]: ERROR:tm:t_forward_nonack: sending request failed Apr 20 20:41:53 sip1 /sbin/opensips[5106]: ERROR:core:tcpconn_async_connect: poll error: flags 1c Apr 20 20:41:53 sip1 /sbin/opensips[5106]: ERROR:core:tcpconn_async_connect: failed to retrieve SO_ERROR [server= 127.0.0.1:5223] (111) Connection refused Thanks, Tito On Mon, Apr 20, 2020 at 7:32 AM Bogdan-Andrei Iancu wrote: > Hi Tito, > > You say OpenSIPS is sending a RESET without being any data exchanged on > the connection? > > What are the logs for the failed send ? > > Regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > > On 4/17/20 11:43 PM, Tito Cumpen wrote: > > Hello, > > I am attempting to use t_relay("tls:domain:port") but I am not having much > success with it on 2.4. opensips sends a syn to the peer then gets an syn > ack and sends a rst . The logs claim that the send failed but it never > opened the socket entirely and did not send the client hello at all. > > Are there any other configs to be considered when attempting this ? > > _______________________________________________ > Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Tue Apr 21 09:02:26 2020 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Tue, 21 Apr 2020 12:02:26 +0300 Subject: [OpenSIPS-Users] OpenSIPS installation with PostgresSQL In-Reply-To: <1950080153.2366807.1587399844986@mail.yahoo.com> References: <129541852.2099756.1587328682918.ref@mail.yahoo.com> <129541852.2099756.1587328682918@mail.yahoo.com> <1985829642.2102360.1587329441034@mail.yahoo.com> <7812bb25-eef4-0b34-04f5-a4f5c5ba589f@opensips.org> <1950080153.2366807.1587399844986@mail.yahoo.com> Message-ID: Hi, and I assume you also locally compiled the postgres module, right ? If you do in the directory with the opensips sources: (cd modules/db_postgres ; make proper) ; make modules module=modules/db_postgres could you post the output here ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com On 4/20/20 7:24 PM, Nayani Nikeshala wrote: > Hello, > > OpenSIPS was compiled by me and I followed the OpenSIPS Installation > Webinar >  tutorial. > > > Thanks! > > On Monday, April 20, 2020, 08:04:10 PM GMT+5:30, Bogdan-Andrei Iancu > wrote: > > > Hi, > > Is OpenSIPS compiled by you or installed from packages (if yes, what > is the repo url)? > > Regards, > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > > On 4/19/20 11:50 PM, Nayani Nikeshala via Users wrote: > Hello, > > I'm trying to install opensips-2.4.7 with psql -8.4.20 on a Centos-6 > platform. I can connect to psql DB as below. > > [root at SIPserver ~]# psql -h localhost -U opensips -d opensips > psql (8.4.20) > Type "help" for help. > > opensips=> > > When I try to start OpenSIP as below, it gives me the following error. > > [root at SIPserver ~]# opensipsctl start > > INFO: Starting OpenSIPS : > > ERROR: PID file /var/run/opensips.pid does not exist -- OpenSIPS start > failed > [root at SIPserver ~]# > > > /var/log/opensips.log shows the below error. > > Apr 20 01:31:20 SIPserver opensips: ERROR:core:sr_load_module: could > not open module : > /usr/local//lib/opensips/modules/db_postgres.so: undefined symbol: > PQconnectdbParams > Apr 20 01:31:20 SIPserver opensips: ERROR:core:load_module: failed to > load module > Apr 20 01:31:20 SIPserver opensips: CRITICAL:core:yyerror: parse error > in config file /usr/local//etc/opensips/opensips.cfg, line 89, column > 13-14: failed to load module db_postgres.so#012 > > I noted that PQconnectdbParams is available from psql-9 onwards and > could not figure out, what does this error mean. > > Could someone assist me please. > > Thanks! > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Tue Apr 21 09:18:33 2020 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Tue, 21 Apr 2020 12:18:33 +0300 Subject: [OpenSIPS-Users] Tls using t_relay In-Reply-To: References: <39b60bcc-945b-f659-ae7f-37f82327c63c@opensips.org> Message-ID: <4e1e6e4c-b955-0f43-00a0-f5d776ea2e83@opensips.org> Hi Tito, Well, from OpenSIPS perceptive it is a "Connection refused", so the destination is rejecting the connect. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com On 4/20/20 11:49 PM, Tito Cumpen wrote: > Yes, > > It closes the socket right after the syn+ack. I can provide the trace > if necessary.Here is an output of the logs > > Apr 20 20:41:53 sip1 /sbin/opensips[5106]: > ERROR:proto_tls:proto_tls_send: connect failed > Apr 20 20:41:53 sip1 /sbin/opensips[5106]: ERROR:tm:msg_send: send() > to 192.xx.xx.xxx:5081 for proto tls/3 failed > Apr 20 20:41:53 sip1 /sbin/opensips[5106]: ERROR:tm:t_forward_nonack: > sending request failed > Apr 20 20:41:53 sip1 /sbin/opensips[5106]: > WARNING:rabbitmq:amqp_check_status: [ID1] socket error: Connection > reset by peer(104) > Apr 20 20:41:53 sip1 /sbin/opensips[5106]: ERROR:rabbitmq:rmq_error: > closing channel: a socket error occurred > Apr 20 20:41:53 sip1 /sbin/opensips[5106]: ERROR:rabbitmq:rmq_error: > closing connection: a socket error occurred > Apr 20 20:41:53 sip1 /sbin/opensips[5106]: should be removing pw > Apr 20 20:41:53 sip1 /sbin/opensips[5106]: new branch at > sip:support.test at test.org within > logic per branch route > Apr 20 20:41:53 sip1 /sbin/opensips[5106]: > ERROR:core:tcp_connect_blocking_timeout: poll error: flags 28 - 4 8 16 32 > Apr 20 20:41:53 sip1 /sbin/opensips[5106]: > ERROR:core:tcp_connect_blocking_timeout: failed to retrieve SO_ERROR > [server=127.0.0.1:443 ] (111) Connection refused > Apr 20 20:41:53 sip1 /sbin/opensips[5106]: > ERROR:proto_tls:tls_sync_connect: tcp_blocking_connect failed > Apr 20 20:41:53 sip1 /sbin/opensips[5106]: > ERROR:proto_tls:proto_tls_send: connect failed > Apr 20 20:41:53 sip1 /sbin/opensips[5106]: ERROR:tm:msg_send: send() > to 127.0.0.1:443 for proto tls/3 failed > Apr 20 20:41:53 sip1 /sbin/opensips[5106]: ERROR:tm:t_forward_nonack: > sending request failed > Apr 20 20:41:53 sip1 /sbin/opensips[5106]: > ERROR:core:tcp_connect_blocking_timeout: poll error: flags 28 - 4 8 16 32 > Apr 20 20:41:53 sip1 /sbin/opensips[5106]: > ERROR:core:tcp_connect_blocking_timeout: failed to retrieve SO_ERROR > [server=127.0.0.1:443 ] (111) Connection refused > Apr 20 20:41:53 sip1 /sbin/opensips[5106]: > ERROR:proto_tls:tls_sync_connect: tcp_blocking_connect failed > Apr 20 20:41:53 sip1 /sbin/opensips[5106]: > ERROR:proto_tls:proto_tls_send: connect failed > Apr 20 20:41:53 sip1 /sbin/opensips[5106]: ERROR:tm:msg_send: send() > to 127.0.0.1:443 for proto tls/3 failed > Apr 20 20:41:53 sip1 /sbin/opensips[5106]: ERROR:tm:t_forward_nonack: > sending request failed > Apr 20 20:41:53 sip1 /sbin/opensips[5106]: > ERROR:core:tcpconn_async_connect: poll error: flags 1c > Apr 20 20:41:53 sip1 /sbin/opensips[5106]: > ERROR:core:tcpconn_async_connect: failed to retrieve SO_ERROR > [server=127.0.0.1:5060 ] (111) Connection refused > Apr 20 20:41:53 sip1 /sbin/opensips[5106]: ERROR:core:proto_tcp_send: > async TCP connect failed > Apr 20 20:41:53 sip1 /sbin/opensips[5106]: ERROR:tm:msg_send: send() > to 127.0.0.1:5060 for proto tcp/2 failed > Apr 20 20:41:53 sip1 /sbin/opensips[5106]: ERROR:tm:t_forward_nonack: > sending request failed > Apr 20 20:41:53 sip1 /sbin/opensips[5106]: > ERROR:core:tcpconn_async_connect: poll error: flags 1c > Apr 20 20:41:53 sip1 /sbin/opensips[5106]: > ERROR:core:tcpconn_async_connect: failed to retrieve SO_ERROR > [server=127.0.0.1:5060 ] (111) Connection refused > Apr 20 20:41:53 sip1 /sbin/opensips[5106]: ERROR:core:proto_tcp_send: > async TCP connect failed > Apr 20 20:41:53 sip1 /sbin/opensips[5106]: ERROR:tm:msg_send: send() > to 127.0.0.1:5060 for proto tcp/2 failed > Apr 20 20:41:53 sip1 /sbin/opensips[5106]: ERROR:tm:t_forward_nonack: > sending request failed > Apr 20 20:41:53 sip1 /sbin/opensips[5106]: > ERROR:core:tcpconn_async_connect: poll error: flags 1c > Apr 20 20:41:53 sip1 /sbin/opensips[5106]: > ERROR:core:tcpconn_async_connect: failed to retrieve SO_ERROR > [server=127.0.0.1:5223 ] (111) Connection refused > Apr 20 20:41:53 sip1 /sbin/opensips[5106]: ERROR:core:proto_tcp_send: > async TCP connect failed > Apr 20 20:41:53 sip1 /sbin/opensips[5106]: ERROR:tm:msg_send: send() > to 127.0.0.1:5223 for proto tcp/2 failed > Apr 20 20:41:53 sip1 /sbin/opensips[5106]: ERROR:tm:t_forward_nonack: > sending request failed > Apr 20 20:41:53 sip1 /sbin/opensips[5106]: > ERROR:core:tcpconn_async_connect: poll error: flags 1c > Apr 20 20:41:53 sip1 /sbin/opensips[5106]: > ERROR:core:tcpconn_async_connect: failed to retrieve SO_ERROR > [server=127.0.0.1:5223 ] (111) Connection refused > > Thanks, > Tito > > On Mon, Apr 20, 2020 at 7:32 AM Bogdan-Andrei Iancu > > wrote: > > Hi Tito, > > You say OpenSIPS is sending a RESET without being any data > exchanged on the connection? > > What are the logs for the failed send ? > > Regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > > On 4/17/20 11:43 PM, Tito Cumpen wrote: >> Hello, >> >> I am attempting to use t_relay("tls:domain:port") but I am not >> having much success with it on 2.4. opensips sends a syn  to the >> peer then gets an syn ack and sends a rst . The logs claim that >> the send failed but it never opened the socket entirely and did >> not send the client hello at all. >> >> Are there any other configs to be considered when attempting this ? >> >> _______________________________________________ >> 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 nnikeshala at yahoo.com Tue Apr 21 14:21:35 2020 From: nnikeshala at yahoo.com (Nayani Nikeshala) Date: Tue, 21 Apr 2020 14:21:35 +0000 (UTC) Subject: [OpenSIPS-Users] OpenSIPS installation with PostgresSQL In-Reply-To: <1582965925.176731.1587476879885@mail.yahoo.com> References: <129541852.2099756.1587328682918.ref@mail.yahoo.com> <129541852.2099756.1587328682918@mail.yahoo.com> <1985829642.2102360.1587329441034@mail.yahoo.com> <7812bb25-eef4-0b34-04f5-a4f5c5ba589f@opensips.org> <1950080153.2366807.1587399844986@mail.yahoo.com> <1582965925.176731.1587476879885@mail.yahoo.com> Message-ID: <1193421104.181467.1587478895108@mail.yahoo.com> Hi Bogdan, I have attached the output log for the above commands.  "make modules module=modules/db_postgres" gave me an error as in the attached file, so I used "make include_modules=db_postgres modules" instead. I could see below in the output.  make[1]: Entering directory `/home/cscore/Downloads/opensips-2.4.7/modules/db_postgres'make[1]: Leaving directory `/home/cscore/Downloads/opensips-2.4.7/modules/db_postgres'make[1]: Entering directory `/home/cscore/Downloads/opensips-2.4.7/modules/db_postgres'Compiling dbase.cCompiling db_postgres.cCompiling pg_con.cCompiling res.cCompiling val.cpg_con.c: In function db_postgres_new_connection:pg_con.c:93: warning: implicit declaration of function PQconnectdbParamspg_con.c:93: warning: assignment makes pointer from integer without a castLinking db_postgres.somake[1]: Leaving directory `/home/cscore/Downloads/opensips-2.4.7/modules/db_postgres' Is this a problem of my psql version(8.4.20), because PQconnectdbParams  is not available in that version.  On Tuesday, April 21, 2020, 02:32:39 PM GMT+5:30, Bogdan-Andrei Iancu wrote: Hi, and I assume you also locally compiled the postgres module, right ? If you do in the directory with the opensips sources: (cd modules/db_postgres ; make proper) ; make modules module=modules/db_postgres could you post the output here ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com On 4/20/20 7:24 PM, Nayani Nikeshala wrote: Hello, OpenSIPS was compiled by me and I followed the OpenSIPS Installation Webinar tutorial.  Thanks! On Monday, April 20, 2020, 08:04:10 PM GMT+5:30, Bogdan-Andrei Iancu wrote: Hi, Is OpenSIPS compiled by you or installed from packages (if yes, what is the repo url)? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com On 4/19/20 11:50 PM, Nayani Nikeshala via Users wrote: Hello, I'm trying to install  opensips-2.4.7 with psql -8.4.20 on a Centos-6 platform. I can connect to psql DB as below. [root at SIPserver ~]# psql -h localhost -U opensips -d opensips psql (8.4.20) Type "help" for help. opensips=>  When I try to start OpenSIP as below, it gives me the following error.  [root at SIPserver ~]# opensipsctl start INFO: Starting OpenSIPS : ERROR: PID file /var/run/opensips.pid does not exist -- OpenSIPS start failed [root at SIPserver ~]# /var/log/opensips.log shows the below error.  Apr 20 01:31:20 SIPserver opensips: ERROR:core:sr_load_module: could not open module :/usr/local//lib/opensips/modules/db_postgres.so: undefined symbol: PQconnectdbParams Apr 20 01:31:20 SIPserver opensips: ERROR:core:load_module: failed to load module Apr 20 01:31:20 SIPserver opensips: CRITICAL:core:yyerror: parse error in config file /usr/local//etc/opensips/opensips.cfg, line 89, column 13-14: failed to load module db_postgres.so#012 I noted that PQconnectdbParams is available from psql-9 onwards and could not figure out, what does this error mean.  Could someone assist me please.  Thanks! _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: output-log.7z Type: application/octet-stream Size: 6572 bytes Desc: not available URL: From bogdan at opensips.org Tue Apr 21 15:27:01 2020 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Tue, 21 Apr 2020 18:27:01 +0300 Subject: [OpenSIPS-Users] OpenSIPS installation with PostgresSQL In-Reply-To: <1193421104.181467.1587478895108@mail.yahoo.com> References: <129541852.2099756.1587328682918.ref@mail.yahoo.com> <129541852.2099756.1587328682918@mail.yahoo.com> <1985829642.2102360.1587329441034@mail.yahoo.com> <7812bb25-eef4-0b34-04f5-a4f5c5ba589f@opensips.org> <1950080153.2366807.1587399844986@mail.yahoo.com> <1582965925.176731.1587476879885@mail.yahoo.com> <1193421104.181467.1587478895108@mail.yahoo.com> Message-ID: Hi Nayani, yeah, that is a rather old and unsupported version of postgres. I think you should try to use a newer version it (maybe a newer version of your distro). Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com On 4/21/20 5:21 PM, Nayani Nikeshala wrote: > Hi Bogdan, > > I have attached the output log for the above commands. > > "make modules module=modules/db_postgres" gave me an error as in the > attached file, so I used "make include_modules=db_postgres modules" > instead. I could see below in the output. > > make[1]: Entering directory > `/home/cscore/Downloads/opensips-2.4.7/modules/db_postgres' > make[1]: Leaving directory > `/home/cscore/Downloads/opensips-2.4.7/modules/db_postgres' > make[1]: Entering directory > `/home/cscore/Downloads/opensips-2.4.7/modules/db_postgres' > Compiling dbase.c > Compiling db_postgres.c > Compiling pg_con.c > Compiling res.c > Compiling val.c > pg_con.c: In function db_postgres_new_connection: > pg_con.c:93: warning: implicit declaration of function PQconnectdbParams > pg_con.c:93: warning: assignment makes pointer from integer without a cast > Linking db_postgres.so > make[1]: Leaving directory > `/home/cscore/Downloads/opensips-2.4.7/modules/db_postgres' > > Is this a problem of my psql version(8.4.20), because > PQconnectdbParams  is not available in that version. > -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Tue Apr 21 15:57:14 2020 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Tue, 21 Apr 2020 18:57:14 +0300 Subject: [OpenSIPS-Users] Announcing SaraPhone, SIP WebRTC Open Source business phone In-Reply-To: References: Message-ID: Hey Giovanni, Wonderful news in dark times !!! Thank you for sharing your work with the rest of the community and of course many thanks to Sara for being your muse ;) I think SaraPhone will be a good replacement for the web embedded sip client we have with the sip.opensips.org free VoIP service - time to upgrade ! Best regards and stay safe, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com On 4/16/20 6:36 PM, Giovanni Maruzzelli wrote: > My fellow VoIPers, > > I am pleased to announce the early availability of: > > SaraPhone > ------------------ > > SaraPhone is a bare bone SIP WebRTC voice phone, complete with most > features real companies want to use in real world: HotDesking, Redial, > BLFs, MWI, DND, PhoneBook, Hold, Transfer, Mute, Attended Transfer, > Notifications, running on all Browsers both on Desktop and SmartPhone. > > SaraPhone is fully integrated with FusionPBX, the full-featured domain > based multi-tenant PBX and voice switch for FreeSwitch. > > Based on SIP.js, SaraPhone works with all WebRTC compliant SIP > proxies, gateways, and servers (Asterisk, OpenSIPS, Kamailio, etc). > > Initial author is Giovanni Maruzzelli, and SaraPhone gets its name > from Giovanni's wife, Sara Hosseini. > > In addition to providing all of the usual DeskPhone functionality, > SaraPhone got: > > * Desktop Notification for Incoming Calls > * Live MWI update > * Real Time BLFs status update > * BLF click to call > * Caller Name and Number Display > * Call Error Cause Display > * AutoAnswer > * Network Disconnect Reload > * Show and Set Caller-ID (incoming-outbound) > > > You an find it in GitHub ( https://github.com/gmaruzz/saraphone ). > > Anyone interested can play with it :). > > Have fun, > giovanni > > > > -- > Sincerely, > > Giovanni Maruzzelli > OpenTelecom.IT > cell: +39 347 266 56 18 > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Tue Apr 21 16:00:03 2020 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Tue, 21 Apr 2020 19:00:03 +0300 Subject: [OpenSIPS-Users] Join us for SIP Chronicles Live #1, featuring Giovanni Maruzzelli In-Reply-To: References: Message-ID: Hi Maxim, Great idea, let's keep the communities connected and up to date - after all this is what we do - we do communication systems :) Giovanni, I will be there ! BTW, is this an interactive session, in the way that questions can be asked? Best regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com On 4/20/20 9:59 PM, Maxim Sobolev wrote: > > Dear Real-Time Friends and Colleagues! > > As many of you we have been totally devastated that we will have no > chance to see you in the next few months to come. :-/ Some people in > the community believe it might be years. I don’t necessarily agree > with that opinion myself. > > > Over the course of the last few years our team had a great time > extending live coverage for some of those events that have been > affected, got some experience and equipment. Instead of just waiting > for the virus to clear, we decided to organize a series of bi-weekly > live casts with some of the speakers that we have hoped to see at > those events presenting their latest developments live and then > answering questions from the audience. > > > So without further ado, let me introduce our first guest Giovanni > Maruzzelli, who is going to introduce his newest project SaraPhone > (https://github.com/gmaruzz/saraphone). > > > Join us this Saturday, April 25th 4:30pm UTC and get a chance to ask > Giovanni a question about his project live: > > > https://youtu.be/mF9elIcVGE8 > > > Or if you miss that opportunity, you can always watch the recording > later on Sippy Labs channel on YouTube and email Giovanni your > question at >. > > > SaraPhone is a bare bone SIP WebRTC phone, complete with most features > real companies want to use in real world: HotDesking, Redial, BLFs, > MWI, DND, PhoneBook, Hold, Transfer, Mute, Attended Transfer, > Notifications, running on all Browsers both on Desktop and SmartPhone. > > SaraPhone is fully integrated with FusionPBX, the full-featured domain > based multi-tenant PBX and voice switch for FreeSwitch. > > Based on SIP.js, SaraPhone works with all WebRTC compliant SIP > proxies, gateways, and servers (Asterisk, OpenSIPS, Kamailio, Janus, etc). > > Initial author is Giovanni Maruzzelli, and SaraPhone gets its name > from Giovanni's wife, Sara Hosseini. > > In addition to providing all of the usual DeskPhone functionality, > SaraPhone got: > > * > > Desktop Notification for Incoming Calls > > * > > Live MWI update > > * > > Real Time BLFs status update > > * > > BLF click to call > > * > > Caller Name and Number Display > > * > > Call Error Cause Display > > * > > AutoAnswer > > * > > Network Disconnect Reload > > * > > Show and Set Caller-ID (incoming-outbound) > > Stay healthy, optimistic and productive! Also share, like and > subscribe. See you soon!!! > > > Regards, > > > Max > > > _______________________________________________ > 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 Apr 21 16:05:43 2020 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Tue, 21 Apr 2020 19:05:43 +0300 Subject: [OpenSIPS-Users] =?utf-8?q?=5BBlog=5D_Call_Center_=E2=80=93_an_ea?= =?utf-8?q?sier_integration_with_OpenSIPS_3=2E1?= Message-ID: <2767a302-7bce-a060-fec1-578848efa330@opensips.org> Flexible wrap-up time? Call dissuading? Pre-call agent announcements? Events on agents activity ? Starting with version 3.1, the Call Center got several improvements to make easier the integration of the module with different external tools for the agent side, to overall improve the agent capabilities and experience in an OpenSIPS based solution. https://blog.opensips.org/2020/04/21/call-center-an-easier-integration-with-opensips-3-1/ Enjoy, -- Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com From nnikeshala at yahoo.com Tue Apr 21 19:35:50 2020 From: nnikeshala at yahoo.com (Nayani Nikeshala) Date: Tue, 21 Apr 2020 19:35:50 +0000 (UTC) Subject: [OpenSIPS-Users] OpenSIPS installation with PostgresSQL In-Reply-To: References: <129541852.2099756.1587328682918.ref@mail.yahoo.com> <129541852.2099756.1587328682918@mail.yahoo.com> <1985829642.2102360.1587329441034@mail.yahoo.com> <7812bb25-eef4-0b34-04f5-a4f5c5ba589f@opensips.org> <1950080153.2366807.1587399844986@mail.yahoo.com> <1582965925.176731.1587476879885@mail.yahoo.com> <1193421104.181467.1587478895108@mail.yahoo.com> Message-ID: <1971406255.325015.1587497750288@mail.yahoo.com> Hi Bogdan, I removed psql 8.4 and installed psql 9.6.  [root at SIPserver opensips-2.4.7]# yum list installed | grep postgrepostgresql96.i686        9.6.17-1PGDG.rhel6postgresql96-devel.i686  9.6.17-1PGDG.rhel6postgresql96-libs.i686   9.6.17-1PGDG.rhel6postgresql96-server.i686 9.6.17-1PGDG.rhel6[root at SIPserver opensips-2.4.7]# When I compile, I get below error. (Below is a part of the error) make[1]: Entering directory `/home/cscore/Downloads/opensips-2.4.7/modules/db_postgres'Compiling dbase.cCompiling pg_con.cCompiling res.cCompiling val.cIn file included from pg_con.c:23:pg_con.h:39:22: error: libpq-fe.h: No such file or directoryIn file included from pg_con.c:23:pg_con.h:54: error: expected specifier-qualifier-list before ‘PGconn’pg_con.c: In function ‘db_postgres_new_connection’:pg_con.c:93: error: ‘struct pg_con’ has no member named ‘con’pg_con.c:93: warning: implicit declaration of function ‘PQconnectdbParams’ libpq-fe.h & pg_config are in the below paths.   [root at SIPserver opensips-2.4.7]# ls -lrt /usr/pgsql-9.6/include/libpq-fe.h-rw-r--r--. 1 root root 21900 Feb 12 02:52 /usr/pgsql-9.6/include/libpq-fe.h[root at SIPserver opensips-2.4.7]# [root at SIPserver opensips-2.4.7]# ls -lrt /usr/pgsql-9.6/bin/pg_config-rwxr-xr-x. 1 root root 24544 Feb 12 02:52 /usr/pgsql-9.6/bin/pg_config Could you assist me with above error.  On Tuesday, April 21, 2020, 08:57:13 PM GMT+5:30, Bogdan-Andrei Iancu wrote: Hi Nayani, yeah, that is a rather old and unsupported version of postgres. I think you should try to use a newer version it (maybe a newer version of your distro). Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com On 4/21/20 5:21 PM, Nayani Nikeshala wrote: Hi Bogdan, I have attached the output log for the above commands.  "make modules module=modules/db_postgres" gave me an error as in the attached file, so I used "make include_modules=db_postgres modules" instead. I could see below in the output.  make[1]: Entering directory `/home/cscore/Downloads/opensips-2.4.7/modules/db_postgres' make[1]: Leaving directory `/home/cscore/Downloads/opensips-2.4.7/modules/db_postgres' make[1]: Entering directory `/home/cscore/Downloads/opensips-2.4.7/modules/db_postgres' Compiling dbase.c Compiling db_postgres.c Compiling pg_con.c Compiling res.c Compiling val.c pg_con.c: In function db_postgres_new_connection: pg_con.c:93: warning: implicit declaration of function PQconnectdbParams pg_con.c:93: warning: assignment makes pointer from integer without a cast Linking db_postgres.so make[1]: Leaving directory `/home/cscore/Downloads/opensips-2.4.7/modules/db_postgres' Is this a problem of my psql version(8.4.20), because PQconnectdbParams  is not available in that version.  -------------- next part -------------- An HTML attachment was scrubbed... URL: From sobomax at sippysoft.com Tue Apr 21 20:48:50 2020 From: sobomax at sippysoft.com (Maxim Sobolev) Date: Tue, 21 Apr 2020 13:48:50 -0700 Subject: [OpenSIPS-Users] Join us for SIP Chronicles Live #1, featuring Giovanni Maruzzelli In-Reply-To: References: Message-ID: Thanks Bogdan, I am glad that you liked the idea! Yes, very good question. We will have a slot (or few) where questions from the audience can be answered interactively. Originally we were planning to take questions over YouTube chat, but maybe it would be also cool if Giovanni can deploy his cool phone so people can actually call in and ask? At which point we could also publish a SIP URI for anyone to ring in directly as well and drill speaker on his answers. Eventually I hope to feel brave enough to deploy Jitsi Meet, but probably not until this whole ordeal is over unless I get some more help from a community, which is also an option. :) -Max On Tue, Apr 21, 2020 at 9:00 AM Bogdan-Andrei Iancu wrote: > Hi Maxim, > > Great idea, let's keep the communities connected and up to date - after > all this is what we do - we do communication systems :) > > Giovanni, I will be there ! > > BTW, is this an interactive session, in the way that questions can be > asked? > > Best regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > > On 4/20/20 9:59 PM, Maxim Sobolev wrote: > > Dear Real-Time Friends and Colleagues! > > As many of you we have been totally devastated that we will have no chance > to see you in the next few months to come. :-/ Some people in the community > believe it might be years. I don’t necessarily agree with that opinion > myself. > > Over the course of the last few years our team had a great time extending > live coverage for some of those events that have been affected, got some > experience and equipment. Instead of just waiting for the virus to clear, > we decided to organize a series of bi-weekly live casts with some of the > speakers that we have hoped to see at those events presenting their latest > developments live and then answering questions from the audience. > > So without further ado, let me introduce our first guest Giovanni > Maruzzelli, who is going to introduce his newest project SaraPhone ( > https://github.com/gmaruzz/saraphone). > > Join us this Saturday, April 25th 4:30pm UTC and get a chance to ask > Giovanni a question about his project live: > > https://youtu.be/mF9elIcVGE8 > > Or if you miss that opportunity, you can always watch the recording later > on Sippy Labs channel on YouTube and email Giovanni your question at < > gmaruzz at gmail.com>. > > SaraPhone is a bare bone SIP WebRTC phone, complete with most features > real companies want to use in real world: HotDesking, Redial, BLFs, MWI, > DND, PhoneBook, Hold, Transfer, Mute, Attended Transfer, Notifications, > running on all Browsers both on Desktop and SmartPhone. > > SaraPhone is fully integrated with FusionPBX, the full-featured domain > based multi-tenant PBX and voice switch for FreeSwitch. > > Based on SIP.js, SaraPhone works with all WebRTC compliant SIP proxies, > gateways, and servers (Asterisk, OpenSIPS, Kamailio, Janus, etc). > > Initial author is Giovanni Maruzzelli, and SaraPhone gets its name from > Giovanni's wife, Sara Hosseini. > > In addition to providing all of the usual DeskPhone functionality, > SaraPhone got: > > - > > Desktop Notification for Incoming Calls > - > > Live MWI update > - > > Real Time BLFs status update > - > > BLF click to call > - > > Caller Name and Number Display > - > > Call Error Cause Display > - > > AutoAnswer > - > > Network Disconnect Reload > - > > Show and Set Caller-ID (incoming-outbound) > > Stay healthy, optimistic and productive! Also share, like and subscribe. > See you soon!!! > > Regards, > > Max > > _______________________________________________ > Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > -- Maksym Sobolyev Sippy Software, Inc. Internet Telephony (VoIP) Experts Tel (Canada): +1-778-783-0474 Tel (Toll-Free): +1-855-747-7779 Fax: +1-866-857-6942 Web: http://www.sippysoft.com MSN: sales at sippysoft.com Skype: SippySoft -------------- next part -------------- An HTML attachment was scrubbed... URL: From tito at xsvoce.com Tue Apr 21 23:22:27 2020 From: tito at xsvoce.com (Tito Cumpen) Date: Tue, 21 Apr 2020 16:22:27 -0700 Subject: [OpenSIPS-Users] Tls using t_relay In-Reply-To: References: <39b60bcc-945b-f659-ae7f-37f82327c63c@opensips.org> <4e1e6e4c-b955-0f43-00a0-f5d776ea2e83@opensips.org> Message-ID: Bogdan, I found out that there was a timer flag for tcp connections that may have been causing an issue tcp_connect_timeout=3 Once I removed this line the tls connection was made fine but now I am seeing opensips send an error message to the client SIP/2.0 500 Server error occurred (1/SL) client---opensips---SIP AS even though the SIP AS sent a 180 response Here are the errors from the log Apr 21 23:12:46 sip1 /sbin/opensips[11376]: ERROR:core:tcp_connect_blocking_timeout: poll error: flags 28 - 4 8 16 32 Apr 21 23:12:46 sip1 /sbin/opensips[11376]: ERROR:core:tcp_connect_blocking_timeout: failed to retrieve SO_ERROR [server=127.0.0.1:443] (111) Connection refused Apr 21 23:12:46 sip1 /sbin/opensips[11376]: ERROR:proto_tls:tls_sync_connect: tcp_blocking_connect failed Apr 21 23:12:46 sip1 /sbin/opensips[11376]: ERROR:proto_tls:proto_tls_send: connect failed Apr 21 23:12:46 sip1 /sbin/opensips[11376]: ERROR:tm:msg_send: send() to 127.0.0.1:443 for proto tls/3 failed Apr 21 23:12:46 sip1 /sbin/opensips[11376]: ERROR:tm:t_forward_nonack: sending request failed Apr 21 23:12:46 sip1 /sbin/opensips[11376]: ERROR:core:tcp_connect_blocking_timeout: poll error: flags 28 - 4 8 16 32 Apr 21 23:12:46 sip1 /sbin/opensips[11376]: ERROR:core:tcp_connect_blocking_timeout: failed to retrieve SO_ERROR [server=127.0.0.1:443] (111) Connection refused Apr 21 23:12:46 sip1 /sbin/opensips[11376]: ERROR:proto_tls:tls_sync_connect: tcp_blocking_connect failed Apr 21 23:12:46 sip1 /sbin/opensips[11376]: ERROR:proto_tls:proto_tls_send: connect failed Apr 21 23:12:46 sip1 /sbin/opensips[11376]: ERROR:tm:msg_send: send() to 127.0.0.1:443 for proto tls/3 failed Apr 21 23:12:46 sip1 /sbin/opensips[11376]: ERROR:tm:t_forward_nonack: sending request failed Apr 21 23:12:46 sip1 /sbin/opensips[11376]: ERROR:core:tcpconn_async_connect: poll error: flags 1c Apr 21 23:12:46 sip1 /sbin/opensips[11376]: ERROR:core:tcpconn_async_connect: failed to retrieve SO_ERROR [server= 127.0.0.1:5060] (111) Connection refused Apr 21 23:12:46 sip1 /sbin/opensips[11376]: ERROR:core:proto_tcp_send: async TCP connect failed Apr 21 23:12:46 sip1 /sbin/opensips[11376]: ERROR:tm:msg_send: send() to 127.0.0.1:5060 for proto tcp/2 failed Apr 21 23:12:46 sip1 /sbin/opensips[11376]: ERROR:tm:t_forward_nonack: sending request failed Apr 21 23:12:46 sip1 /sbin/opensips[11376]: ERROR:core:tcpconn_async_connect: poll error: flags 1c Apr 21 23:12:46 sip1 /sbin/opensips[11376]: ERROR:core:tcpconn_async_connect: failed to retrieve SO_ERROR [server= 127.0.0.1:5060] (111) Connection refused Apr 21 23:12:46 sip1 /sbin/opensips[11376]: ERROR:core:proto_tcp_send: async TCP connect failed Apr 21 23:12:46 sip1 /sbin/opensips[11376]: ERROR:tm:msg_send: send() to 127.0.0.1:5060 for proto tcp/2 failed Apr 21 23:12:46 sip1 /sbin/opensips[11376]: ERROR:tm:t_forward_nonack: sending request failed Apr 21 23:12:46 sip1 /sbin/opensips[11376]: ERROR:core:tcpconn_async_connect: poll error: flags 1c Apr 21 23:12:46 sip1 /sbin/opensips[11376]: ERROR:core:tcpconn_async_connect: failed to retrieve SO_ERROR [server= 127.0.0.1:5223] (111) Connection refused Apr 21 23:12:46 sip1 /sbin/opensips[11376]: ERROR:core:proto_tcp_send: async TCP connect failed Apr 21 23:12:46 sip1 /sbin/opensips[11376]: ERROR:tm:msg_send: send() to 127.0.0.1:5223 for proto tcp/2 failed Apr 21 23:12:46 sip1 /sbin/opensips[11376]: ERROR:tm:t_forward_nonack: sending request failed Apr 21 23:12:46 sip1 /sbin/opensips[11376]: ERROR:core:tcpconn_async_connect: poll error: flags 1c Apr 21 23:12:46 sip1 /sbin/opensips[11376]: ERROR:core:tcpconn_async_connect: failed to retrieve SO_ERROR [server= 127.0.0.1:5223] (111) Connection refused Apr 21 23:12:46 sip1 /sbin/opensips[11376]: ERROR:core:proto_tcp_send: async TCP connect failed Apr 21 23:12:46 sip1 /sbin/opensips[11376]: ERROR:tm:msg_send: send() to 127.0.0.1:5223 for proto tcp/2 failed On Tue, Apr 21, 2020 at 11:26 AM Tito Cumpen wrote: > Hey Bogdan, > > Here is the capture I took from using t_relay("tls:domain:port") > > As you can see the client side (opensips) does not proceed with allowing > the socket to open. > > Thanks, > Tito > > On Tue, Apr 21, 2020 at 2:18 AM Bogdan-Andrei Iancu > wrote: > >> Hi Tito, >> >> Well, from OpenSIPS perceptive it is a "Connection refused", so the >> destination is rejecting the connect. >> >> Regards, >> >> Bogdan-Andrei Iancu >> >> OpenSIPS Founder and Developer >> https://www.opensips-solutions.com >> >> On 4/20/20 11:49 PM, Tito Cumpen wrote: >> >> Yes, >> >> It closes the socket right after the syn+ack. I can provide the trace if >> necessary.Here is an output of the logs >> >> Apr 20 20:41:53 sip1 /sbin/opensips[5106]: >> ERROR:proto_tls:proto_tls_send: connect failed >> Apr 20 20:41:53 sip1 /sbin/opensips[5106]: ERROR:tm:msg_send: send() to >> 192.xx.xx.xxx:5081 for proto tls/3 failed >> Apr 20 20:41:53 sip1 /sbin/opensips[5106]: ERROR:tm:t_forward_nonack: >> sending request failed >> Apr 20 20:41:53 sip1 /sbin/opensips[5106]: >> WARNING:rabbitmq:amqp_check_status: [ID1] socket error: Connection reset by >> peer(104) >> Apr 20 20:41:53 sip1 /sbin/opensips[5106]: ERROR:rabbitmq:rmq_error: >> closing channel: a socket error occurred >> Apr 20 20:41:53 sip1 /sbin/opensips[5106]: ERROR:rabbitmq:rmq_error: >> closing connection: a socket error occurred >> Apr 20 20:41:53 sip1 /sbin/opensips[5106]: should be removing pw >> Apr 20 20:41:53 sip1 /sbin/opensips[5106]: new branch at >> sip:support.test at test.org within logic per branch route >> Apr 20 20:41:53 sip1 /sbin/opensips[5106]: >> ERROR:core:tcp_connect_blocking_timeout: poll error: flags 28 - 4 8 16 32 >> Apr 20 20:41:53 sip1 /sbin/opensips[5106]: >> ERROR:core:tcp_connect_blocking_timeout: failed to retrieve SO_ERROR >> [server=127.0.0.1:443] (111) Connection refused >> Apr 20 20:41:53 sip1 /sbin/opensips[5106]: >> ERROR:proto_tls:tls_sync_connect: tcp_blocking_connect failed >> Apr 20 20:41:53 sip1 /sbin/opensips[5106]: >> ERROR:proto_tls:proto_tls_send: connect failed >> Apr 20 20:41:53 sip1 /sbin/opensips[5106]: ERROR:tm:msg_send: send() to >> 127.0.0.1:443 for proto tls/3 failed >> Apr 20 20:41:53 sip1 /sbin/opensips[5106]: ERROR:tm:t_forward_nonack: >> sending request failed >> Apr 20 20:41:53 sip1 /sbin/opensips[5106]: >> ERROR:core:tcp_connect_blocking_timeout: poll error: flags 28 - 4 8 16 32 >> Apr 20 20:41:53 sip1 /sbin/opensips[5106]: >> ERROR:core:tcp_connect_blocking_timeout: failed to retrieve SO_ERROR >> [server=127.0.0.1:443] (111) Connection refused >> Apr 20 20:41:53 sip1 /sbin/opensips[5106]: >> ERROR:proto_tls:tls_sync_connect: tcp_blocking_connect failed >> Apr 20 20:41:53 sip1 /sbin/opensips[5106]: >> ERROR:proto_tls:proto_tls_send: connect failed >> Apr 20 20:41:53 sip1 /sbin/opensips[5106]: ERROR:tm:msg_send: send() to >> 127.0.0.1:443 for proto tls/3 failed >> Apr 20 20:41:53 sip1 /sbin/opensips[5106]: ERROR:tm:t_forward_nonack: >> sending request failed >> Apr 20 20:41:53 sip1 /sbin/opensips[5106]: >> ERROR:core:tcpconn_async_connect: poll error: flags 1c >> Apr 20 20:41:53 sip1 /sbin/opensips[5106]: >> ERROR:core:tcpconn_async_connect: failed to retrieve SO_ERROR [server= >> 127.0.0.1:5060] (111) Connection refused >> Apr 20 20:41:53 sip1 /sbin/opensips[5106]: ERROR:core:proto_tcp_send: >> async TCP connect failed >> Apr 20 20:41:53 sip1 /sbin/opensips[5106]: ERROR:tm:msg_send: send() to >> 127.0.0.1:5060 for proto tcp/2 failed >> Apr 20 20:41:53 sip1 /sbin/opensips[5106]: ERROR:tm:t_forward_nonack: >> sending request failed >> Apr 20 20:41:53 sip1 /sbin/opensips[5106]: >> ERROR:core:tcpconn_async_connect: poll error: flags 1c >> Apr 20 20:41:53 sip1 /sbin/opensips[5106]: >> ERROR:core:tcpconn_async_connect: failed to retrieve SO_ERROR [server= >> 127.0.0.1:5060] (111) Connection refused >> Apr 20 20:41:53 sip1 /sbin/opensips[5106]: ERROR:core:proto_tcp_send: >> async TCP connect failed >> Apr 20 20:41:53 sip1 /sbin/opensips[5106]: ERROR:tm:msg_send: send() to >> 127.0.0.1:5060 for proto tcp/2 failed >> Apr 20 20:41:53 sip1 /sbin/opensips[5106]: ERROR:tm:t_forward_nonack: >> sending request failed >> Apr 20 20:41:53 sip1 /sbin/opensips[5106]: >> ERROR:core:tcpconn_async_connect: poll error: flags 1c >> Apr 20 20:41:53 sip1 /sbin/opensips[5106]: >> ERROR:core:tcpconn_async_connect: failed to retrieve SO_ERROR [server= >> 127.0.0.1:5223] (111) Connection refused >> Apr 20 20:41:53 sip1 /sbin/opensips[5106]: ERROR:core:proto_tcp_send: >> async TCP connect failed >> Apr 20 20:41:53 sip1 /sbin/opensips[5106]: ERROR:tm:msg_send: send() to >> 127.0.0.1:5223 for proto tcp/2 failed >> Apr 20 20:41:53 sip1 /sbin/opensips[5106]: ERROR:tm:t_forward_nonack: >> sending request failed >> Apr 20 20:41:53 sip1 /sbin/opensips[5106]: >> ERROR:core:tcpconn_async_connect: poll error: flags 1c >> Apr 20 20:41:53 sip1 /sbin/opensips[5106]: >> ERROR:core:tcpconn_async_connect: failed to retrieve SO_ERROR [server= >> 127.0.0.1:5223] (111) Connection refused >> >> Thanks, >> Tito >> >> On Mon, Apr 20, 2020 at 7:32 AM Bogdan-Andrei Iancu >> wrote: >> >>> Hi Tito, >>> >>> You say OpenSIPS is sending a RESET without being any data exchanged on >>> the connection? >>> >>> What are the logs for the failed send ? >>> >>> Regards, >>> >>> Bogdan-Andrei Iancu >>> >>> OpenSIPS Founder and Developer >>> https://www.opensips-solutions.com >>> >>> On 4/17/20 11:43 PM, Tito Cumpen wrote: >>> >>> Hello, >>> >>> I am attempting to use t_relay("tls:domain:port") but I am not having >>> much success with it on 2.4. opensips sends a syn to the peer then gets an >>> syn ack and sends a rst . The logs claim that the send failed but it never >>> opened the socket entirely and did not send the client hello at all. >>> >>> Are there any other configs to be considered when attempting this ? >>> >>> _______________________________________________ >>> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >>> >>> >> -------------- next part -------------- An HTML attachment was scrubbed... URL: From ag at ag-projects.com Wed Apr 22 01:57:23 2020 From: ag at ag-projects.com (Adrian Georgescu) Date: Tue, 21 Apr 2020 22:57:23 -0300 Subject: [OpenSIPS-Users] Join us for SIP Chronicles Live #1, featuring Giovanni Maruzzelli In-Reply-To: References: Message-ID: <51438375-4A99-45BE-8D7D-DEB184DE8A17@ag-projects.com> You can try out SylkServer / sip2sip.info Adrian > On 21 Apr 2020, at 17:48, Maxim Sobolev wrote: > > Thanks Bogdan, I am glad that you liked the idea! Yes, very good question. We will have a slot (or few) where questions from the audience can be answered interactively. Originally we were planning to take questions over YouTube chat, but maybe it would be also cool if Giovanni can deploy his cool phone so people can actually call in and ask? At which point we could also publish a SIP URI for anyone to ring in directly as well and drill speaker on his answers. > > Eventually I hope to feel brave enough to deploy Jitsi Meet, but probably not until this whole ordeal is over unless I get some more help from a community, which is also an option. :) > > -Max > > On Tue, Apr 21, 2020 at 9:00 AM Bogdan-Andrei Iancu > wrote: > Hi Maxim, > > Great idea, let's keep the communities connected and up to date - after all this is what we do - we do communication systems :) > > Giovanni, I will be there ! > > BTW, is this an interactive session, in the way that questions can be asked? > > Best regards, > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > > On 4/20/20 9:59 PM, Maxim Sobolev wrote: >> Dear Real-Time Friends and Colleagues! >> >> As many of you we have been totally devastated that we will have no chance to see you in the next few months to come. :-/ Some people in the community believe it might be years. I don’t necessarily agree with that opinion myself. >> >> Over the course of the last few years our team had a great time extending live coverage for some of those events that have been affected, got some experience and equipment. Instead of just waiting for the virus to clear, we decided to organize a series of bi-weekly live casts with some of the speakers that we have hoped to see at those events presenting their latest developments live and then answering questions from the audience. >> >> So without further ado, let me introduce our first guest Giovanni Maruzzelli, who is going to introduce his newest project SaraPhone (https://github.com/gmaruzz/saraphone ). >> >> Join us this Saturday, April 25th 4:30pm UTC and get a chance to ask Giovanni a question about his project live: >> >> https://youtu.be/mF9elIcVGE8 >> >> Or if you miss that opportunity, you can always watch the recording later on Sippy Labs channel on YouTube and email Giovanni your question at >. >> >> SaraPhone is a bare bone SIP WebRTC phone, complete with most features real companies want to use in real world: HotDesking, Redial, BLFs, MWI, DND, PhoneBook, Hold, Transfer, Mute, Attended Transfer, Notifications, running on all Browsers both on Desktop and SmartPhone. >> >> SaraPhone is fully integrated with FusionPBX, the full-featured domain based multi-tenant PBX and voice switch for FreeSwitch. >> >> Based on SIP.js, SaraPhone works with all WebRTC compliant SIP proxies, gateways, and servers (Asterisk, OpenSIPS, Kamailio, Janus, etc). >> >> Initial author is Giovanni Maruzzelli, and SaraPhone gets its name from Giovanni's wife, Sara Hosseini. >> >> In addition to providing all of the usual DeskPhone functionality, SaraPhone got: >> >> Desktop Notification for Incoming Calls >> Live MWI update >> Real Time BLFs status update >> BLF click to call >> Caller Name and Number Display >> Call Error Cause Display >> AutoAnswer >> Network Disconnect Reload >> Show and Set Caller-ID (incoming-outbound) >> >> Stay healthy, optimistic and productive! Also share, like and subscribe. See you soon!!! >> >> Regards, >> >> Max >> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > -- > Maksym Sobolyev > Sippy Software, Inc. > Internet Telephony (VoIP) Experts > Tel (Canada): +1-778-783-0474 > Tel (Toll-Free): +1-855-747-7779 > Fax: +1-866-857-6942 > Web: http://www.sippysoft.com > MSN: sales at sippysoft.com > Skype: SippySoft > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Wed Apr 22 08:23:55 2020 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Wed, 22 Apr 2020 11:23:55 +0300 Subject: [OpenSIPS-Users] OpenSIPS installation with PostgresSQL In-Reply-To: <1971406255.325015.1587497750288@mail.yahoo.com> References: <129541852.2099756.1587328682918.ref@mail.yahoo.com> <129541852.2099756.1587328682918@mail.yahoo.com> <1985829642.2102360.1587329441034@mail.yahoo.com> <7812bb25-eef4-0b34-04f5-a4f5c5ba589f@opensips.org> <1950080153.2366807.1587399844986@mail.yahoo.com> <1582965925.176731.1587476879885@mail.yahoo.com> <1193421104.181467.1587478895108@mail.yahoo.com> <1971406255.325015.1587497750288@mail.yahoo.com> Message-ID: <15e3ac1f-2e01-a4fb-147a-5ac32411c98d@opensips.org> It may sound weird, but maybe you into the other extrema, with some too new version :D... Is it possible to try some 9.3 or so ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com On 4/21/20 10:35 PM, Nayani Nikeshala wrote: > Hi Bogdan, > > I removed psql 8.4 and installed psql 9.6. > > [root at SIPserver opensips-2.4.7]# yum list installed | grep postgre > postgresql96.i686        9.6.17-1PGDG.rhel6 > postgresql96-devel.i686  9.6.17-1PGDG.rhel6 > postgresql96-libs.i686   9.6.17-1PGDG.rhel6 > postgresql96-server.i686 9.6.17-1PGDG.rhel6 > [root at SIPserver opensips-2.4.7]# > > > When I compile, I get below error. (Below is a part of the error) > > make[1]: Entering directory > `/home/cscore/Downloads/opensips-2.4.7/modules/db_postgres' > Compiling dbase.c > Compiling pg_con.c > Compiling res.c > Compiling val.c > In file included from pg_con.c:23: > pg_con.h:39:22: error: libpq-fe.h: No such file or directory > In file included from pg_con.c:23: > pg_con.h:54: error: expected specifier-qualifier-list before ‘PGconn’ > pg_con.c: In function ‘db_postgres_new_connection’: > pg_con.c:93: error: ‘struct pg_con’ has no member named ‘con’ > pg_con.c:93: warning: implicit declaration of function ‘PQconnectdbParams’ > > libpq-fe.h & pg_config are in the below paths. > > [root at SIPserver opensips-2.4.7]# ls -lrt /usr/pgsql-9.6/include/libpq-fe.h > -rw-r--r--. 1 root root 21900 Feb 12 02:52 > /usr/pgsql-9.6/include/libpq-fe.h > [root at SIPserver opensips-2.4.7]# > > [root at SIPserver opensips-2.4.7]# ls -lrt /usr/pgsql-9.6/bin/pg_config > -rwxr-xr-x. 1 root root 24544 Feb 12 02:52 /usr/pgsql-9.6/bin/pg_config > > Could you assist me with above error. > > > > On Tuesday, April 21, 2020, 08:57:13 PM GMT+5:30, Bogdan-Andrei Iancu > wrote: > > > Hi Nayani, > > yeah, that is a rather old and unsupported version of postgres. I > think you should try to use a newer version it (maybe a newer version > of your distro). > > Regards, > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > > On 4/21/20 5:21 PM, Nayani Nikeshala wrote: > Hi Bogdan, > > I have attached the output log for the above commands. > > "make modules module=modules/db_postgres" gave me an error as in the > attached file, so I used "make include_modules=db_postgres modules" > instead. I could see below in the output. > > make[1]: Entering directory > `/home/cscore/Downloads/opensips-2.4.7/modules/db_postgres' > make[1]: Leaving directory > `/home/cscore/Downloads/opensips-2.4.7/modules/db_postgres' > make[1]: Entering directory > `/home/cscore/Downloads/opensips-2.4.7/modules/db_postgres' > Compiling dbase.c > Compiling db_postgres.c > Compiling pg_con.c > Compiling res.c > Compiling val.c > pg_con.c: In function db_postgres_new_connection: > pg_con.c:93: warning: implicit declaration of function PQconnectdbParams > pg_con.c:93: warning: assignment makes pointer from integer without a cast > Linking db_postgres.so > make[1]: Leaving directory > `/home/cscore/Downloads/opensips-2.4.7/modules/db_postgres' > > Is this a problem of my psql version(8.4.20), because > PQconnectdbParams  is not available in that version. > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From gmaruzz at gmail.com Wed Apr 22 12:08:56 2020 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Wed, 22 Apr 2020 14:08:56 +0200 Subject: [OpenSIPS-Users] Announcing SaraPhone, SIP WebRTC Open Source business phone In-Reply-To: References: Message-ID: On Tue, Apr 21, 2020 at 5:57 PM Bogdan-Andrei Iancu wrote: > > I think SaraPhone will be a good replacement for the web embedded sip > client we have with the sip.opensips.org free VoIP service - time to > upgrade ! > > This is a very very nice news! Please consider me **completely** available for any integration/customization etc it may need!! Yay! -- Sincerely, Giovanni Maruzzelli OpenTelecom.IT cell: +39 347 266 56 18 -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Wed Apr 22 12:28:43 2020 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Wed, 22 Apr 2020 15:28:43 +0300 Subject: [OpenSIPS-Users] Announcing SaraPhone, SIP WebRTC Open Source business phone In-Reply-To: References: Message-ID: <008dd7b7-e6d5-d5dc-3fea-2e601a01bb6d@opensips.org> Oh Giovanni, be sure I will not be shy in asking for the help, just stay tune. Best regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com On 4/22/20 3:08 PM, Giovanni Maruzzelli wrote: > On Tue, Apr 21, 2020 at 5:57 PM Bogdan-Andrei Iancu > > wrote: > > > I think SaraPhone will be a good replacement for the web embedded > sip client we have with the sip.opensips.org > free VoIP service - time to upgrade ! > > > This is a very very nice news! > > Please consider me **completely** available for any > integration/customization etc it may need!! > > Yay! > > -- > Sincerely, > > Giovanni Maruzzelli > OpenTelecom.IT > cell: +39 347 266 56 18 > -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Wed Apr 22 12:43:32 2020 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Wed, 22 Apr 2020 15:43:32 +0300 Subject: [OpenSIPS-Users] Join us for SIP Chronicles Live #1, featuring Giovanni Maruzzelli In-Reply-To: References: Message-ID: Just a crazy idea, we can use SaraPhone as end points (on Giovannni's side  and also on participants side) and have in the middle an OpenSIPS to do call-center/call queuing ;) BTW, how comes that a Saturday was picked for the events? not like anyone is planning to go out somewhere :P. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com On 4/21/20 11:48 PM, Maxim Sobolev wrote: > Thanks Bogdan, I am glad that you liked the idea! Yes, very good > question. We will have a slot (or few) where questions from the > audience can be answered interactively. Originally we were planning to > take questions over YouTube chat, but maybe it would be also cool if > Giovanni can deploy his cool phone so people can actually call in and > ask? At which point we could also publish a SIP URI for anyone to  > ring in directly as well and drill speaker on his answers. > > Eventually I hope to feel brave enough to deploy Jitsi Meet, but > probably not until this whole ordeal is over unless I get some more > help from a community, which is also an option. :) > > -Max > > On Tue, Apr 21, 2020 at 9:00 AM Bogdan-Andrei Iancu > > wrote: > > Hi Maxim, > > Great idea, let's keep the communities connected and up to date - > after all this is what we do - we do communication systems :) > > Giovanni, I will be there ! > > BTW, is this an interactive session, in the way that questions can > be asked? > > Best regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > > On 4/20/20 9:59 PM, Maxim Sobolev wrote: >> >> Dear Real-Time Friends and Colleagues! >> >> As many of you we have been totally devastated that we will have >> no chance to see you in the next few months to come. :-/ Some >> people in the community believe it might be years. I don’t >> necessarily agree with that opinion myself. >> >> >> Over the course of the last few years our team had a great time >> extending live coverage for some of those events that have been >> affected, got some experience and equipment. Instead of just >> waiting for the virus to clear, we decided to organize a series >> of bi-weekly live casts with some of the speakers that we have >> hoped to see at those events presenting their latest developments >> live and then answering questions from the audience. >> >> >> So without further ado, let me introduce our first guest Giovanni >> Maruzzelli, who is going to introduce his newest project >> SaraPhone (https://github.com/gmaruzz/saraphone). >> >> >> Join us this Saturday, April 25th 4:30pm UTC and get a chance to >> ask Giovanni a question about his project live: >> >> >> https://youtu.be/mF9elIcVGE8 >> >> >> Or if you miss that opportunity, you can always watch the >> recording later on Sippy Labs channel on YouTube and email >> Giovanni your question at > >. >> >> >> SaraPhone is a bare bone SIP WebRTC phone, complete with most >> features real companies want to use in real world: HotDesking, >> Redial, BLFs, MWI, DND, PhoneBook, Hold, Transfer, Mute, Attended >> Transfer, Notifications, running on all Browsers both on Desktop >> and SmartPhone. >> >> SaraPhone is fully integrated with FusionPBX, the full-featured >> domain based multi-tenant PBX and voice switch for FreeSwitch. >> >> Based on SIP.js, SaraPhone works with all WebRTC compliant SIP >> proxies, gateways, and servers (Asterisk, OpenSIPS, Kamailio, >> Janus, etc). >> >> Initial author is Giovanni Maruzzelli, and SaraPhone gets its >> name from Giovanni's wife, Sara Hosseini. >> >> In addition to providing all of the usual DeskPhone >> functionality, SaraPhone got: >> >> * >> >> Desktop Notification for Incoming Calls >> >> * >> >> Live MWI update >> >> * >> >> Real Time BLFs status update >> >> * >> >> BLF click to call >> >> * >> >> Caller Name and Number Display >> >> * >> >> Call Error Cause Display >> >> * >> >> AutoAnswer >> >> * >> >> Network Disconnect Reload >> >> * >> >> Show and Set Caller-ID (incoming-outbound) >> >> Stay healthy, optimistic and productive! Also share, like and >> subscribe. See you soon!!! >> >> >> Regards, >> >> >> Max >> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > -- > Maksym Sobolyev > Sippy Software, Inc. > Internet Telephony (VoIP) Experts > Tel (Canada): +1-778-783-0474 > Tel (Toll-Free): +1-855-747-7779 > Fax: +1-866-857-6942 > Web: http://www.sippysoft.com > MSN: sales at sippysoft.com > Skype: SippySoft -------------- next part -------------- An HTML attachment was scrubbed... URL: From nnikeshala at yahoo.com Wed Apr 22 12:44:21 2020 From: nnikeshala at yahoo.com (Nayani Nikeshala) Date: Wed, 22 Apr 2020 12:44:21 +0000 (UTC) Subject: [OpenSIPS-Users] OpenSIPS installation with PostgresSQL In-Reply-To: <15e3ac1f-2e01-a4fb-147a-5ac32411c98d@opensips.org> References: <129541852.2099756.1587328682918.ref@mail.yahoo.com> <129541852.2099756.1587328682918@mail.yahoo.com> <1985829642.2102360.1587329441034@mail.yahoo.com> <7812bb25-eef4-0b34-04f5-a4f5c5ba589f@opensips.org> <1950080153.2366807.1587399844986@mail.yahoo.com> <1582965925.176731.1587476879885@mail.yahoo.com> <1193421104.181467.1587478895108@mail.yahoo.com> <1971406255.325015.1587497750288@mail.yahoo.com> <15e3ac1f-2e01-a4fb-147a-5ac32411c98d@opensips.org> Message-ID: <1119346896.578368.1587559461192@mail.yahoo.com> Hi Bogdan, I found an old email, where he has faced a similar kind of an issue with PSQL 9.2 also. I believe, it will not get resolved with PSQL version.  https://opensips.org/pipermail/devel/2015-June/017666.html According to the email, he has resolved it by modifying makefile, but I'm not sure, how he did that. On Wednesday, April 22, 2020, 01:54:09 PM GMT+5:30, Bogdan-Andrei Iancu wrote: It may sound weird, but maybe you into the other extrema, with some too new version :D... Is it possible to try some 9.3 or so ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com On 4/21/20 10:35 PM, Nayani Nikeshala wrote: Hi Bogdan, I removed psql 8.4 and installed psql 9.6.  [root at SIPserver opensips-2.4.7]# yum list installed | grep postgre postgresql96.i686        9.6.17-1PGDG.rhel6 postgresql96-devel.i686  9.6.17-1PGDG.rhel6 postgresql96-libs.i686   9.6.17-1PGDG.rhel6 postgresql96-server.i686 9.6.17-1PGDG.rhel6 [root at SIPserver opensips-2.4.7]# When I compile, I get below error. (Below is a part of the error) make[1]: Entering directory `/home/cscore/Downloads/opensips-2.4.7/modules/db_postgres' Compiling dbase.c Compiling pg_con.c Compiling res.c Compiling val.c In file included from pg_con.c:23: pg_con.h:39:22: error: libpq-fe.h: No such file or directory In file included from pg_con.c:23: pg_con.h:54: error: expected specifier-qualifier-list before ‘PGconn’ pg_con.c: In function ‘db_postgres_new_connection’: pg_con.c:93: error: ‘struct pg_con’ has no member named ‘con’ pg_con.c:93: warning: implicit declaration of function ‘PQconnectdbParams’ libpq-fe.h & pg_config are in the below paths.   [root at SIPserver opensips-2.4.7]# ls -lrt /usr/pgsql-9.6/include/libpq-fe.h -rw-r--r--. 1 root root 21900 Feb 12 02:52 /usr/pgsql-9.6/include/libpq-fe.h [root at SIPserver opensips-2.4.7]# [root at SIPserver opensips-2.4.7]# ls -lrt /usr/pgsql-9.6/bin/pg_config -rwxr-xr-x. 1 root root 24544 Feb 12 02:52 /usr/pgsql-9.6/bin/pg_config Could you assist me with above error.  On Tuesday, April 21, 2020, 08:57:13 PM GMT+5:30, Bogdan-Andrei Iancu wrote: Hi Nayani, yeah, that is a rather old and unsupported version of postgres. I think you should try to use a newer version it (maybe a newer version of your distro). Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com On 4/21/20 5:21 PM, Nayani Nikeshala wrote: Hi Bogdan, I have attached the output log for the above commands.  "make modules module=modules/db_postgres" gave me an error as in the attached file, so I used "make include_modules=db_postgres modules" instead. I could see below in the output.  make[1]: Entering directory `/home/cscore/Downloads/opensips-2.4.7/modules/db_postgres' make[1]: Leaving directory `/home/cscore/Downloads/opensips-2.4.7/modules/db_postgres' make[1]: Entering directory `/home/cscore/Downloads/opensips-2.4.7/modules/db_postgres' Compiling dbase.c Compiling db_postgres.c Compiling pg_con.c Compiling res.c Compiling val.c pg_con.c: In function db_postgres_new_connection: pg_con.c:93: warning: implicit declaration of function PQconnectdbParams pg_con.c:93: warning: assignment makes pointer from integer without a cast Linking db_postgres.so make[1]: Leaving directory `/home/cscore/Downloads/opensips-2.4.7/modules/db_postgres' Is this a problem of my psql version(8.4.20), because PQconnectdbParams  is not available in that version.  -------------- next part -------------- An HTML attachment was scrubbed... URL: From gmaruzz at gmail.com Wed Apr 22 17:27:43 2020 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Wed, 22 Apr 2020 19:27:43 +0200 Subject: [OpenSIPS-Users] Join us for SIP Chronicles Live #1, featuring Giovanni Maruzzelli In-Reply-To: References: Message-ID: On Wed, Apr 22, 2020 at 2:43 PM Bogdan-Andrei Iancu wrote: > Just a crazy idea, we can use SaraPhone as end points (on Giovannni's > side and also on participants side) and have in the middle an OpenSIPS to > do call-center/call queuing ;) > > BTW, how comes that a Saturday was picked for the events? not like anyone > is planning to go out somewhere :P. > > On Wed, Apr 22, 2020 at 2:43 PM Bogdan-Andrei Iancu wrote: > Just a crazy idea, we can use SaraPhone as end points (on Giovannni's > side and also on participants side) and have in the middle an OpenSIPS to > do call-center/call queuing ;) > > BTW, how comes that a Saturday was picked for the events? not like anyone > is planning to go out somewhere :P. > Yep, not going anywhere :)) I am presently in the mountain, and my upload bandwidth is kind of limited, but I sawe today is enough for a low res videocall. So, I don't know how it will go, because we need: 1) the video with Max, to be rebroadcasted in youtube 2) some video to show SaraPhone to work 3) additional streams for voice? So, I would try to do something similar: I have a test FusionPBX/SaraPhone in DigitalOcean (hypercheap, like $5/month) We can leverage that for doing demos, and calls, and all that... (at least is already working, and do not necessitate further working) Friday I will understand how Max wants to do the video broadcast If needed, we can use a videoconferencing webrtc proof of concept I have, so I will share in videoconferencing the SaraPhone, Max will be at other end of conferencing, and broadcast to youtube... At the same time, I will give away login and password for the FusionPBX?SaraPhone in DigitalOcean so people can use SaraPhone to call and ask questions. Let's not try to federate domains, no time/energy for doing that. We can plan another performance/happening/modernart, when we'll be ready with SaraPhone for OpenSIPS. :)) -giovanni > Regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > > On 4/21/20 11:48 PM, Maxim Sobolev wrote: > > Thanks Bogdan, I am glad that you liked the idea! Yes, very good question. > We will have a slot (or few) where questions from the audience can be > answered interactively. Originally we were planning to take questions over > YouTube chat, but maybe it would be also cool if Giovanni can deploy his > cool phone so people can actually call in and ask? At which point we could > also publish a SIP URI for anyone to ring in directly as well and drill > speaker on his answers. > > Eventually I hope to feel brave enough to deploy Jitsi Meet, but probably > not until this whole ordeal is over unless I get some more help from a > community, which is also an option. :) > > -Max > > On Tue, Apr 21, 2020 at 9:00 AM Bogdan-Andrei Iancu > wrote: > >> Hi Maxim, >> >> Great idea, let's keep the communities connected and up to date - after >> all this is what we do - we do communication systems :) >> >> Giovanni, I will be there ! >> >> BTW, is this an interactive session, in the way that questions can be >> asked? >> >> Best regards, >> >> Bogdan-Andrei Iancu >> >> OpenSIPS Founder and Developer >> https://www.opensips-solutions.com >> >> On 4/20/20 9:59 PM, Maxim Sobolev wrote: >> >> Dear Real-Time Friends and Colleagues! >> >> As many of you we have been totally devastated that we will have no >> chance to see you in the next few months to come. :-/ Some people in the >> community believe it might be years. I don’t necessarily agree with that >> opinion myself. >> >> Over the course of the last few years our team had a great time extending >> live coverage for some of those events that have been affected, got some >> experience and equipment. Instead of just waiting for the virus to clear, >> we decided to organize a series of bi-weekly live casts with some of the >> speakers that we have hoped to see at those events presenting their latest >> developments live and then answering questions from the audience. >> >> So without further ado, let me introduce our first guest Giovanni >> Maruzzelli, who is going to introduce his newest project SaraPhone ( >> https://github.com/gmaruzz/saraphone). >> >> Join us this Saturday, April 25th 4:30pm UTC and get a chance to ask >> Giovanni a question about his project live: >> >> https://youtu.be/mF9elIcVGE8 >> >> Or if you miss that opportunity, you can always watch the recording later >> on Sippy Labs channel on YouTube and email Giovanni your question at < >> gmaruzz at gmail.com>. >> >> SaraPhone is a bare bone SIP WebRTC phone, complete with most features >> real companies want to use in real world: HotDesking, Redial, BLFs, MWI, >> DND, PhoneBook, Hold, Transfer, Mute, Attended Transfer, Notifications, >> running on all Browsers both on Desktop and SmartPhone. >> >> SaraPhone is fully integrated with FusionPBX, the full-featured domain >> based multi-tenant PBX and voice switch for FreeSwitch. >> >> Based on SIP.js, SaraPhone works with all WebRTC compliant SIP proxies, >> gateways, and servers (Asterisk, OpenSIPS, Kamailio, Janus, etc). >> >> Initial author is Giovanni Maruzzelli, and SaraPhone gets its name from >> Giovanni's wife, Sara Hosseini. >> >> In addition to providing all of the usual DeskPhone functionality, >> SaraPhone got: >> >> - >> >> Desktop Notification for Incoming Calls >> - >> >> Live MWI update >> - >> >> Real Time BLFs status update >> - >> >> BLF click to call >> - >> >> Caller Name and Number Display >> - >> >> Call Error Cause Display >> - >> >> AutoAnswer >> - >> >> Network Disconnect Reload >> - >> >> Show and Set Caller-ID (incoming-outbound) >> >> Stay healthy, optimistic and productive! Also share, like and subscribe. >> See you soon!!! >> >> Regards, >> >> Max >> >> _______________________________________________ >> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> >> > > -- > Maksym Sobolyev > Sippy Software, Inc. > Internet Telephony (VoIP) Experts > Tel (Canada): +1-778-783-0474 > Tel (Toll-Free): +1-855-747-7779 > Fax: +1-866-857-6942 > Web: http://www.sippysoft.com > MSN: sales at sippysoft.com > Skype: SippySoft > > > -- Sincerely, Giovanni Maruzzelli OpenTelecom.IT cell: +39 347 266 56 18 -------------- next part -------------- An HTML attachment was scrubbed... URL: From williamj at exetel.com.au Thu Apr 23 04:04:39 2020 From: williamj at exetel.com.au (William Jin) Date: Thu, 23 Apr 2020 04:04:39 +0000 Subject: [OpenSIPS-Users] opensips 3.0.2 100% CPU after enable tls? Message-ID: Hi, Linux platform: Debian 9 (stretch) opensips -V version: opensips 3.0.2 (x86_64/linux) flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, HP_MALLOC, DBG_MALLOC, FAST_LOCK-ADAPTIVE_WAIT ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535 poll method support: poll, epoll, sigio_rt, select. main.c compiled on with gcc 6.3.0 related config: listen = tls:xxx.xxx.xxx.xxx:5061 anycast ####TLS loadmodule "tls_mgm.so" loadmodule "proto_tls.so" modparam("tls_mgm", "server_domain", "sip1") modparam("tls_mgm", "match_ip_address", "[sip1]xx.xx.xx.xx:5061") modparam("tls_mgm", "match_sip_domain", "[sip1]xxx.xxx.example.com") modparam("tls_mgm", "verify_cert", "[sip1]1") modparam("tls_mgm", "require_cert", "[sip1]0") modparam("tls_mgm", "tls_method", "[sip1]SSLv23") modparam("tls_mgm", "ciphers_list", "[sip1]AES256-GCM-SHA384,AES256-SHA256,AES256-SHA,CAMELLIA256-SHA,AES128-SHA,SEED-SHA,CAMELLIA128-SHA,RC4-SHA,DES-CBC3-SHA") modparam("tls_mgm", "certificate", "[sip1]/etc/opensips/tls/mycerts/selfsignedcert.pem") modparam("tls_mgm", "private_key", "[sip1]/etc/opensips/tls/mycerts/unsecuredkey.pem") opensips-cli -x trap {pid} result attached Can someone shed some light on it? Thanks. -- Regards, William Jin -------------- next part -------------- An HTML attachment was scrubbed... URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: gdb_opensips_20200423_135258 Type: application/octet-stream Size: 8985 bytes Desc: gdb_opensips_20200423_135258 URL: From anexiole at gmail.com Thu Apr 23 06:21:46 2020 From: anexiole at gmail.com (Gordon Yeong) Date: Thu, 23 Apr 2020 16:21:46 +1000 Subject: [OpenSIPS-Users] Testing ENUM queries Message-ID: hi there, I am going to start adding ENUM queries into my setup. My production set up doesn't quite let me run sipp or pjsip2. Thus, I figured one way to do this is to write a script (likely in perl) to call use the enum_query() function in the Enum module ( https://opensips.org/docs/modules/3.0.x/enum.html#overview) independently. 1. Is this possible? 2. If it is not in perl , then is it possible to write a C program that uses the Enum module? If so, what libraries should I include? Thank you Gordon Yeong -perl, rails, elixir, python, devops - db engines. -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Thu Apr 23 08:39:22 2020 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Thu, 23 Apr 2020 11:39:22 +0300 Subject: [OpenSIPS-Users] Tls using t_relay In-Reply-To: References: <39b60bcc-945b-f659-ae7f-37f82327c63c@opensips.org> <4e1e6e4c-b955-0f43-00a0-f5d776ea2e83@opensips.org> Message-ID: Hi Tito, Note that the tcp_connect_timeout is in milliseconds, so maybe 3 ms is too short for getting back the SYN ACK. The logs are keep reporting the failed connect. You say the connect is ok, the INVITE is sent forward to callee and there is also an 180 response? ....and then you get the 500 reply ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com On 4/22/20 2:22 AM, Tito Cumpen wrote: > Bogdan, > > I found out that there was a timer flag for tcp connections that may > have been causing an issue > tcp_connect_timeout=3 > Once I removed this line the tls connection was made fine but now I am > seeing opensips send an error message to the client > > SIP/2.0 500 Server error occurred (1/SL) > client---opensips---SIP AS > even though the SIP AS sent a 180 response > > > Here are the errors from the log > > Apr 21 23:12:46 sip1 /sbin/opensips[11376]: > ERROR:core:tcp_connect_blocking_timeout: poll error: flags 28 - 4 8 16 32 > Apr 21 23:12:46 sip1 /sbin/opensips[11376]: > ERROR:core:tcp_connect_blocking_timeout: failed to retrieve SO_ERROR > [server=127.0.0.1:443 ] (111) Connection refused > Apr 21 23:12:46 sip1 /sbin/opensips[11376]: > ERROR:proto_tls:tls_sync_connect: tcp_blocking_connect failed > Apr 21 23:12:46 sip1 /sbin/opensips[11376]: > ERROR:proto_tls:proto_tls_send: connect failed > Apr 21 23:12:46 sip1 /sbin/opensips[11376]: ERROR:tm:msg_send: send() > to 127.0.0.1:443 for proto tls/3 failed > Apr 21 23:12:46 sip1 /sbin/opensips[11376]: ERROR:tm:t_forward_nonack: > sending request failed > Apr 21 23:12:46 sip1 /sbin/opensips[11376]: > ERROR:core:tcp_connect_blocking_timeout: poll error: flags 28 - 4 8 16 32 > Apr 21 23:12:46 sip1 /sbin/opensips[11376]: > ERROR:core:tcp_connect_blocking_timeout: failed to retrieve SO_ERROR > [server=127.0.0.1:443 ] (111) Connection refused > Apr 21 23:12:46 sip1 /sbin/opensips[11376]: > ERROR:proto_tls:tls_sync_connect: tcp_blocking_connect failed > Apr 21 23:12:46 sip1 /sbin/opensips[11376]: > ERROR:proto_tls:proto_tls_send: connect failed > Apr 21 23:12:46 sip1 /sbin/opensips[11376]: ERROR:tm:msg_send: send() > to 127.0.0.1:443 for proto tls/3 failed > Apr 21 23:12:46 sip1 /sbin/opensips[11376]: ERROR:tm:t_forward_nonack: > sending request failed > Apr 21 23:12:46 sip1 /sbin/opensips[11376]: > ERROR:core:tcpconn_async_connect: poll error: flags 1c > Apr 21 23:12:46 sip1 /sbin/opensips[11376]: > ERROR:core:tcpconn_async_connect: failed to retrieve SO_ERROR > [server=127.0.0.1:5060 ] (111) Connection refused > Apr 21 23:12:46 sip1 /sbin/opensips[11376]: ERROR:core:proto_tcp_send: > async TCP connect failed > Apr 21 23:12:46 sip1 /sbin/opensips[11376]: ERROR:tm:msg_send: send() > to 127.0.0.1:5060 for proto tcp/2 failed > Apr 21 23:12:46 sip1 /sbin/opensips[11376]: ERROR:tm:t_forward_nonack: > sending request failed > Apr 21 23:12:46 sip1 /sbin/opensips[11376]: > ERROR:core:tcpconn_async_connect: poll error: flags 1c > Apr 21 23:12:46 sip1 /sbin/opensips[11376]: > ERROR:core:tcpconn_async_connect: failed to retrieve SO_ERROR > [server=127.0.0.1:5060 ] (111) Connection refused > Apr 21 23:12:46 sip1 /sbin/opensips[11376]: ERROR:core:proto_tcp_send: > async TCP connect failed > Apr 21 23:12:46 sip1 /sbin/opensips[11376]: ERROR:tm:msg_send: send() > to 127.0.0.1:5060 for proto tcp/2 failed > Apr 21 23:12:46 sip1 /sbin/opensips[11376]: ERROR:tm:t_forward_nonack: > sending request failed > Apr 21 23:12:46 sip1 /sbin/opensips[11376]: > ERROR:core:tcpconn_async_connect: poll error: flags 1c > Apr 21 23:12:46 sip1 /sbin/opensips[11376]: > ERROR:core:tcpconn_async_connect: failed to retrieve SO_ERROR > [server=127.0.0.1:5223 ] (111) Connection refused > Apr 21 23:12:46 sip1 /sbin/opensips[11376]: ERROR:core:proto_tcp_send: > async TCP connect failed > Apr 21 23:12:46 sip1 /sbin/opensips[11376]: ERROR:tm:msg_send: send() > to 127.0.0.1:5223 for proto tcp/2 failed > Apr 21 23:12:46 sip1 /sbin/opensips[11376]: ERROR:tm:t_forward_nonack: > sending request failed > Apr 21 23:12:46 sip1 /sbin/opensips[11376]: > ERROR:core:tcpconn_async_connect: poll error: flags 1c > Apr 21 23:12:46 sip1 /sbin/opensips[11376]: > ERROR:core:tcpconn_async_connect: failed to retrieve SO_ERROR > [server=127.0.0.1:5223 ] (111) Connection refused > Apr 21 23:12:46 sip1 /sbin/opensips[11376]: ERROR:core:proto_tcp_send: > async TCP connect failed > Apr 21 23:12:46 sip1 /sbin/opensips[11376]: ERROR:tm:msg_send: send() > to 127.0.0.1:5223 for proto tcp/2 failed > > On Tue, Apr 21, 2020 at 11:26 AM Tito Cumpen > wrote: > > Hey Bogdan, > > Here is the capture I took from using t_relay("tls:domain:port") > > As you can see the client side (opensips) does not proceed with > allowing the socket to open. > > Thanks, > Tito > > On Tue, Apr 21, 2020 at 2:18 AM Bogdan-Andrei Iancu > > wrote: > > Hi Tito, > > Well, from OpenSIPS perceptive it is a "Connection refused", > so the destination is rejecting the connect. > > Regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Thu Apr 23 08:43:27 2020 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Thu, 23 Apr 2020 11:43:27 +0300 Subject: [OpenSIPS-Users] Testing ENUM queries In-Reply-To: References: Message-ID: Hi, I'm not sure I understand what you want to achieve here... .You want to use OpenSIPS for enum queries, but from some other application, not via SIP? Like your app needs to do the ENUM, so it somehow uses OpenSIPS to do it? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com On 4/23/20 9:21 AM, Gordon Yeong wrote: > hi there, > I am going to start adding ENUM queries into my setup. > My production set up doesn't quite let me run sipp or pjsip2. > Thus, I figured one way to do this is to write a script (likely in > perl) to call use the > enum_query() function in the Enum module > (https://opensips.org/docs/modules/3.0.x/enum.html#overview) > independently. > > 1. Is this possible? > 2. If it is not in perl , then is it possible to write a C program > that uses the Enum module? If so, what libraries should I include? > > Thank you > > Gordon Yeong > -perl, rails, elixir, python, devops > - db engines. > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Thu Apr 23 08:49:11 2020 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Thu, 23 Apr 2020 11:49:11 +0300 Subject: [OpenSIPS-Users] opensips 3.0.2 100% CPU after enable tls? In-Reply-To: References: Message-ID: <18b56215-a2ed-6cd2-8730-de499ee521d5@opensips.org> Hi William, What GIT revision of OpenSIPS do you use? (this is exposed by the "opensips -V") Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com On 4/23/20 7:04 AM, William Jin wrote: > Hi, > > Linux platform: Debian 9 (stretch) > > opensips -V > version: opensips 3.0.2 (x86_64/linux) > flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, > Q_MALLOC, F_MALLOC, HP_MALLOC, DBG_MALLOC, FAST_LOCK-ADAPTIVE_WAIT > ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, > MAX_URI_SIZE 1024, BUF_SIZE 65535 > poll method support: poll, epoll, sigio_rt, select. > main.c compiled on  with gcc 6.3.0 > > related config: > > listen = tls:xxx.xxx.xxx.xxx:5061 anycast > > > ####TLS > loadmodule "tls_mgm.so" > loadmodule "proto_tls.so" > > modparam("tls_mgm", "server_domain", "sip1") > modparam("tls_mgm", "match_ip_address", "[sip1]xx.xx.xx.xx:5061") > modparam("tls_mgm", "match_sip_domain", "[sip1]xxx.xxx.example.com") > > modparam("tls_mgm", "verify_cert", "[sip1]1") > modparam("tls_mgm", "require_cert", "[sip1]0") > modparam("tls_mgm", "tls_method", "[sip1]SSLv23") > modparam("tls_mgm", "ciphers_list", > "[sip1]AES256-GCM-SHA384,AES256-SHA256,AES256-SHA,CAMELLIA256-SHA,AES128-SHA,SEED-SHA,CAMELLIA128-SHA,RC4-SHA,DES-CBC3-SHA") > > > modparam("tls_mgm", "certificate", > "[sip1]/etc/opensips/tls/mycerts/selfsignedcert.pem") > modparam("tls_mgm", "private_key", > "[sip1]/etc/opensips/tls/mycerts/unsecuredkey.pem") > > > opensips-cli -x trap {pid} result attached > > Can someone shed some light on it? Thanks. > > > -- > Regards, > William Jin > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Thu Apr 23 08:50:49 2020 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Thu, 23 Apr 2020 11:50:49 +0300 Subject: [OpenSIPS-Users] OpenSIPS installation with PostgresSQL In-Reply-To: <1119346896.578368.1587559461192@mail.yahoo.com> References: <129541852.2099756.1587328682918.ref@mail.yahoo.com> <129541852.2099756.1587328682918@mail.yahoo.com> <1985829642.2102360.1587329441034@mail.yahoo.com> <7812bb25-eef4-0b34-04f5-a4f5c5ba589f@opensips.org> <1950080153.2366807.1587399844986@mail.yahoo.com> <1582965925.176731.1587476879885@mail.yahoo.com> <1193421104.181467.1587478895108@mail.yahoo.com> <1971406255.325015.1587497750288@mail.yahoo.com> <15e3ac1f-2e01-a4fb-147a-5ac32411c98d@opensips.org> <1119346896.578368.1587559461192@mail.yahoo.com> Message-ID: <2d3c288a-4f71-d78b-6d83-84e2731fc2e5@opensips.org> Hi, I have an 9.5.19 (on Ubuntu) and works ok. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com On 4/22/20 3:44 PM, Nayani Nikeshala wrote: > Hi Bogdan, > > I found an old email, where he has faced a similar kind of an issue > with PSQL 9.2 also. I believe, it will not get resolved with PSQL > version. > > https://opensips.org/pipermail/devel/2015-June/017666.html > > According to the email, he has resolved it by modifying makefile, but > I'm not sure, how he did that. > > > On Wednesday, April 22, 2020, 01:54:09 PM GMT+5:30, Bogdan-Andrei > Iancu wrote: > > > It may sound weird, but maybe you into the other extrema, with some > too new version :D... > > Is it possible to try some 9.3 or so ? > > Regards, > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > > On 4/21/20 10:35 PM, Nayani Nikeshala wrote: > Hi Bogdan, > > I removed psql 8.4 and installed psql 9.6. > > [root at SIPserver opensips-2.4.7]# yum list installed | grep postgre > postgresql96.i686        9.6.17-1PGDG.rhel6 > postgresql96-devel.i686  9.6.17-1PGDG.rhel6 > postgresql96-libs.i686   9.6.17-1PGDG.rhel6 > postgresql96-server.i686 9.6.17-1PGDG.rhel6 > [root at SIPserver opensips-2.4.7]# > > > When I compile, I get below error. (Below is a part of the error) > > make[1]: Entering directory > `/home/cscore/Downloads/opensips-2.4.7/modules/db_postgres' > Compiling dbase.c > Compiling pg_con.c > Compiling res.c > Compiling val.c > In file included from pg_con.c:23: > pg_con.h:39:22: error: libpq-fe.h: No such file or directory > In file included from pg_con.c:23: > pg_con.h:54: error: expected specifier-qualifier-list before ‘PGconn’ > pg_con.c: In function ‘db_postgres_new_connection’: > pg_con.c:93: error: ‘struct pg_con’ has no member named ‘con’ > pg_con.c:93: warning: implicit declaration of function ‘PQconnectdbParams’ > > libpq-fe.h & pg_config are in the below paths. > > [root at SIPserver opensips-2.4.7]# ls -lrt /usr/pgsql-9.6/include/libpq-fe.h > -rw-r--r--. 1 root root 21900 Feb 12 02:52 > /usr/pgsql-9.6/include/libpq-fe.h > [root at SIPserver opensips-2.4.7]# > > [root at SIPserver opensips-2.4.7]# ls -lrt /usr/pgsql-9.6/bin/pg_config > -rwxr-xr-x. 1 root root 24544 Feb 12 02:52 /usr/pgsql-9.6/bin/pg_config > > Could you assist me with above error. > > > > On Tuesday, April 21, 2020, 08:57:13 PM GMT+5:30, Bogdan-Andrei Iancu > wrote: > > > Hi Nayani, > > yeah, that is a rather old and unsupported version of postgres. I > think you should try to use a newer version it (maybe a newer version > of your distro). > > Regards, > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > > On 4/21/20 5:21 PM, Nayani Nikeshala wrote: > Hi Bogdan, > > I have attached the output log for the above commands. > > "make modules module=modules/db_postgres" gave me an error as in the > attached file, so I used "make include_modules=db_postgres modules" > instead. I could see below in the output. > > make[1]: Entering directory > `/home/cscore/Downloads/opensips-2.4.7/modules/db_postgres' > make[1]: Leaving directory > `/home/cscore/Downloads/opensips-2.4.7/modules/db_postgres' > make[1]: Entering directory > `/home/cscore/Downloads/opensips-2.4.7/modules/db_postgres' > Compiling dbase.c > Compiling db_postgres.c > Compiling pg_con.c > Compiling res.c > Compiling val.c > pg_con.c: In function db_postgres_new_connection: > pg_con.c:93: warning: implicit declaration of function PQconnectdbParams > pg_con.c:93: warning: assignment makes pointer from integer without a cast > Linking db_postgres.so > make[1]: Leaving directory > `/home/cscore/Downloads/opensips-2.4.7/modules/db_postgres' > > Is this a problem of my psql version(8.4.20), because > PQconnectdbParams  is not available in that version. > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Thu Apr 23 10:49:03 2020 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Thu, 23 Apr 2020 13:49:03 +0300 Subject: [OpenSIPS-Users] Access to listen/advertised IP Addresses In-Reply-To: <5957bcba-f283-50e4-19a7-4b4d43f8b987@opensips.org> References: <5eb6a6ca-83aa-c59e-2f2a-2c4f216a4118@opensips.org> <5957bcba-f283-50e4-19a7-4b4d43f8b987@opensips.org> Message-ID: Hi Mark, check these new variables $socket_in() and $socket_out() in 3.1 https://www.opensips.org/Documentation/Script-CoreVar-3-1#socket_in https://www.opensips.org/Documentation/Script-CoreVar-3-1#socket_out Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com On 4/7/20 7:29 PM, Bogdan-Andrei Iancu wrote: > No need, just use in script, where ever you need > $socket_in(advertised_ip) and it will be evaluated for the current > socket (used for receiving the request) > > Regardsm > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > OpenSIPS Summit, Amsterdam, May 2020 > https://www.opensips.org/events/Summit-2020Amsterdam/ > > On 4/7/20 6:56 PM, Mark Farmer wrote: >> I was thinking something like: >> >> modparam("cfgutils", "varset", "extip=s:$in_socket.advertised_ip") >> >> >> On Tue, 7 Apr 2020 at 14:40, Bogdan-Andrei Iancu > > wrote: >> >> Hi Mark, >> >> ingenious solution :) >> >> In regards to the proposed solution, I do not understand the >> question about varset (cfgutils), as there is no relation between >> the script vars and these new $socket vars. Maybe I'm missing >> something from your question. >> >> Regards, >> >> Bogdan-Andrei Iancu >> >> OpenSIPS Founder and Developer >> https://www.opensips-solutions.com >> OpenSIPS Summit, Amsterdam, May 2020 >> https://www.opensips.org/events/Summit-2020Amsterdam/ >> >> On 4/7/20 2:10 PM, Mark Farmer wrote: >>> Hi Bogdan >>> >>> The root of my issue is that I need 2 variables containing the >>> IP's of my 2 interfaces (mhomed=yes) but the advertised address >>> of the NAT'd DMZ interface while keeping changes per server to a >>> bare minimum to ease deployment. >>> >>> I actually solved my issue by using include_file and using >>> cfgutils to set 2 script variables. So now all deployment >>> changes are confined to a much simpler/smaller file. >>> >>> However, the proposed changes would make things even nicer. >>> Would cfgutils be able to accept those variables as parameters >>> to the 'varset' function? >>> >>> Regards >>> Mark. >>> >>> >>> >>> On Tue, 7 Apr 2020 at 11:44, Bogdan-Andrei Iancu >>> > wrote: >>> >>> Hi guys, >>> >>> Maybe adding a new core variable like $in_socket.XXXX, to >>> give access to various fields, like $in_socket.ip, >>> $in_socket.port, $in_socket.advertised_ip, etc. This will >>> replace the $Ri and $Rp >>> >>> And we can also add $out_socket, that will similarly replace >>> the $fs (forced socket) >>> >>> Regards, >>> >>> Bogdan-Andrei Iancu >>> >>> OpenSIPS Founder and Developer >>> https://www.opensips-solutions.com >>> OpenSIPS Summit, Amsterdam, May 2020 >>> https://www.opensips.org/events/Summit-2020Amsterdam/ >>> >>> On 4/6/20 6:00 PM, Johan De Clercq wrote: >>>> It,s not exposed I think. I can’t find it back either >>>> >>>> Outlook voor iOS downloaden >>>> ------------------------------------------------------------------------ >>>> *Van:* Users >>>> namens David >>>> Villasmil >>>> >>>> *Verzonden:* Monday, April 6, 2020 4:49:36 PM >>>> *Aan:* OpenSIPS users mailling list >>>> >>>> *Onderwerp:* Re: [OpenSIPS-Users] Access to >>>> listen/advertised IP Addresses >>>> No, you’re right. It’s not in the core variables and I >>>> can’t find it either. Which makes me think it’s either not >>>> exposed or somewhere in a module (it’s not in proto_udp) >>>> >>>> I will research a little to try and find it.. >>>> >>>> On Mon, 6 Apr 2020 at 14:04, Mark Farmer >>> > wrote: >>>> >>>> Thanks David. But I see no reference to the same >>>> variable in OpenSIPS. >>>> >>>> https://www.opensips.org/Documentation/Script-CoreVar-2-4 >>>> >>>> Am I missing something? >>>> >>>> >>>> On Mon, 6 Apr 2020 at 13:45, David Villasmil >>>> >>> > wrote: >>>> >>>> Right here: >>>> >>>> https://www.kamailio.org/wiki/cookbooks/5.2.x/pseudovariables#rai_-_received_advertised_ip_address >>>> >>>> Regards, >>>> >>>> David Villasmil >>>> email: david.villasmil.work at gmail.com >>>> >>>> phone: +34669448337 >>>> >>>> >>>> On Mon, Apr 6, 2020 at 12:08 PM Mark Farmer >>>> > wrote: >>>> >>>> Many thanks for the reply. >>>> >>>> $Ri is certainly useful when the request comes >>>> from a non-natted interface. Thanks for >>>> pointing that out :) >>>> >>>> Is there a way to reference the advertised IP >>>> address defined in the listen statement? >>>> >>>> listen=udp:xxx.xxx.xxx.xxx:5060 as >>>> xxx.xxx.xxx.xxx:5060 >>>> >>>> Thanks >>>> Mark. >>>> >>>> >>>> On Thu, 2 Apr 2020 at 17:32, Sharad Kumar via >>>> Users >>> > wrote: >>>> >>>> Hi Mark, >>>> >>>>  If your initial goal is to get the >>>> interface IP where request is received then >>>> you can try these variables. >>>> >>>> *$Ri* - reference to IP address of the >>>> interface where the request has been received >>>> >>>> *$Rp* - reference to the port where the >>>> message was received >>>> >>>> _______________________________________________ >>>> Users mailing list >>>> Users at lists.opensips.org >>>> >>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>> >>>> >>>> >>>> -- >>>> Mark Farmer >>>> farmorg at gmail.com >>>> _______________________________________________ >>>> Users mailing list >>>> Users at lists.opensips.org >>>> >>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>> >>>> _______________________________________________ >>>> Users mailing list >>>> Users at lists.opensips.org >>>> >>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>> >>>> >>>> >>>> -- >>>> Mark Farmer >>>> farmorg at gmail.com >>>> _______________________________________________ >>>> Users mailing list >>>> Users at lists.opensips.org >>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>> >>>> -- >>>> 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 >>> >>> >>> >>> -- >>> Mark Farmer >>> farmorg at gmail.com >>> >>> _______________________________________________ >>> Users mailing list >>> Users at lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> >> >> -- >> Mark Farmer >> farmorg at gmail.com > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From volga629 at networklab.ca Thu Apr 23 12:26:50 2020 From: volga629 at networklab.ca (volga629) Date: Thu, 23 Apr 2020 09:26:50 -0300 Subject: [OpenSIPS-Users] string comparison Message-ID: <5e6e5656-a785-5b86-a6e9-c591f4442249@skillsearch.ca> An HTML attachment was scrubbed... URL: From john.quick at smartvox.co.uk Thu Apr 23 12:41:00 2020 From: john.quick at smartvox.co.uk (John Quick) Date: Thu, 23 Apr 2020 13:41:00 +0100 Subject: [OpenSIPS-Users] string comparison Message-ID: <002701d6196c$71dbf050$5593d0f0$@smartvox.co.uk> Hi Volga Please could you always format your emails to the users group as plain text, not HTML. I had to copy and paste then save and re-open just to read your question. I would think the most likely explanation for the string comparison failing is that you are comparing an $avp with a $var They are different - in particular, an AVP can hold several values, somewhat like an array type in other languages. Can you try the same test but with a var for both sides of the comparison. John Quick Smartvox Limited From liviu at opensips.org Thu Apr 23 13:34:38 2020 From: liviu at opensips.org (Liviu Chircu) Date: Thu, 23 Apr 2020 16:34:38 +0300 Subject: [OpenSIPS-Users] string comparison In-Reply-To: <5e6e5656-a785-5b86-a6e9-c591f4442249@skillsearch.ca> References: <5e6e5656-a785-5b86-a6e9-c591f4442249@skillsearch.ca> Message-ID: <9362669e-d19f-a1f3-f90c-f53dd7668fb4@opensips.org> On 23.04.2020 15:26, volga629 via Users wrote: > > $var(usr_check_tls) = "tls_" + $(tU{s.select,0,%}); > > Second  var > > cache_fetch("redis:wss-grp","tls-frompbx", $avp(tls-frompbx)) > > This return always false > > if($avp(tls-frompbx)==$var(usr_check_tls)) { > Hi, Volga! Maybe there is some route being called in the meantime or some other logic.  The comparison operators are working well on all latest 2.4+ versions, you can test for yourself with a simple opensips.cfg:     $var(x) = "201%123";     $var(usr_check_tls) = "tls_" + $(var(x){s.select,0,%});     $avp(tls-frompbx) = "tls_201";     $avp(tls-frompbx) = "tls_201";     assert($avp(tls-frompbx) == $var(usr_check_tls), "avp-var-1"); Regards, -- Liviu Chircu www.twitter.com/liviuchircu | www.opensips-solutions.com OpenSIPS Summit, Amsterdam, Fall 2020 www.opensips.org/events -------------- next part -------------- An HTML attachment was scrubbed... URL: From alex.a at gtanetworkconsulting.com Thu Apr 23 13:37:44 2020 From: alex.a at gtanetworkconsulting.com (Alex A) Date: Thu, 23 Apr 2020 09:37:44 -0400 Subject: [OpenSIPS-Users] High Volume Accouting backend options In-Reply-To: <171a268e042.fa35ce09171614.5445550439030139138@gtanetworkconsulting.com> References: <171a268e042.fa35ce09171614.5445550439030139138@gtanetworkconsulting.com> Message-ID: <171a7422dea.1247b83848131.3910812018483636749@gtanetworkconsulting.com> Hi, We are looking to deploy accounting/homer integration on Opensips 3.0.2. As the first step deployed acc module with pgsql backend. The config seem to be pretty straight-forward - see attached. It appears that as soon as volume hits about 30-35k in_use transactions - the server stops replying to new requests (or give 408 Timeout) and syslog gets filled with: Apr 22 10:19:38 opensip1 opensips: Apr 22 10:19:38 [19258] CRITICAL:tm:set_timer: set_timer for 1 list called on a "detached" timer -- ignoring: 0x7fb63b993cf8 Apr 22 10:19:40 opensip1 opensips: Apr 22 10:19:40 [19255] CRITICAL:tm:set_timer: set_timer for 1 list called on a "detached" timer -- ignoring: 0x7fb638cf9a40 Apr 22 10:19:40 opensip1 opensips: Apr 22 10:19:40 [19260] CRITICAL:tm:set_timer: set_timer for 1 list called on a "detached" timer -- ignoring: 0x7fb63f3b23c8 Apr 22 10:19:49 opensip1 opensips: Apr 22 10:19:49 [19258] CRITICAL:tm:set_timer: set_timer for 1 list called on a "detached" timer -- ignoring: 0x7fb63eb93a80 Apr 22 10:20:01 opensip1 opensips: Apr 22 10:20:01 [19267] CRITICAL:tm:set_timer: set_timer for 1 list called on a "detached" timer -- ignoring: 0x7fb5de690700 Although, the remote Postgres service is on SSD with relatively small network latency, it appears to be the bottleneck. The initial assumption was the Opensips acc uses no-blocking SQL calls (since cdrs are not real-time). Another observation: opensips only opens 20 SQL connections to postgres via tcp 5432.  I have tried playing with db_max_async_connections, however to no avail. Any suggestions to troubleshoot ? or any alternatives for accounting in high volume applications would be greatly appreciated. Thank you. -------------- next part -------------- An HTML attachment was scrubbed... URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: opensips.cfg Type: application/octet-stream Size: 7774 bytes Desc: not available URL: From volga629 at networklab.ca Thu Apr 23 13:43:09 2020 From: volga629 at networklab.ca (volga629) Date: Thu, 23 Apr 2020 10:43:09 -0300 Subject: [OpenSIPS-Users] string comparison In-Reply-To: <002701d6196c$71dbf050$5593d0f0$@smartvox.co.uk> References: <002701d6196c$71dbf050$5593d0f0$@smartvox.co.uk> Message-ID: <57c9bc1f-41f1-6925-439e-eca135a87b62@skillsearch.ca> Hello John, I tested var to var and test still failing Tried  if($var(tls-frompbx)=="$var(usr_check_tls)") or this  if($var(tls-frompbx)==$var(usr_check_tls)) volga629 On 4/23/20 9:41 AM, John Quick wrote: > Hi Volga > > Please could you always format your emails to the users group as plain text, > not HTML. > I had to copy and paste then save and re-open just to read your question. > > I would think the most likely explanation for the string comparison failing > is that you are comparing an $avp with a $var > They are different - in particular, an AVP can hold several values, somewhat > like an array type in other languages. > Can you try the same test but with a var for both sides of the comparison. > > John Quick > Smartvox Limited > > > From ivailod at telera.eu Thu Apr 23 14:05:53 2020 From: ivailod at telera.eu (Ivailo Dobrev) Date: Thu, 23 Apr 2020 17:05:53 +0300 Subject: [OpenSIPS-Users] string comparison In-Reply-To: <57c9bc1f-41f1-6925-439e-eca135a87b62@skillsearch.ca> References: <002701d6196c$71dbf050$5593d0f0$@smartvox.co.uk> <57c9bc1f-41f1-6925-439e-eca135a87b62@skillsearch.ca> Message-ID: <278ea628-15e4-1340-287c-b821183dae9b@telera.eu> Hi Volga, xlog  md5 of both vars before if statement. I'm pretty sure they will be different. Maybe not printable character ? On 4/23/20 4:43 PM, volga629 via Users wrote: > Hello John, > > I tested var to var > > and test still failing > > Tried > >  if($var(tls-frompbx)=="$var(usr_check_tls)") > > or this > >  if($var(tls-frompbx)==$var(usr_check_tls)) > > volga629 > > > On 4/23/20 9:41 AM, John Quick wrote: >> Hi Volga >> >> Please could you always format your emails to the users group as >> plain text, >> not HTML. >> I had to copy and paste then save and re-open just to read your >> question. >> >> I would think the most likely explanation for the string comparison >> failing >> is that you are comparing an $avp with a $var >> They are different - in particular, an AVP can hold several values, >> somewhat >> like an array type in other languages. >> Can you try the same test but with a var for both sides of the >> comparison. >> >> John Quick >> Smartvox Limited >> >> >> > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users From john.quick at smartvox.co.uk Thu Apr 23 14:26:49 2020 From: john.quick at smartvox.co.uk (John Quick) Date: Thu, 23 Apr 2020 15:26:49 +0100 Subject: [OpenSIPS-Users] string comparison In-Reply-To: <57c9bc1f-41f1-6925-439e-eca135a87b62@skillsearch.ca> References: <002701d6196c$71dbf050$5593d0f0$@smartvox.co.uk> <57c9bc1f-41f1-6925-439e-eca135a87b62@skillsearch.ca> Message-ID: <002d01d6197b$39f4e1b0$addea510$@smartvox.co.uk> Hi Volga, I just added a check on one of my test servers, making use of some existing code that sets a variable $var(srctype) to various values including: if (is_registered("location")) $var(srctype) = "regd-user"; The new code I added just below looks like this: $var(teststr) = "regd-user"; if ($var(srctype) == $var(teststr)) xlog("L_WARN", "## SOURCE IS A REGISTERED USER ##\n"); Then I registered an IP Phone and made a call. Output to my log file: 2020-04-23 15:10:40 ## SOURCE IS A REGISTERED USER ## So the basic test for comparing strings should work. There must be some other explanation. I have known situations where it was necessary to cast a string value to be recognised as an integer, but you are only testing strings. Could there be a leading or trailing space? Perhaps you need to use a trim function. Check out the string transformations: https://www.opensips.org/Documentation/Script-Tran-2-4#toc1 To understand what is happening you could try using regex test, =~ instead of equality, == You could report the length of the string values stored in the two vars, using {s.len} transformation (see above link). John Quick Smartvox Limited -----Original Message----- From: volga629 Sent: 23 April 2020 14:43 To: john.quick at smartvox.co.uk; users at lists.opensips.org Subject: Re: [OpenSIPS-Users] string comparison Hello John, I tested var to var and test still failing Tried if($var(tls-frompbx)=="$var(usr_check_tls)") or this if($var(tls-frompbx)==$var(usr_check_tls)) volga629 On 4/23/20 9:41 AM, John Quick wrote: > Hi Volga > > Please could you always format your emails to the users group as plain > text, not HTML. > I had to copy and paste then save and re-open just to read your question. > > I would think the most likely explanation for the string comparison > failing is that you are comparing an $avp with a $var They are > different - in particular, an AVP can hold several values, somewhat > like an array type in other languages. > Can you try the same test but with a var for both sides of the comparison. > > John Quick > Smartvox Limited > > > From volga629 at networklab.ca Thu Apr 23 14:31:25 2020 From: volga629 at networklab.ca (volga629) Date: Thu, 23 Apr 2020 11:31:25 -0300 Subject: [OpenSIPS-Users] string comparison In-Reply-To: <278ea628-15e4-1340-287c-b821183dae9b@telera.eu> References: <002701d6196c$71dbf050$5593d0f0$@smartvox.co.uk> <57c9bc1f-41f1-6925-439e-eca135a87b62@skillsearch.ca> <278ea628-15e4-1340-287c-b821183dae9b@telera.eu> Message-ID: <48421947-340b-8cee-45de-fe3ccb3c4457@skillsearch.ca> Hello Ivailo, I tested md5 and  values are the same. xlog("MD5 over cached user ~> $(avp(tls-frompbx){s.md5}) checked user ~> $(avp(usr_check_tls){s.md5})\n"); Log Apr 23 16:23:11 pr1 /usr/sbin/opensips[12598]: ---setting as BLF callee Apr 23 16:23:11 pr1 /usr/sbin/opensips[12598]: Number of calls from user 100 at demo.sip.lan is 2 Apr 23 16:23:11 pr1 /usr/sbin/opensips[12598]: Number of calls from domain demo.sip.lan is 0 Apr 23 16:23:11 pr1 /usr/sbin/opensips[12598]: Is [INVITE] from fs ~> 192.168.50.12 and sip:201%40demo.sip.lan at 192.168.50.10:5060 Apr 23 16:23:11 pr1 /usr/sbin/opensips[12598]: looking up [sip:201%40demo.sip.lan at 192.168.50.10:5060] Apr 23 16:23:11 pr1 /usr/sbin/opensips[12598]: WSS: [INVITE] found transport ~> tls Apr 23 16:23:11 pr1 /usr/sbin/opensips[12598]: [INVITE] fetched var ~> tls_201 Apr 23 16:23:11 pr1 /usr/sbin/opensips[12598]: MD5 over cached user ~> b384583ffd6d280b18286afa30399850 checked user ~> b384583ffd6d280b18286afa30399850 Apr 23 16:23:11 pr1 /usr/sbin/opensips[12598]: [INVITE] Call going from LAN SouceIP <192.168.50.12> to WAN with transport set ~> [tcp MobileFlag] original transport [tls_201] and checked transport [tls_201] Apr 23 16:23:11 pr1 /usr/sbin/opensips[12583]: OnReply_Route3: [INVITE] fetched var ~> plain_100 Apr 23 16:23:11 pr1 /usr/sbin/opensips[12583]: OnReply_Route3: [INVITE] and call status 183 and transport plain_100 Apr 23 16:23:11 pr1 /usr/sbin/opensips[12583]: OnReply_Route3: [INVITE] [183] Default match [FS ~> Client] Apr 23 16:23:12 pr1 /usr/sbin/opensips[12591]: OnReply_Route3: [INVITE] fetched var ~> plain_100 volga629 On 4/23/20 11:05 AM, Ivailo Dobrev wrote: > Hi Volga, > > xlog  md5 of both vars before if statement. I'm pretty sure they will > be different. Maybe not printable character ? > > On 4/23/20 4:43 PM, volga629 via Users wrote: >> Hello John, >> >> I tested var to var >> >> and test still failing >> >> Tried >> >>  if($var(tls-frompbx)=="$var(usr_check_tls)") >> >> or this >> >>  if($var(tls-frompbx)==$var(usr_check_tls)) >> >> volga629 >> >> >> On 4/23/20 9:41 AM, John Quick wrote: >>> Hi Volga >>> >>> Please could you always format your emails to the users group as >>> plain text, >>> not HTML. >>> I had to copy and paste then save and re-open just to read your >>> question. >>> >>> I would think the most likely explanation for the string comparison >>> failing >>> is that you are comparing an $avp with a $var >>> They are different - in particular, an AVP can hold several values, >>> somewhat >>> like an array type in other languages. >>> Can you try the same test but with a var for both sides of the >>> comparison. >>> >>> John Quick >>> Smartvox Limited >>> >>> >>> >> >> _______________________________________________ >> 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 ivailod at telera.eu Thu Apr 23 14:44:22 2020 From: ivailod at telera.eu (Ivailo Dobrev) Date: Thu, 23 Apr 2020 17:44:22 +0300 Subject: [OpenSIPS-Users] string comparison In-Reply-To: <48421947-340b-8cee-45de-fe3ccb3c4457@skillsearch.ca> References: <002701d6196c$71dbf050$5593d0f0$@smartvox.co.uk> <57c9bc1f-41f1-6925-439e-eca135a87b62@skillsearch.ca> <278ea628-15e4-1340-287c-b821183dae9b@telera.eu> <48421947-340b-8cee-45de-fe3ccb3c4457@skillsearch.ca> Message-ID: <07e30839-11a6-a2f9-1169-a3168140048b@telera.eu> Hi Volga Maybe I'm missing something but "if($avp(tls-frompbx)==$var(usr_check_tls))" !!! AVP = VAR !!! In xlog both vars are AVP. On 4/23/20 5:31 PM, volga629 wrote: > Hello Ivailo, > > I tested md5 and  values are the same. > > > xlog("MD5 over cached user ~> $(avp(tls-frompbx){s.md5}) checked user > ~> $(avp(usr_check_tls){s.md5})\n"); > > > Log > > Apr 23 16:23:11 pr1 /usr/sbin/opensips[12598]: ---setting as BLF > callee > Apr 23 16:23:11 pr1 /usr/sbin/opensips[12598]: Number of calls from > user 100 at demo.sip.lan is 2 > Apr 23 16:23:11 pr1 /usr/sbin/opensips[12598]: Number of calls from > domain demo.sip.lan is 0 > Apr 23 16:23:11 pr1 /usr/sbin/opensips[12598]: Is [INVITE] from fs ~> > 192.168.50.12 and sip:201%40demo.sip.lan at 192.168.50.10:5060 > Apr 23 16:23:11 pr1 /usr/sbin/opensips[12598]: looking up > [sip:201%40demo.sip.lan at 192.168.50.10:5060] > Apr 23 16:23:11 pr1 /usr/sbin/opensips[12598]: WSS: [INVITE] found > transport ~> tls > Apr 23 16:23:11 pr1 /usr/sbin/opensips[12598]: [INVITE] fetched var ~> > tls_201 > Apr 23 16:23:11 pr1 /usr/sbin/opensips[12598]: MD5 over cached user ~> > b384583ffd6d280b18286afa30399850 checked user ~> > b384583ffd6d280b18286afa30399850 > Apr 23 16:23:11 pr1 /usr/sbin/opensips[12598]: [INVITE] Call going > from LAN SouceIP <192.168.50.12> to WAN with transport set ~> [tcp > MobileFlag] original transport [tls_201] and checked transport [tls_201] > Apr 23 16:23:11 pr1 /usr/sbin/opensips[12583]: OnReply_Route3: > [INVITE] fetched var ~> plain_100 > Apr 23 16:23:11 pr1 /usr/sbin/opensips[12583]: OnReply_Route3: > [INVITE] and call status 183 and transport plain_100 > Apr 23 16:23:11 pr1 /usr/sbin/opensips[12583]: OnReply_Route3: > [INVITE] [183] Default match [FS ~> Client] > Apr 23 16:23:12 pr1 /usr/sbin/opensips[12591]: OnReply_Route3: > [INVITE] fetched var ~> plain_100 > > > volga629 > > > On 4/23/20 11:05 AM, Ivailo Dobrev wrote: >> Hi Volga, >> >> xlog  md5 of both vars before if statement. I'm pretty sure they will >> be different. Maybe not printable character ? >> >> On 4/23/20 4:43 PM, volga629 via Users wrote: >>> Hello John, >>> >>> I tested var to var >>> >>> and test still failing >>> >>> Tried >>> >>>  if($var(tls-frompbx)=="$var(usr_check_tls)") >>> >>> or this >>> >>>  if($var(tls-frompbx)==$var(usr_check_tls)) >>> >>> volga629 >>> >>> >>> On 4/23/20 9:41 AM, John Quick wrote: >>>> Hi Volga >>>> >>>> Please could you always format your emails to the users group as >>>> plain text, >>>> not HTML. >>>> I had to copy and paste then save and re-open just to read your >>>> question. >>>> >>>> I would think the most likely explanation for the string comparison >>>> failing >>>> is that you are comparing an $avp with a $var >>>> They are different - in particular, an AVP can hold several values, >>>> somewhat >>>> like an array type in other languages. >>>> Can you try the same test but with a var for both sides of the >>>> comparison. >>>> >>>> John Quick >>>> Smartvox Limited >>>> >>>> >>>> >>> >>> _______________________________________________ >>> 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 volga629 at networklab.ca Thu Apr 23 14:48:16 2020 From: volga629 at networklab.ca (volga629) Date: Thu, 23 Apr 2020 11:48:16 -0300 Subject: [OpenSIPS-Users] string comparison In-Reply-To: <07e30839-11a6-a2f9-1169-a3168140048b@telera.eu> References: <002701d6196c$71dbf050$5593d0f0$@smartvox.co.uk> <57c9bc1f-41f1-6925-439e-eca135a87b62@skillsearch.ca> <278ea628-15e4-1340-287c-b821183dae9b@telera.eu> <48421947-340b-8cee-45de-fe3ccb3c4457@skillsearch.ca> <07e30839-11a6-a2f9-1169-a3168140048b@telera.eu> Message-ID: <821a3d30-63b0-edd0-f417-6d2c0ef86cd4@skillsearch.ca> Yes, I set all to avp volga629 On 4/23/20 11:44 AM, Ivailo Dobrev wrote: > > Hi Volga > > Maybe I'm missing something but > "if($avp(tls-frompbx)==$var(usr_check_tls))" !!! AVP = VAR !!! > > In xlog both vars are AVP. > > > On 4/23/20 5:31 PM, volga629 wrote: > >> Hello Ivailo, >> >> I tested md5 and  values are the same. >> >> >> xlog("MD5 over cached user ~> $(avp(tls-frompbx){s.md5}) checked user >> ~> $(avp(usr_check_tls){s.md5})\n"); >> >> >> Log >> >> Apr 23 16:23:11 pr1 /usr/sbin/opensips[12598]: ---setting as BLF >> callee >> Apr 23 16:23:11 pr1 /usr/sbin/opensips[12598]: Number of calls from >> user 100 at demo.sip.lan is 2 >> Apr 23 16:23:11 pr1 /usr/sbin/opensips[12598]: Number of calls from >> domain demo.sip.lan is 0 >> Apr 23 16:23:11 pr1 /usr/sbin/opensips[12598]: Is [INVITE] from fs ~> >> 192.168.50.12 and sip:201%40demo.sip.lan at 192.168.50.10:5060 >> Apr 23 16:23:11 pr1 /usr/sbin/opensips[12598]: looking up >> [sip:201%40demo.sip.lan at 192.168.50.10:5060] >> Apr 23 16:23:11 pr1 /usr/sbin/opensips[12598]: WSS: [INVITE] found >> transport ~> tls >> Apr 23 16:23:11 pr1 /usr/sbin/opensips[12598]: [INVITE] fetched var >> ~> tls_201 >> Apr 23 16:23:11 pr1 /usr/sbin/opensips[12598]: MD5 over cached user >> ~> b384583ffd6d280b18286afa30399850 checked user ~> >> b384583ffd6d280b18286afa30399850 >> Apr 23 16:23:11 pr1 /usr/sbin/opensips[12598]: [INVITE] Call going >> from LAN SouceIP <192.168.50.12> to WAN with transport set ~> [tcp >> MobileFlag] original transport [tls_201] and checked transport [tls_201] >> Apr 23 16:23:11 pr1 /usr/sbin/opensips[12583]: OnReply_Route3: >> [INVITE] fetched var ~> plain_100 >> Apr 23 16:23:11 pr1 /usr/sbin/opensips[12583]: OnReply_Route3: >> [INVITE] and call status 183 and transport plain_100 >> Apr 23 16:23:11 pr1 /usr/sbin/opensips[12583]: OnReply_Route3: >> [INVITE] [183] Default match [FS ~> Client] >> Apr 23 16:23:12 pr1 /usr/sbin/opensips[12591]: OnReply_Route3: >> [INVITE] fetched var ~> plain_100 >> >> >> volga629 >> >> >> On 4/23/20 11:05 AM, Ivailo Dobrev wrote: >>> Hi Volga, >>> >>> xlog  md5 of both vars before if statement. I'm pretty sure they >>> will be different. Maybe not printable character ? >>> >>> On 4/23/20 4:43 PM, volga629 via Users wrote: >>>> Hello John, >>>> >>>> I tested var to var >>>> >>>> and test still failing >>>> >>>> Tried >>>> >>>>  if($var(tls-frompbx)=="$var(usr_check_tls)") >>>> >>>> or this >>>> >>>>  if($var(tls-frompbx)==$var(usr_check_tls)) >>>> >>>> volga629 >>>> >>>> >>>> On 4/23/20 9:41 AM, John Quick wrote: >>>>> Hi Volga >>>>> >>>>> Please could you always format your emails to the users group as >>>>> plain text, >>>>> not HTML. >>>>> I had to copy and paste then save and re-open just to read your >>>>> question. >>>>> >>>>> I would think the most likely explanation for the string >>>>> comparison failing >>>>> is that you are comparing an $avp with a $var >>>>> They are different - in particular, an AVP can hold several >>>>> values, somewhat >>>>> like an array type in other languages. >>>>> Can you try the same test but with a var for both sides of the >>>>> comparison. >>>>> >>>>> John Quick >>>>> Smartvox Limited >>>>> >>>>> >>>>> >>>> >>>> _______________________________________________ >>>> Users mailing list >>>> Users at lists.opensips.org >>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >>> _______________________________________________ >>> Users mailing list >>> Users at lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users From bogdan at opensips.org Thu Apr 23 14:53:33 2020 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Thu, 23 Apr 2020 17:53:33 +0300 Subject: [OpenSIPS-Users] High Volume Accouting backend options In-Reply-To: <171a7422dea.1247b83848131.3910812018483636749@gtanetworkconsulting.com> References: <171a268e042.fa35ce09171614.5445550439030139138@gtanetworkconsulting.com> <171a7422dea.1247b83848131.3910812018483636749@gtanetworkconsulting.com> Message-ID: <1204dcfb-db50-1b41-7b87-0050bce972db@opensips.org> Hi Alex, Typical approach in this case is to do the accounting via a very fast backend (like db_flatstore, into a text file) and import the files off-sync into db (like every 5 mins). Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com On 4/23/20 4:37 PM, Alex A wrote: > Hi, > > > We are looking to deploy accounting/homer integration on Opensips > 3.0.2. > As the first step deployed acc module with pgsql backend. > > The config seem to be pretty straight-forward - see attached. > > It appears that as soon as volume hits about 30-35k in_use > transactions - the server stops replying to new requests (or give > 408 Timeout) and syslog gets filled with: > > Apr 22 10:19:38 opensip1 opensips: Apr 22 10:19:38 [19258] > CRITICAL:tm:set_timer: set_timer for 1 list called on a "detached" > timer -- ignoring: 0x7fb63b993cf8 > Apr 22 10:19:40 opensip1 opensips: Apr 22 10:19:40 [19255] > CRITICAL:tm:set_timer: set_timer for 1 list called on a "detached" > timer -- ignoring: 0x7fb638cf9a40 > Apr 22 10:19:40 opensip1 opensips: Apr 22 10:19:40 [19260] > CRITICAL:tm:set_timer: set_timer for 1 list called on a "detached" > timer -- ignoring: 0x7fb63f3b23c8 > Apr 22 10:19:49 opensip1 opensips: Apr 22 10:19:49 [19258] > CRITICAL:tm:set_timer: set_timer for 1 list called on a "detached" > timer -- ignoring: 0x7fb63eb93a80 > Apr 22 10:20:01 opensip1 opensips: Apr 22 10:20:01 [19267] > CRITICAL:tm:set_timer: set_timer for 1 list called on a "detached" > timer -- ignoring: 0x7fb5de690700 > > > Although, the remote Postgres service is on SSD with relatively > small network latency, it appears to be the bottleneck. > The initial assumption was the Opensips acc uses no-blocking SQL > calls (since cdrs are not real-time). > > > Another observation: > opensips only opens 20 SQL connections to postgres via tcp 5432.  > I have tried playing with db_max_async_connections, however to no > avail. > > > Any suggestions to troubleshoot ? or any alternatives for > accounting in high volume applications would be greatly appreciated. > > > 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 Ben.Newlin at genesys.com Thu Apr 23 15:02:26 2020 From: Ben.Newlin at genesys.com (Ben Newlin) Date: Thu, 23 Apr 2020 15:02:26 +0000 Subject: [OpenSIPS-Users] High Volume Accouting backend options In-Reply-To: <1204dcfb-db50-1b41-7b87-0050bce972db@opensips.org> References: <171a268e042.fa35ce09171614.5445550439030139138@gtanetworkconsulting.com> <171a7422dea.1247b83848131.3910812018483636749@gtanetworkconsulting.com> <1204dcfb-db50-1b41-7b87-0050bce972db@opensips.org> Message-ID: As far as I know, the db_postgres module does not implement async support. So the CDR calls to DB *will* be blocking and the max_async parameter has no effect. This is one of the reasons we use Bogdan’s suggested approach; it makes the write of the CDR local and very fast and they can then be moved off asynchronously. The actual number of connections open to the DB will be one per each OpenSIPS process that uses the DB. Often this is only the listener processes, so if you have 20 DB connections open consistently I’m guessing you are running with 20 children/listeners? Ben Newlin From: Users on behalf of Bogdan-Andrei Iancu Reply-To: OpenSIPS users mailling list Date: Thursday, April 23, 2020 at 10:55 AM To: OpenSIPS users mailling list , Alex A Subject: Re: [OpenSIPS-Users] High Volume Accouting backend options Hi Alex, Typical approach in this case is to do the accounting via a very fast backend (like db_flatstore, into a text file) and import the files off-sync into db (like every 5 mins). Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com On 4/23/20 4:37 PM, Alex A wrote: Hi, We are looking to deploy accounting/homer integration on Opensips 3.0.2. As the first step deployed acc module with pgsql backend. The config seem to be pretty straight-forward - see attached. It appears that as soon as volume hits about 30-35k in_use transactions - the server stops replying to new requests (or give 408 Timeout) and syslog gets filled with: Apr 22 10:19:38 opensip1 opensips: Apr 22 10:19:38 [19258] CRITICAL:tm:set_timer: set_timer for 1 list called on a "detached" timer -- ignoring: 0x7fb63b993cf8 Apr 22 10:19:40 opensip1 opensips: Apr 22 10:19:40 [19255] CRITICAL:tm:set_timer: set_timer for 1 list called on a "detached" timer -- ignoring: 0x7fb638cf9a40 Apr 22 10:19:40 opensip1 opensips: Apr 22 10:19:40 [19260] CRITICAL:tm:set_timer: set_timer for 1 list called on a "detached" timer -- ignoring: 0x7fb63f3b23c8 Apr 22 10:19:49 opensip1 opensips: Apr 22 10:19:49 [19258] CRITICAL:tm:set_timer: set_timer for 1 list called on a "detached" timer -- ignoring: 0x7fb63eb93a80 Apr 22 10:20:01 opensip1 opensips: Apr 22 10:20:01 [19267] CRITICAL:tm:set_timer: set_timer for 1 list called on a "detached" timer -- ignoring: 0x7fb5de690700 Although, the remote Postgres service is on SSD with relatively small network latency, it appears to be the bottleneck. The initial assumption was the Opensips acc uses no-blocking SQL calls (since cdrs are not real-time). Another observation: opensips only opens 20 SQL connections to postgres via tcp 5432. I have tried playing with db_max_async_connections, however to no avail. Any suggestions to troubleshoot ? or any alternatives for accounting in high volume applications would be greatly appreciated. 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 volga629 at networklab.ca Thu Apr 23 15:26:22 2020 From: volga629 at networklab.ca (volga629) Date: Thu, 23 Apr 2020 12:26:22 -0300 Subject: [OpenSIPS-Users] string comparison In-Reply-To: <002d01d6197b$39f4e1b0$addea510$@smartvox.co.uk> References: <002701d6196c$71dbf050$5593d0f0$@smartvox.co.uk> <57c9bc1f-41f1-6925-439e-eca135a87b62@skillsearch.ca> <002d01d6197b$39f4e1b0$addea510$@smartvox.co.uk> Message-ID: Hello John, Thank you  for reply. I wonder if it something to do that variable contain numbers and letters. Example: tls_201 I wonder if I can compare md5 . volga629 On 4/23/20 11:26 AM, John Quick wrote: > Hi Volga, > > I just added a check on one of my test servers, making use of some existing code that sets a variable $var(srctype) to various values including: > > if (is_registered("location")) > $var(srctype) = "regd-user"; > > The new code I added just below looks like this: > > $var(teststr) = "regd-user"; > if ($var(srctype) == $var(teststr)) > xlog("L_WARN", "## SOURCE IS A REGISTERED USER ##\n"); > > Then I registered an IP Phone and made a call. Output to my log file: > 2020-04-23 15:10:40 ## SOURCE IS A REGISTERED USER ## > > So the basic test for comparing strings should work. There must be some other explanation. > I have known situations where it was necessary to cast a string value to be recognised as an integer, but you are only testing strings. > Could there be a leading or trailing space? Perhaps you need to use a trim function. Check out the string transformations: > https://www.opensips.org/Documentation/Script-Tran-2-4#toc1 > To understand what is happening you could try using regex test, =~ instead of equality, == > You could report the length of the string values stored in the two vars, using {s.len} transformation (see above link). > > John Quick > Smartvox Limited > > > -----Original Message----- > From: volga629 > Sent: 23 April 2020 14:43 > To: john.quick at smartvox.co.uk; users at lists.opensips.org > Subject: Re: [OpenSIPS-Users] string comparison > > Hello John, > > I tested var to var > > and test still failing > > Tried > > if($var(tls-frompbx)=="$var(usr_check_tls)") > > or this > > if($var(tls-frompbx)==$var(usr_check_tls)) > > volga629 > > > On 4/23/20 9:41 AM, John Quick wrote: >> Hi Volga >> >> Please could you always format your emails to the users group as plain >> text, not HTML. >> I had to copy and paste then save and re-open just to read your question. >> >> I would think the most likely explanation for the string comparison >> failing is that you are comparing an $avp with a $var They are >> different - in particular, an AVP can hold several values, somewhat >> like an array type in other languages. >> Can you try the same test but with a var for both sides of the comparison. >> >> John Quick >> Smartvox Limited >> >> >> > From nnikeshala at yahoo.com Thu Apr 23 19:18:26 2020 From: nnikeshala at yahoo.com (Nayani Nikeshala) Date: Thu, 23 Apr 2020 19:18:26 +0000 (UTC) Subject: [OpenSIPS-Users] OpenSIPS installation with PostgresSQL In-Reply-To: <2d3c288a-4f71-d78b-6d83-84e2731fc2e5@opensips.org> References: <129541852.2099756.1587328682918.ref@mail.yahoo.com> <129541852.2099756.1587328682918@mail.yahoo.com> <1985829642.2102360.1587329441034@mail.yahoo.com> <7812bb25-eef4-0b34-04f5-a4f5c5ba589f@opensips.org> <1950080153.2366807.1587399844986@mail.yahoo.com> <1582965925.176731.1587476879885@mail.yahoo.com> <1193421104.181467.1587478895108@mail.yahoo.com> <1971406255.325015.1587497750288@mail.yahoo.com> <15e3ac1f-2e01-a4fb-147a-5ac32411c98d@opensips.org> <1119346896.578368.1587559461192@mail.yahoo.com> <2d3c288a-4f71-d78b-6d83-84e2731fc2e5@opensips.org> Message-ID: <1876180299.356916.1587669506692@mail.yahoo.com> Hi Bogdan, I could fix the problem by modifying Makefile of postgres module. Thanks for the support. On Thursday, April 23, 2020, 02:21:00 PM GMT+5:30, Bogdan-Andrei Iancu wrote: Hi, I have an 9.5.19 (on Ubuntu) and works ok. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com On 4/22/20 3:44 PM, Nayani Nikeshala wrote: Hi Bogdan, I found an old email, where he has faced a similar kind of an issue with PSQL 9.2 also. I believe, it will not get resolved with PSQL version.  https://opensips.org/pipermail/devel/2015-June/017666.html According to the email, he has resolved it by modifying makefile, but I'm not sure, how he did that. On Wednesday, April 22, 2020, 01:54:09 PM GMT+5:30, Bogdan-Andrei Iancu wrote: It may sound weird, but maybe you into the other extrema, with some too new version :D... Is it possible to try some 9.3 or so ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com On 4/21/20 10:35 PM, Nayani Nikeshala wrote: Hi Bogdan, I removed psql 8.4 and installed psql 9.6.  [root at SIPserver opensips-2.4.7]# yum list installed | grep postgre postgresql96.i686        9.6.17-1PGDG.rhel6 postgresql96-devel.i686  9.6.17-1PGDG.rhel6 postgresql96-libs.i686   9.6.17-1PGDG.rhel6 postgresql96-server.i686 9.6.17-1PGDG.rhel6 [root at SIPserver opensips-2.4.7]# When I compile, I get below error. (Below is a part of the error) make[1]: Entering directory `/home/cscore/Downloads/opensips-2.4.7/modules/db_postgres' Compiling dbase.c Compiling pg_con.c Compiling res.c Compiling val.c In file included from pg_con.c:23: pg_con.h:39:22: error: libpq-fe.h: No such file or directory In file included from pg_con.c:23: pg_con.h:54: error: expected specifier-qualifier-list before ‘PGconn’ pg_con.c: In function ‘db_postgres_new_connection’: pg_con.c:93: error: ‘struct pg_con’ has no member named ‘con’ pg_con.c:93: warning: implicit declaration of function ‘PQconnectdbParams’ libpq-fe.h & pg_config are in the below paths.   [root at SIPserver opensips-2.4.7]# ls -lrt /usr/pgsql-9.6/include/libpq-fe.h -rw-r--r--. 1 root root 21900 Feb 12 02:52 /usr/pgsql-9.6/include/libpq-fe.h [root at SIPserver opensips-2.4.7]# [root at SIPserver opensips-2.4.7]# ls -lrt /usr/pgsql-9.6/bin/pg_config -rwxr-xr-x. 1 root root 24544 Feb 12 02:52 /usr/pgsql-9.6/bin/pg_config Could you assist me with above error.  On Tuesday, April 21, 2020, 08:57:13 PM GMT+5:30, Bogdan-Andrei Iancu wrote: Hi Nayani, yeah, that is a rather old and unsupported version of postgres. I think you should try to use a newer version it (maybe a newer version of your distro). Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com On 4/21/20 5:21 PM, Nayani Nikeshala wrote: Hi Bogdan, I have attached the output log for the above commands.  "make modules module=modules/db_postgres" gave me an error as in the attached file, so I used "makeinclude_modules=db_postgres modules" instead. I could see below in the output.  make[1]: Entering directory `/home/cscore/Downloads/opensips-2.4.7/modules/db_postgres' make[1]: Leaving directory `/home/cscore/Downloads/opensips-2.4.7/modules/db_postgres' make[1]: Entering directory `/home/cscore/Downloads/opensips-2.4.7/modules/db_postgres' Compiling dbase.c Compiling db_postgres.c Compiling pg_con.c Compiling res.c Compiling val.c pg_con.c: In function db_postgres_new_connection: pg_con.c:93: warning: implicit declaration of function PQconnectdbParams pg_con.c:93: warning: assignment makes pointer from integer without a cast Linking db_postgres.so make[1]: Leaving directory `/home/cscore/Downloads/opensips-2.4.7/modules/db_postgres' Is this a problem of my psql version(8.4.20), because PQconnectdbParams  is not available in that version.  -------------- next part -------------- An HTML attachment was scrubbed... URL: From tito at xsvoce.com Thu Apr 23 19:48:32 2020 From: tito at xsvoce.com (Tito Cumpen) Date: Thu, 23 Apr 2020 12:48:32 -0700 Subject: [OpenSIPS-Users] Tls using t_relay In-Reply-To: References: <39b60bcc-945b-f659-ae7f-37f82327c63c@opensips.org> <4e1e6e4c-b955-0f43-00a0-f5d776ea2e83@opensips.org> Message-ID: Hey Bogdan, Yes it seems like it continued down my routing script and tried to send it locally as well which is why I saw these Apr 21 23:12:46 sip1 /sbin/opensips[11376]: ERROR:tm:msg_send: send() to 127.0.0.1:443 for proto tls/3 failed when using the t_relay command. The remote end was sending provisionals but the failure above was causing the internal error. I've found a solution by using force_send_socket(tls:MY_IP:443); my tls port and then rewriting the host port to the destination. rewritehostport("SIPAS:port"); should I be using the exit command after a t_relay to avoid this? Thanks, Tito On Thu, Apr 23, 2020 at 1:39 AM Bogdan-Andrei Iancu wrote: > Hi Tito, > > Note that the tcp_connect_timeout is in milliseconds, so maybe 3 ms is too > short for getting back the SYN ACK. > > The logs are keep reporting the failed connect. You say the connect is ok, > the INVITE is sent forward to callee and there is also an 180 response? > ....and then you get the 500 reply ? > > Regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > > On 4/22/20 2:22 AM, Tito Cumpen wrote: > > Bogdan, > > I found out that there was a timer flag for tcp connections that may have > been causing an issue > tcp_connect_timeout=3 > Once I removed this line the tls connection was made fine but now I am > seeing opensips send an error message to the client > > SIP/2.0 500 Server error occurred (1/SL) > client---opensips---SIP AS > even though the SIP AS sent a 180 response > > > Here are the errors from the log > > Apr 21 23:12:46 sip1 /sbin/opensips[11376]: > ERROR:core:tcp_connect_blocking_timeout: poll error: flags 28 - 4 8 16 32 > Apr 21 23:12:46 sip1 /sbin/opensips[11376]: > ERROR:core:tcp_connect_blocking_timeout: failed to retrieve SO_ERROR > [server=127.0.0.1:443] (111) Connection refused > Apr 21 23:12:46 sip1 /sbin/opensips[11376]: > ERROR:proto_tls:tls_sync_connect: tcp_blocking_connect failed > Apr 21 23:12:46 sip1 /sbin/opensips[11376]: > ERROR:proto_tls:proto_tls_send: connect failed > Apr 21 23:12:46 sip1 /sbin/opensips[11376]: ERROR:tm:msg_send: send() to > 127.0.0.1:443 for proto tls/3 failed > Apr 21 23:12:46 sip1 /sbin/opensips[11376]: ERROR:tm:t_forward_nonack: > sending request failed > Apr 21 23:12:46 sip1 /sbin/opensips[11376]: > ERROR:core:tcp_connect_blocking_timeout: poll error: flags 28 - 4 8 16 32 > Apr 21 23:12:46 sip1 /sbin/opensips[11376]: > ERROR:core:tcp_connect_blocking_timeout: failed to retrieve SO_ERROR > [server=127.0.0.1:443] (111) Connection refused > Apr 21 23:12:46 sip1 /sbin/opensips[11376]: > ERROR:proto_tls:tls_sync_connect: tcp_blocking_connect failed > Apr 21 23:12:46 sip1 /sbin/opensips[11376]: > ERROR:proto_tls:proto_tls_send: connect failed > Apr 21 23:12:46 sip1 /sbin/opensips[11376]: ERROR:tm:msg_send: send() to > 127.0.0.1:443 for proto tls/3 failed > Apr 21 23:12:46 sip1 /sbin/opensips[11376]: ERROR:tm:t_forward_nonack: > sending request failed > Apr 21 23:12:46 sip1 /sbin/opensips[11376]: > ERROR:core:tcpconn_async_connect: poll error: flags 1c > Apr 21 23:12:46 sip1 /sbin/opensips[11376]: > ERROR:core:tcpconn_async_connect: failed to retrieve SO_ERROR [server= > 127.0.0.1:5060] (111) Connection refused > Apr 21 23:12:46 sip1 /sbin/opensips[11376]: ERROR:core:proto_tcp_send: > async TCP connect failed > Apr 21 23:12:46 sip1 /sbin/opensips[11376]: ERROR:tm:msg_send: send() to > 127.0.0.1:5060 for proto tcp/2 failed > Apr 21 23:12:46 sip1 /sbin/opensips[11376]: ERROR:tm:t_forward_nonack: > sending request failed > Apr 21 23:12:46 sip1 /sbin/opensips[11376]: > ERROR:core:tcpconn_async_connect: poll error: flags 1c > Apr 21 23:12:46 sip1 /sbin/opensips[11376]: > ERROR:core:tcpconn_async_connect: failed to retrieve SO_ERROR [server= > 127.0.0.1:5060] (111) Connection refused > Apr 21 23:12:46 sip1 /sbin/opensips[11376]: ERROR:core:proto_tcp_send: > async TCP connect failed > Apr 21 23:12:46 sip1 /sbin/opensips[11376]: ERROR:tm:msg_send: send() to > 127.0.0.1:5060 for proto tcp/2 failed > Apr 21 23:12:46 sip1 /sbin/opensips[11376]: ERROR:tm:t_forward_nonack: > sending request failed > Apr 21 23:12:46 sip1 /sbin/opensips[11376]: > ERROR:core:tcpconn_async_connect: poll error: flags 1c > Apr 21 23:12:46 sip1 /sbin/opensips[11376]: > ERROR:core:tcpconn_async_connect: failed to retrieve SO_ERROR [server= > 127.0.0.1:5223] (111) Connection refused > Apr 21 23:12:46 sip1 /sbin/opensips[11376]: ERROR:core:proto_tcp_send: > async TCP connect failed > Apr 21 23:12:46 sip1 /sbin/opensips[11376]: ERROR:tm:msg_send: send() to > 127.0.0.1:5223 for proto tcp/2 failed > Apr 21 23:12:46 sip1 /sbin/opensips[11376]: ERROR:tm:t_forward_nonack: > sending request failed > Apr 21 23:12:46 sip1 /sbin/opensips[11376]: > ERROR:core:tcpconn_async_connect: poll error: flags 1c > Apr 21 23:12:46 sip1 /sbin/opensips[11376]: > ERROR:core:tcpconn_async_connect: failed to retrieve SO_ERROR [server= > 127.0.0.1:5223] (111) Connection refused > Apr 21 23:12:46 sip1 /sbin/opensips[11376]: ERROR:core:proto_tcp_send: > async TCP connect failed > Apr 21 23:12:46 sip1 /sbin/opensips[11376]: ERROR:tm:msg_send: send() to > 127.0.0.1:5223 for proto tcp/2 failed > > On Tue, Apr 21, 2020 at 11:26 AM Tito Cumpen wrote: > >> Hey Bogdan, >> >> Here is the capture I took from using t_relay("tls:domain:port") >> >> As you can see the client side (opensips) does not proceed with allowing >> the socket to open. >> >> Thanks, >> Tito >> >> On Tue, Apr 21, 2020 at 2:18 AM Bogdan-Andrei Iancu >> wrote: >> >>> Hi Tito, >>> >>> Well, from OpenSIPS perceptive it is a "Connection refused", so the >>> destination is rejecting the connect. >>> >>> Regards, >>> >>> Bogdan-Andrei Iancu >>> >>> OpenSIPS Founder and Developer >>> https://www.opensips-solutions.com >>> >>> >>> > -------------- next part -------------- An HTML attachment was scrubbed... URL: From williamj at exetel.com.au Thu Apr 23 22:22:44 2020 From: williamj at exetel.com.au (William Jin) Date: Thu, 23 Apr 2020 22:22:44 +0000 Subject: [OpenSIPS-Users] opensips 3.0.2 100% CPU after enable tls? In-Reply-To: <18b56215-a2ed-6cd2-8730-de499ee521d5@opensips.org> References: , <18b56215-a2ed-6cd2-8730-de499ee521d5@opensips.org> Message-ID: I am using the debian apt repo, not from git. -- Regards, William Jin ________________________________ From: Bogdan-Andrei Iancu Sent: Thursday, 23 April 2020 6:49 PM To: OpenSIPS users mailling list ; William Jin Subject: Re: [OpenSIPS-Users] opensips 3.0.2 100% CPU after enable tls? Hi William, What GIT revision of OpenSIPS do you use? (this is exposed by the "opensips -V") Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com On 4/23/20 7:04 AM, William Jin wrote: Hi, Linux platform: Debian 9 (stretch) opensips -V version: opensips 3.0.2 (x86_64/linux) flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, HP_MALLOC, DBG_MALLOC, FAST_LOCK-ADAPTIVE_WAIT ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535 poll method support: poll, epoll, sigio_rt, select. main.c compiled on with gcc 6.3.0 related config: listen = tls:xxx.xxx.xxx.xxx:5061 anycast ####TLS loadmodule "tls_mgm.so" loadmodule "proto_tls.so" modparam("tls_mgm", "server_domain", "sip1") modparam("tls_mgm", "match_ip_address", "[sip1]xx.xx.xx.xx:5061") modparam("tls_mgm", "match_sip_domain", "[sip1]xxx.xxx.example.com") modparam("tls_mgm", "verify_cert", "[sip1]1") modparam("tls_mgm", "require_cert", "[sip1]0") modparam("tls_mgm", "tls_method", "[sip1]SSLv23") modparam("tls_mgm", "ciphers_list", "[sip1]AES256-GCM-SHA384,AES256-SHA256,AES256-SHA,CAMELLIA256-SHA,AES128-SHA,SEED-SHA,CAMELLIA128-SHA,RC4-SHA,DES-CBC3-SHA") modparam("tls_mgm", "certificate", "[sip1]/etc/opensips/tls/mycerts/selfsignedcert.pem") modparam("tls_mgm", "private_key", "[sip1]/etc/opensips/tls/mycerts/unsecuredkey.pem") opensips-cli -x trap {pid} result attached Can someone shed some light on it? Thanks. -- Regards, William Jin _______________________________________________ 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 alex.a at gtanetworkconsulting.com Thu Apr 23 22:42:41 2020 From: alex.a at gtanetworkconsulting.com (Alex A) Date: Thu, 23 Apr 2020 18:42:41 -0400 Subject: [OpenSIPS-Users] High Volume Accouting backend options In-Reply-To: <1204dcfb-db50-1b41-7b87-0050bce972db@opensips.org> References: <171a268e042.fa35ce09171614.5445550439030139138@gtanetworkconsulting.com> <171a7422dea.1247b83848131.3910812018483636749@gtanetworkconsulting.com> <1204dcfb-db50-1b41-7b87-0050bce972db@opensips.org> Message-ID: <2e84fdf8-db20-412a-8f4e-a0c3ff426c19@gtanetworkconsulting.com> Thank you I try it out via rabbitMQ event subscription On Apr 23, 2020, 10:53 AM, at 10:53 AM, Bogdan-Andrei Iancu wrote: >Hi Alex, > >Typical approach in this case is to do the accounting via a very fast >backend (like db_flatstore, into a text file) and import the files >off-sync into db (like every 5 mins). > >Regards, > >Bogdan-Andrei Iancu > >OpenSIPS Founder and Developer > https://www.opensips-solutions.com > >On 4/23/20 4:37 PM, Alex A wrote: >> Hi, >> >> >> We are looking to deploy accounting/homer integration on Opensips >> 3.0.2. >> As the first step deployed acc module with pgsql backend. >> >> The config seem to be pretty straight-forward - see attached. >> >> It appears that as soon as volume hits about 30-35k in_use >> transactions - the server stops replying to new requests (or give >> 408 Timeout) and syslog gets filled with: >> >> Apr 22 10:19:38 opensip1 opensips: Apr 22 10:19:38 [19258] >> CRITICAL:tm:set_timer: set_timer for 1 list called on a >"detached" >> timer -- ignoring: 0x7fb63b993cf8 >> Apr 22 10:19:40 opensip1 opensips: Apr 22 10:19:40 [19255] >> CRITICAL:tm:set_timer: set_timer for 1 list called on a >"detached" >> timer -- ignoring: 0x7fb638cf9a40 >> Apr 22 10:19:40 opensip1 opensips: Apr 22 10:19:40 [19260] >> CRITICAL:tm:set_timer: set_timer for 1 list called on a >"detached" >> timer -- ignoring: 0x7fb63f3b23c8 >> Apr 22 10:19:49 opensip1 opensips: Apr 22 10:19:49 [19258] >> CRITICAL:tm:set_timer: set_timer for 1 list called on a >"detached" >> timer -- ignoring: 0x7fb63eb93a80 >> Apr 22 10:20:01 opensip1 opensips: Apr 22 10:20:01 [19267] >> CRITICAL:tm:set_timer: set_timer for 1 list called on a >"detached" >> timer -- ignoring: 0x7fb5de690700 >> >> >> Although, the remote Postgres service is on SSD with relatively >> small network latency, it appears to be the bottleneck. >> The initial assumption was the Opensips acc uses no-blocking SQL >> calls (since cdrs are not real-time). >> >> >> Another observation: >> opensips only opens 20 SQL connections to postgres via tcp 5432.  >> I have tried playing with db_max_async_connections, however to no >> avail. >> >> >> Any suggestions to troubleshoot ? or any alternatives for >> accounting in high volume applications would be greatly >appreciated. >> >> >> 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 anexiole at gmail.com Fri Apr 24 05:05:06 2020 From: anexiole at gmail.com (Gordon Yeong) Date: Fri, 24 Apr 2020 15:05:06 +1000 Subject: [OpenSIPS-Users] Testing ENUM queries In-Reply-To: References: Message-ID: The reason why i am doing this is because I would like to because to query my ENUM server using a standalone script which reuses the OpenSips Enum module. I read a little more and figured it's prolly not worth it as the ENUM module exists in Opensips so that we can just do ENUM queries within OpenSips. Any direct queries to any ENUM server I guess could be done directly instead. Sorry, guys, My mistake. > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From spanda at 3clogic.com Fri Apr 24 05:55:45 2020 From: spanda at 3clogic.com (Sasmita Panda) Date: Fri, 24 Apr 2020 11:25:45 +0530 Subject: [OpenSIPS-Users] need some help to parse the data I got after doing lookup in location table . Message-ID: Hi , I have a 2 user , 1 is register from browser and another is register from a cisco phone . The protocol through which both get register is different . browser get register through WSS and cisco phone get register through UDP . Now while an INVITE comes for any of the user after doing lookup , while giving call to browser I wanted to involve Rtpengine in for media encryption but while giving call to cisco phone no need of Rtpengine involvement . My question is , how I will differentiate both user ? My knowledge says I have to parse the contact I got from lookup , parse that contact and find which user register through which protocol and accordingly differentiate them . If so , then how I will parse the contact ? Please do help me . Thank you in advance . *Thanks & Regards* *Sasmita Panda* *Senior Network Testing and Software Engineer* *3CLogic , ph:07827611765* -------------- next part -------------- An HTML attachment was scrubbed... URL: From tito at xsvoce.com Fri Apr 24 07:08:48 2020 From: tito at xsvoce.com (Tito Cumpen) Date: Fri, 24 Apr 2020 00:08:48 -0700 Subject: [OpenSIPS-Users] need some help to parse the data I got after doing lookup in location table . In-Reply-To: References: Message-ID: Sasmita, Look at the following tutorial. https://www.opensips.org/Documentation/Tutorials-WebSocket-2-2 which considers adding a branch flag to your destinations with websocket so they are sent an SDP with DTLS and AVPF. I am not sure if this works while using parallel forking all that well at this point. It didn't when I tried it a while back and I was instructed to use serial forking by setting different priorities on a per destination type. On Thu, Apr 23, 2020 at 10:58 PM Sasmita Panda wrote: > Hi , > > > I have a 2 user , 1 is register from browser and another is register from > a cisco phone . The protocol through which both get register is different . > browser get register through WSS and cisco phone get register through UDP > . > > Now while an INVITE comes for any of the user after doing lookup , while > giving call to browser I wanted to involve Rtpengine in for media > encryption but while giving call to cisco phone no need of Rtpengine > involvement . > > My question is , how I will differentiate both user ? My knowledge says I > have to parse the contact I got from lookup , parse that contact and find > which user register through which protocol and accordingly differentiate > them . > If so , then how I will parse the contact ? Please do help me . > > Thank you in advance . > > *Thanks & Regards* > *Sasmita Panda* > *Senior Network Testing and Software Engineer* > *3CLogic , ph:07827611765* > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -------------- next part -------------- An HTML attachment was scrubbed... URL: From callum.guy at x-on.co.uk Fri Apr 24 07:13:06 2020 From: callum.guy at x-on.co.uk (Callum Guy) Date: Fri, 24 Apr 2020 08:13:06 +0100 Subject: [OpenSIPS-Users] need some help to parse the data I got after doing lookup in location table . In-Reply-To: References: Message-ID: Hi Sasmita, I would advise that you capture this information in a branch flag during registration, these will be stored in location and retrieved when performing a matching lookup(). So: if ($pr == "ws" || $pr == "wss") { setbflag(SRC_WS); } save("location_table") When you do the lookup this bflag will be loaded and can be utilised to route as required. Good luck On Fri, 24 Apr 2020 at 06:57, Sasmita Panda wrote: > Hi , > > > I have a 2 user , 1 is register from browser and another is register from > a cisco phone . The protocol through which both get register is different . > browser get register through WSS and cisco phone get register through UDP > . > > Now while an INVITE comes for any of the user after doing lookup , while > giving call to browser I wanted to involve Rtpengine in for media > encryption but while giving call to cisco phone no need of Rtpengine > involvement . > > My question is , how I will differentiate both user ? My knowledge says I > have to parse the contact I got from lookup , parse that contact and find > which user register through which protocol and accordingly differentiate > them . > If so , then how I will parse the contact ? Please do help me . > > Thank you in advance . > > *Thanks & Regards* > *Sasmita Panda* > *Senior Network Testing and Software Engineer* > *3CLogic , ph:07827611765* > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -- *0333 332 0000  |  x-on.co.uk   |   **      **  |  Coronavirus * THE ITSPA AWARDS 2020 AND Best ITSP - Mid Market, Best Software and Best Vertical Solution are trade marks of the Internet Telephony Services Providers' Association, used under licence. X-on is a trading name of Storacall Technology Ltd a limited company registered in England and Wales. Registered Office : Avaland House, 110 London Road, Apsley, Hemel Hempstead, Herts, HP3 9SD. Company Registration No. 2578478. The information in this e-mail is confidential and for use by the addressee(s) only. If you are not the intended recipient, please notify X-on immediately on +44(0)333 332 0000 and delete the message from your computer. If you are not a named addressee you must not use, disclose, disseminate, distribute, copy, print or reply to this email. Views or opinions expressed by an individual within this email may not necessarily reflect the views of X-on or its associated companies. Although X-on routinely screens for viruses, addressees should scan this email and any attachments for viruses. X-on makes no representation or warranty as to the absence of viruses in this email or any attachments. -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Fri Apr 24 08:14:56 2020 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Fri, 24 Apr 2020 11:14:56 +0300 Subject: [OpenSIPS-Users] opensips 3.0.2 100% CPU after enable tls? In-Reply-To: References: <18b56215-a2ed-6cd2-8730-de499ee521d5@opensips.org> Message-ID: <4bb55ace-480e-4e01-6953-ee296372d426@opensips.org> And it is the release or nightly build ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com On 4/24/20 1:22 AM, William Jin wrote: > I am using the debian apt repo, not from git. > > > -- > Regards, > William Jin > ------------------------------------------------------------------------ > *From:* Bogdan-Andrei Iancu > *Sent:* Thursday, 23 April 2020 6:49 PM > *To:* OpenSIPS users mailling list ; William > Jin > *Subject:* Re: [OpenSIPS-Users] opensips 3.0.2 100% CPU after enable tls? > Hi William, > > What GIT revision of OpenSIPS do you use? (this is exposed by the > "opensips -V") > > Regards, > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > > On 4/23/20 7:04 AM, William Jin wrote: >> Hi, >> >> Linux platform: Debian 9 (stretch) >> >> opensips -V >> version: opensips 3.0.2 (x86_64/linux) >> flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, >> Q_MALLOC, F_MALLOC, HP_MALLOC, DBG_MALLOC, FAST_LOCK-ADAPTIVE_WAIT >> ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, >> MAX_URI_SIZE 1024, BUF_SIZE 65535 >> poll method support: poll, epoll, sigio_rt, select. >> main.c compiled on  with gcc 6.3.0 >> >> related config: >> >> listen = tls:xxx.xxx.xxx.xxx:5061 anycast >> >> >> ####TLS >> loadmodule "tls_mgm.so" >> loadmodule "proto_tls.so" >> >> modparam("tls_mgm", "server_domain", "sip1") >> modparam("tls_mgm", "match_ip_address", "[sip1]xx.xx.xx.xx:5061") >> modparam("tls_mgm", "match_sip_domain", "[sip1]xxx.xxx.example.com") >> >> modparam("tls_mgm", "verify_cert", "[sip1]1") >> modparam("tls_mgm", "require_cert", "[sip1]0") >> modparam("tls_mgm", "tls_method", "[sip1]SSLv23") >> modparam("tls_mgm", "ciphers_list", >> "[sip1]AES256-GCM-SHA384,AES256-SHA256,AES256-SHA,CAMELLIA256-SHA,AES128-SHA,SEED-SHA,CAMELLIA128-SHA,RC4-SHA,DES-CBC3-SHA") >> >> >> modparam("tls_mgm", "certificate", >> "[sip1]/etc/opensips/tls/mycerts/selfsignedcert.pem") >> modparam("tls_mgm", "private_key", >> "[sip1]/etc/opensips/tls/mycerts/unsecuredkey.pem") >> >> >> opensips-cli -x trap {pid} result attached >> >> Can someone shed some light on it? Thanks. >> >> >> -- >> Regards, >> William Jin >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Fri Apr 24 08:16:05 2020 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Fri, 24 Apr 2020 11:16:05 +0300 Subject: [OpenSIPS-Users] High Volume Accouting backend options In-Reply-To: <2e84fdf8-db20-412a-8f4e-a0c3ff426c19@gtanetworkconsulting.com> References: <171a268e042.fa35ce09171614.5445550439030139138@gtanetworkconsulting.com> <171a7422dea.1247b83848131.3910812018483636749@gtanetworkconsulting.com> <1204dcfb-db50-1b41-7b87-0050bce972db@opensips.org> <2e84fdf8-db20-412a-8f4e-a0c3ff426c19@gtanetworkconsulting.com> Message-ID: <4b82db4c-f246-5ab2-41e0-4987026f8db3@opensips.org> Just note that RMQ may be lighter than SQL DBs, but much slower than db_flatstore. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com On 4/24/20 1:42 AM, Alex A wrote: > Thank you > > I try it out via rabbitMQ event subscription > > On Apr 23, 2020, at 10:53 AM, Bogdan-Andrei Iancu > wrote: > > Hi Alex, > > Typical approach in this case is to do the accounting via a very > fast backend (like db_flatstore, into a text file) and import the > files off-sync into db (like every 5 mins). > > Regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > > On 4/23/20 4:37 PM, Alex A wrote: >> Hi, >> >> >> We are looking to deploy accounting/homer integration on >> Opensips 3.0.2. >> As the first step deployed acc module with pgsql backend. >> >> The config seem to be pretty straight-forward - see attached. >> >> It appears that as soon as volume hits about 30-35k in_use >> transactions - the server stops replying to new requests (or >> give 408 Timeout) and syslog gets filled with: >> >> Apr 22 10:19:38 opensip1 opensips: Apr 22 10:19:38 [19258] >> CRITICAL:tm:set_timer: set_timer for 1 list called on a >> "detached" timer -- ignoring: 0x7fb63b993cf8 >> Apr 22 10:19:40 opensip1 opensips: Apr 22 10:19:40 [19255] >> CRITICAL:tm:set_timer: set_timer for 1 list called on a >> "detached" timer -- ignoring: 0x7fb638cf9a40 >> Apr 22 10:19:40 opensip1 opensips: Apr 22 10:19:40 [19260] >> CRITICAL:tm:set_timer: set_timer for 1 list called on a >> "detached" timer -- ignoring: 0x7fb63f3b23c8 >> Apr 22 10:19:49 opensip1 opensips: Apr 22 10:19:49 [19258] >> CRITICAL:tm:set_timer: set_timer for 1 list called on a >> "detached" timer -- ignoring: 0x7fb63eb93a80 >> Apr 22 10:20:01 opensip1 opensips: Apr 22 10:20:01 [19267] >> CRITICAL:tm:set_timer: set_timer for 1 list called on a >> "detached" timer -- ignoring: 0x7fb5de690700 >> >> >> Although, the remote Postgres service is on SSD with >> relatively small network latency, it appears to be the >> bottleneck. >> The initial assumption was the Opensips acc uses no-blocking >> SQL calls (since cdrs are not real-time). >> >> >> Another observation: >> opensips only opens 20 SQL connections to postgres via tcp >> 5432.  I have tried playing with db_max_async_connections, >> however to no avail. >> >> >> Any suggestions to troubleshoot ? or any alternatives for >> accounting in high volume applications would be greatly >> appreciated. >> >> >> 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 Fri Apr 24 08:21:22 2020 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Fri, 24 Apr 2020 11:21:22 +0300 Subject: [OpenSIPS-Users] OpenSIPS installation with PostgresSQL In-Reply-To: <1876180299.356916.1587669506692@mail.yahoo.com> References: <129541852.2099756.1587328682918.ref@mail.yahoo.com> <129541852.2099756.1587328682918@mail.yahoo.com> <1985829642.2102360.1587329441034@mail.yahoo.com> <7812bb25-eef4-0b34-04f5-a4f5c5ba589f@opensips.org> <1950080153.2366807.1587399844986@mail.yahoo.com> <1582965925.176731.1587476879885@mail.yahoo.com> <1193421104.181467.1587478895108@mail.yahoo.com> <1971406255.325015.1587497750288@mail.yahoo.com> <15e3ac1f-2e01-a4fb-147a-5ac32411c98d@opensips.org> <1119346896.578368.1587559461192@mail.yahoo.com> <2d3c288a-4f71-d78b-6d83-84e2731fc2e5@opensips.org> <1876180299.356916.1587669506692@mail.yahoo.com> Message-ID: <057323a0-7346-dc4d-2180-4fb7137dd574@opensips.org> Hi Nayani, What was the fix you did ? the path to the postgres lib? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com On 4/23/20 10:18 PM, Nayani Nikeshala wrote: > Hi Bogdan, > > I could fix the problem by modifying Makefile of postgres module. > Thanks for the support. > > > On Thursday, April 23, 2020, 02:21:00 PM GMT+5:30, Bogdan-Andrei Iancu > wrote: > > > Hi, > > I have an 9.5.19 (on Ubuntu) and works ok. > > Regards, > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > > On 4/22/20 3:44 PM, Nayani Nikeshala wrote: > Hi Bogdan, > > I found an old email, where he has faced a similar kind of an issue > with PSQL 9.2 also. I believe, it will not get resolved with PSQL > version. > > https://opensips.org/pipermail/devel/2015-June/017666.html > > According to the email, he has resolved it by modifying makefile, but > I'm not sure, how he did that. > > > On Wednesday, April 22, 2020, 01:54:09 PM GMT+5:30, Bogdan-Andrei > Iancu wrote: > > > It may sound weird, but maybe you into the other extrema, with some > too new version :D... > > Is it possible to try some 9.3 or so ? > > Regards, > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > > On 4/21/20 10:35 PM, Nayani Nikeshala wrote: > Hi Bogdan, > > I removed psql 8.4 and installed psql 9.6. > > [root at SIPserver opensips-2.4.7]# yum list installed | grep postgre > postgresql96.i686 9.6.17-1PGDG.rhel6 > postgresql96-devel.i686 9.6.17-1PGDG.rhel6 > postgresql96-libs.i686  9.6.17-1PGDG.rhel6 > postgresql96-server.i686 9.6.17-1PGDG.rhel6 > [root at SIPserver opensips-2.4.7]# > > > When I compile, I get below error. (Below is a part of the error) > > make[1]: Entering directory > `/home/cscore/Downloads/opensips-2.4.7/modules/db_postgres' > Compiling dbase.c > Compiling pg_con.c > Compiling res.c > Compiling val.c > In file included from pg_con.c:23: > pg_con.h:39:22: error: libpq-fe.h: No such file or directory > In file included from pg_con.c:23: > pg_con.h:54: error: expected specifier-qualifier-list before ‘PGconn’ > pg_con.c: In function ‘db_postgres_new_connection’: > pg_con.c:93: error: ‘struct pg_con’ has no member named ‘con’ > pg_con.c:93: warning: implicit declaration of function ‘PQconnectdbParams’ > > libpq-fe.h & pg_config are in the below paths. > > [root at SIPserver opensips-2.4.7]# ls -lrt /usr/pgsql-9.6/include/libpq-fe.h > -rw-r--r--. 1 root root 21900 Feb 12 02:52 > /usr/pgsql-9.6/include/libpq-fe.h > [root at SIPserver opensips-2.4.7]# > > [root at SIPserver opensips-2.4.7]# ls -lrt /usr/pgsql-9.6/bin/pg_config > -rwxr-xr-x. 1 root root 24544 Feb 12 02:52 /usr/pgsql-9.6/bin/pg_config > > Could you assist me with above error. > > > > On Tuesday, April 21, 2020, 08:57:13 PM GMT+5:30, Bogdan-Andrei Iancu > wrote: > > > Hi Nayani, > > yeah, that is a rather old and unsupported version of postgres. I > think you should try to use a newer version it (maybe a newer version > of your distro). > > Regards, > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > > On 4/21/20 5:21 PM, Nayani Nikeshala wrote: > Hi Bogdan, > > I have attached the output log for the above commands. > > "make modules module=modules/db_postgres" gave me an error as in the > attached file, so I used "make include_modules=db_postgres modules" > instead. I could see below in the output. > > make[1]: Entering directory > `/home/cscore/Downloads/opensips-2.4.7/modules/db_postgres' > make[1]: Leaving directory > `/home/cscore/Downloads/opensips-2.4.7/modules/db_postgres' > make[1]: Entering directory > `/home/cscore/Downloads/opensips-2.4.7/modules/db_postgres' > Compiling dbase.c > Compiling db_postgres.c > Compiling pg_con.c > Compiling res.c > Compiling val.c > pg_con.c: In function db_postgres_new_connection: > pg_con.c:93: warning: implicit declaration of function PQconnectdbParams > pg_con.c:93: warning: assignment makes pointer from integer without a cast > Linking db_postgres.so > make[1]: Leaving directory > `/home/cscore/Downloads/opensips-2.4.7/modules/db_postgres' > > Is this a problem of my psql version(8.4.20), because > PQconnectdbParams  is not available in that version. > > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From volga629 at networklab.ca Fri Apr 24 10:22:49 2020 From: volga629 at networklab.ca (volga629) Date: Fri, 24 Apr 2020 07:22:49 -0300 Subject: [OpenSIPS-Users] string comparison In-Reply-To: <9362669e-d19f-a1f3-f90c-f53dd7668fb4@opensips.org> References: <5e6e5656-a785-5b86-a6e9-c591f4442249@skillsearch.ca> <9362669e-d19f-a1f3-f90c-f53dd7668fb4@opensips.org> Message-ID: <75852689-9b39-65f3-d071-2bdb1523dccd@skillsearch.ca> An HTML attachment was scrubbed... URL: From liviu at opensips.org Fri Apr 24 10:57:11 2020 From: liviu at opensips.org (Liviu Chircu) Date: Fri, 24 Apr 2020 13:57:11 +0300 Subject: [OpenSIPS-Users] string comparison In-Reply-To: <75852689-9b39-65f3-d071-2bdb1523dccd@skillsearch.ca> References: <5e6e5656-a785-5b86-a6e9-c591f4442249@skillsearch.ca> <9362669e-d19f-a1f3-f90c-f53dd7668fb4@opensips.org> <75852689-9b39-65f3-d071-2bdb1523dccd@skillsearch.ca> Message-ID: <7d08e2ce-5c2f-f4f1-8692-09611bb15082@opensips.org> On 24.04.2020 13:22, volga629 wrote: > if($var(tls-frompbx)==$var(usr_check_tls)) { >                 $var(transport) = "tls"; >                 if(!codec_exists("opus")) { >                     $var(codec_flag_frompbx_tls) = "transcode-opus"; >                 } >                 rtpengine_offer("replace-origin > replace-session-connection $var(codec_flag_frompbx_tls) RTP/SAVP > ICE=remove $var(dir)"); >             } Please add an "else" to this "if" block exactly as follows, run it and provide the output:             if($var(tls-frompbx)==$var(usr_check_tls)) {                 $var(transport) = "tls";                 if(!codec_exists("opus")) {                     $var(codec_flag_frompbx_tls) = "transcode-opus";                 }                 rtpengine_offer("replace-origin replace-session-connection $var(codec_flag_frompbx_tls) RTP/SAVP ICE=remove $var(dir)");             } else {                 xlog("L_INFO", "DBG: TLS strings differ: '$var(tls-frompbx)'/'$var(usr_check_tls)', '$(var(tls-frompbx){s.len})'/'$(var(usr_check_tls){s.len})', '$(var(tls-frompbx){s.md5})'/'$(var(usr_check_tls){s.md5})'\n");             } Also, what is the output of "opensips -V"? Best regards, -- Liviu Chircu www.twitter.com/liviuchircu | www.opensips-solutions.com OpenSIPS Summit, Amsterdam, Fall 2020 www.opensips.org/events -------------- next part -------------- An HTML attachment was scrubbed... URL: From callum.guy at x-on.co.uk Fri Apr 24 15:31:48 2020 From: callum.guy at x-on.co.uk (Callum Guy) Date: Fri, 24 Apr 2020 16:31:48 +0100 Subject: [OpenSIPS-Users] disable_503_translation Message-ID: Hi All, I've been hunting a minor memory leak in my config and wanted to check in with the devs in case it is related to ues of the parameter: disable_503_translation=yes Here is the implementation link to save you a few seconds: https://github.com/OpenSIPS/opensips/blob/7dd1151341b8229cd30e335b246e56938551f6bd/msg_translator.c#L2425 Is there any chance that this is failing to free the tiny amount it's allocating on each use? All the best, Callum -- *0333 332 0000  |  x-on.co.uk   |   **      **  |  Coronavirus * THE ITSPA AWARDS 2020 AND Best ITSP - Mid Market, Best Software and Best Vertical Solution are trade marks of the Internet Telephony Services Providers' Association, used under licence. X-on is a trading name of Storacall Technology Ltd a limited company registered in England and Wales. Registered Office : Avaland House, 110 London Road, Apsley, Hemel Hempstead, Herts, HP3 9SD. Company Registration No. 2578478. The information in this e-mail is confidential and for use by the addressee(s) only. If you are not the intended recipient, please notify X-on immediately on +44(0)333 332 0000 and delete the message from your computer. If you are not a named addressee you must not use, disclose, disseminate, distribute, copy, print or reply to this email. Views or opinions expressed by an individual within this email may not necessarily reflect the views of X-on or its associated companies. Although X-on routinely screens for viruses, addressees should scan this email and any attachments for viruses. X-on makes no representation or warranty as to the absence of viruses in this email or any attachments. -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Fri Apr 24 15:50:06 2020 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Fri, 24 Apr 2020 18:50:06 +0300 Subject: [OpenSIPS-Users] disable_503_translation In-Reply-To: References: Message-ID: <55435248-82fe-e1b7-782a-587ecad53b51@opensips.org> Hi Callum, I 99.9999999999999999999% that that setting cannot generate a leak :) Best regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com On 4/24/20 6:31 PM, Callum Guy wrote: > Hi All, > > I've been hunting a minor memory leak in my config and wanted to check > in with the devs in case it is related to ues of the parameter: > > disable_503_translation=yes > > Here is the implementation link to save you a few seconds: > > https://github.com/OpenSIPS/opensips/blob/7dd1151341b8229cd30e335b246e56938551f6bd/msg_translator.c#L2425 > > Is there any chance that this is failing to free the tiny amount it's > allocating on each use? > > All the best, > > Callum > > > > *^0333 332 0000  | x-on.co.uk | > _**_^ > **^  | > Coronavirus > * > > THE ITSPA AWARDS 2020 AND Best > ITSP - Mid Market, Best Software and Best Vertical Solution are trade > marks of the Internet Telephony Services Providers' Association, used > under licence. > > X-on is a trading name of Storacall Technology Ltd a limited company > registered in England and Wales. > Registered Office : Avaland House, 110 London Road, Apsley, Hemel > Hempstead, Herts, HP3 9SD. Company Registration No. 2578478. > The information in this e-mail is confidential and for use by the > addressee(s) only. If you are not the intended recipient, please > notify X-on immediately on +44(0)333 332 0000 and delete the > message from your computer. If you are not a named addressee you must > not use, disclose, disseminate, distribute, copy, print or reply to > this email. Views or opinions expressed by an individual > within this email may not necessarily reflect the views of X-on or its > associated companies. Although X-on routinely screens for viruses, > addressees should scan this email and any attachments > for viruses. X-on makes no representation or warranty as to the > absence of viruses in this email or any attachments. > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From nnikeshala at yahoo.com Fri Apr 24 16:53:09 2020 From: nnikeshala at yahoo.com (Nayani Nikeshala) Date: Fri, 24 Apr 2020 16:53:09 +0000 (UTC) Subject: [OpenSIPS-Users] OpenSIPS installation with PostgresSQL In-Reply-To: <057323a0-7346-dc4d-2180-4fb7137dd574@opensips.org> References: <129541852.2099756.1587328682918.ref@mail.yahoo.com> <129541852.2099756.1587328682918@mail.yahoo.com> <1985829642.2102360.1587329441034@mail.yahoo.com> <7812bb25-eef4-0b34-04f5-a4f5c5ba589f@opensips.org> <1950080153.2366807.1587399844986@mail.yahoo.com> <1582965925.176731.1587476879885@mail.yahoo.com> <1193421104.181467.1587478895108@mail.yahoo.com> <1971406255.325015.1587497750288@mail.yahoo.com> <15e3ac1f-2e01-a4fb-147a-5ac32411c98d@opensips.org> <1119346896.578368.1587559461192@mail.yahoo.com> <2d3c288a-4f71-d78b-6d83-84e2731fc2e5@opensips.org> <1876180299.356916.1587669506692@mail.yahoo.com> <057323a0-7346-dc4d-2180-4fb7137dd574@opensips.org> Message-ID: <1567006032.186555.1587747189795@mail.yahoo.com> Hi Bogdan, I added the path of libpq-fe.h to the Makefile.  [root at SIPserver include]# ls -lrt /usr/pgsql-9.6/include/libpq-fe.h-rw-r--r--. 1 root root 21900 Feb 12 02:52 /usr/pgsql-9.6/include/libpq-fe.h On Friday, April 24, 2020, 01:51:35 PM GMT+5:30, Bogdan-Andrei Iancu wrote: Hi Nayani, What was the fix you did ? the path to the postgres lib? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com On 4/23/20 10:18 PM, Nayani Nikeshala wrote: Hi Bogdan, I could fix the problem by modifying Makefile of postgres module. Thanks for the support. On Thursday, April 23, 2020, 02:21:00 PM GMT+5:30, Bogdan-Andrei Iancu wrote: Hi, I have an 9.5.19 (on Ubuntu) and works ok. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com On 4/22/20 3:44 PM, Nayani Nikeshala wrote: Hi Bogdan, I found an old email, where he has faced a similar kind of an issue with PSQL 9.2 also. I believe, it will not get resolved with PSQL version.  https://opensips.org/pipermail/devel/2015-June/017666.html According to the email, he has resolved it by modifying makefile, but I'm not sure, how he did that. On Wednesday, April 22, 2020, 01:54:09 PM GMT+5:30, Bogdan-Andrei Iancu wrote: It may sound weird, but maybe you into the other extrema, with some too new version :D... Is it possible to try some 9.3 or so ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com On 4/21/20 10:35 PM, Nayani Nikeshala wrote: Hi Bogdan, I removed psql 8.4 and installed psql 9.6.  [root at SIPserver opensips-2.4.7]# yum list installed | grep postgre postgresql96.i686        9.6.17-1PGDG.rhel6 postgresql96-devel.i686  9.6.17-1PGDG.rhel6 postgresql96-libs.i686   9.6.17-1PGDG.rhel6 postgresql96-server.i686 9.6.17-1PGDG.rhel6 [root at SIPserver opensips-2.4.7]# When I compile, I get below error. (Below is a part of the error) make[1]: Entering directory `/home/cscore/Downloads/opensips-2.4.7/modules/db_postgres' Compiling dbase.c Compiling pg_con.c Compiling res.c Compiling val.c In file included from pg_con.c:23: pg_con.h:39:22: error: libpq-fe.h: No such file or directory In file included from pg_con.c:23: pg_con.h:54: error: expected specifier-qualifier-list before ‘PGconn’ pg_con.c: In function ‘db_postgres_new_connection’: pg_con.c:93: error: ‘struct pg_con’ has no member named ‘con’ pg_con.c:93: warning: implicit declaration of function ‘PQconnectdbParams’ libpq-fe.h & pg_config are in the below paths.   [root at SIPserver opensips-2.4.7]# ls -lrt /usr/pgsql-9.6/include/libpq-fe.h -rw-r--r--. 1 root root 21900 Feb 12 02:52 /usr/pgsql-9.6/include/libpq-fe.h [root at SIPserver opensips-2.4.7]# [root at SIPserver opensips-2.4.7]# ls -lrt /usr/pgsql-9.6/bin/pg_config -rwxr-xr-x. 1 root root 24544 Feb 12 02:52 /usr/pgsql-9.6/bin/pg_config Could you assist me with above error.  On Tuesday, April 21, 2020, 08:57:13 PM GMT+5:30, Bogdan-Andrei Iancu wrote: Hi Nayani, yeah, that is a rather old and unsupported version of postgres. I think you should try to use a newer version it (maybe a newer version of your distro). Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com On 4/21/20 5:21 PM, Nayani Nikeshala wrote: Hi Bogdan, I have attached the output log for the above commands.  "make modules module=modules/db_postgres" gave me an error as in the attached file, so I used "makeinclude_modules=db_postgres modules" instead. I could see below in the output.  make[1]: Entering directory `/home/cscore/Downloads/opensips-2.4.7/modules/db_postgres' make[1]: Leaving directory `/home/cscore/Downloads/opensips-2.4.7/modules/db_postgres' make[1]: Entering directory `/home/cscore/Downloads/opensips-2.4.7/modules/db_postgres' Compiling dbase.c Compiling db_postgres.c Compiling pg_con.c Compiling res.c Compiling val.c pg_con.c: In function db_postgres_new_connection: pg_con.c:93: warning: implicit declaration of function PQconnectdbParams pg_con.c:93: warning: assignment makes pointer from integer without a cast Linking db_postgres.so make[1]: Leaving directory `/home/cscore/Downloads/opensips-2.4.7/modules/db_postgres' Is this a problem of my psql version(8.4.20), because PQconnectdbParams  is not available in that version.  -------------- next part -------------- An HTML attachment was scrubbed... URL: From sobomax at sippysoft.com Fri Apr 24 21:02:57 2020 From: sobomax at sippysoft.com (Maxim Sobolev) Date: Fri, 24 Apr 2020 14:02:57 -0700 Subject: [OpenSIPS-Users] Join us for SIP Chronicles Live #1a, featuring Bogdan and Razvan!!! In-Reply-To: References: Message-ID: Folks, We were frankly positively overwhelmed with the amount of interest for such an event from the OpenSIPS community!!! So here we go, instead of pushing it back another two weeks we decided to deliver two shows tomorrow instead of just one! Back-To-back as we say. With about 1 hour break in between to stretch your legs and refill the popcorn bowl. In the second part of the first episode Bogdan and Razvan present their latest work on CallCenter functionality and demonstrate how they can make good use of SaraPhone to make it even cooler. 7pm UTC, it would be a good show to put you into sleep mode eventually. https://youtu.be/cea0B-oad3w See you tomorrow! -Max -------------- next part -------------- An HTML attachment was scrubbed... URL: From williamj at exetel.com.au Fri Apr 24 21:12:33 2020 From: williamj at exetel.com.au (William Jin) Date: Fri, 24 Apr 2020 21:12:33 +0000 Subject: [OpenSIPS-Users] opensips 3.0.2 100% CPU after enable tls? In-Reply-To: <4bb55ace-480e-4e01-6953-ee296372d426@opensips.org> References: <18b56215-a2ed-6cd2-8730-de499ee521d5@opensips.org> , <4bb55ace-480e-4e01-6953-ee296372d426@opensips.org> Message-ID: It's the release. -- Regards, William Jin ________________________________ From: Bogdan-Andrei Iancu Sent: Friday, 24 April 2020 6:14 PM To: William Jin ; OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] opensips 3.0.2 100% CPU after enable tls? And it is the release or nightly build ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com On 4/24/20 1:22 AM, William Jin wrote: I am using the debian apt repo, not from git. -- Regards, William Jin ________________________________ From: Bogdan-Andrei Iancu Sent: Thursday, 23 April 2020 6:49 PM To: OpenSIPS users mailling list ; William Jin Subject: Re: [OpenSIPS-Users] opensips 3.0.2 100% CPU after enable tls? Hi William, What GIT revision of OpenSIPS do you use? (this is exposed by the "opensips -V") Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com On 4/23/20 7:04 AM, William Jin wrote: Hi, Linux platform: Debian 9 (stretch) opensips -V version: opensips 3.0.2 (x86_64/linux) flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, HP_MALLOC, DBG_MALLOC, FAST_LOCK-ADAPTIVE_WAIT ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535 poll method support: poll, epoll, sigio_rt, select. main.c compiled on with gcc 6.3.0 related config: listen = tls:xxx.xxx.xxx.xxx:5061 anycast ####TLS loadmodule "tls_mgm.so" loadmodule "proto_tls.so" modparam("tls_mgm", "server_domain", "sip1") modparam("tls_mgm", "match_ip_address", "[sip1]xx.xx.xx.xx:5061") modparam("tls_mgm", "match_sip_domain", "[sip1]xxx.xxx.example.com") modparam("tls_mgm", "verify_cert", "[sip1]1") modparam("tls_mgm", "require_cert", "[sip1]0") modparam("tls_mgm", "tls_method", "[sip1]SSLv23") modparam("tls_mgm", "ciphers_list", "[sip1]AES256-GCM-SHA384,AES256-SHA256,AES256-SHA,CAMELLIA256-SHA,AES128-SHA,SEED-SHA,CAMELLIA128-SHA,RC4-SHA,DES-CBC3-SHA") modparam("tls_mgm", "certificate", "[sip1]/etc/opensips/tls/mycerts/selfsignedcert.pem") modparam("tls_mgm", "private_key", "[sip1]/etc/opensips/tls/mycerts/unsecuredkey.pem") opensips-cli -x trap {pid} result attached Can someone shed some light on it? Thanks. -- Regards, William Jin _______________________________________________ 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 Sat Apr 25 04:36:06 2020 From: osas at voipembedded.com (Ovidiu Sas) Date: Sat, 25 Apr 2020 00:36:06 -0400 Subject: [OpenSIPS-Users] Raise event from MI/FIFO interface(externally) In-Reply-To: <5bab39fb-8c23-1982-652a-ae9af0bf0d0f@opensips.org> References: <9b3e0dbc-0190-4d15-96b5-c04edcaca90f@opensips.org> <5bab39fb-8c23-1982-652a-ae9af0bf0d0f@opensips.org> Message-ID: This feature was requested here: https://github.com/OpenSIPS/opensips/issues/1526 and now is available in 3.1. -ovidiu On Tue, Nov 6, 2018 at 4:37 PM Bogdan-Andrei Iancu wrote: > > Thanks Sammy for the follow up. > > For the sake of the completion of this discussion, just update here with the feature request link, so people can follow it later. > > Thanks and regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > http://www.opensips-solutions.com > OpenSIPS Bootcamp 2018 > http://opensips.org/training/OpenSIPS_Bootcamp_2018/ > > On 11/05/2018 05:29 PM, SamyGo wrote: > > Hi Bogdan, > Sure thing I'd open up a feature request. Yes thats what we decided to go with to use sipsak or something like that to trigger SIP packet and make use of it. > Apologize for late reply; thanks a lot for your response. > > Regards, > Sammy > > On Tue, Oct 16, 2018 at 4:29 AM Bogdan-Andrei Iancu wrote: >> >> Hi Sammy, >> >> The Event Interface philosophy is to allow external apps to get access to the events generated by OpenSIPS; basically a data flow from OpenSIPS to outside world. So, there is no way to trigger an event from outside (via MI). >> Still, your example is valid, so please open a feature request on github and we could add a "raise_event" MI function. >> >> As really (really) dirty hack, you can use (if possible) some SIP OPTIONS to trigger the event. From the web send an OPTIONS via UDP, with a special RURI -> you identify the RURI in script and do a raise_event() in script. >> >> Regards, >> >> Bogdan-Andrei Iancu >> >> OpenSIPS Founder and Developer >> http://www.opensips-solutions.com >> OpenSIPS Bootcamp 2018 >> http://opensips.org/training/OpenSIPS_Bootcamp_2018/ >> >> On 10/11/2018 07:35 PM, SamyGo wrote: >> >> Hi, >> >> I'm trying to find out document pages regarding raising events from outside the OpenSIPS via fifo/mi_* modules. All I have read so far is opensips can send events and their data OUT to external "subscribing" applications. There is even a fifo command to subscribe for an event from FIFO layer. >> >> Kindly guide me as how do I tell OpenSIPS that a particular Event has triggered. >> >> My usage scenario is (OpenSIPS 2.2.7) a caller is waiting_on_event("OPENSIPS_BOOTCAMP"); >> Now some external web monitoring bot just realized Bootcamp has started and it wants to raise this event to the waiting caller ! >> >> opensipsctl fifo raise_event OPENSIPS_BOOTCAMP 22Oct18 Romania >> >> Best Regards, >> Sammy >> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -- VoIP Embedded, Inc. http://www.voipembedded.com From volga629 at networklab.ca Sun Apr 26 13:35:43 2020 From: volga629 at networklab.ca (volga629) Date: Sun, 26 Apr 2020 10:35:43 -0300 Subject: [OpenSIPS-Users] string comparison In-Reply-To: <7d08e2ce-5c2f-f4f1-8692-09611bb15082@opensips.org> References: <5e6e5656-a785-5b86-a6e9-c591f4442249@skillsearch.ca> <9362669e-d19f-a1f3-f90c-f53dd7668fb4@opensips.org> <75852689-9b39-65f3-d071-2bdb1523dccd@skillsearch.ca> <7d08e2ce-5c2f-f4f1-8692-09611bb15082@opensips.org> Message-ID: Hello Liviu, I restructured config file to better see what was evaluated and I see in debug right now clearly that opensips is not evaluate properly. Oh I don't see something. Debug: https://pastebin.com/GHmbb0Uf Config https://pastebin.com/P60Bh9En volga629 On 4/24/20 7:57 AM, Liviu Chircu wrote: > On 24.04.2020 13:22, volga629 wrote: >> if($var(tls-frompbx)==$var(usr_check_tls)) { >>                 $var(transport) = "tls"; >>                 if(!codec_exists("opus")) { >>                     $var(codec_flag_frompbx_tls) = "transcode-opus"; >>                 } >>                 rtpengine_offer("replace-origin >> replace-session-connection $var(codec_flag_frompbx_tls) RTP/SAVP >> ICE=remove $var(dir)"); >>             } > > Please add an "else" to this "if" block exactly as follows, run it and > provide the output: > >             if($var(tls-frompbx)==$var(usr_check_tls)) { >                 $var(transport) = "tls"; >                 if(!codec_exists("opus")) { >                     $var(codec_flag_frompbx_tls) = "transcode-opus"; >                 } >                 rtpengine_offer("replace-origin > replace-session-connection $var(codec_flag_frompbx_tls) RTP/SAVP > ICE=remove $var(dir)"); >             } else { >                 xlog("L_INFO", "DBG: TLS strings differ: > '$var(tls-frompbx)'/'$var(usr_check_tls)', > '$(var(tls-frompbx){s.len})'/'$(var(usr_check_tls){s.len})', > '$(var(tls-frompbx){s.md5})'/'$(var(usr_check_tls){s.md5})'\n"); >             } > > Also, what is the output of "opensips -V"? > > Best regards, > > -- > Liviu Chircu > www.twitter.com/liviuchircu |www.opensips-solutions.com > > OpenSIPS Summit, Amsterdam, Fall 2020 > www.opensips.org/events > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users From volga629 at networklab.ca Sun Apr 26 13:55:31 2020 From: volga629 at networklab.ca (volga629) Date: Sun, 26 Apr 2020 10:55:31 -0300 Subject: [OpenSIPS-Users] string comparison In-Reply-To: References: <5e6e5656-a785-5b86-a6e9-c591f4442249@skillsearch.ca> <9362669e-d19f-a1f3-f90c-f53dd7668fb4@opensips.org> <75852689-9b39-65f3-d071-2bdb1523dccd@skillsearch.ca> <7d08e2ce-5c2f-f4f1-8692-09611bb15082@opensips.org> Message-ID: <15d214bb-e448-cbe3-e2c0-f4e9ded0e2ce@skillsearch.ca> Hello Liviu, Forgot mention version. [root at pr1 ~]# opensips -V version: opensips 3.1.0-dev (x86_64/linux) flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, HP_MALLOC, DBG_MALLOC, CC_O0, FAST_LOCK-ADAPTIVE_WAIT ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535 poll method support: poll, epoll, sigio_rt, select. main.c compiled on 10:46:54 Jan 29 2020 with gcc 8 volga629 On 4/26/20 10:35 AM, volga629 via Users wrote: > Hello Liviu, > > I restructured config file to better see what was evaluated and I see > in debug right now clearly that opensips is not evaluate properly. > > Oh I don't see something. > > Debug: > > > https://pastebin.com/GHmbb0Uf > > Config > > > https://pastebin.com/P60Bh9En > > > volga629 > > > On 4/24/20 7:57 AM, Liviu Chircu wrote: >> On 24.04.2020 13:22, volga629 wrote: >>> if($var(tls-frompbx)==$var(usr_check_tls)) { >>>                 $var(transport) = "tls"; >>>                 if(!codec_exists("opus")) { >>>                     $var(codec_flag_frompbx_tls) = "transcode-opus"; >>>                 } >>>                 rtpengine_offer("replace-origin >>> replace-session-connection $var(codec_flag_frompbx_tls) RTP/SAVP >>> ICE=remove $var(dir)"); >>>             } >> >> Please add an "else" to this "if" block exactly as follows, run it >> and provide the output: >> >>             if($var(tls-frompbx)==$var(usr_check_tls)) { >>                 $var(transport) = "tls"; >>                 if(!codec_exists("opus")) { >>                     $var(codec_flag_frompbx_tls) = "transcode-opus"; >>                 } >>                 rtpengine_offer("replace-origin >> replace-session-connection $var(codec_flag_frompbx_tls) RTP/SAVP >> ICE=remove $var(dir)"); >>             } else { >>                 xlog("L_INFO", "DBG: TLS strings differ: >> '$var(tls-frompbx)'/'$var(usr_check_tls)', >> '$(var(tls-frompbx){s.len})'/'$(var(usr_check_tls){s.len})', >> '$(var(tls-frompbx){s.md5})'/'$(var(usr_check_tls){s.md5})'\n"); >>             } >> >> Also, what is the output of "opensips -V"? >> >> Best regards, >> >> -- >> Liviu Chircu >> www.twitter.com/liviuchircu  |www.opensips-solutions.com >> >> OpenSIPS Summit, Amsterdam, Fall 2020 >>    www.opensips.org/events >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users From bogdan at opensips.org Mon Apr 27 08:33:43 2020 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Mon, 27 Apr 2020 11:33:43 +0300 Subject: [OpenSIPS-Users] OpenSIPS installation with PostgresSQL In-Reply-To: <1567006032.186555.1587747189795@mail.yahoo.com> References: <129541852.2099756.1587328682918.ref@mail.yahoo.com> <129541852.2099756.1587328682918@mail.yahoo.com> <1985829642.2102360.1587329441034@mail.yahoo.com> <7812bb25-eef4-0b34-04f5-a4f5c5ba589f@opensips.org> <1950080153.2366807.1587399844986@mail.yahoo.com> <1582965925.176731.1587476879885@mail.yahoo.com> <1193421104.181467.1587478895108@mail.yahoo.com> <1971406255.325015.1587497750288@mail.yahoo.com> <15e3ac1f-2e01-a4fb-147a-5ac32411c98d@opensips.org> <1119346896.578368.1587559461192@mail.yahoo.com> <2d3c288a-4f71-d78b-6d83-84e2731fc2e5@opensips.org> <1876180299.356916.1587669506692@mail.yahoo.com> <057323a0-7346-dc4d-2180-4fb7137dd574@opensips.org> <1567006032.186555.1587747189795@mail.yahoo.com> Message-ID: <9d1512ea-2aef-9e45-f165-1cb0e60bf687@opensips.org> Thanks Nayani, This is a bit strante as teh libpq-fe.h file is not directly included by OpenSIPS, but it is via the pq-con.h file. And both are in the same directory pf libpq, directory which should be detected by the OpenSIPS Makefile. Could you run and post the output of: (cd modules/db_postgres/ ; make proper; NICER=0 make ) Please run it as it is, including the brackets Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com On 4/24/20 7:53 PM, Nayani Nikeshala wrote: > Hi Bogdan, > > I added the path of libpq-fe.h to the Makefile. > > [root at SIPserver include]# ls -lrt /usr/pgsql-9.6/include/libpq-fe.h > -rw-r--r--. 1 root root 21900 Feb 12 02:52 > /usr/pgsql-9.6/include/libpq-fe.h > > > > On Friday, April 24, 2020, 01:51:35 PM GMT+5:30, Bogdan-Andrei Iancu > wrote: > > > Hi Nayani, > > What was the fix you did ? the path to the postgres lib? > > Regards, > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > > On 4/23/20 10:18 PM, Nayani Nikeshala wrote: > Hi Bogdan, > > I could fix the problem by modifying Makefile of postgres module. > Thanks for the support. > > > On Thursday, April 23, 2020, 02:21:00 PM GMT+5:30, Bogdan-Andrei Iancu > wrote: > > > Hi, > > I have an 9.5.19 (on Ubuntu) and works ok. > > Regards, > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > > On 4/22/20 3:44 PM, Nayani Nikeshala wrote: > Hi Bogdan, > > I found an old email, where he has faced a similar kind of an issue > with PSQL 9.2 also. I believe, it will not get resolved with PSQL > version. > > https://opensips.org/pipermail/devel/2015-June/017666.html > > According to the email, he has resolved it by modifying makefile, but > I'm not sure, how he did that. > > > On Wednesday, April 22, 2020, 01:54:09 PM GMT+5:30, Bogdan-Andrei > Iancu wrote: > > > It may sound weird, but maybe you into the other extrema, with some > too new version :D... > > Is it possible to try some 9.3 or so ? > > Regards, > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > > On 4/21/20 10:35 PM, Nayani Nikeshala wrote: > Hi Bogdan, > > I removed psql 8.4 and installed psql 9.6. > > [root at SIPserver opensips-2.4.7]# yum list installed | grep postgre > postgresql96.i686 9.6.17-1PGDG.rhel6 > postgresql96-devel.i686 9.6.17-1PGDG.rhel6 > postgresql96-libs.i686  9.6.17-1PGDG.rhel6 > postgresql96-server.i686 9.6.17-1PGDG.rhel6 > [root at SIPserver opensips-2.4.7]# > > > When I compile, I get below error. (Below is a part of the error) > > make[1]: Entering directory > `/home/cscore/Downloads/opensips-2.4.7/modules/db_postgres' > Compiling dbase.c > Compiling pg_con.c > Compiling res.c > Compiling val.c > In file included from pg_con.c:23: > pg_con.h:39:22: error: libpq-fe.h: No such file or directory > In file included from pg_con.c:23: > pg_con.h:54: error: expected specifier-qualifier-list before ‘PGconn’ > pg_con.c: In function ‘db_postgres_new_connection’: > pg_con.c:93: error: ‘struct pg_con’ has no member named ‘con’ > pg_con.c:93: warning: implicit declaration of function ‘PQconnectdbParams’ > > libpq-fe.h & pg_config are in the below paths. > > [root at SIPserver opensips-2.4.7]# ls -lrt /usr/pgsql-9.6/include/libpq-fe.h > -rw-r--r--. 1 root root 21900 Feb 12 02:52 > /usr/pgsql-9.6/include/libpq-fe.h > [root at SIPserver opensips-2.4.7]# > > [root at SIPserver opensips-2.4.7]# ls -lrt /usr/pgsql-9.6/bin/pg_config > -rwxr-xr-x. 1 root root 24544 Feb 12 02:52 /usr/pgsql-9.6/bin/pg_config > > Could you assist me with above error. > > > > On Tuesday, April 21, 2020, 08:57:13 PM GMT+5:30, Bogdan-Andrei Iancu > wrote: > > > Hi Nayani, > > yeah, that is a rather old and unsupported version of postgres. I > think you should try to use a newer version it (maybe a newer version > of your distro). > > Regards, > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > > On 4/21/20 5:21 PM, Nayani Nikeshala wrote: > Hi Bogdan, > > I have attached the output log for the above commands. > > "make modules module=modules/db_postgres" gave me an error as in the > attached file, so I used "make include_modules=db_postgres modules" > instead. I could see below in the output. > > make[1]: Entering directory > `/home/cscore/Downloads/opensips-2.4.7/modules/db_postgres' > make[1]: Leaving directory > `/home/cscore/Downloads/opensips-2.4.7/modules/db_postgres' > make[1]: Entering directory > `/home/cscore/Downloads/opensips-2.4.7/modules/db_postgres' > Compiling dbase.c > Compiling db_postgres.c > Compiling pg_con.c > Compiling res.c > Compiling val.c > pg_con.c: In function db_postgres_new_connection: > pg_con.c:93: warning: implicit declaration of function PQconnectdbParams > pg_con.c:93: warning: assignment makes pointer from integer without a cast > Linking db_postgres.so > make[1]: Leaving directory > `/home/cscore/Downloads/opensips-2.4.7/modules/db_postgres' > > Is this a problem of my psql version(8.4.20), because > PQconnectdbParams  is not available in that version. > > > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Mon Apr 27 08:42:35 2020 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Mon, 27 Apr 2020 11:42:35 +0300 Subject: [OpenSIPS-Users] Tls using t_relay In-Reply-To: References: <39b60bcc-945b-f659-ae7f-37f82327c63c@opensips.org> <4e1e6e4c-b955-0f43-00a0-f5d776ea2e83@opensips.org> Message-ID: Hey Tito, You lost me a bit with your scenario.....You say you do t_relay() twice in the script for the same request? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com On 4/23/20 10:48 PM, Tito Cumpen wrote: > Hey Bogdan, > > Yes it seems like it continued down my routing script and tried to > send it locally as well > which is why I saw these > Apr 21 23:12:46 sip1 /sbin/opensips[11376]: ERROR:tm:msg_send: send() > to 127.0.0.1:443  for proto tls/3 failed > when using the t_relay command. The remote end was sending > provisionals but the failure above was causing the internal error. > > I've found a solution by using force_send_socket(tls:MY_IP:443); my > tls port > > > and then rewriting the host port to the destination. >  rewritehostport("SIPAS:port"); > > should I be using the exit command after a t_relay to avoid this? > > Thanks, > Tito > > > On Thu, Apr 23, 2020 at 1:39 AM Bogdan-Andrei Iancu > > wrote: > > Hi Tito, > > Note that the tcp_connect_timeout is in milliseconds, so maybe 3 > ms is too short for getting back the SYN ACK. > > The logs are keep reporting the failed connect. You say the > connect is ok, the INVITE is sent forward to callee and there is > also an 180 response? ....and then you get the 500 reply ? > > Regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > > On 4/22/20 2:22 AM, Tito Cumpen wrote: >> Bogdan, >> >> I found out that there was a timer flag for tcp connections that >> may have been causing an issue >> tcp_connect_timeout=3 >> Once I removed this line the tls connection was made fine but now >> I am seeing opensips send an error message to the client >> >> SIP/2.0 500 Server error occurred (1/SL) >> client---opensips---SIP AS >> even though the SIP AS sent a 180 response >> >> >> Here are the errors from the log >> >> Apr 21 23:12:46 sip1 /sbin/opensips[11376]: >> ERROR:core:tcp_connect_blocking_timeout: poll error: flags 28 - 4 >> 8 16 32 >> Apr 21 23:12:46 sip1 /sbin/opensips[11376]: >> ERROR:core:tcp_connect_blocking_timeout: failed to retrieve >> SO_ERROR [server=127.0.0.1:443 ] (111) >> Connection refused >> Apr 21 23:12:46 sip1 /sbin/opensips[11376]: >> ERROR:proto_tls:tls_sync_connect: tcp_blocking_connect failed >> Apr 21 23:12:46 sip1 /sbin/opensips[11376]: >> ERROR:proto_tls:proto_tls_send: connect failed >> Apr 21 23:12:46 sip1 /sbin/opensips[11376]: ERROR:tm:msg_send: >> send() to 127.0.0.1:443 for proto tls/3 failed >> Apr 21 23:12:46 sip1 /sbin/opensips[11376]: >> ERROR:tm:t_forward_nonack: sending request failed >> Apr 21 23:12:46 sip1 /sbin/opensips[11376]: >> ERROR:core:tcp_connect_blocking_timeout: poll error: flags 28 - 4 >> 8 16 32 >> Apr 21 23:12:46 sip1 /sbin/opensips[11376]: >> ERROR:core:tcp_connect_blocking_timeout: failed to retrieve >> SO_ERROR [server=127.0.0.1:443 ] (111) >> Connection refused >> Apr 21 23:12:46 sip1 /sbin/opensips[11376]: >> ERROR:proto_tls:tls_sync_connect: tcp_blocking_connect failed >> Apr 21 23:12:46 sip1 /sbin/opensips[11376]: >> ERROR:proto_tls:proto_tls_send: connect failed >> Apr 21 23:12:46 sip1 /sbin/opensips[11376]: ERROR:tm:msg_send: >> send() to 127.0.0.1:443 for proto tls/3 failed >> Apr 21 23:12:46 sip1 /sbin/opensips[11376]: >> ERROR:tm:t_forward_nonack: sending request failed >> Apr 21 23:12:46 sip1 /sbin/opensips[11376]: >> ERROR:core:tcpconn_async_connect: poll error: flags 1c >> Apr 21 23:12:46 sip1 /sbin/opensips[11376]: >> ERROR:core:tcpconn_async_connect: failed to retrieve SO_ERROR >> [server=127.0.0.1:5060 ] (111) Connection >> refused >> Apr 21 23:12:46 sip1 /sbin/opensips[11376]: >> ERROR:core:proto_tcp_send: async TCP connect failed >> Apr 21 23:12:46 sip1 /sbin/opensips[11376]: ERROR:tm:msg_send: >> send() to 127.0.0.1:5060 for proto tcp/2 >> failed >> Apr 21 23:12:46 sip1 /sbin/opensips[11376]: >> ERROR:tm:t_forward_nonack: sending request failed >> Apr 21 23:12:46 sip1 /sbin/opensips[11376]: >> ERROR:core:tcpconn_async_connect: poll error: flags 1c >> Apr 21 23:12:46 sip1 /sbin/opensips[11376]: >> ERROR:core:tcpconn_async_connect: failed to retrieve SO_ERROR >> [server=127.0.0.1:5060 ] (111) Connection >> refused >> Apr 21 23:12:46 sip1 /sbin/opensips[11376]: >> ERROR:core:proto_tcp_send: async TCP connect failed >> Apr 21 23:12:46 sip1 /sbin/opensips[11376]: ERROR:tm:msg_send: >> send() to 127.0.0.1:5060 for proto tcp/2 >> failed >> Apr 21 23:12:46 sip1 /sbin/opensips[11376]: >> ERROR:tm:t_forward_nonack: sending request failed >> Apr 21 23:12:46 sip1 /sbin/opensips[11376]: >> ERROR:core:tcpconn_async_connect: poll error: flags 1c >> Apr 21 23:12:46 sip1 /sbin/opensips[11376]: >> ERROR:core:tcpconn_async_connect: failed to retrieve SO_ERROR >> [server=127.0.0.1:5223 ] (111) Connection >> refused >> Apr 21 23:12:46 sip1 /sbin/opensips[11376]: >> ERROR:core:proto_tcp_send: async TCP connect failed >> Apr 21 23:12:46 sip1 /sbin/opensips[11376]: ERROR:tm:msg_send: >> send() to 127.0.0.1:5223 for proto tcp/2 >> failed >> Apr 21 23:12:46 sip1 /sbin/opensips[11376]: >> ERROR:tm:t_forward_nonack: sending request failed >> Apr 21 23:12:46 sip1 /sbin/opensips[11376]: >> ERROR:core:tcpconn_async_connect: poll error: flags 1c >> Apr 21 23:12:46 sip1 /sbin/opensips[11376]: >> ERROR:core:tcpconn_async_connect: failed to retrieve SO_ERROR >> [server=127.0.0.1:5223 ] (111) Connection >> refused >> Apr 21 23:12:46 sip1 /sbin/opensips[11376]: >> ERROR:core:proto_tcp_send: async TCP connect failed >> Apr 21 23:12:46 sip1 /sbin/opensips[11376]: ERROR:tm:msg_send: >> send() to 127.0.0.1:5223 for proto tcp/2 >> failed >> >> On Tue, Apr 21, 2020 at 11:26 AM Tito Cumpen > > wrote: >> >> Hey Bogdan, >> >> Here is the capture I took from using t_relay("tls:domain:port") >> >> As you can see the client side (opensips) does not proceed >> with allowing the socket to open. >> >> Thanks, >> Tito >> >> On Tue, Apr 21, 2020 at 2:18 AM Bogdan-Andrei Iancu >> > wrote: >> >> Hi Tito, >> >> Well, from OpenSIPS perceptive it is a "Connection >> refused", so the destination is rejecting the connect. >> >> Regards, >> >> Bogdan-Andrei Iancu >> >> OpenSIPS Founder and Developer >> https://www.opensips-solutions.com >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Mon Apr 27 09:10:34 2020 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Mon, 27 Apr 2020 12:10:34 +0300 Subject: [OpenSIPS-Users] opensips 3.0.2 100% CPU after enable tls? In-Reply-To: References: <18b56215-a2ed-6cd2-8730-de499ee521d5@opensips.org> <4bb55ace-480e-4e01-6953-ee296372d426@opensips.org> Message-ID: <98c1752c-3a4f-0b1e-55e1-05b0cbdba44a@opensips.org> Hi William, Please use the nightly builds for 3.0 - there is a fix which didn't make it into the release package. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com On 4/25/20 12:12 AM, William Jin wrote: > It's the release. > > -- > Regards, > William Jin > ------------------------------------------------------------------------ > *From:* Bogdan-Andrei Iancu > *Sent:* Friday, 24 April 2020 6:14 PM > *To:* William Jin ; OpenSIPS users mailling > list > *Subject:* Re: [OpenSIPS-Users] opensips 3.0.2 100% CPU after enable tls? > And it is the release or nightly build ? > > Regards, > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > > On 4/24/20 1:22 AM, William Jin wrote: >> I am using the debian apt repo, not from git. >> >> >> -- >> Regards, >> William Jin >> ------------------------------------------------------------------------ >> *From:* Bogdan-Andrei Iancu >> >> *Sent:* Thursday, 23 April 2020 6:49 PM >> *To:* OpenSIPS users mailling list >> ; William Jin >> >> *Subject:* Re: [OpenSIPS-Users] opensips 3.0.2 100% CPU after enable >> tls? >> Hi William, >> >> What GIT revision of OpenSIPS do you use? (this is exposed by the >> "opensips -V") >> >> Regards, >> Bogdan-Andrei Iancu >> >> OpenSIPS Founder and Developer >> https://www.opensips-solutions.com >> >> On 4/23/20 7:04 AM, William Jin wrote: >>> Hi, >>> >>> Linux platform: Debian 9 (stretch) >>> >>> opensips -V >>> version: opensips 3.0.2 (x86_64/linux) >>> flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, >>> Q_MALLOC, F_MALLOC, HP_MALLOC, DBG_MALLOC, FAST_LOCK-ADAPTIVE_WAIT >>> ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN >>> 16, MAX_URI_SIZE 1024, BUF_SIZE 65535 >>> poll method support: poll, epoll, sigio_rt, select. >>> main.c compiled on  with gcc 6.3.0 >>> >>> related config: >>> >>> listen = tls:xxx.xxx.xxx.xxx:5061 anycast >>> >>> >>> ####TLS >>> loadmodule "tls_mgm.so" >>> loadmodule "proto_tls.so" >>> >>> modparam("tls_mgm", "server_domain", "sip1") >>> modparam("tls_mgm", "match_ip_address", "[sip1]xx.xx.xx.xx:5061") >>> modparam("tls_mgm", "match_sip_domain", "[sip1]xxx.xxx.example.com") >>> >>> modparam("tls_mgm", "verify_cert", "[sip1]1") >>> modparam("tls_mgm", "require_cert", "[sip1]0") >>> modparam("tls_mgm", "tls_method", "[sip1]SSLv23") >>> modparam("tls_mgm", "ciphers_list", >>> "[sip1]AES256-GCM-SHA384,AES256-SHA256,AES256-SHA,CAMELLIA256-SHA,AES128-SHA,SEED-SHA,CAMELLIA128-SHA,RC4-SHA,DES-CBC3-SHA") >>> >>> >>> modparam("tls_mgm", "certificate", >>> "[sip1]/etc/opensips/tls/mycerts/selfsignedcert.pem") >>> modparam("tls_mgm", "private_key", >>> "[sip1]/etc/opensips/tls/mycerts/unsecuredkey.pem") >>> >>> >>> opensips-cli -x trap {pid} result attached >>> >>> Can someone shed some light on it? Thanks. >>> >>> >>> -- >>> Regards, >>> William Jin >>> >>> _______________________________________________ >>> Users mailing list >>> Users at lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Mon Apr 27 09:46:50 2020 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Mon, 27 Apr 2020 12:46:50 +0300 Subject: [OpenSIPS-Users] Problem with add_rr_param() in v2.4.7 In-Reply-To: <001801d6135a$d1db1a70$75914f50$@smartvox.co.uk> References: <001b01d61270$cd7d8200$68788600$@smartvox.co.uk> <001801d6135a$d1db1a70$75914f50$@smartvox.co.uk> Message-ID: <52762473-5103-8b19-a598-be92f42a3da0@opensips.org> Hi John, Using the latest 2.4 from GIT with the default cfg, I simply added your lines instead on the record_route();     # record routing     if (!is_method("REGISTER|MESSAGE")) {         record_route_preset("127.0.0.1:5060;transport=udp", "127.0.0.1:5060");         add_rr_param(";r2=on");     } As a result, in the outbound INVITE, I got the RR hdrs: Record-Route: Record-Route: Again, could you come up (starting from the default opensips.cfg) with a minimal cfg to show your issue ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com On 4/15/20 10:19 PM, John Quick wrote: > Hi Bogdan, > > The only things that are unusual are: > 1) transport protocol conversion between TLS and UDP so it requires double RR's. > 2) it must use FQDN instead of IP address in the Record Route headers for the TLS interface > > The script adds two RR headers like this: > > record_route_preset(":5061;transport=tls", ":5060"); > add_rr_param(";r2=on"); > > There is also one more call to add_rr_param() after those two lines, but none of the parameters is added. > > For calls going in the other direction: > record_route_preset(":5060", ":5061;transport=tls"); > add_rr_param(";r2=on"); > > When testing with v2.4.7, the listen statements were like this: > auto_aliases=no > alias=udp::5060 > alias=tls::5061 > listen=udp::5060 > listen=tls::5061 > > One idea I had is that it might work to use record_route() instead of record_route_preset() provided I changed the second listen statement to this: > listen=tls::5061 AS :5061 > > ...but I haven't tested to see if the fault only happens with record_route_preset() and not with record_route(). > > John Quick > Smartvox Limited > > > -----Original Message----- > From: Bogdan-Andrei Iancu > Sent: 15 April 2020 18:37 > To: john.quick at smartvox.co.uk; OpenSIPS users mailling list > Subject: Re: [OpenSIPS-Users] Problem with add_rr_param() in v2.4.7 > > Hi John, > > So you say you experience a regression between 2.4.6 and 2.4.7.... Any particularities in terms of how you do the record_routing() and add_rr_param(), like sequence, other msg changes or signaling ? > > Any simple way to reproduce the issue? > > Regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > > On 4/14/20 6:24 PM, John Quick wrote: >> I've tried running the same opensips.cfg script in v2.4.6 and then >> 2.4.7 When it is changed to 2.4.7, the function add_rr_param() does nothing. >> When run under 2.4.6 it updates the Record-Route header as you would expect. >> >> John Quick >> Smartvox Limited >> Web: www.smartvox.co.uk >> >> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users From volga629 at networklab.ca Mon Apr 27 09:58:29 2020 From: volga629 at networklab.ca (volga629) Date: Mon, 27 Apr 2020 06:58:29 -0300 Subject: [OpenSIPS-Users] ms teams ACK In-Reply-To: <76b1f17e-21c7-8ccb-d5c9-abe54701d564@skillsearch.ca> References: <94a0f640-5400-9124-0a98-96556fa000b6@skillsearch.ca> <6f80948f-fa0a-696f-eb11-2a46e2f49591@skillsearch.ca> <0c16126f-931a-d8a6-5aaa-07a67a8727d1@skillsearch.ca> <1587240834447-0.post@n2.nabble.com> <4e26f956-8513-3863-8ced-67bcac216861@skillsearch.ca> <76b1f17e-21c7-8ccb-d5c9-abe54701d564@skillsearch.ca> Message-ID: Hello Everyone, Thank you all for the help. Here are some highlight 1. Avoid call NAT in reply with 200 OK with option 1 or 3 2. Preset routing header toward Microsoft Teams. 3. Use record route toward you asterisk or feeeswitch. 4. Append contact header in keepalive messages. 5. Use dialplan feature to remove append + to normalize phone numbers. volga629 On 4/19/20 9:40 AM, volga629 via Users wrote: > > Hello Johan, > > Thank you for reply. > > > The only NAT problem can be on MS Teams Client,  because on Opensips > side pretty sure all good. > > > volga629 > > > On 4/19/20 3:04 AM, Johan De Clercq wrote: >> Can’t it be a NAT problem? The IP address where the bye is coming >> from doesn’t seem a pstnhub to me. >> >> Outlook voor iOS downloaden >> ------------------------------------------------------------------------ >> *Van:* Users namens volga629 via >> Users >> *Verzonden:* Saturday, April 18, 2020 11:01:19 PM >> *Aan:* OpenSIPS users mailling list ; >> Alexey Vasilyev >> *Onderwerp:* Re: [OpenSIPS-Users] ms teams ACK >> >> Hello Alexey, >> >> Thank you on reply, >> >> I undone all changes regard headers changes and MS Teams send BYE >> directly to asterisk. >> >> No Route header present. >> >> But INVITE ACK 183 180 all travel with  proper routing information. >> >> >> 2020/04/18 17:54:28.599711 190.109.70.77:5060 -> 190.109.68.250:5060 >> BYE sip:11988582770 at 190.109.68.250:5060 >> SIP/2.0 >> FROM: >> ;tag=4d7fb0763c224e39a13a03c669c4b387 >> TO: >> ;tag=as41e97ff5 >> CSEQ: 3 BYE >> CALL-ID: 2e6c1a8d2383a4752403e94512ced077 at 190.109.70.77 >> >> MAX-FORWARDS: 69 >> Via: SIP/2.0/UDP >> 190.109.70.77:5060;branch=z9hG4bK050e.e400e373.0;i=66c9c603 >> VIA: SIP/2.0/TLS >> 52.114.14.70:5061;rport=8208;received=52.114.14.70;branch=z9hG4bK9594cd7 >> REASON: >> Q.850;cause=18;text="fcb37a2a-4bc4-49b6-a5e3-aabddc8f7a22;Call >> Controller timed out while waiting for acknowledgement." >> CONTACT: >> >> >> CONTENT-LENGTH: 0 >> USER-AGENT: Microsoft.PSTNHub.SIPProxy v.2020.4.13.7 i.ASSE.3 >> ALLOW: INVITE,ACK,OPTIONS,CANCEL,BYE,NOTIFY >> >> >> volga629 >> >> >> On 4/18/20 5:13 PM, Alexey Vasilyev wrote: >>> Hi volga629, >>> >>> There were nothing special for ACK. You don't need to change >>> To/From/Contact. All the necessary steps were in the article >>> https://blog.opensips.org/2019/09/16/opensips-as-ms-teams-sbc/ and for most >>> people it still works. >>> So I'm not sure, that MS changed anything, because all the hardware SBCs >>> should change behaviour, so they need new firmware. SBC vendors should >>> inform customers to update etc. So this is not so simple process. And it >>> definitely make no sense for anybody. >>> And in the test lab for the article I've used absolutely the same >>> architecture with asterisk, the only difference was RTPEngine to transcode >>> SRTP-RTP. >>> And within test lab I've tested not only calls, but transfers worked fine >>> too. >>> >>> >>> >>> ----- >>> --- >>> Alexey Vasilyev >>> -- >>> Sent from:http://opensips-open-sip-server.1449251.n2.nabble.com/OpenSIPS-Users-f1449235.html >>> >>> _______________________________________________ >>> Users mailing list >>> Users at lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users From farmorg at gmail.com Mon Apr 27 13:30:56 2020 From: farmorg at gmail.com (Mark Farmer) Date: Mon, 27 Apr 2020 14:30:56 +0100 Subject: [OpenSIPS-Users] Access to listen/advertised IP Addresses In-Reply-To: References: <5eb6a6ca-83aa-c59e-2f2a-2c4f216a4118@opensips.org> <5957bcba-f283-50e4-19a7-4b4d43f8b987@opensips.org> Message-ID: Hi Bogdan That looks great! Exactly what I was looking for :) Will this be available in 3.1 only? My solution currently uses 2.4 but I'm hoping to be able to upgrade it to 3.1 before go live. Nice work and thank you. Mark. On Thu, 23 Apr 2020 at 11:49, Bogdan-Andrei Iancu wrote: > Hi Mark, > > check these new variables $socket_in() and $socket_out() in 3.1 > > https://www.opensips.org/Documentation/Script-CoreVar-3-1#socket_in > > https://www.opensips.org/Documentation/Script-CoreVar-3-1#socket_out > > Regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > > On 4/7/20 7:29 PM, Bogdan-Andrei Iancu wrote: > > No need, just use in script, where ever you need $socket_in(advertised_ip) > and it will be evaluated for the current socket (used for receiving the > request) > > Regardsm > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > OpenSIPS Summit, Amsterdam, May 2020 > https://www.opensips.org/events/Summit-2020Amsterdam/ > > On 4/7/20 6:56 PM, Mark Farmer wrote: > > I was thinking something like: > > modparam("cfgutils", "varset", "extip=s:$in_socket.advertised_ip") > > > On Tue, 7 Apr 2020 at 14:40, Bogdan-Andrei Iancu > wrote: > >> Hi Mark, >> >> ingenious solution :) >> >> In regards to the proposed solution, I do not understand the question >> about varset (cfgutils), as there is no relation between the script vars >> and these new $socket vars. Maybe I'm missing something from your question. >> >> Regards, >> >> Bogdan-Andrei Iancu >> >> OpenSIPS Founder and Developer >> https://www.opensips-solutions.com >> OpenSIPS Summit, Amsterdam, May 2020 >> https://www.opensips.org/events/Summit-2020Amsterdam/ >> >> On 4/7/20 2:10 PM, Mark Farmer wrote: >> >> Hi Bogdan >> >> The root of my issue is that I need 2 variables containing the IP's of my >> 2 interfaces (mhomed=yes) but the advertised address of the NAT'd DMZ >> interface while keeping changes per server to a bare minimum to ease >> deployment. >> >> I actually solved my issue by using include_file and using cfgutils to >> set 2 script variables. So now all deployment changes are confined to a >> much simpler/smaller file. >> >> However, the proposed changes would make things even nicer. Would >> cfgutils be able to accept those variables as parameters to the 'varset' >> function? >> >> Regards >> Mark. >> >> >> >> On Tue, 7 Apr 2020 at 11:44, Bogdan-Andrei Iancu >> wrote: >> >>> Hi guys, >>> >>> Maybe adding a new core variable like $in_socket.XXXX, to give access to >>> various fields, like $in_socket.ip, $in_socket.port, $in_socket.advertised_ip, >>> etc. This will replace the $Ri and $Rp >>> >>> And we can also add $out_socket, that will similarly replace the $fs >>> (forced socket) >>> >>> Regards, >>> >>> Bogdan-Andrei Iancu >>> >>> OpenSIPS Founder and Developer >>> https://www.opensips-solutions.com >>> OpenSIPS Summit, Amsterdam, May 2020 >>> https://www.opensips.org/events/Summit-2020Amsterdam/ >>> >>> On 4/6/20 6:00 PM, Johan De Clercq wrote: >>> >>> It,s not exposed I think. I can’t find it back either >>> >>> Outlook voor iOS downloaden >>> ------------------------------ >>> *Van:* Users >>> namens David Villasmil >>> >>> *Verzonden:* Monday, April 6, 2020 4:49:36 PM >>> *Aan:* OpenSIPS users mailling list >>> >>> *Onderwerp:* Re: [OpenSIPS-Users] Access to listen/advertised IP >>> Addresses >>> >>> No, you’re right. It’s not in the core variables and I can’t find it >>> either. Which makes me think it’s either not exposed or somewhere in a >>> module (it’s not in proto_udp) >>> >>> I will research a little to try and find it.. >>> >>> On Mon, 6 Apr 2020 at 14:04, Mark Farmer wrote: >>> >>> Thanks David. But I see no reference to the same variable in OpenSIPS. >>> >>> https://www.opensips.org/Documentation/Script-CoreVar-2-4 >>> >>> Am I missing something? >>> >>> >>> On Mon, 6 Apr 2020 at 13:45, David Villasmil < >>> david.villasmil.work at gmail.com> wrote: >>> >>> Right here: >>> >>> >>> https://www.kamailio.org/wiki/cookbooks/5.2.x/pseudovariables#rai_-_received_advertised_ip_address >>> >>> Regards, >>> >>> David Villasmil >>> email: david.villasmil.work at gmail.com >>> phone: +34669448337 >>> >>> >>> On Mon, Apr 6, 2020 at 12:08 PM Mark Farmer wrote: >>> >>> Many thanks for the reply. >>> >>> $Ri is certainly useful when the request comes from a non-natted >>> interface. Thanks for pointing that out :) >>> >>> Is there a way to reference the advertised IP address defined in the >>> listen statement? >>> >>> listen=udp:xxx.xxx.xxx.xxx:5060 as xxx.xxx.xxx.xxx:5060 >>> >>> Thanks >>> Mark. >>> >>> >>> On Thu, 2 Apr 2020 at 17:32, Sharad Kumar via Users < >>> users at lists.opensips.org> wrote: >>> >>> Hi Mark, >>> >>> If your initial goal is to get the interface IP where request is >>> received then you can try these variables. >>> >>> *$Ri* - reference to IP address of the interface where the request has >>> been received >>> >>> *$Rp* - reference to the port where the message was received >>> _______________________________________________ >>> Users mailing list >>> Users at lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >>> >>> >>> -- >>> Mark Farmer >>> farmorg at gmail.com >>> _______________________________________________ >>> Users mailing list >>> Users at lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >>> _______________________________________________ >>> Users mailing list >>> Users at lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >>> >>> >>> -- >>> Mark Farmer >>> farmorg at gmail.com >>> _______________________________________________ >>> Users mailing list >>> Users at lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >>> -- >>> Regards, >>> >>> David Villasmil >>> email: david.villasmil.work at gmail.com >>> phone: +34669448337 >>> >>> _______________________________________________ >>> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >>> >>> _______________________________________________ >>> Users mailing list >>> Users at lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >> >> >> -- >> Mark Farmer >> farmorg at gmail.com >> >> _______________________________________________ >> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> >> > > -- > Mark Farmer > farmorg at gmail.com > > > > _______________________________________________ > Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > -- Mark Farmer farmorg at gmail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Mon Apr 27 14:38:05 2020 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Mon, 27 Apr 2020 17:38:05 +0300 Subject: [OpenSIPS-Users] Access to listen/advertised IP Addresses In-Reply-To: References: <5eb6a6ca-83aa-c59e-2f2a-2c4f216a4118@opensips.org> <5957bcba-f283-50e4-19a7-4b4d43f8b987@opensips.org> Message-ID: <69288ddb-c0bf-7a7f-1776-f426fe7d73b9@opensips.org> Hi Mark, Only fixes are backported to the existing releases, so these new variables are available only starting 3.1 Best regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com On 4/27/20 4:30 PM, Mark Farmer wrote: > Hi Bogdan > > That looks great! Exactly what I was looking for :) > > Will this be available in 3.1 only? My solution currently uses 2.4 but > I'm hoping to be able to upgrade it to 3.1 before go live. > > Nice work and thank you. > Mark. > > > > On Thu, 23 Apr 2020 at 11:49, Bogdan-Andrei Iancu > wrote: > > Hi Mark, > > check these new variables $socket_in() and $socket_out() in 3.1 > > https://www.opensips.org/Documentation/Script-CoreVar-3-1#socket_in > > https://www.opensips.org/Documentation/Script-CoreVar-3-1#socket_out > > Regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > > On 4/7/20 7:29 PM, Bogdan-Andrei Iancu wrote: >> No need, just use in script, where ever you need >> $socket_in(advertised_ip) and it will be evaluated for the >> current socket (used for receiving the request) >> >> Regardsm >> Bogdan-Andrei Iancu >> >> OpenSIPS Founder and Developer >> https://www.opensips-solutions.com >> OpenSIPS Summit, Amsterdam, May 2020 >> https://www.opensips.org/events/Summit-2020Amsterdam/ >> >> On 4/7/20 6:56 PM, Mark Farmer wrote: >>> I was thinking something like: >>> >>> modparam("cfgutils", "varset", "extip=s:$in_socket.advertised_ip") >>> >>> >>> On Tue, 7 Apr 2020 at 14:40, Bogdan-Andrei Iancu >>> > wrote: >>> >>> Hi Mark, >>> >>> ingenious solution :) >>> >>> In regards to the proposed solution, I do not understand the >>> question about varset (cfgutils), as there is no relation >>> between the script vars and these new $socket vars. Maybe >>> I'm missing something from your question. >>> >>> Regards, >>> >>> Bogdan-Andrei Iancu >>> >>> OpenSIPS Founder and Developer >>> https://www.opensips-solutions.com >>> OpenSIPS Summit, Amsterdam, May 2020 >>> https://www.opensips.org/events/Summit-2020Amsterdam/ >>> >>> On 4/7/20 2:10 PM, Mark Farmer wrote: >>>> Hi Bogdan >>>> >>>> The root of my issue is that I need 2 variables containing >>>> the IP's of my 2 interfaces (mhomed=yes) but the advertised >>>> address of the NAT'd DMZ interface while keeping changes >>>> per server to a bare minimum to ease deployment. >>>> >>>> I actually solved my issue by using include_file and using >>>> cfgutils to set 2 script variables. So now all deployment >>>> changes are confined to a much simpler/smaller file. >>>> >>>> However, the proposed changes would make things even nicer. >>>> Would cfgutils be able to accept those variables as >>>> parameters to the 'varset' function? >>>> >>>> Regards >>>> Mark. >>>> >>>> >>>> >>>> On Tue, 7 Apr 2020 at 11:44, Bogdan-Andrei Iancu >>>> > wrote: >>>> >>>> Hi guys, >>>> >>>> Maybe adding a new core variable like $in_socket.XXXX, >>>> to give access to various fields, like $in_socket.ip, >>>> $in_socket.port, $in_socket.advertised_ip, etc. This >>>> will replace the $Ri and $Rp >>>> >>>> And we can also add $out_socket, that will similarly >>>> replace the $fs (forced socket) >>>> >>>> Regards, >>>> >>>> Bogdan-Andrei Iancu >>>> >>>> OpenSIPS Founder and Developer >>>> https://www.opensips-solutions.com >>>> OpenSIPS Summit, Amsterdam, May 2020 >>>> https://www.opensips.org/events/Summit-2020Amsterdam/ >>>> >>>> On 4/6/20 6:00 PM, Johan De Clercq wrote: >>>>> It,s not exposed I think. I can’t find it back either >>>>> >>>>> Outlook voor iOS downloaden >>>>> ------------------------------------------------------------------------ >>>>> *Van:* Users >>>>> namens David >>>>> Villasmil >>>>> >>>>> *Verzonden:* Monday, April 6, 2020 4:49:36 PM >>>>> *Aan:* OpenSIPS users mailling list >>>>> >>>>> >>>>> *Onderwerp:* Re: [OpenSIPS-Users] Access to >>>>> listen/advertised IP Addresses >>>>> No, you’re right. It’s not in the core variables and I >>>>> can’t find it either. Which makes me think it’s either >>>>> not exposed or somewhere in a module (it’s not in >>>>> proto_udp) >>>>> >>>>> I will research a little to try and find it.. >>>>> >>>>> On Mon, 6 Apr 2020 at 14:04, Mark Farmer >>>>> > wrote: >>>>> >>>>> Thanks David. But I see no reference to the same >>>>> variable in OpenSIPS. >>>>> >>>>> https://www.opensips.org/Documentation/Script-CoreVar-2-4 >>>>> >>>>> Am I missing something? >>>>> >>>>> >>>>> On Mon, 6 Apr 2020 at 13:45, David Villasmil >>>>> >>>> > wrote: >>>>> >>>>> Right here: >>>>> >>>>> https://www.kamailio.org/wiki/cookbooks/5.2.x/pseudovariables#rai_-_received_advertised_ip_address >>>>> >>>>> Regards, >>>>> >>>>> David Villasmil >>>>> email: david.villasmil.work at gmail.com >>>>> >>>>> phone: +34669448337 >>>>> >>>>> >>>>> On Mon, Apr 6, 2020 at 12:08 PM Mark Farmer >>>>> > >>>>> wrote: >>>>> >>>>> Many thanks for the reply. >>>>> >>>>> $Ri is certainly useful when the request >>>>> comes from a non-natted interface. Thanks >>>>> for pointing that out :) >>>>> >>>>> Is there a way to reference the advertised >>>>> IP address defined in the listen statement? >>>>> >>>>> listen=udp:xxx.xxx.xxx.xxx:5060 as >>>>> xxx.xxx.xxx.xxx:5060 >>>>> >>>>> Thanks >>>>> Mark. >>>>> >>>>> >>>>> On Thu, 2 Apr 2020 at 17:32, Sharad Kumar >>>>> via Users >>>> > wrote: >>>>> >>>>> Hi Mark, >>>>> >>>>>  If your initial goal is to get the >>>>> interface IP where request is received >>>>> then you can try these variables. >>>>> >>>>> *$Ri* - reference to IP address of the >>>>> interface where the request has been >>>>> received >>>>> >>>>> *$Rp* - reference to the port where >>>>> the message was received >>>>> >>>>> _______________________________________________ >>>>> Users mailing list >>>>> Users at lists.opensips.org >>>>> >>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>>> >>>>> >>>>> >>>>> -- >>>>> Mark Farmer >>>>> farmorg at gmail.com >>>>> _______________________________________________ >>>>> Users mailing list >>>>> Users at lists.opensips.org >>>>> >>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>>> >>>>> _______________________________________________ >>>>> Users mailing list >>>>> Users at lists.opensips.org >>>>> >>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>>> >>>>> >>>>> >>>>> -- >>>>> Mark Farmer >>>>> farmorg at gmail.com >>>>> _______________________________________________ >>>>> Users mailing list >>>>> Users at lists.opensips.org >>>>> >>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>>> >>>>> -- >>>>> 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 >>>> >>>> >>>> >>>> -- >>>> Mark Farmer >>>> farmorg at gmail.com >>>> >>>> _______________________________________________ >>>> Users mailing list >>>> Users at lists.opensips.org >>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >>> >>> >>> -- >>> Mark Farmer >>> farmorg at gmail.com >> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > -- > Mark Farmer > farmorg at gmail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From williamj at exetel.com.au Mon Apr 27 22:28:46 2020 From: williamj at exetel.com.au (William Jin) Date: Mon, 27 Apr 2020 22:28:46 +0000 Subject: [OpenSIPS-Users] opensips 3.0.2 100% CPU after enable tls? In-Reply-To: <98c1752c-3a4f-0b1e-55e1-05b0cbdba44a@opensips.org> References: <18b56215-a2ed-6cd2-8730-de499ee521d5@opensips.org> <4bb55ace-480e-4e01-6953-ee296372d426@opensips.org> , <98c1752c-3a4f-0b1e-55e1-05b0cbdba44a@opensips.org> Message-ID: Ok, I will try that and let you know the result And BTW, does the 2.4.7 (release) also has this issue? -- Regards, William Jin ________________________________ From: Bogdan-Andrei Iancu Sent: Monday, 27 April 2020 7:10 PM To: William Jin ; OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] opensips 3.0.2 100% CPU after enable tls? Hi William, Please use the nightly builds for 3.0 - there is a fix which didn't make it into the release package. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com On 4/25/20 12:12 AM, William Jin wrote: It's the release. -- Regards, William Jin ________________________________ From: Bogdan-Andrei Iancu Sent: Friday, 24 April 2020 6:14 PM To: William Jin ; OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] opensips 3.0.2 100% CPU after enable tls? And it is the release or nightly build ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com On 4/24/20 1:22 AM, William Jin wrote: I am using the debian apt repo, not from git. -- Regards, William Jin ________________________________ From: Bogdan-Andrei Iancu Sent: Thursday, 23 April 2020 6:49 PM To: OpenSIPS users mailling list ; William Jin Subject: Re: [OpenSIPS-Users] opensips 3.0.2 100% CPU after enable tls? Hi William, What GIT revision of OpenSIPS do you use? (this is exposed by the "opensips -V") Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com On 4/23/20 7:04 AM, William Jin wrote: Hi, Linux platform: Debian 9 (stretch) opensips -V version: opensips 3.0.2 (x86_64/linux) flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, HP_MALLOC, DBG_MALLOC, FAST_LOCK-ADAPTIVE_WAIT ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535 poll method support: poll, epoll, sigio_rt, select. main.c compiled on with gcc 6.3.0 related config: listen = tls:xxx.xxx.xxx.xxx:5061 anycast ####TLS loadmodule "tls_mgm.so" loadmodule "proto_tls.so" modparam("tls_mgm", "server_domain", "sip1") modparam("tls_mgm", "match_ip_address", "[sip1]xx.xx.xx.xx:5061") modparam("tls_mgm", "match_sip_domain", "[sip1]xxx.xxx.example.com") modparam("tls_mgm", "verify_cert", "[sip1]1") modparam("tls_mgm", "require_cert", "[sip1]0") modparam("tls_mgm", "tls_method", "[sip1]SSLv23") modparam("tls_mgm", "ciphers_list", "[sip1]AES256-GCM-SHA384,AES256-SHA256,AES256-SHA,CAMELLIA256-SHA,AES128-SHA,SEED-SHA,CAMELLIA128-SHA,RC4-SHA,DES-CBC3-SHA") modparam("tls_mgm", "certificate", "[sip1]/etc/opensips/tls/mycerts/selfsignedcert.pem") modparam("tls_mgm", "private_key", "[sip1]/etc/opensips/tls/mycerts/unsecuredkey.pem") opensips-cli -x trap {pid} result attached Can someone shed some light on it? Thanks. -- Regards, William Jin _______________________________________________ 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 anexiole at gmail.com Tue Apr 28 01:41:54 2020 From: anexiole at gmail.com (Gordon Yeong) Date: Tue, 28 Apr 2020 11:41:54 +1000 Subject: [OpenSIPS-Users] HOW to simulate ENUM server queries Message-ID: hi guys, I have a dev machine which has opensips installed. I am going to develop a function to be used in opensips.cfg which uses the ENUM module's functions (ie. enum_query, i_enum_query) but I do not have access to my live ENUM server. I have some sample ENUM results which i can alter its resulting contents for my test purposes but does anyone know how can we mock up the ENUM Server module in opensips? As in, where can I feed my mocked up test results so that the enum_query, i_enum_query functions off the Enum module could pickup and return? Also, I just need to see how exactly the functions, enum_query, i_enum_query return data look like. Regards, Gordon Yeong -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Tue Apr 28 07:19:19 2020 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Tue, 28 Apr 2020 10:19:19 +0300 Subject: [OpenSIPS-Users] opensips 3.0.2 100% CPU after enable tls? In-Reply-To: References: <18b56215-a2ed-6cd2-8730-de499ee521d5@opensips.org> <4bb55ace-480e-4e01-6953-ee296372d426@opensips.org> <98c1752c-3a4f-0b1e-55e1-05b0cbdba44a@opensips.org> Message-ID: <672fb9bb-b06f-0772-8d23-eb861027b118@opensips.org> Similarly, try the nightly build Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com On 4/28/20 1:28 AM, William Jin wrote: > Ok, I will try that and let you know the result > And BTW, does the 2.4.7 (release) also has this issue? > > > -- > Regards, > William Jin > ------------------------------------------------------------------------ > *From:* Bogdan-Andrei Iancu > *Sent:* Monday, 27 April 2020 7:10 PM > *To:* William Jin ; OpenSIPS users mailling > list > *Subject:* Re: [OpenSIPS-Users] opensips 3.0.2 100% CPU after enable tls? > Hi William, > > Please use the nightly builds for 3.0 - there is a fix which didn't > make it into the release package. > > Regards, > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > > On 4/25/20 12:12 AM, William Jin wrote: >> It's the release. >> >> -- >> Regards, >> William Jin >> ------------------------------------------------------------------------ >> *From:* Bogdan-Andrei Iancu >> >> *Sent:* Friday, 24 April 2020 6:14 PM >> *To:* William Jin >> ; OpenSIPS users mailling list >> >> *Subject:* Re: [OpenSIPS-Users] opensips 3.0.2 100% CPU after enable >> tls? >> And it is the release or nightly build ? >> >> Regards, >> Bogdan-Andrei Iancu >> >> OpenSIPS Founder and Developer >> https://www.opensips-solutions.com >> >> On 4/24/20 1:22 AM, William Jin wrote: >>> I am using the debian apt repo, not from git. >>> >>> >>> -- >>> Regards, >>> William Jin >>> ------------------------------------------------------------------------ >>> *From:* Bogdan-Andrei Iancu >>> >>> *Sent:* Thursday, 23 April 2020 6:49 PM >>> *To:* OpenSIPS users mailling list >>> ; William Jin >>> >>> *Subject:* Re: [OpenSIPS-Users] opensips 3.0.2 100% CPU after enable >>> tls? >>> Hi William, >>> >>> What GIT revision of OpenSIPS do you use? (this is exposed by the >>> "opensips -V") >>> >>> Regards, >>> Bogdan-Andrei Iancu >>> >>> OpenSIPS Founder and Developer >>> https://www.opensips-solutions.com >>> >>> On 4/23/20 7:04 AM, William Jin wrote: >>>> Hi, >>>> >>>> Linux platform: Debian 9 (stretch) >>>> >>>> opensips -V >>>> version: opensips 3.0.2 (x86_64/linux) >>>> flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, >>>> Q_MALLOC, F_MALLOC, HP_MALLOC, DBG_MALLOC, FAST_LOCK-ADAPTIVE_WAIT >>>> ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN >>>> 16, MAX_URI_SIZE 1024, BUF_SIZE 65535 >>>> poll method support: poll, epoll, sigio_rt, select. >>>> main.c compiled on  with gcc 6.3.0 >>>> >>>> related config: >>>> >>>> listen = tls:xxx.xxx.xxx.xxx:5061 anycast >>>> >>>> >>>> ####TLS >>>> loadmodule "tls_mgm.so" >>>> loadmodule "proto_tls.so" >>>> >>>> modparam("tls_mgm", "server_domain", "sip1") >>>> modparam("tls_mgm", "match_ip_address", "[sip1]xx.xx.xx.xx:5061") >>>> modparam("tls_mgm", "match_sip_domain", "[sip1]xxx.xxx.example.com") >>>> >>>> modparam("tls_mgm", "verify_cert", "[sip1]1") >>>> modparam("tls_mgm", "require_cert", "[sip1]0") >>>> modparam("tls_mgm", "tls_method", "[sip1]SSLv23") >>>> modparam("tls_mgm", "ciphers_list", >>>> "[sip1]AES256-GCM-SHA384,AES256-SHA256,AES256-SHA,CAMELLIA256-SHA,AES128-SHA,SEED-SHA,CAMELLIA128-SHA,RC4-SHA,DES-CBC3-SHA") >>>> >>>> >>>> modparam("tls_mgm", "certificate", >>>> "[sip1]/etc/opensips/tls/mycerts/selfsignedcert.pem") >>>> modparam("tls_mgm", "private_key", >>>> "[sip1]/etc/opensips/tls/mycerts/unsecuredkey.pem") >>>> >>>> >>>> opensips-cli -x trap {pid} result attached >>>> >>>> Can someone shed some light on it? Thanks. >>>> >>>> >>>> -- >>>> Regards, >>>> William Jin >>>> >>>> _______________________________________________ >>>> 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 Apr 28 09:01:19 2020 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Tue, 28 Apr 2020 12:01:19 +0300 Subject: [OpenSIPS-Users] disable_503_translation In-Reply-To: References: <55435248-82fe-e1b7-782a-587ecad53b51@opensips.org> Message-ID: Hi Callum, Maybe a better approach will be try to identify the potential leak as described here -> https://opensips.org/Documentation/TroubleShooting-OutOfMem Best regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com On 4/28/20 1:38 AM, Callum Guy wrote: > Hi Bogdan, > > I'm still searching for my memory leak, just downgraded from 3.0.2 to > 3.0.1 and should be able to confirm if that has made a difference in a > day or two. > > During the hunt I've been reviewing the changelog commits - wondered > if you could check params var created at the link below: > > https://github.com/OpenSIPS/opensips/blob/559921962c69575bcdc90c1f03102e6d5c56d776/modules/dialog/dlg_req_within.c#L489 > > My understanding of memory management in C isn't great but it looks to > be allocating without a free? > > Many thanks, > > Callum > > > On Fri, 24 Apr 2020 at 17:16, Callum Guy wrote: >> Thanks Bogdan, I realise it was a long shot! There are only minor differences between this and another system i am running so I'm going through eliminating the config differences. >> >> Version wise the stable one is 3.0.1 and the leaky one is 3.0.2 >> >> Slow progress but I will get there eventually! >> >> On Fri, 24 Apr 2020, 16:50 Bogdan-Andrei Iancu, wrote: >>> Hi Callum, >>> >>> I 99.9999999999999999999% that that setting cannot generate a leak :) >>> >>> Best regards, >>> >>> Bogdan-Andrei Iancu >>> >>> OpenSIPS Founder and Developer >>> https://www.opensips-solutions.com >>> >>> On 4/24/20 6:31 PM, Callum Guy wrote: >>> >>> Hi All, >>> >>> I've been hunting a minor memory leak in my config and wanted to check in with the devs in case it is related to ues of the parameter: >>> >>> disable_503_translation=yes >>> >>> Here is the implementation link to save you a few seconds: >>> >>> https://github.com/OpenSIPS/opensips/blob/7dd1151341b8229cd30e335b246e56938551f6bd/msg_translator.c#L2425 >>> >>> Is there any chance that this is failing to free the tiny amount it's allocating on each use? >>> >>> All the best, >>> >>> Callum >>> >>> >>> 0333 332 0000 | x-on.co.uk | | Coronavirus >>> >>> THE ITSPA AWARDS 2020 AND Best ITSP - Mid Market, Best Software and Best Vertical Solution are trade marks of the Internet Telephony Services Providers' Association, used under licence. >>> >>> X-on is a trading name of Storacall Technology Ltd a limited company registered in England and Wales. >>> Registered Office : Avaland House, 110 London Road, Apsley, Hemel Hempstead, Herts, HP3 9SD. Company Registration No. 2578478. >>> The information in this e-mail is confidential and for use by the addressee(s) only. If you are not the intended recipient, please notify X-on immediately on +44(0)333 332 0000 and delete the >>> message from your computer. If you are not a named addressee you must not use, disclose, disseminate, distribute, copy, print or reply to this email. Views or opinions expressed by an individual >>> within this email may not necessarily reflect the views of X-on or its associated companies. Although X-on routinely screens for viruses, addressees should scan this email and any attachments >>> for viruses. X-on makes no representation or warranty as to the absence of viruses in this email or any attachments. >>> >>> >>> _______________________________________________ >>> Users mailing list >>> Users at lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >>> From callum.guy at x-on.co.uk Tue Apr 28 10:21:51 2020 From: callum.guy at x-on.co.uk (Callum Guy) Date: Tue, 28 Apr 2020 11:21:51 +0100 Subject: [OpenSIPS-Users] disable_503_translation In-Reply-To: References: <55435248-82fe-e1b7-782a-587ecad53b51@opensips.org> Message-ID: OK will do, I've been hesitant to configure the debugger on a production system in case it caused any performance/stability issues - is that a reason for concern? For now I'll see if I can configure a test setup and see if I can duplicate the problem there. Thanks for your time. On Tue, 28 Apr 2020 at 10:01, Bogdan-Andrei Iancu wrote: > > Hi Callum, > > Maybe a better approach will be try to identify the potential leak as > described here -> > https://opensips.org/Documentation/TroubleShooting-OutOfMem > > Best regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > > On 4/28/20 1:38 AM, Callum Guy wrote: > > Hi Bogdan, > > > > I'm still searching for my memory leak, just downgraded from 3.0.2 to > > 3.0.1 and should be able to confirm if that has made a difference in a > > day or two. > > > > During the hunt I've been reviewing the changelog commits - wondered > > if you could check params var created at the link below: > > > > https://github.com/OpenSIPS/opensips/blob/559921962c69575bcdc90c1f03102e6d5c56d776/modules/dialog/dlg_req_within.c#L489 > > > > My understanding of memory management in C isn't great but it looks to > > be allocating without a free? > > > > Many thanks, > > > > Callum > > > > > > On Fri, 24 Apr 2020 at 17:16, Callum Guy wrote: > >> Thanks Bogdan, I realise it was a long shot! There are only minor differences between this and another system i am running so I'm going through eliminating the config differences. > >> > >> Version wise the stable one is 3.0.1 and the leaky one is 3.0.2 > >> > >> Slow progress but I will get there eventually! > >> > >> On Fri, 24 Apr 2020, 16:50 Bogdan-Andrei Iancu, wrote: > >>> Hi Callum, > >>> > >>> I 99.9999999999999999999% that that setting cannot generate a leak :) > >>> > >>> Best regards, > >>> > >>> Bogdan-Andrei Iancu > >>> > >>> OpenSIPS Founder and Developer > >>> https://www.opensips-solutions.com > >>> > >>> On 4/24/20 6:31 PM, Callum Guy wrote: > >>> > >>> Hi All, > >>> > >>> I've been hunting a minor memory leak in my config and wanted to check in with the devs in case it is related to ues of the parameter: > >>> > >>> disable_503_translation=yes > >>> > >>> Here is the implementation link to save you a few seconds: > >>> > >>> https://github.com/OpenSIPS/opensips/blob/7dd1151341b8229cd30e335b246e56938551f6bd/msg_translator.c#L2425 > >>> > >>> Is there any chance that this is failing to free the tiny amount it's allocating on each use? > >>> > >>> All the best, > >>> > >>> Callum > >>> > >>> > >>> 0333 332 0000 | x-on.co.uk | | Coronavirus > >>> > >>> THE ITSPA AWARDS 2020 AND Best ITSP - Mid Market, Best Software and Best Vertical Solution are trade marks of the Internet Telephony Services Providers' Association, used under licence. > >>> > >>> X-on is a trading name of Storacall Technology Ltd a limited company registered in England and Wales. > >>> Registered Office : Avaland House, 110 London Road, Apsley, Hemel Hempstead, Herts, HP3 9SD. Company Registration No. 2578478. > >>> The information in this e-mail is confidential and for use by the addressee(s) only. If you are not the intended recipient, please notify X-on immediately on +44(0)333 332 0000 and delete the > >>> message from your computer. If you are not a named addressee you must not use, disclose, disseminate, distribute, copy, print or reply to this email. Views or opinions expressed by an individual > >>> within this email may not necessarily reflect the views of X-on or its associated companies. Although X-on routinely screens for viruses, addressees should scan this email and any attachments > >>> for viruses. X-on makes no representation or warranty as to the absence of viruses in this email or any attachments. > >>> > >>> > >>> _______________________________________________ > >>> Users mailing list > >>> Users at lists.opensips.org > >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users > >>> > >>> > -- *0333 332 0000  |  x-on.co.uk   |   **      **  |  Coronavirus * THE ITSPA AWARDS 2020 AND Best ITSP - Mid Market, Best Software and Best Vertical Solution are trade marks of the Internet Telephony Services Providers' Association, used under licence. X-on is a trading name of Storacall Technology Ltd a limited company registered in England and Wales. Registered Office : Avaland House, 110 London Road, Apsley, Hemel Hempstead, Herts, HP3 9SD. Company Registration No. 2578478. The information in this e-mail is confidential and for use by the addressee(s) only. If you are not the intended recipient, please notify X-on immediately on +44(0)333 332 0000 and delete the message from your computer. If you are not a named addressee you must not use, disclose, disseminate, distribute, copy, print or reply to this email. Views or opinions expressed by an individual within this email may not necessarily reflect the views of X-on or its associated companies. Although X-on routinely screens for viruses, addressees should scan this email and any attachments for viruses. X-on makes no representation or warranty as to the absence of viruses in this email or any attachments. From bogdan at opensips.org Tue Apr 28 11:18:42 2020 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Tue, 28 Apr 2020 14:18:42 +0300 Subject: [OpenSIPS-Users] disable_503_translation In-Reply-To: References: <55435248-82fe-e1b7-782a-587ecad53b51@opensips.org> Message-ID: Just be sure you are not logging all the debug ops, but only the dump - in this case you should be safe . Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com On 4/28/20 1:21 PM, Callum Guy wrote: > OK will do, I've been hesitant to configure the debugger on a > production system in case it caused any performance/stability issues - > is that a reason for concern? > > For now I'll see if I can configure a test setup and see if I can > duplicate the problem there. > > Thanks for your time. > > > On Tue, 28 Apr 2020 at 10:01, Bogdan-Andrei Iancu wrote: >> Hi Callum, >> >> Maybe a better approach will be try to identify the potential leak as >> described here -> >> https://opensips.org/Documentation/TroubleShooting-OutOfMem >> >> Best regards, >> >> Bogdan-Andrei Iancu >> >> OpenSIPS Founder and Developer >> https://www.opensips-solutions.com >> >> On 4/28/20 1:38 AM, Callum Guy wrote: >>> Hi Bogdan, >>> >>> I'm still searching for my memory leak, just downgraded from 3.0.2 to >>> 3.0.1 and should be able to confirm if that has made a difference in a >>> day or two. >>> >>> During the hunt I've been reviewing the changelog commits - wondered >>> if you could check params var created at the link below: >>> >>> https://github.com/OpenSIPS/opensips/blob/559921962c69575bcdc90c1f03102e6d5c56d776/modules/dialog/dlg_req_within.c#L489 >>> >>> My understanding of memory management in C isn't great but it looks to >>> be allocating without a free? >>> >>> Many thanks, >>> >>> Callum >>> >>> >>> On Fri, 24 Apr 2020 at 17:16, Callum Guy wrote: >>>> Thanks Bogdan, I realise it was a long shot! There are only minor differences between this and another system i am running so I'm going through eliminating the config differences. >>>> >>>> Version wise the stable one is 3.0.1 and the leaky one is 3.0.2 >>>> >>>> Slow progress but I will get there eventually! >>>> >>>> On Fri, 24 Apr 2020, 16:50 Bogdan-Andrei Iancu, wrote: >>>>> Hi Callum, >>>>> >>>>> I 99.9999999999999999999% that that setting cannot generate a leak :) >>>>> >>>>> Best regards, >>>>> >>>>> Bogdan-Andrei Iancu >>>>> >>>>> OpenSIPS Founder and Developer >>>>> https://www.opensips-solutions.com >>>>> >>>>> On 4/24/20 6:31 PM, Callum Guy wrote: >>>>> >>>>> Hi All, >>>>> >>>>> I've been hunting a minor memory leak in my config and wanted to check in with the devs in case it is related to ues of the parameter: >>>>> >>>>> disable_503_translation=yes >>>>> >>>>> Here is the implementation link to save you a few seconds: >>>>> >>>>> https://github.com/OpenSIPS/opensips/blob/7dd1151341b8229cd30e335b246e56938551f6bd/msg_translator.c#L2425 >>>>> >>>>> Is there any chance that this is failing to free the tiny amount it's allocating on each use? >>>>> >>>>> All the best, >>>>> >>>>> Callum >>>>> >>>>> >>>>> 0333 332 0000 | x-on.co.uk | | Coronavirus >>>>> >>>>> THE ITSPA AWARDS 2020 AND Best ITSP - Mid Market, Best Software and Best Vertical Solution are trade marks of the Internet Telephony Services Providers' Association, used under licence. >>>>> >>>>> X-on is a trading name of Storacall Technology Ltd a limited company registered in England and Wales. >>>>> Registered Office : Avaland House, 110 London Road, Apsley, Hemel Hempstead, Herts, HP3 9SD. Company Registration No. 2578478. >>>>> The information in this e-mail is confidential and for use by the addressee(s) only. If you are not the intended recipient, please notify X-on immediately on +44(0)333 332 0000 and delete the >>>>> message from your computer. If you are not a named addressee you must not use, disclose, disseminate, distribute, copy, print or reply to this email. Views or opinions expressed by an individual >>>>> within this email may not necessarily reflect the views of X-on or its associated companies. Although X-on routinely screens for viruses, addressees should scan this email and any attachments >>>>> for viruses. X-on makes no representation or warranty as to the absence of viruses in this email or any attachments. >>>>> >>>>> >>>>> _______________________________________________ >>>>> Users mailing list >>>>> Users at lists.opensips.org >>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>>> >>>>> From john.quick at smartvox.co.uk Tue Apr 28 13:58:34 2020 From: john.quick at smartvox.co.uk (John Quick) Date: Tue, 28 Apr 2020 14:58:34 +0100 Subject: [OpenSIPS-Users] Problem with add_rr_param() in v2.4.7 Message-ID: <001301d61d65$1b946220$52bd2660$@smartvox.co.uk> Bogdan, Thanks for checking. If possible, I will provide a minimal cfg file to demonstrate the problem. I'm not able to use the original server for this because it belongs to a customer, has been reverted to v2.4.6 and is in constant use. On that server, OpenSIPS was installed using apt-get - i.e. from a binary package. I don't know if that could be relevant. On a different server, I will compile a fresh version of 2.4.7 from source. Then see if I can reproduce the fault and report back here. John Quick Smartvox Limited From Santi.Anton at quarea.com Tue Apr 28 15:00:29 2020 From: Santi.Anton at quarea.com (Santi Anton) Date: Tue, 28 Apr 2020 08:00:29 -0700 (MST) Subject: [OpenSIPS-Users] ms teams ACK In-Reply-To: References: <94a0f640-5400-9124-0a98-96556fa000b6@skillsearch.ca> <6f80948f-fa0a-696f-eb11-2a46e2f49591@skillsearch.ca> <0c16126f-931a-d8a6-5aaa-07a67a8727d1@skillsearch.ca> <1587240834447-0.post@n2.nabble.com> <4e26f956-8513-3863-8ced-67bcac216861@skillsearch.ca> <76b1f17e-21c7-8ccb-d5c9-abe54701d564@skillsearch.ca> Message-ID: <1588086029217-0.post@n2.nabble.com> Hello, Same case here. I'm sending the ACK to MSTeams but after 20 seconds Opensips gets a BYE with the same reason as you. In my scenario both Opensips and Asterisk have public IP addresses and there is no NAT between them. Hope we can find why is MSTeams ignoring the ACK or what are we doing wrong. Regards, -- Sent from: http://opensips-open-sip-server.1449251.n2.nabble.com/OpenSIPS-Users-f1449235.html From bogdan at opensips.org Tue Apr 28 17:23:50 2020 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Tue, 28 Apr 2020 20:23:50 +0300 Subject: [OpenSIPS-Users] [Blog] DFKS or the "Key to Synchronization" Message-ID: <12015047-b8bd-9986-70bb-f04fe32c8cc5@opensips.org> This ability to control Class 5 features from multiple places raises the problem of synchronizing the features' status, as at the end of the day we want to experience an unified behavior of the feature. Even if it is not an IETF standard, an answer to this issue is provided by the Broadsoft's Device Feature Key Synchronization protocol (or shortly, DFKS) https://blog.opensips.org/2020/04/28/dfks-or-the-key-to-synchronization/ Enjoy it, -- Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com From venefax at gmail.com Wed Apr 29 03:17:56 2020 From: venefax at gmail.com (Saint Michael) Date: Tue, 28 Apr 2020 23:17:56 -0400 Subject: [OpenSIPS-Users] CDR capturing not working Message-ID: I checked the database and I missed the CDR for many calls. Later I found them sitting in files in /var/log/opensips I run it manually cdr_worker.sh and got this error ./cdr_worker.sh lockfile: Sorry, giving up on "/var/run/opensips_cdr_worker.lock"which means is not firing because the lock is being held, but there is no traffic. I stopped opensips and still the same error , so the script is not working as it should. The locking mechanism is stuck. I restarted opensips, placed another call, and odbc did not get executed, so recycling opensips does not free the lock. Any idea how can I get my CDR inserted 100% for sure into my database? -------------- next part -------------- An HTML attachment was scrubbed... URL: From nesken at gmail.com Wed Apr 29 05:35:12 2020 From: nesken at gmail.com (Dioris Moreno) Date: Wed, 29 Apr 2020 00:35:12 -0500 Subject: [OpenSIPS-Users] opensips-mi-client Message-ID: I just published to npm opensips-mi-client, a node client library for the OpenSIPS Management Interface 3.0. This library wraps all OpenSIPS 3.0 MI functions in a comprehensive and easy-to-use way. At this moment opensips-mi-client only supports http transport. I hope this library will facilitate the development of applications that require to connect to OpenSIPS MI. https://npmjs.com/package/opensips-mi-client Regards, *Dioris Moreno* *Systems Architect*Libereco Systems dmoreno at libersys.io +507 6150-5756 P Tome en consideración el medio ambiente antes de imprimir este correo electrónico. -------------- next part -------------- An HTML attachment was scrubbed... URL: From tpaivaa at gmail.com Wed Apr 29 05:59:41 2020 From: tpaivaa at gmail.com (Tomi Hakkarainen) Date: Wed, 29 Apr 2020 08:59:41 +0300 Subject: [OpenSIPS-Users] opensips-mi-client In-Reply-To: References: Message-ID: <38387EA2-EAF5-4DF5-B06E-71DD6BDD9B42@gmail.com> Nice! And I really like the doc showing examples of use and responses -> Well done Br, Tomi On 29. Apr 2020, at 8.35, Dioris Moreno wrote: I just published to npm opensips-mi-client, a node client library for the OpenSIPS Management Interface 3.0. This library wraps all OpenSIPS 3.0 MI functions in a comprehensive and easy-to-use way. At this moment opensips-mi-client only supports http transport. I hope this library will facilitate the development of applications that require to connect to OpenSIPS MI. https://npmjs.com/package/opensips-mi-client Regards, Dioris Moreno Systems Architect Libereco Systems dmoreno at libersys.io +507 6150-5756 P Tome en consideración el medio ambiente antes de imprimir este correo electrónico. _______________________________________________ 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 spanda at 3clogic.com Wed Apr 29 07:16:11 2020 From: spanda at 3clogic.com (Sasmita Panda) Date: Wed, 29 Apr 2020 12:46:11 +0530 Subject: [OpenSIPS-Users] Need help to use alias db . Message-ID: Hi All , For the first time I am trying to use alias db . Before processing any request I wanted to do alias db lookup , if find the corresponding domain in the alias db then process the request otherwise not . How will I do this ? *Thanks & Regards* *Sasmita Panda* *Senior Network Testing and Software Engineer* *3CLogic , ph:07827611765* -------------- next part -------------- An HTML attachment was scrubbed... URL: From Santi.Anton at quarea.com Wed Apr 29 08:11:10 2020 From: Santi.Anton at quarea.com (Santi Anton) Date: Wed, 29 Apr 2020 01:11:10 -0700 (MST) Subject: [OpenSIPS-Users] ms teams ACK In-Reply-To: <1588086029217-0.post@n2.nabble.com> References: <6f80948f-fa0a-696f-eb11-2a46e2f49591@skillsearch.ca> <0c16126f-931a-d8a6-5aaa-07a67a8727d1@skillsearch.ca> <1587240834447-0.post@n2.nabble.com> <4e26f956-8513-3863-8ced-67bcac216861@skillsearch.ca> <76b1f17e-21c7-8ccb-d5c9-abe54701d564@skillsearch.ca> <1588086029217-0.post@n2.nabble.com> Message-ID: <1588147870705-0.post@n2.nabble.com> Hello, If it helps, these are the ACK sent to MSTeams and BYE received after 20 seconds. sbc.xxxxxxxx.com:5067 Opensips TLS interface 195.xx.xx.xx:5068 Opensips UDP interface 80.xx.xx.xx:5060 Asterisk ACK sip:sip.pstnhub.microsoft.com;x-i=0c135a71-ddec-454c-a506-91079cb038d7;x-c=385a4453fc885c0bb6276a40d7796d3a/s/1/e1c45d052842415e9c0b3b0b04e32791 SIP/2.0 Via: SIP/2.0/TLS 195.xx.xx.xx:5067;branch=z9hG4bK2f83.ee24a46.2 Via: SIP/2.0/UDP 80.xx.xx.xx:5060;received=80.xx.xx.xx;branch=z9hG4bK0a1d079a;rport=5060 Route: Max-Forwards: 69 From: ;tag=as7659f57d To: ;tag=50fe42a7fdb64fe583c1089a0b47f44a Contact: Call-ID: 433371746cb44b7d76fdc42013bf92b8 at 80.xx.xx.xx:5060:5060 CSeq: 102 ACK User-Agent: FPBX-12.0.76.6(11.20.0) Content-Length: 0 BYE sip:935441221 at 80.xx.xx.xx:5060 SIP/2.0 FROM: ;tag=50fe42a7fdb64fe583c1089a0b47f44a TO: ;tag=as7659f57d CSEQ: 3 BYE CALL-ID: 433371746cb44b7d76fdc42013bf92b8 at 80.xx.xx.xx:5060 MAX-FORWARDS: 69 VIA: SIP/2.0/TLS 52.114.75.24:5061;branch=z9hG4bKf36698cf REASON: Q.850;cause=18;text="0c135a71-ddec-454c-a506-91079cb038d7;*Call Controller timed out while waiting for acknowledgement.*" ROUTE: , CONTACT: CONTENT-LENGTH: 0 USER-AGENT: Microsoft.PSTNHub.SIPProxy v.2020.4.20.2 i.EUWE.6 ALLOW: INVITE,ACK,OPTIONS,CANCEL,BYE,NOTIFY Best Regards, -- Sent from: http://opensips-open-sip-server.1449251.n2.nabble.com/OpenSIPS-Users-f1449235.html From bogdan at opensips.org Wed Apr 29 12:48:25 2020 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Wed, 29 Apr 2020 15:48:25 +0300 Subject: [OpenSIPS-Users] CDR capturing not working In-Reply-To: References: Message-ID: Hi there, What is this cdr_worker.sh file you are talking about ?? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com On 4/29/20 6:17 AM, Saint Michael wrote: > I checked the database and I missed the CDR for many calls.  Later I > found them sitting in files in /var/log/opensips > I run it manually cdr_worker.sh and got this error ./cdr_worker.sh > lockfile: Sorry, giving up on "/var/run/opensips_cdr_worker.lock"which > means is not firing because the lock is being held, but there is no > traffic. I stopped opensips and still the same error , so the script > is not working as it should. The locking mechanism is stuck. I > restarted opensips, placed another call, and odbc did not get > executed, so recycling opensips does not free the lock. > Any idea how can I get my CDR inserted 100% for sure into my database? > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Wed Apr 29 12:51:05 2020 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Wed, 29 Apr 2020 15:51:05 +0300 Subject: [OpenSIPS-Users] Need help to use alias db . In-Reply-To: References: Message-ID: Hi, The aliase_db module does full aliasing, user+domain to another user+domain, not only domains. And you can test the return code of the lookup function to see if any matching and translation was done or not. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com On 4/29/20 10:16 AM, Sasmita Panda wrote: > Hi All , > > For the first time I am trying to use alias db . Before processing any > request I wanted to do alias db lookup , if find the corresponding > domain in the alias db then process the request otherwise not . > > How will I do this ? > > */Thanks & Regards/* > /Sasmita Panda/ > /Senior Network Testing and Software Engineer/ > /3CLogic , ph:07827611765/ > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Wed Apr 29 12:52:39 2020 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Wed, 29 Apr 2020 15:52:39 +0300 Subject: [OpenSIPS-Users] opensips-mi-client In-Reply-To: References: Message-ID: <0a066571-8039-3542-f943-c665ea6a6470@opensips.org> Nice job Dioris, Is it fine with you to link your project from the MI manual page in 3.0, so people will be aware of this option ? Best regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com On 4/29/20 8:35 AM, Dioris Moreno wrote: > I just published to npm opensips-mi-client, a node client library for > the OpenSIPS Management Interface 3.0. > This library wraps all OpenSIPS 3.0 MI functions in a comprehensive > and easy-to-use way. At this moment opensips-mi-client only supports > http transport. > I hope this library will facilitate the development of applications > that require to connect to OpenSIPS MI. > > https://npmjs.com/package/opensips-mi-client > > Regards, > > *Dioris Moreno* > /Systems Architect > /Libereco Systems > dmoreno at libersys.io > +507 6150-5756 > > > > > P  Tome en consideración el medio ambiente antes de imprimir este > correo electrónico. > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From Ben.Newlin at genesys.com Wed Apr 29 13:05:17 2020 From: Ben.Newlin at genesys.com (Ben Newlin) Date: Wed, 29 Apr 2020 13:05:17 +0000 Subject: [OpenSIPS-Users] ms teams ACK In-Reply-To: <1588147870705-0.post@n2.nabble.com> References: <6f80948f-fa0a-696f-eb11-2a46e2f49591@skillsearch.ca> <0c16126f-931a-d8a6-5aaa-07a67a8727d1@skillsearch.ca> <1587240834447-0.post@n2.nabble.com> <4e26f956-8513-3863-8ced-67bcac216861@skillsearch.ca> <76b1f17e-21c7-8ccb-d5c9-abe54701d564@skillsearch.ca> <1588086029217-0.post@n2.nabble.com> <1588147870705-0.post@n2.nabble.com> Message-ID: <6BD54BE3-B931-4997-A774-4A17BE50A925@genesys.com> A BYE ~30s after the ACK usually means the ACK was not accepted or associated with the call. This is usually due to some problem with the ACK. The ACK responses must be built according to specific requirements based on the INVITE and 200 OK values, so it is not possible to say for sure whether there is anything wrong with the ACK with only the ACK and BYE. Ben Newlin On 4/29/20, 4:14 AM, "Users on behalf of Santi Anton" wrote: Hello, If it helps, these are the ACK sent to MSTeams and BYE received after 20 seconds. sbc.xxxxxxxx.com:5067 Opensips TLS interface 195.xx.xx.xx:5068 Opensips UDP interface 80.xx.xx.xx:5060 Asterisk ACK sip:sip.pstnhub.microsoft.com;x-i=0c135a71-ddec-454c-a506-91079cb038d7;x-c=385a4453fc885c0bb6276a40d7796d3a/s/1/e1c45d052842415e9c0b3b0b04e32791 SIP/2.0 Via: SIP/2.0/TLS 195.xx.xx.xx:5067;branch=z9hG4bK2f83.ee24a46.2 Via: SIP/2.0/UDP 80.xx.xx.xx:5060;received=80.xx.xx.xx;branch=z9hG4bK0a1d079a;rport=5060 Route: Max-Forwards: 69 From: ;tag=as7659f57d To: ;tag=50fe42a7fdb64fe583c1089a0b47f44a Contact: Call-ID: 433371746cb44b7d76fdc42013bf92b8 at 80.xx.xx.xx:5060:5060 CSeq: 102 ACK User-Agent: FPBX-12.0.76.6(11.20.0) Content-Length: 0 BYE sip:935441221 at 80.xx.xx.xx:5060 SIP/2.0 FROM: ;tag=50fe42a7fdb64fe583c1089a0b47f44a TO: ;tag=as7659f57d CSEQ: 3 BYE CALL-ID: 433371746cb44b7d76fdc42013bf92b8 at 80.xx.xx.xx:5060 MAX-FORWARDS: 69 VIA: SIP/2.0/TLS 52.114.75.24:5061;branch=z9hG4bKf36698cf REASON: Q.850;cause=18;text="0c135a71-ddec-454c-a506-91079cb038d7;*Call Controller timed out while waiting for acknowledgement.*" ROUTE: , CONTACT: CONTENT-LENGTH: 0 USER-AGENT: Microsoft.PSTNHub.SIPProxy v.2020.4.20.2 i.EUWE.6 ALLOW: INVITE,ACK,OPTIONS,CANCEL,BYE,NOTIFY Best Regards, -- Sent from: http://opensips-open-sip-server.1449251.n2.nabble.com/OpenSIPS-Users-f1449235.html _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users From kevin.vines at gmail.com Wed Apr 29 13:36:02 2020 From: kevin.vines at gmail.com (Kevin Vines) Date: Wed, 29 Apr 2020 09:36:02 -0400 Subject: [OpenSIPS-Users] ms teams ACK In-Reply-To: <1588147870705-0.post@n2.nabble.com> Message-ID: <9l42eb5800khb17q70tt2k45.1588167362939@gmail.com> The call IDs don't match. Your ACK has an extra :5060 appended. Just a thought... Kevin V. +15195726354   Original Message   From: Santi.Anton at quarea.com Sent: April 29, 2020 4:13 a.m. To: users at lists.opensips.org Reply to: users at lists.opensips.org Subject: Re: [OpenSIPS-Users] ms teams ACK Hello, If it helps, these are the ACK sent to MSTeams and BYE received after 20 seconds. sbc.xxxxxxxx.com:5067 Opensips TLS interface 195.xx.xx.xx:5068 Opensips UDP interface 80.xx.xx.xx:5060 Asterisk ACK sip:sip.pstnhub.microsoft.com;x-i=0c135a71-ddec-454c-a506-91079cb038d7;x-c=385a4453fc885c0bb6276a40d7796d3a/s/1/e1c45d052842415e9c0b3b0b04e32791 SIP/2.0 Via: SIP/2.0/TLS 195.xx.xx.xx:5067;branch=z9hG4bK2f83.ee24a46.2 Via: SIP/2.0/UDP 80.xx.xx.xx:5060;received=80.xx.xx.xx;branch=z9hG4bK0a1d079a;rport=5060 Route: Max-Forwards: 69 From: ;tag=as7659f57d To: ;tag=50fe42a7fdb64fe583c1089a0b47f44a Contact: Call-ID: 433371746cb44b7d76fdc42013bf92b8 at 80.xx.xx.xx:5060:5060 CSeq: 102 ACK User-Agent: FPBX-12.0.76.6(11.20.0) Content-Length: 0 BYE sip:935441221 at 80.xx.xx.xx:5060 SIP/2.0 FROM: ;tag=50fe42a7fdb64fe583c1089a0b47f44a TO: ;tag=as7659f57d CSEQ: 3 BYE CALL-ID: 433371746cb44b7d76fdc42013bf92b8 at 80.xx.xx.xx:5060 MAX-FORWARDS: 69 VIA: SIP/2.0/TLS 52.114.75.24:5061;branch=z9hG4bKf36698cf REASON: Q.850;cause=18;text="0c135a71-ddec-454c-a506-91079cb038d7;*Call Controller timed out while waiting for acknowledgement.*" ROUTE: , CONTACT: CONTENT-LENGTH: 0 USER-AGENT: Microsoft.PSTNHub.SIPProxy v.2020.4.20.2 i.EUWE.6 ALLOW: INVITE,ACK,OPTIONS,CANCEL,BYE,NOTIFY Best Regards, -- Sent from: http://opensips-open-sip-server.1449251.n2.nabble.com/OpenSIPS-Users-f1449235.html _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users From Santi.Anton at quarea.com Wed Apr 29 13:59:08 2020 From: Santi.Anton at quarea.com (Santi Anton) Date: Wed, 29 Apr 2020 06:59:08 -0700 (MST) Subject: [OpenSIPS-Users] ms teams ACK In-Reply-To: <6BD54BE3-B931-4997-A774-4A17BE50A925@genesys.com> References: <0c16126f-931a-d8a6-5aaa-07a67a8727d1@skillsearch.ca> <1587240834447-0.post@n2.nabble.com> <4e26f956-8513-3863-8ced-67bcac216861@skillsearch.ca> <76b1f17e-21c7-8ccb-d5c9-abe54701d564@skillsearch.ca> <1588086029217-0.post@n2.nabble.com> <1588147870705-0.post@n2.nabble.com> <6BD54BE3-B931-4997-A774-4A17BE50A925@genesys.com> Message-ID: <1588168748870-0.post@n2.nabble.com> Hello Ben, I agree. These are the INVITE and 200 OK. I only could see that to and from tag an Call-ID are the same. I think that in the article there is nothing about modifying ACK headers to make it work. INVITE sip:5009 at sip.pstnhub.microsoft.com SIP/2.0 Record-Route: Record-Route: Via: SIP/2.0/TLS 195.xx.xx.xx:5067;branch=z9hG4bK2f83.ee24a46.0 Via: SIP/2.0/UDP 80.xx.xx.xx.xx:5060;received=80.xx.xx.xx.xx;branch=z9hG4bK4df6ca5c;rport=5060 Max-Forwards: 69 From: ;tag=as7659f57d To: Contact: Call-ID: 433371746cb44b7d76fdc42013bf92b8 at 80.xx.xx.xx.xx:5060 CSeq: 102 INVITE User-Agent: FPBX-12.0.76.6(11.20.0) Date: Mon, 27 Apr 2020 16:25:24 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 1470 SIP/2.0 200 OK FROM: ;tag=as7659f57d TO: ;tag=50fe42a7fdb64fe583c1089a0b47f44a CSEQ: 102 INVITE CALL-ID: 433371746cb44b7d76fdc42013bf92b8 at 80.xx.xx.xx.xx:5060 VIA: SIP/2.0/TLS 195.xx.xx.xx:5067;branch=z9hG4bK2f83.ee24a46.0,SIP/2.0/UDP 80.xx.xx.xx.xx:5060;received=80.xx.xx.xx.xx;branch=z9hG4bK4df6ca5c;rport=5060 RECORD-ROUTE: ,, CONTACT: CONTENT-LENGTH: 827 SUPPORTED: timer CONTENT-TYPE: application/sdp ALLOW: INVITE,ACK,OPTIONS,CANCEL,BYE,NOTIFY SESSION-EXPIRES: 3600;refresher=uas SERVER: Microsoft.PSTNHub.SIPProxy v.2020.4.20.2 i.EUWE.6 Best Regards, -- Sent from: http://opensips-open-sip-server.1449251.n2.nabble.com/OpenSIPS-Users-f1449235.html From Ben.Newlin at genesys.com Wed Apr 29 14:11:48 2020 From: Ben.Newlin at genesys.com (Ben Newlin) Date: Wed, 29 Apr 2020 14:11:48 +0000 Subject: [OpenSIPS-Users] ms teams ACK In-Reply-To: <1588168748870-0.post@n2.nabble.com> References: <0c16126f-931a-d8a6-5aaa-07a67a8727d1@skillsearch.ca> <1587240834447-0.post@n2.nabble.com> <4e26f956-8513-3863-8ced-67bcac216861@skillsearch.ca> <76b1f17e-21c7-8ccb-d5c9-abe54701d564@skillsearch.ca> <1588086029217-0.post@n2.nabble.com> <1588147870705-0.post@n2.nabble.com> <6BD54BE3-B931-4997-A774-4A17BE50A925@genesys.com> <1588168748870-0.post@n2.nabble.com> Message-ID: I have not read the MS Teams article, but from a SIP perspective your ACK is malformed. The Request URI in the ACK - and any other sequential requests - must be the Contact URI from the 200 OK response. Your ACK Request URI has the parameters from the Contact header, but the domain/host is different. I'm guessing your script may be modifying that value as it passes the ACK through? Also, your ACK does not contain the complete Route set from the 200 OK, it only contains the first entry. Ben Newlin On 4/29/20, 10:00 AM, "Users on behalf of Santi Anton" wrote: Hello Ben, I agree. These are the INVITE and 200 OK. I only could see that to and from tag an Call-ID are the same. I think that in the article there is nothing about modifying ACK headers to make it work. INVITE sip:5009 at sip.pstnhub.microsoft.com SIP/2.0 Record-Route: Record-Route: Via: SIP/2.0/TLS 195.xx.xx.xx:5067;branch=z9hG4bK2f83.ee24a46.0 Via: SIP/2.0/UDP 80.xx.xx.xx.xx:5060;received=80.xx.xx.xx.xx;branch=z9hG4bK4df6ca5c;rport=5060 Max-Forwards: 69 From: ;tag=as7659f57d To: Contact: Call-ID: 433371746cb44b7d76fdc42013bf92b8 at 80.xx.xx.xx.xx:5060 CSeq: 102 INVITE User-Agent: FPBX-12.0.76.6(11.20.0) Date: Mon, 27 Apr 2020 16:25:24 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 1470 SIP/2.0 200 OK FROM: ;tag=as7659f57d TO: ;tag=50fe42a7fdb64fe583c1089a0b47f44a CSEQ: 102 INVITE CALL-ID: 433371746cb44b7d76fdc42013bf92b8 at 80.xx.xx.xx.xx:5060 VIA: SIP/2.0/TLS 195.xx.xx.xx:5067;branch=z9hG4bK2f83.ee24a46.0,SIP/2.0/UDP 80.xx.xx.xx.xx:5060;received=80.xx.xx.xx.xx;branch=z9hG4bK4df6ca5c;rport=5060 RECORD-ROUTE: ,, CONTACT: CONTENT-LENGTH: 827 SUPPORTED: timer CONTENT-TYPE: application/sdp ALLOW: INVITE,ACK,OPTIONS,CANCEL,BYE,NOTIFY SESSION-EXPIRES: 3600;refresher=uas SERVER: Microsoft.PSTNHub.SIPProxy v.2020.4.20.2 i.EUWE.6 Best Regards, -- Sent from: http://opensips-open-sip-server.1449251.n2.nabble.com/OpenSIPS-Users-f1449235.html _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users From volga629 at networklab.ca Wed Apr 29 14:17:02 2020 From: volga629 at networklab.ca (volga629) Date: Wed, 29 Apr 2020 11:17:02 -0300 Subject: [OpenSIPS-Users] ms teams ACK In-Reply-To: <1588168748870-0.post@n2.nabble.com> References: <0c16126f-931a-d8a6-5aaa-07a67a8727d1@skillsearch.ca> <1587240834447-0.post@n2.nabble.com> <4e26f956-8513-3863-8ced-67bcac216861@skillsearch.ca> <76b1f17e-21c7-8ccb-d5c9-abe54701d564@skillsearch.ca> <1588086029217-0.post@n2.nabble.com> <1588147870705-0.post@n2.nabble.com> <6BD54BE3-B931-4997-A774-4A17BE50A925@genesys.com> <1588168748870-0.post@n2.nabble.com> Message-ID: <8be2f9bf-2dea-7429-6a92-f4d6be46ce9d@skillsearch.ca> Try check you preset route. volga629 On 4/29/20 10:59 AM, Santi Anton wrote: > Hello Ben, > > I agree. These are the INVITE and 200 OK. I only could see that to and from > tag an Call-ID are the same. I think that in the article there is nothing > about modifying ACK headers to make it work. > > > INVITE sip:5009 at sip.pstnhub.microsoft.com SIP/2.0 > Record-Route: > > Record-Route: > Via: SIP/2.0/TLS 195.xx.xx.xx:5067;branch=z9hG4bK2f83.ee24a46.0 > Via: SIP/2.0/UDP > 80.xx.xx.xx.xx:5060;received=80.xx.xx.xx.xx;branch=z9hG4bK4df6ca5c;rport=5060 > Max-Forwards: 69 > From: ;tag=as7659f57d > To: > Contact: > Call-ID: 433371746cb44b7d76fdc42013bf92b8 at 80.xx.xx.xx.xx:5060 > CSeq: 102 INVITE > User-Agent: FPBX-12.0.76.6(11.20.0) > Date: Mon, 27 Apr 2020 16:25:24 GMT > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, > PUBLISH, MESSAGE > Supported: replaces, timer > Content-Type: application/sdp > Content-Length: 1470 > > > SIP/2.0 200 OK > FROM: ;tag=as7659f57d > TO: ;tag=50fe42a7fdb64fe583c1089a0b47f44a > CSEQ: 102 INVITE > CALL-ID: 433371746cb44b7d76fdc42013bf92b8 at 80.xx.xx.xx.xx:5060 > VIA: SIP/2.0/TLS 195.xx.xx.xx:5067;branch=z9hG4bK2f83.ee24a46.0,SIP/2.0/UDP > 80.xx.xx.xx.xx:5060;received=80.xx.xx.xx.xx;branch=z9hG4bK4df6ca5c;rport=5060 > RECORD-ROUTE: > ,, > CONTACT: > > CONTENT-LENGTH: 827 > SUPPORTED: timer > CONTENT-TYPE: application/sdp > ALLOW: INVITE,ACK,OPTIONS,CANCEL,BYE,NOTIFY > SESSION-EXPIRES: 3600;refresher=uas > SERVER: Microsoft.PSTNHub.SIPProxy v.2020.4.20.2 i.EUWE.6 > > Best Regards, > > > > -- > Sent from: http://opensips-open-sip-server.1449251.n2.nabble.com/OpenSIPS-Users-f1449235.html > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users From Santi.Anton at quarea.com Wed Apr 29 14:30:36 2020 From: Santi.Anton at quarea.com (Santi Anton) Date: Wed, 29 Apr 2020 07:30:36 -0700 (MST) Subject: [OpenSIPS-Users] ms teams ACK In-Reply-To: <9l42eb5800khb17q70tt2k45.1588167362939@gmail.com> References: <0c16126f-931a-d8a6-5aaa-07a67a8727d1@skillsearch.ca> <1587240834447-0.post@n2.nabble.com> <4e26f956-8513-3863-8ced-67bcac216861@skillsearch.ca> <76b1f17e-21c7-8ccb-d5c9-abe54701d564@skillsearch.ca> <1588086029217-0.post@n2.nabble.com> <1588147870705-0.post@n2.nabble.com> <9l42eb5800khb17q70tt2k45.1588167362939@gmail.com> Message-ID: <1588170636087-0.post@n2.nabble.com> Hi Kevin, Thanks, in that case it has been a copy/paste error. I checked it again and all Call-IDs are exactly the same. -- Sent from: http://opensips-open-sip-server.1449251.n2.nabble.com/OpenSIPS-Users-f1449235.html From Santi.Anton at quarea.com Wed Apr 29 14:49:52 2020 From: Santi.Anton at quarea.com (Santi Anton) Date: Wed, 29 Apr 2020 07:49:52 -0700 (MST) Subject: [OpenSIPS-Users] ms teams ACK In-Reply-To: References: <1587240834447-0.post@n2.nabble.com> <4e26f956-8513-3863-8ced-67bcac216861@skillsearch.ca> <76b1f17e-21c7-8ccb-d5c9-abe54701d564@skillsearch.ca> <1588086029217-0.post@n2.nabble.com> <1588147870705-0.post@n2.nabble.com> <6BD54BE3-B931-4997-A774-4A17BE50A925@genesys.com> <1588168748870-0.post@n2.nabble.com> Message-ID: <1588171792626-0.post@n2.nabble.com> Hi Ben, I found the mistake. As you said, the ACK Request URI was different from 200 OK contact header, checking my script I noticed that ACK was being routed by dynamic routing module changing its Request URI. Thanks for the hint. -- Sent from: http://opensips-open-sip-server.1449251.n2.nabble.com/OpenSIPS-Users-f1449235.html From james at ip-sentinel.com Wed Apr 29 15:00:38 2020 From: james at ip-sentinel.com (James Hogbin) Date: Wed, 29 Apr 2020 15:00:38 +0000 Subject: [OpenSIPS-Users] ms teams ACK In-Reply-To: <1588171792626-0.post@n2.nabble.com> References: <1587240834447-0.post@n2.nabble.com> <4e26f956-8513-3863-8ced-67bcac216861@skillsearch.ca> <76b1f17e-21c7-8ccb-d5c9-abe54701d564@skillsearch.ca> <1588086029217-0.post@n2.nabble.com> <1588147870705-0.post@n2.nabble.com> <6BD54BE3-B931-4997-A774-4A17BE50A925@genesys.com> <1588168748870-0.post@n2.nabble.com> <1588171792626-0.post@n2.nabble.com> Message-ID: <21CE11F1-3FE1-48AA-A208-68B0EFC21BB1@ip-sentinel.com> Precisely my issue. Do you have an example opensips.conf you are could share? James James Hogbin Director IP Sentinel t. +44 (0)20 3011 4150 m. +44 7786910895 w. https://www.ip-sentinel.com > On 29 Apr 2020, at 15:49, Santi Anton wrote: > > Hi Ben, > > I found the mistake. As you said, the ACK Request URI was different from 200 > OK contact header, checking my script I noticed that ACK was being routed by > dynamic routing module changing its Request URI. > > Thanks for the hint. > > > > -- > Sent from: http://opensips-open-sip-server.1449251.n2.nabble.com/OpenSIPS-Users-f1449235.html > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users IP Sentinel Disclaimer This communication is for the information of the person to whom it has been delivered and neither it nor any of its contents should be passed on to or used by any other person. IP Sentinel Ltd is a limited company registered in England and Wales under Registered Number 08648097. Registered Office: Newnhams Wood, Horsted Keynes, West Sussex, RH17 7BT. Disclaimer: Q3dhRSrm_disclaimer -------------- next part -------------- An HTML attachment was scrubbed... URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: 120042916004202563.png Type: image/png Size: 3317 bytes Desc: not available URL: From Santi.Anton at quarea.com Wed Apr 29 15:51:56 2020 From: Santi.Anton at quarea.com (Santi Anton) Date: Wed, 29 Apr 2020 08:51:56 -0700 (MST) Subject: [OpenSIPS-Users] ms teams ACK In-Reply-To: <21CE11F1-3FE1-48AA-A208-68B0EFC21BB1@ip-sentinel.com> References: <76b1f17e-21c7-8ccb-d5c9-abe54701d564@skillsearch.ca> <1588086029217-0.post@n2.nabble.com> <1588147870705-0.post@n2.nabble.com> <6BD54BE3-B931-4997-A774-4A17BE50A925@genesys.com> <1588168748870-0.post@n2.nabble.com> <1588171792626-0.post@n2.nabble.com> <21CE11F1-3FE1-48AA-A208-68B0EFC21BB1@ip-sentinel.com> Message-ID: <1588175516087-0.post@n2.nabble.com> Hi James, Check where you are calling the do_routing function (if you are using dynamic routing module) and be sure that only INVITEs are routed by this function. My mistake was that ACK (and lately I noticed that BYEs too) was being routed by do_routing changing the Request URI. -- Sent from: http://opensips-open-sip-server.1449251.n2.nabble.com/OpenSIPS-Users-f1449235.html From alex.a at gtanetworkconsulting.com Wed Apr 29 18:38:42 2020 From: alex.a at gtanetworkconsulting.com (Alex A) Date: Wed, 29 Apr 2020 14:38:42 -0400 Subject: [OpenSIPS-Users] Drouting failover by carrier only Message-ID: <171c73bdfcf.f647ce1b105300.4293922564087136871@gtanetworkconsulting.com> Hi Everyone, Is it possible to failover to next carrier (instead next gateway) while using drouting? I got the below to work; however currently, use_next_gw gets the next gateway in the list, so if gwlist= #0,#3  and one of the carriers has multiple gateway IPs, the retry happens many times to the same carrier: route[droute] {         xlog("DRoute GATEWAY: source:$si - $fU -> $rU\n");         if (!do_routing(0,"F")) {                 xlog("DRoute GATEWAY: Failed. source:$si -$fU -> $rU\n");                 send_reply(500, "No Gateway to Route found");                 exit;         }         route(relay);         exit;         } route[relay] {         if (is_method("INVITE")) {                 t_on_failure("missed_call");         }         if (!t_relay()) {                 if (use_next_gw()) {                       xlog("L_INFO","Next Gateway: From=$fu, To=$tu,RU=$ru, CI=$ci IP=$si\n");                       t_on_failure("missed_call");                       route(relay);                       exit;                 }                 else {                         send_reply(503, "Service not available, no more gws");                         exit;                 }         };         exit; } failure_route[missed_call] {                 #if (use_next_gw(, , $var(carrier_attrs))) {                 if (use_next_gw(, $avp(gw_id),$avp(carrier_id))) {                       xlog("MissedCall--Next Gateway: From=$fu, To=$tu,RU=$ru, CI=$ci IP=$si\n");                       xlog("Carrier attributes of current gateway: $avp(gw_id). carrier: $avp(carrier_id)\n");                       t_on_failure("missed_call");                       route(relay);                       exit;                 }                 else {                         send_reply(503, "Service not available, no more gws");                         exit;                 } Thank you. -------------- next part -------------- An HTML attachment was scrubbed... URL: From Ben.Newlin at genesys.com Wed Apr 29 18:50:38 2020 From: Ben.Newlin at genesys.com (Ben Newlin) Date: Wed, 29 Apr 2020 18:50:38 +0000 Subject: [OpenSIPS-Users] Drouting failover by carrier only In-Reply-To: <171c73bdfcf.f647ce1b105300.4293922564087136871@gtanetworkconsulting.com> References: <171c73bdfcf.f647ce1b105300.4293922564087136871@gtanetworkconsulting.com> Message-ID: We also had a need to do this and did not find a way to do it via drouting directly, but it will work with drouting with a little help. First, we specify the carrier_id_avp param [1]. After we call do_routing, we copy out the carrier IDs from that avp into our own AVP. Then we use route_to_carrier [2] for each carrier ID in the list. To continue routing within the carrier, you can still do use_next_gw. When that returns false (no more gateways), or if you want to skip to the next carrier you just call route_to_carrier again with the next ID in your list. This solved two problem for us: * allows us to fail over by carrier instead of just by gateway * allows us to call do_routing multiple times for a call with different groups and aggregate the results [1] - https://opensips.org/docs/modules/3.0.x/drouting.html#param_carrier_id_avp [2] - https://opensips.org/docs/modules/3.0.x/drouting.html#func_route_to_carrier Ben Newlin From: Users on behalf of Alex A Reply-To: OpenSIPS users mailling list Date: Wednesday, April 29, 2020 at 2:40 PM To: users Subject: [OpenSIPS-Users] Drouting failover by carrier only Hi Everyone, Is it possible to failover to next carrier (instead next gateway) while using drouting? I got the below to work; however currently, use_next_gw gets the next gateway in the list, so if gwlist= #0,#3 and one of the carriers has multiple gateway IPs, the retry happens many times to the same carrier: route[droute] { xlog("DRoute GATEWAY: source:$si - $fU -> $rU\n"); if (!do_routing(0,"F")) { xlog("DRoute GATEWAY: Failed. source:$si -$fU -> $rU\n"); send_reply(500, "No Gateway to Route found"); exit; } route(relay); exit; } route[relay] { if (is_method("INVITE")) { t_on_failure("missed_call"); } if (!t_relay()) { if (use_next_gw()) { xlog("L_INFO","Next Gateway: From=$fu, To=$tu,RU=$ru, CI=$ci IP=$si\n"); t_on_failure("missed_call"); route(relay); exit; } else { send_reply(503, "Service not available, no more gws"); exit; } }; exit; } failure_route[missed_call] { #if (use_next_gw(, , $var(carrier_attrs))) { if (use_next_gw(, $avp(gw_id),$avp(carrier_id))) { xlog("MissedCall--Next Gateway: From=$fu, To=$tu,RU=$ru, CI=$ci IP=$si\n"); xlog("Carrier attributes of current gateway: $avp(gw_id). carrier: $avp(carrier_id)\n"); t_on_failure("missed_call"); route(relay); exit; } else { send_reply(503, "Service not available, no more gws"); exit; } Thank you. -------------- next part -------------- An HTML attachment was scrubbed... URL: From williamj at exetel.com.au Thu Apr 30 00:00:17 2020 From: williamj at exetel.com.au (William Jin) Date: Thu, 30 Apr 2020 00:00:17 +0000 Subject: [OpenSIPS-Users] opensips 3.0.2 100% CPU after enable tls? In-Reply-To: <672fb9bb-b06f-0772-8d23-eb861027b118@opensips.org> References: <18b56215-a2ed-6cd2-8730-de499ee521d5@opensips.org> <4bb55ace-480e-4e01-6953-ee296372d426@opensips.org> <98c1752c-3a4f-0b1e-55e1-05b0cbdba44a@opensips.org> , <672fb9bb-b06f-0772-8d23-eb861027b118@opensips.org> Message-ID: ok, thanks. I can confirm the nightly build works without any issue. -- Regards, William Jin ________________________________ From: Bogdan-Andrei Iancu Sent: Tuesday, 28 April 2020 5:19 PM To: William Jin ; OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] opensips 3.0.2 100% CPU after enable tls? Similarly, try the nightly build Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com On 4/28/20 1:28 AM, William Jin wrote: Ok, I will try that and let you know the result And BTW, does the 2.4.7 (release) also has this issue? -- Regards, William Jin ________________________________ From: Bogdan-Andrei Iancu Sent: Monday, 27 April 2020 7:10 PM To: William Jin ; OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] opensips 3.0.2 100% CPU after enable tls? Hi William, Please use the nightly builds for 3.0 - there is a fix which didn't make it into the release package. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com On 4/25/20 12:12 AM, William Jin wrote: It's the release. -- Regards, William Jin ________________________________ From: Bogdan-Andrei Iancu Sent: Friday, 24 April 2020 6:14 PM To: William Jin ; OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] opensips 3.0.2 100% CPU after enable tls? And it is the release or nightly build ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com On 4/24/20 1:22 AM, William Jin wrote: I am using the debian apt repo, not from git. -- Regards, William Jin ________________________________ From: Bogdan-Andrei Iancu Sent: Thursday, 23 April 2020 6:49 PM To: OpenSIPS users mailling list ; William Jin Subject: Re: [OpenSIPS-Users] opensips 3.0.2 100% CPU after enable tls? Hi William, What GIT revision of OpenSIPS do you use? (this is exposed by the "opensips -V") Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com On 4/23/20 7:04 AM, William Jin wrote: Hi, Linux platform: Debian 9 (stretch) opensips -V version: opensips 3.0.2 (x86_64/linux) flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, Q_MALLOC, F_MALLOC, HP_MALLOC, DBG_MALLOC, FAST_LOCK-ADAPTIVE_WAIT ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535 poll method support: poll, epoll, sigio_rt, select. main.c compiled on with gcc 6.3.0 related config: listen = tls:xxx.xxx.xxx.xxx:5061 anycast ####TLS loadmodule "tls_mgm.so" loadmodule "proto_tls.so" modparam("tls_mgm", "server_domain", "sip1") modparam("tls_mgm", "match_ip_address", "[sip1]xx.xx.xx.xx:5061") modparam("tls_mgm", "match_sip_domain", "[sip1]xxx.xxx.example.com") modparam("tls_mgm", "verify_cert", "[sip1]1") modparam("tls_mgm", "require_cert", "[sip1]0") modparam("tls_mgm", "tls_method", "[sip1]SSLv23") modparam("tls_mgm", "ciphers_list", "[sip1]AES256-GCM-SHA384,AES256-SHA256,AES256-SHA,CAMELLIA256-SHA,AES128-SHA,SEED-SHA,CAMELLIA128-SHA,RC4-SHA,DES-CBC3-SHA") modparam("tls_mgm", "certificate", "[sip1]/etc/opensips/tls/mycerts/selfsignedcert.pem") modparam("tls_mgm", "private_key", "[sip1]/etc/opensips/tls/mycerts/unsecuredkey.pem") opensips-cli -x trap {pid} result attached Can someone shed some light on it? Thanks. -- Regards, William Jin _______________________________________________ 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 spanda at 3clogic.com Thu Apr 30 06:33:22 2020 From: spanda at 3clogic.com (Sasmita Panda) Date: Thu, 30 Apr 2020 12:03:22 +0530 Subject: [OpenSIPS-Users] Need help to use alias db . In-Reply-To: References: Message-ID: Ok . But username cant be fixed . That can be anything . When I am adding aliase in the config file that only matches the domain and process the request . Isn't the aliase_db do the same thing ? *Thanks & Regards* *Sasmita Panda* *Senior Network Testing and Software Engineer* *3CLogic , ph:07827611765* On Wed, Apr 29, 2020 at 6:21 PM Bogdan-Andrei Iancu wrote: > Hi, > > The aliase_db module does full aliasing, user+domain to another > user+domain, not only domains. And you can test the return code of the > lookup function to see if any matching and translation was done or not. > > Regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > > On 4/29/20 10:16 AM, Sasmita Panda wrote: > > Hi All , > > For the first time I am trying to use alias db . Before processing any > request I wanted to do alias db lookup , if find the corresponding domain > in the alias db then process the request otherwise not . > > How will I do this ? > > *Thanks & Regards* > *Sasmita Panda* > *Senior Network Testing and Software Engineer* > *3CLogic , ph:07827611765* > > _______________________________________________ > Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Thu Apr 30 07:26:00 2020 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Thu, 30 Apr 2020 10:26:00 +0300 Subject: [OpenSIPS-Users] Need help to use alias db . In-Reply-To: References: Message-ID: <25fc2c22-39a8-4e75-386f-cd61fc4435bb@opensips.org> Hi Sasmita, To be honest I do not understand what you want to achieve. Could you provide an example of aliasing you want to have ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com On 4/30/20 9:33 AM, Sasmita Panda wrote: > Ok . But username cant be fixed . That can be anything . > > When I am adding aliase in the config file that only matches the > domain and process the request . Isn't the aliase_db do the same thing ? > > > */Thanks & Regards/* > /Sasmita Panda/ > /Senior Network Testing and Software Engineer/ > /3CLogic , ph:07827611765/ > > > On Wed, Apr 29, 2020 at 6:21 PM Bogdan-Andrei Iancu > > wrote: > > Hi, > > The aliase_db module does full aliasing, user+domain to another > user+domain, not only domains. And you can test the return code of > the lookup function to see if any matching and translation was > done or not. > > Regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > > On 4/29/20 10:16 AM, Sasmita Panda wrote: >> Hi All , >> >> For the first time I am trying to use alias db . Before >> processing any request I wanted to do alias db lookup , if find >> the corresponding domain in the alias db then process the >> request otherwise not . >> >> How will I do this ? >> >> */Thanks & Regards/* >> /Sasmita Panda/ >> /Senior Network Testing and Software Engineer/ >> /3CLogic , ph:07827611765/ >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Thu Apr 30 07:35:33 2020 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Thu, 30 Apr 2020 10:35:33 +0300 Subject: [OpenSIPS-Users] opensips 3.0.2 100% CPU after enable tls? In-Reply-To: References: <18b56215-a2ed-6cd2-8730-de499ee521d5@opensips.org> <4bb55ace-480e-4e01-6953-ee296372d426@opensips.org> <98c1752c-3a4f-0b1e-55e1-05b0cbdba44a@opensips.org> <672fb9bb-b06f-0772-8d23-eb861027b118@opensips.org> Message-ID: Thanks for the confirmation, we will soon get new minor releases out and get the release packages also fixed. Best regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com On 4/30/20 3:00 AM, William Jin wrote: > ok, thanks. > > I can confirm the nightly build works without any issue. > > -- > Regards, > William Jin > ------------------------------------------------------------------------ > *From:* Bogdan-Andrei Iancu > *Sent:* Tuesday, 28 April 2020 5:19 PM > *To:* William Jin ; OpenSIPS users mailling > list > *Subject:* Re: [OpenSIPS-Users] opensips 3.0.2 100% CPU after enable tls? > Similarly, try the nightly build > > Regards, > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > > On 4/28/20 1:28 AM, William Jin wrote: >> Ok, I will try that and let you know the result >> And BTW, does the 2.4.7 (release) also has this issue? >> >> >> -- >> Regards, >> William Jin >> ------------------------------------------------------------------------ >> *From:* Bogdan-Andrei Iancu >> >> *Sent:* Monday, 27 April 2020 7:10 PM >> *To:* William Jin >> ; OpenSIPS users mailling list >> >> *Subject:* Re: [OpenSIPS-Users] opensips 3.0.2 100% CPU after enable >> tls? >> Hi William, >> >> Please use the nightly builds for 3.0 - there is a fix which didn't >> make it into the release package. >> >> Regards, >> Bogdan-Andrei Iancu >> >> OpenSIPS Founder and Developer >> https://www.opensips-solutions.com >> >> On 4/25/20 12:12 AM, William Jin wrote: >>> It's the release. >>> >>> -- >>> Regards, >>> William Jin >>> ------------------------------------------------------------------------ >>> *From:* Bogdan-Andrei Iancu >>> >>> *Sent:* Friday, 24 April 2020 6:14 PM >>> *To:* William Jin >>> ; OpenSIPS users mailling list >>> >>> *Subject:* Re: [OpenSIPS-Users] opensips 3.0.2 100% CPU after enable >>> tls? >>> And it is the release or nightly build ? >>> >>> Regards, >>> Bogdan-Andrei Iancu >>> >>> OpenSIPS Founder and Developer >>> https://www.opensips-solutions.com >>> >>> On 4/24/20 1:22 AM, William Jin wrote: >>>> I am using the debian apt repo, not from git. >>>> >>>> >>>> -- >>>> Regards, >>>> William Jin >>>> ------------------------------------------------------------------------ >>>> *From:* Bogdan-Andrei Iancu >>>> >>>> *Sent:* Thursday, 23 April 2020 6:49 PM >>>> *To:* OpenSIPS users mailling list >>>> ; William Jin >>>> >>>> *Subject:* Re: [OpenSIPS-Users] opensips 3.0.2 100% CPU after >>>> enable tls? >>>> Hi William, >>>> >>>> What GIT revision of OpenSIPS do you use? (this is exposed by the >>>> "opensips -V") >>>> >>>> Regards, >>>> Bogdan-Andrei Iancu >>>> >>>> OpenSIPS Founder and Developer >>>> https://www.opensips-solutions.com >>>> >>>> On 4/23/20 7:04 AM, William Jin wrote: >>>>> Hi, >>>>> >>>>> Linux platform: Debian 9 (stretch) >>>>> >>>>> opensips -V >>>>> version: opensips 3.0.2 (x86_64/linux) >>>>> flags: STATS: On, DISABLE_NAGLE, USE_MCAST, SHM_MMAP, PKG_MALLOC, >>>>> Q_MALLOC, F_MALLOC, HP_MALLOC, DBG_MALLOC, FAST_LOCK-ADAPTIVE_WAIT >>>>> ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN >>>>> 16, MAX_URI_SIZE 1024, BUF_SIZE 65535 >>>>> poll method support: poll, epoll, sigio_rt, select. >>>>> main.c compiled on  with gcc 6.3.0 >>>>> >>>>> related config: >>>>> >>>>> listen = tls:xxx.xxx.xxx.xxx:5061 anycast >>>>> >>>>> >>>>> ####TLS >>>>> loadmodule "tls_mgm.so" >>>>> loadmodule "proto_tls.so" >>>>> >>>>> modparam("tls_mgm", "server_domain", "sip1") >>>>> modparam("tls_mgm", "match_ip_address", "[sip1]xx.xx.xx.xx:5061") >>>>> modparam("tls_mgm", "match_sip_domain", "[sip1]xxx.xxx.example.com") >>>>> >>>>> modparam("tls_mgm", "verify_cert", "[sip1]1") >>>>> modparam("tls_mgm", "require_cert", "[sip1]0") >>>>> modparam("tls_mgm", "tls_method", "[sip1]SSLv23") >>>>> modparam("tls_mgm", "ciphers_list", >>>>> "[sip1]AES256-GCM-SHA384,AES256-SHA256,AES256-SHA,CAMELLIA256-SHA,AES128-SHA,SEED-SHA,CAMELLIA128-SHA,RC4-SHA,DES-CBC3-SHA") >>>>> >>>>> >>>>> modparam("tls_mgm", "certificate", >>>>> "[sip1]/etc/opensips/tls/mycerts/selfsignedcert.pem") >>>>> modparam("tls_mgm", "private_key", >>>>> "[sip1]/etc/opensips/tls/mycerts/unsecuredkey.pem") >>>>> >>>>> >>>>> opensips-cli -x trap {pid} result attached >>>>> >>>>> Can someone shed some light on it? Thanks. >>>>> >>>>> >>>>> -- >>>>> Regards, >>>>> William Jin >>>>> >>>>> _______________________________________________ >>>>> Users mailing list >>>>> Users at lists.opensips.org >>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>> >>> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Thu Apr 30 07:40:21 2020 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Thu, 30 Apr 2020 10:40:21 +0300 Subject: [OpenSIPS-Users] Drouting failover by carrier only In-Reply-To: <171c73bdfcf.f647ce1b105300.4293922564087136871@gtanetworkconsulting.com> References: <171c73bdfcf.f647ce1b105300.4293922564087136871@gtanetworkconsulting.com> Message-ID: <1a2f5cf6-8ab9-d33f-dece-57bbfdb505da@opensips.org> Hi Alex, Have you tried the "use only the first GW" flag in the carrier definition? See https://opensips.org/html/docs/modules/3.0.x/drouting.html#idp25339408, "Carriers" subsection: flags : 0x1 - use weight for sorting the list and not definition order; 0x2 - use only the first gateway from the carrier (depending on the sorting); 0x4 - disable the usage of this carrier Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com On 4/29/20 9:38 PM, Alex A wrote: > Hi Everyone, > > Is it possible to failover to next carrier (instead next gateway) > while using drouting? > > > I got the below to work; however currently, use_next_gw gets the next > gateway in the list, so > if gwlist= #0,#3 > > and one of the carriers has multiple gateway IPs, the retry happens > many times to the same carrier: > > > route[droute] { > >         xlog("DRoute GATEWAY: source:$si - $fU -> $rU\n"); >         if (!do_routing(0,"F")) { >                 xlog("DRoute GATEWAY: Failed. source:$si -$fU -> $rU\n"); >                 send_reply(500, "No Gateway to Route found"); >                 exit; >         } >         route(relay); >         exit; > >         } > > route[relay] { > >         if (is_method("INVITE")) { >                 t_on_failure("missed_call"); >         } > >         if (!t_relay()) { >                 if (use_next_gw()) { >                       xlog("L_INFO","Next Gateway: From=$fu, > To=$tu,RU=$ru, CI=$ci IP=$si\n"); >                       t_on_failure("missed_call"); >                       route(relay); >                       exit; >                 } >                 else { >                         send_reply(503, "Service not available, no > more gws"); >                         exit; >                 } >         }; >         exit; > } > > > failure_route[missed_call] { > >                 #if (use_next_gw(, , $var(carrier_attrs))) { >                 if (use_next_gw(, $avp(gw_id),$avp(carrier_id))) { >                       xlog("MissedCall--Next Gateway: From=$fu, > To=$tu,RU=$ru, CI=$ci IP=$si\n"); >                       xlog("Carrier attributes of current gateway: > $avp(gw_id). carrier: $avp(carrier_id)\n"); > >                       t_on_failure("missed_call"); >                       route(relay); >                       exit; >                 } >                 else { >                         send_reply(503, "Service not available, no > more gws"); >                         exit; >                 } > > > > > 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 spanda at 3clogic.com Thu Apr 30 08:08:40 2020 From: spanda at 3clogic.com (Sasmita Panda) Date: Thu, 30 Apr 2020 13:38:40 +0530 Subject: [OpenSIPS-Users] Need help to use alias db . In-Reply-To: <25fc2c22-39a8-4e75-386f-cd61fc4435bb@opensips.org> References: <25fc2c22-39a8-4e75-386f-cd61fc4435bb@opensips.org> Message-ID: I have one opensips server , and I have mapped 2 domain into that . In the config file I have added the alias list for that domain . for example : alias="freeswitch-registrar.xyz.com" alias="freeswitch-cisco.xyz.com" route{ if (!uri==myself) { send_reply("403","Rely forbidden"); exit; } ...................... } If any request coming to this has the request uri matches with the alias added then that request get processed else it gives Error message . Now if I am mapping another domain to the server but I dont want to change the config and restart the service again . I want any other way to do dynamically so that whenever I will add any domain opensips will read that and process the request without service restart . How will I achive this ? *Thanks & Regards* *Sasmita Panda* *Senior Network Testing and Software Engineer* *3CLogic , ph:07827611765* On Thu, Apr 30, 2020 at 12:56 PM Bogdan-Andrei Iancu wrote: > Hi Sasmita, > > To be honest I do not understand what you want to achieve. Could you > provide an example of aliasing you want to have ? > > Regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > > On 4/30/20 9:33 AM, Sasmita Panda wrote: > > Ok . But username cant be fixed . That can be anything . > > When I am adding aliase in the config file that only matches the domain > and process the request . Isn't the aliase_db do the same thing ? > > > *Thanks & Regards* > *Sasmita Panda* > *Senior Network Testing and Software Engineer* > *3CLogic , ph:07827611765* > > > On Wed, Apr 29, 2020 at 6:21 PM Bogdan-Andrei Iancu > wrote: > >> Hi, >> >> The aliase_db module does full aliasing, user+domain to another >> user+domain, not only domains. And you can test the return code of the >> lookup function to see if any matching and translation was done or not. >> >> Regards, >> >> Bogdan-Andrei Iancu >> >> OpenSIPS Founder and Developer >> https://www.opensips-solutions.com >> >> On 4/29/20 10:16 AM, Sasmita Panda wrote: >> >> Hi All , >> >> For the first time I am trying to use alias db . Before processing any >> request I wanted to do alias db lookup , if find the corresponding domain >> in the alias db then process the request otherwise not . >> >> How will I do this ? >> >> *Thanks & Regards* >> *Sasmita Panda* >> *Senior Network Testing and Software Engineer* >> *3CLogic , ph:07827611765* >> >> _______________________________________________ >> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: From gmaruzz at gmail.com Thu Apr 30 08:14:58 2020 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Thu, 30 Apr 2020 10:14:58 +0200 Subject: [OpenSIPS-Users] Need help to use alias db . In-Reply-To: References: <25fc2c22-39a8-4e75-386f-cd61fc4435bb@opensips.org> Message-ID: you reload the domain table via MI interface each time you change it -giovanni On Thu, Apr 30, 2020 at 10:11 AM Sasmita Panda wrote: > I have one opensips server , and I have mapped 2 domain into that . In the > config file I have added the alias list for that domain . > for example : > alias="freeswitch-registrar.xyz.com" > alias="freeswitch-cisco.xyz.com" > > route{ > if (!uri==myself) { > send_reply("403","Rely forbidden"); > exit; > } > ...................... > } > If any request coming to this has the request uri matches with the alias > added then that request get processed else it gives Error message . > > Now if I am mapping another domain to the server but I dont want to change > the config and restart the service again . I want any other way to do > dynamically so that whenever I will add any domain opensips will read that > and process the request without service restart . > > > How will I achive this ? > > > *Thanks & Regards* > *Sasmita Panda* > *Senior Network Testing and Software Engineer* > *3CLogic , ph:07827611765* > > > On Thu, Apr 30, 2020 at 12:56 PM Bogdan-Andrei Iancu > wrote: > >> Hi Sasmita, >> >> To be honest I do not understand what you want to achieve. Could you >> provide an example of aliasing you want to have ? >> >> Regards, >> >> Bogdan-Andrei Iancu >> >> OpenSIPS Founder and Developer >> https://www.opensips-solutions.com >> >> On 4/30/20 9:33 AM, Sasmita Panda wrote: >> >> Ok . But username cant be fixed . That can be anything . >> >> When I am adding aliase in the config file that only matches the domain >> and process the request . Isn't the aliase_db do the same thing ? >> >> >> *Thanks & Regards* >> *Sasmita Panda* >> *Senior Network Testing and Software Engineer* >> *3CLogic , ph:07827611765* >> >> >> On Wed, Apr 29, 2020 at 6:21 PM Bogdan-Andrei Iancu >> wrote: >> >>> Hi, >>> >>> The aliase_db module does full aliasing, user+domain to another >>> user+domain, not only domains. And you can test the return code of the >>> lookup function to see if any matching and translation was done or not. >>> >>> Regards, >>> >>> Bogdan-Andrei Iancu >>> >>> OpenSIPS Founder and Developer >>> https://www.opensips-solutions.com >>> >>> On 4/29/20 10:16 AM, Sasmita Panda wrote: >>> >>> Hi All , >>> >>> For the first time I am trying to use alias db . Before processing any >>> request I wanted to do alias db lookup , if find the corresponding domain >>> in the alias db then process the request otherwise not . >>> >>> How will I do this ? >>> >>> *Thanks & Regards* >>> *Sasmita Panda* >>> *Senior Network Testing and Software Engineer* >>> *3CLogic , ph:07827611765* >>> >>> _______________________________________________ >>> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >>> >>> >> _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -- Sincerely, Giovanni Maruzzelli OpenTelecom.IT cell: +39 347 266 56 18 -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Thu Apr 30 08:16:04 2020 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Thu, 30 Apr 2020 11:16:04 +0300 Subject: [OpenSIPS-Users] Need help to use alias db . In-Reply-To: References: <25fc2c22-39a8-4e75-386f-cd61fc4435bb@opensips.org> Message-ID: <14c1204d-ac1e-bd6a-dce6-1957036816da@opensips.org> Just use the domain module for that: https://opensips.org/html/docs/modules/2.4.x/domain.html#func_is_uri_host_local Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com On 4/30/20 11:08 AM, Sasmita Panda wrote: > I have one opensips server , and I have mapped 2 domain into that . In > the config file I have added the alias list for that domain . > for example : > alias="freeswitch-registrar.xyz.com " > alias="freeswitch-cisco.xyz.com " > > route{ >   if (!uri==myself) { >                                 send_reply("403","Rely forbidden"); >                                 exit; >                         } > ...................... > } > If any request coming to this has the request uri matches with the > alias added then that request get processed else it gives Error message . > > Now if I am mapping another domain to the server but I dont want to > change the config and restart the service again . I want any other way > to do dynamically so that whenever I will add any domain opensips will > read that and process the request without service restart . > > > How will I achive this ? > > > */Thanks & Regards/* > /Sasmita Panda/ > /Senior Network Testing and Software Engineer/ > /3CLogic , ph:07827611765/ > > > On Thu, Apr 30, 2020 at 12:56 PM Bogdan-Andrei Iancu > > wrote: > > Hi Sasmita, > > To be honest I do not understand what you want to achieve. Could > you provide an example of aliasing you want to have ? > > Regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > > On 4/30/20 9:33 AM, Sasmita Panda wrote: >> Ok . But username cant be fixed . That can be anything . >> >> When I am adding aliase in the config file that only matches the >> domain and process the request . Isn't the aliase_db do the same >> thing ? >> >> >> */Thanks & Regards/* >> /Sasmita Panda/ >> /Senior Network Testing and Software Engineer/ >> /3CLogic , ph:07827611765/ >> >> >> On Wed, Apr 29, 2020 at 6:21 PM Bogdan-Andrei Iancu >> > wrote: >> >> Hi, >> >> The aliase_db module does full aliasing, user+domain to >> another user+domain, not only domains. And you can test the >> return code of the lookup function to see if any matching and >> translation was done or not. >> >> Regards, >> >> Bogdan-Andrei Iancu >> >> OpenSIPS Founder and Developer >> https://www.opensips-solutions.com >> >> On 4/29/20 10:16 AM, Sasmita Panda wrote: >>> Hi All , >>> >>> For the first time I am trying to use alias db . Before >>> processing any request I wanted to do alias db lookup , if >>> find the corresponding domain in the alias db then process >>> the request otherwise not . >>> >>> How will I do this ? >>> >>> */Thanks & Regards/* >>> /Sasmita Panda/ >>> /Senior Network Testing and Software Engineer/ >>> /3CLogic , ph:07827611765/ >>> >>> _______________________________________________ >>> Users mailing list >>> Users at lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: From alex.a at gtanetworkconsulting.com Thu Apr 30 11:01:42 2020 From: alex.a at gtanetworkconsulting.com (Alex A) Date: Thu, 30 Apr 2020 07:01:42 -0400 Subject: [OpenSIPS-Users] Drouting failover by carrier only In-Reply-To: <1a2f5cf6-8ab9-d33f-dece-57bbfdb505da@opensips.org> References: <171c73bdfcf.f647ce1b105300.4293922564087136871@gtanetworkconsulting.com> <1a2f5cf6-8ab9-d33f-dece-57bbfdb505da@opensips.org> Message-ID: <171cabfd988.10ad0f9ff193758.1421691352906670212@gtanetworkconsulting.com> Hi Bogdan, Will "use only the first gateway from the carrier"  allow for round-robin for the regular calls (ie. does it choose the first gw randomly ) ? Thank you. Alex ---- On Thu, 30 Apr 2020 03:40:21 -0400 Bogdan-Andrei Iancu wrote ---- Hi Alex, Have you tried the "use only the first GW" flag in the carrier definition? See https://opensips.org/html/docs/modules/3.0.x/drouting.html#idp25339408, "Carriers" subsection: flags : 0x1 - use weight for sorting the list and not definition order; 0x2 - use only the first gateway from the carrier (depending on the sorting); 0x4 - disable the usage of this carrier Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com On 4/29/20 9:38 PM, Alex A wrote: Hi Everyone, Is it possible to failover to next carrier (instead next gateway) while using drouting? I got the below to work; however currently, use_next_gw gets the next gateway in the list, so if gwlist= #0,#3  and one of the carriers has multiple gateway IPs, the retry happens many times to the same carrier: route[droute] {         xlog("DRoute GATEWAY: source:$si - $fU -> $rU\n");         if (!do_routing(0,"F")) {                 xlog("DRoute GATEWAY: Failed. source:$si -$fU -> $rU\n");                 send_reply(500, "No Gateway to Route found");                 exit;         }         route(relay);         exit;         } route[relay] {         if (is_method("INVITE")) {                 t_on_failure("missed_call");         }         if (!t_relay()) {                 if (use_next_gw()) {                       xlog("L_INFO","Next Gateway: From=$fu, To=$tu,RU=$ru, CI=$ci IP=$si\n");                       t_on_failure("missed_call");                       route(relay);                       exit;                 }                 else {                         send_reply(503, "Service not available, no more gws");                         exit;                 }         };         exit; } failure_route[missed_call] {                 #if (use_next_gw(, , $var(carrier_attrs))) {                 if (use_next_gw(, $avp(gw_id),$avp(carrier_id))) {                       xlog("MissedCall--Next Gateway: From=$fu, To=$tu,RU=$ru, CI=$ci IP=$si\n");                       xlog("Carrier attributes of current gateway: $avp(gw_id). carrier: $avp(carrier_id)\n");                       t_on_failure("missed_call");                       route(relay);                       exit;                 }                 else {                         send_reply(503, "Service not available, no more gws");                         exit;                 } Thank you. _______________________________________________ Users mailing list mailto:Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From alex.a at gtanetworkconsulting.com Thu Apr 30 11:12:04 2020 From: alex.a at gtanetworkconsulting.com (Alex A) Date: Thu, 30 Apr 2020 07:12:04 -0400 Subject: [OpenSIPS-Users] Drouting failover by carrier only In-Reply-To: References: <171c73bdfcf.f647ce1b105300.4293922564087136871@gtanetworkconsulting.com> Message-ID: <171cac9548e.10c7a5752194891.3787572842061681149@gtanetworkconsulting.com> Thank you for the tip, Ben. Do you by chance have a script snipped for this scenario? For some reason, the drouting part:         if (!do_routing(0,"F")) {                 xlog("DRoute GATEWAY: Failed. source:$si -$fU -> $rU\n");                 send_reply(500, "No Gateway to Route found");                 exit;         }         xlog("Before Entering Relay\n CarrierID_avp: $avp(carrier_id), GW ID avp: $avp(gw_id)");         route(relay); Both AVPs return a single value instead of the a list for me, so I must be missing something. Thanks. ---- On Wed, 29 Apr 2020 14:50:38 -0400 Ben Newlin wrote ---- We also had a need to do this and did not find a way to do it via drouting directly, but it will work with drouting with a little help.   First, we specify the carrier_id_avp param [1]. After we call do_routing, we copy out the carrier IDs from that avp into our own AVP. Then we use route_to_carrier [2] for each carrier ID in the list. To continue routing within the carrier, you can still do use_next_gw. When that returns false (no more gateways), or if you want to skip to the next carrier you just call route_to_carrier again with the next ID in your list.   This solved two problem for us: * allows us to fail over by carrier instead of just by gateway * allows us to call do_routing multiple times for a call with different groups and aggregate the results   [1] - https://opensips.org/docs/modules/3.0.x/drouting.html#param_carrier_id_avp [2] - https://opensips.org/docs/modules/3.0.x/drouting.html#func_route_to_carrier     Ben Newlin   From: Users on behalf of Alex A Reply-To: OpenSIPS users mailling list Date: Wednesday, April 29, 2020 at 2:40 PM To: users Subject: [OpenSIPS-Users] Drouting failover by carrier only   Hi Everyone,   Is it possible to failover to next carrier (instead next gateway) while using drouting?     I got the below to work; however currently, use_next_gw gets the next gateway in the list, so if gwlist= #0,#3    and one of the carriers has multiple gateway IPs, the retry happens many times to the same carrier:     route[droute] {           xlog("DRoute GATEWAY: source:$si - $fU -> $rU\n");         if (!do_routing(0,"F")) {                 xlog("DRoute GATEWAY: Failed. source:$si -$fU -> $rU\n");                 send_reply(500, "No Gateway to Route found");                 exit;         }         route(relay);         exit;           }   route[relay] {           if (is_method("INVITE")) {                 t_on_failure("missed_call");         }           if (!t_relay()) {                 if (use_next_gw()) {                       xlog("L_INFO","Next Gateway: From=$fu, To=$tu,RU=$ru, CI=$ci IP=$si\n");                       t_on_failure("missed_call");                       route(relay);                       exit;                 }                 else {                         send_reply(503, "Service not available, no more gws");                         exit;                 }         };         exit; }     failure_route[missed_call] {                   #if (use_next_gw(, , $var(carrier_attrs))) {                 if (use_next_gw(, $avp(gw_id),$avp(carrier_id))) {                       xlog("MissedCall--Next Gateway: From=$fu, To=$tu,RU=$ru, CI=$ci IP=$si\n");                       xlog("Carrier attributes of current gateway: $avp(gw_id). carrier: $avp(carrier_id)\n");                         t_on_failure("missed_call");                       route(relay);                       exit;                 }                 else {                         send_reply(503, "Service not available, no more gws");                         exit;                 }         Thank you.       _______________________________________________ Users mailing list mailto:Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From spanda at 3clogic.com Thu Apr 30 11:15:05 2020 From: spanda at 3clogic.com (Sasmita Panda) Date: Thu, 30 Apr 2020 16:45:05 +0530 Subject: [OpenSIPS-Users] Need help to use alias db . In-Reply-To: References: <25fc2c22-39a8-4e75-386f-cd61fc4435bb@opensips.org> Message-ID: Thank u so much . I am just wandering . In opensips 3.0 there is opensips_cli . Is this possible through cli command also ? I haven't tried yet . *Thanks & Regards* *Sasmita Panda* *Senior Network Testing and Software Engineer* *3CLogic , ph:07827611765* On Thu, Apr 30, 2020 at 1:46 PM Giovanni Maruzzelli wrote: > you reload the domain table via MI interface each time you change it > -giovanni > > > > > On Thu, Apr 30, 2020 at 10:11 AM Sasmita Panda wrote: > >> I have one opensips server , and I have mapped 2 domain into that . In >> the config file I have added the alias list for that domain . >> for example : >> alias="freeswitch-registrar.xyz.com" >> alias="freeswitch-cisco.xyz.com" >> >> route{ >> if (!uri==myself) { >> send_reply("403","Rely forbidden"); >> exit; >> } >> ...................... >> } >> If any request coming to this has the request uri matches with the alias >> added then that request get processed else it gives Error message . >> >> Now if I am mapping another domain to the server but I dont want to >> change the config and restart the service again . I want any other way to >> do dynamically so that whenever I will add any domain opensips will read >> that and process the request without service restart . >> >> >> How will I achive this ? >> >> >> *Thanks & Regards* >> *Sasmita Panda* >> *Senior Network Testing and Software Engineer* >> *3CLogic , ph:07827611765* >> >> >> On Thu, Apr 30, 2020 at 12:56 PM Bogdan-Andrei Iancu >> wrote: >> >>> Hi Sasmita, >>> >>> To be honest I do not understand what you want to achieve. Could you >>> provide an example of aliasing you want to have ? >>> >>> Regards, >>> >>> Bogdan-Andrei Iancu >>> >>> OpenSIPS Founder and Developer >>> https://www.opensips-solutions.com >>> >>> On 4/30/20 9:33 AM, Sasmita Panda wrote: >>> >>> Ok . But username cant be fixed . That can be anything . >>> >>> When I am adding aliase in the config file that only matches the domain >>> and process the request . Isn't the aliase_db do the same thing ? >>> >>> >>> *Thanks & Regards* >>> *Sasmita Panda* >>> *Senior Network Testing and Software Engineer* >>> *3CLogic , ph:07827611765* >>> >>> >>> On Wed, Apr 29, 2020 at 6:21 PM Bogdan-Andrei Iancu >>> wrote: >>> >>>> Hi, >>>> >>>> The aliase_db module does full aliasing, user+domain to another >>>> user+domain, not only domains. And you can test the return code of the >>>> lookup function to see if any matching and translation was done or not. >>>> >>>> Regards, >>>> >>>> Bogdan-Andrei Iancu >>>> >>>> OpenSIPS Founder and Developer >>>> https://www.opensips-solutions.com >>>> >>>> On 4/29/20 10:16 AM, Sasmita Panda wrote: >>>> >>>> Hi All , >>>> >>>> For the first time I am trying to use alias db . Before processing any >>>> request I wanted to do alias db lookup , if find the corresponding domain >>>> in the alias db then process the request otherwise not . >>>> >>>> How will I do this ? >>>> >>>> *Thanks & Regards* >>>> *Sasmita Panda* >>>> *Senior Network Testing and Software Engineer* >>>> *3CLogic , ph:07827611765* >>>> >>>> _______________________________________________ >>>> Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>> >>>> >>>> >>> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> > > > -- > Sincerely, > > Giovanni Maruzzelli > OpenTelecom.IT > cell: +39 347 266 56 18 > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -------------- next part -------------- An HTML attachment was scrubbed... URL: From gmaruzz at gmail.com Thu Apr 30 11:16:59 2020 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Thu, 30 Apr 2020 13:16:59 +0200 Subject: [OpenSIPS-Users] Need help to use alias db . In-Reply-To: References: <25fc2c22-39a8-4e75-386f-cd61fc4435bb@opensips.org> Message-ID: On Thu, Apr 30, 2020 at 1:15 PM Sasmita Panda wrote: > Thank u so much . I am just wandering . > In opensips 3.0 there is opensips_cli . Is this possible through cli > command also ? I haven't tried yet . > > may I suggest reading the documentation? ;) -giovanni -- Sincerely, Giovanni Maruzzelli OpenTelecom.IT cell: +39 347 266 56 18 -------------- next part -------------- An HTML attachment was scrubbed... URL: From alex.a at gtanetworkconsulting.com Thu Apr 30 13:06:58 2020 From: alex.a at gtanetworkconsulting.com (Alex A) Date: Thu, 30 Apr 2020 09:06:58 -0400 Subject: [OpenSIPS-Users] Drouting failover by carrier only In-Reply-To: <171cac9548e.10c7a5752194891.3787572842061681149@gtanetworkconsulting.com> References: <171c73bdfcf.f647ce1b105300.4293922564087136871@gtanetworkconsulting.com> <171cac9548e.10c7a5752194891.3787572842061681149@gtanetworkconsulting.com> Message-ID: <171cb32856d.10c7a77ae209716.2354084514008881204@gtanetworkconsulting.com> Setting the "First Only" flag on the carrier seem to be done the trick for me. It round-robins, while failing over to another carrier directly. Thank you for your help. ---- On Thu, 30 Apr 2020 07:12:04 -0400 Alex A wrote ---- Thank you for the tip, Ben. Do you by chance have a script snipped for this scenario? For some reason, the drouting part:         if (!do_routing(0,"F")) {                 xlog("DRoute GATEWAY: Failed. source:$si -$fU -> $rU\n");                 send_reply(500, "No Gateway to Route found");                 exit;         }         xlog("Before Entering Relay\n CarrierID_avp: $avp(carrier_id), GW ID avp: $avp(gw_id)");         route(relay); Both AVPs return a single value instead of the a list for me, so I must be missing something. Thanks. ---- On Wed, 29 Apr 2020 14:50:38 -0400 Ben Newlin wrote ---- We also had a need to do this and did not find a way to do it via drouting directly, but it will work with drouting with a little help.   First, we specify the carrier_id_avp param [1]. After we call do_routing, we copy out the carrier IDs from that avp into our own AVP. Then we use route_to_carrier [2] for each carrier ID in the list. To continue routing within the carrier, you can still do use_next_gw. When that returns false (no more gateways), or if you want to skip to the next carrier you just call route_to_carrier again with the next ID in your list.   This solved two problem for us: * allows us to fail over by carrier instead of just by gateway * allows us to call do_routing multiple times for a call with different groups and aggregate the results   [1] - https://opensips.org/docs/modules/3.0.x/drouting.html#param_carrier_id_avp [2] - https://opensips.org/docs/modules/3.0.x/drouting.html#func_route_to_carrier     Ben Newlin   From: Users on behalf of Alex A Reply-To: OpenSIPS users mailling list Date: Wednesday, April 29, 2020 at 2:40 PM To: users Subject: [OpenSIPS-Users] Drouting failover by carrier only   Hi Everyone,   Is it possible to failover to next carrier (instead next gateway) while using drouting?     I got the below to work; however currently, use_next_gw gets the next gateway in the list, so if gwlist= #0,#3    and one of the carriers has multiple gateway IPs, the retry happens many times to the same carrier:     route[droute] {           xlog("DRoute GATEWAY: source:$si - $fU -> $rU\n");         if (!do_routing(0,"F")) {                 xlog("DRoute GATEWAY: Failed. source:$si -$fU -> $rU\n");                 send_reply(500, "No Gateway to Route found");                 exit;         }         route(relay);         exit;           }   route[relay] {           if (is_method("INVITE")) {                 t_on_failure("missed_call");         }           if (!t_relay()) {                 if (use_next_gw()) {                       xlog("L_INFO","Next Gateway: From=$fu, To=$tu,RU=$ru, CI=$ci IP=$si\n");                       t_on_failure("missed_call");                       route(relay);                       exit;                 }                 else {                         send_reply(503, "Service not available, no more gws");                         exit;                 }         };         exit; }     failure_route[missed_call] {                   #if (use_next_gw(, , $var(carrier_attrs))) {                 if (use_next_gw(, $avp(gw_id),$avp(carrier_id))) {                       xlog("MissedCall--Next Gateway: From=$fu, To=$tu,RU=$ru, CI=$ci IP=$si\n");                       xlog("Carrier attributes of current gateway: $avp(gw_id). carrier: $avp(carrier_id)\n");                         t_on_failure("missed_call");                       route(relay);                       exit;                 }                 else {                         send_reply(503, "Service not available, no more gws");                         exit;                 }         Thank you.       _______________________________________________ Users mailing list mailto:Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users _______________________________________________ Users mailing list mailto:Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From Johan at democon.be Thu Apr 30 13:10:41 2020 From: Johan at democon.be (Johan De Clercq) Date: Thu, 30 Apr 2020 15:10:41 +0200 Subject: [OpenSIPS-Users] Drouting failover by carrier only In-Reply-To: <171cb32856d.10c7a77ae209716.2354084514008881204@gtanetworkconsulting.com> References: <171c73bdfcf.f647ce1b105300.4293922564087136871@gtanetworkconsulting.com> <171cac9548e.10c7a5752194891.3787572842061681149@gtanetworkconsulting.com> <171cb32856d.10c7a77ae209716.2354084514008881204@gtanetworkconsulting.com> Message-ID: on what version is this ? Op do 30 apr. 2020 om 15:09 schreef Alex A : > Setting the "First Only" flag on the carrier seem to be done the trick for > me. > It round-robins, while failing over to another carrier directly. > > > Thank you for your help. > > > > > ---- On Thu, 30 Apr 2020 07:12:04 -0400 *Alex A > >* > wrote ---- > > Thank you for the tip, Ben. > > Do you by chance have a script snipped for this scenario? > > For some reason, the drouting part: > if (!do_routing(0,"F")) { > xlog("DRoute GATEWAY: Failed. source:$si -$fU -> $rU\n"); > send_reply(500, "No Gateway to Route found"); > exit; > } > xlog("Before Entering Relay\n CarrierID_avp: $avp(carrier_id), GW > ID avp: $avp(gw_id)"); > route(relay); > > Both AVPs return a single value instead of the a list for me, so I must be > missing something. > > > Thanks. > > > > > ---- On Wed, 29 Apr 2020 14:50:38 -0400 *Ben Newlin > >* wrote ---- > > We also had a need to do this and did not find a way to do it via drouting > directly, but it will work with drouting with a little help. > > > > First, we specify the carrier_id_avp param [1]. After we call do_routing, > we copy out the carrier IDs from that avp into our own AVP. Then we use > route_to_carrier [2] for each carrier ID in the list. To continue routing > within the carrier, you can still do use_next_gw. When that returns false > (no more gateways), or if you want to skip to the next carrier you just > call route_to_carrier again with the next ID in your list. > > > > This solved two problem for us: > > * allows us to fail over by carrier instead of just by gateway > > * allows us to call do_routing multiple times for a call with different > groups and aggregate the results > > > > [1] - > https://opensips.org/docs/modules/3.0.x/drouting.html#param_carrier_id_avp > > [2] - > https://opensips.org/docs/modules/3.0.x/drouting.html#func_route_to_carrier > > > > > > Ben Newlin > > > > *From: *Users on behalf of Alex A < > alex.a at gtanetworkconsulting.com> > *Reply-To: *OpenSIPS users mailling list > *Date: *Wednesday, April 29, 2020 at 2:40 PM > *To: *users > *Subject: *[OpenSIPS-Users] Drouting failover by carrier only > > > > Hi Everyone, > > > > Is it possible to failover to next carrier (instead next gateway) while > using drouting? > > > > > > I got the below to work; however currently, use_next_gw gets the next > gateway in the list, so > > if gwlist= #0,#3 > > > > and one of the carriers has multiple gateway IPs, the retry happens many > times to the same carrier: > > > > > > route[droute] { > > > > xlog("DRoute GATEWAY: source:$si - $fU -> $rU\n"); > > if (!do_routing(0,"F")) { > > xlog("DRoute GATEWAY: Failed. source:$si -$fU -> $rU\n"); > > send_reply(500, "No Gateway to Route found"); > > exit; > > } > > route(relay); > > exit; > > > > } > > > > route[relay] { > > > > if (is_method("INVITE")) { > > t_on_failure("missed_call"); > > } > > > > if (!t_relay()) { > > if (use_next_gw()) { > > xlog("L_INFO","Next Gateway: From=$fu, > To=$tu,RU=$ru, CI=$ci IP=$si\n"); > > t_on_failure("missed_call"); > > route(relay); > > exit; > > } > > else { > > send_reply(503, "Service not available, no more > gws"); > > exit; > > } > > }; > > exit; > > } > > > > > > failure_route[missed_call] { > > > > #if (use_next_gw(, , $var(carrier_attrs))) { > > if (use_next_gw(, $avp(gw_id),$avp(carrier_id))) { > > xlog("MissedCall--Next Gateway: From=$fu, > To=$tu,RU=$ru, CI=$ci IP=$si\n"); > > xlog("Carrier attributes of current gateway: > $avp(gw_id). carrier: $avp(carrier_id)\n"); > > > > t_on_failure("missed_call"); > > route(relay); > > exit; > > } > > else { > > send_reply(503, "Service not available, no more > gws"); > > exit; > > } > > > > > > > > > > Thank you. > > > > > > > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -------------- next part -------------- An HTML attachment was scrubbed... URL: From alex.a at gtanetworkconsulting.com Thu Apr 30 13:29:05 2020 From: alex.a at gtanetworkconsulting.com (Alex A) Date: Thu, 30 Apr 2020 09:29:05 -0400 Subject: [OpenSIPS-Users] Drouting failover by carrier only In-Reply-To: References: <171c73bdfcf.f647ce1b105300.4293922564087136871@gtanetworkconsulting.com> <171cac9548e.10c7a5752194891.3787572842061681149@gtanetworkconsulting.com> <171cb32856d.10c7a77ae209716.2354084514008881204@gtanetworkconsulting.com> Message-ID: <9261b64e-131f-4876-86fe-3bedea457321@gtanetworkconsulting.com> 3.0.2 On Apr 30, 2020, 9:12 AM, at 9:12 AM, Johan De Clercq wrote: >on what version is this ? > >Op do 30 apr. 2020 om 15:09 schreef Alex A >>: > >> Setting the "First Only" flag on the carrier seem to be done the >trick for >> me. >> It round-robins, while failing over to another carrier directly. >> >> >> Thank you for your help. >> >> >> >> >> ---- On Thu, 30 Apr 2020 07:12:04 -0400 *Alex A >> >* >> wrote ---- >> >> Thank you for the tip, Ben. >> >> Do you by chance have a script snipped for this scenario? >> >> For some reason, the drouting part: >> if (!do_routing(0,"F")) { >> xlog("DRoute GATEWAY: Failed. source:$si -$fU -> >$rU\n"); >> send_reply(500, "No Gateway to Route found"); >> exit; >> } >> xlog("Before Entering Relay\n CarrierID_avp: >$avp(carrier_id), GW >> ID avp: $avp(gw_id)"); >> route(relay); >> >> Both AVPs return a single value instead of the a list for me, so I >must be >> missing something. >> >> >> Thanks. >> >> >> >> >> ---- On Wed, 29 Apr 2020 14:50:38 -0400 *Ben Newlin >> >* wrote ---- >> >> We also had a need to do this and did not find a way to do it via >drouting >> directly, but it will work with drouting with a little help. >> >> >> >> First, we specify the carrier_id_avp param [1]. After we call >do_routing, >> we copy out the carrier IDs from that avp into our own AVP. Then we >use >> route_to_carrier [2] for each carrier ID in the list. To continue >routing >> within the carrier, you can still do use_next_gw. When that returns >false >> (no more gateways), or if you want to skip to the next carrier you >just >> call route_to_carrier again with the next ID in your list. >> >> >> >> This solved two problem for us: >> >> * allows us to fail over by carrier instead of just by gateway >> >> * allows us to call do_routing multiple times for a call with >different >> groups and aggregate the results >> >> >> >> [1] - >> >https://opensips.org/docs/modules/3.0.x/drouting.html#param_carrier_id_avp >> >> [2] - >> >https://opensips.org/docs/modules/3.0.x/drouting.html#func_route_to_carrier >> >> >> >> >> >> Ben Newlin >> >> >> >> *From: *Users on behalf of Alex A >< >> alex.a at gtanetworkconsulting.com> >> *Reply-To: *OpenSIPS users mailling list >> *Date: *Wednesday, April 29, 2020 at 2:40 PM >> *To: *users >> *Subject: *[OpenSIPS-Users] Drouting failover by carrier only >> >> >> >> Hi Everyone, >> >> >> >> Is it possible to failover to next carrier (instead next gateway) >while >> using drouting? >> >> >> >> >> >> I got the below to work; however currently, use_next_gw gets the next >> gateway in the list, so >> >> if gwlist= #0,#3 >> >> >> >> and one of the carriers has multiple gateway IPs, the retry happens >many >> times to the same carrier: >> >> >> >> >> >> route[droute] { >> >> >> >> xlog("DRoute GATEWAY: source:$si - $fU -> $rU\n"); >> >> if (!do_routing(0,"F")) { >> >> xlog("DRoute GATEWAY: Failed. source:$si -$fU -> >$rU\n"); >> >> send_reply(500, "No Gateway to Route found"); >> >> exit; >> >> } >> >> route(relay); >> >> exit; >> >> >> >> } >> >> >> >> route[relay] { >> >> >> >> if (is_method("INVITE")) { >> >> t_on_failure("missed_call"); >> >> } >> >> >> >> if (!t_relay()) { >> >> if (use_next_gw()) { >> >> xlog("L_INFO","Next Gateway: From=$fu, >> To=$tu,RU=$ru, CI=$ci IP=$si\n"); >> >> t_on_failure("missed_call"); >> >> route(relay); >> >> exit; >> >> } >> >> else { >> >> send_reply(503, "Service not available, no >more >> gws"); >> >> exit; >> >> } >> >> }; >> >> exit; >> >> } >> >> >> >> >> >> failure_route[missed_call] { >> >> >> >> #if (use_next_gw(, , $var(carrier_attrs))) { >> >> if (use_next_gw(, $avp(gw_id),$avp(carrier_id))) { >> >> xlog("MissedCall--Next Gateway: From=$fu, >> To=$tu,RU=$ru, CI=$ci IP=$si\n"); >> >> xlog("Carrier attributes of current gateway: >> $avp(gw_id). carrier: $avp(carrier_id)\n"); >> >> >> >> t_on_failure("missed_call"); >> >> route(relay); >> >> exit; >> >> } >> >> else { >> >> send_reply(503, "Service not available, no >more >> gws"); >> >> exit; >> >> } >> >> >> >> >> >> >> >> >> >> 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 >> >> >> >> _______________________________________________ >> 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 Johan at democon.be Thu Apr 30 13:55:25 2020 From: Johan at democon.be (Johan De Clercq) Date: Thu, 30 Apr 2020 15:55:25 +0200 Subject: [OpenSIPS-Users] Drouting failover by carrier only In-Reply-To: <9261b64e-131f-4876-86fe-3bedea457321@gtanetworkconsulting.com> References: <171c73bdfcf.f647ce1b105300.4293922564087136871@gtanetworkconsulting.com> <171cac9548e.10c7a5752194891.3787572842061681149@gtanetworkconsulting.com> <171cb32856d.10c7a77ae209716.2354084514008881204@gtanetworkconsulting.com> <9261b64e-131f-4876-86fe-3bedea457321@gtanetworkconsulting.com> Message-ID: Can you please send your routing rules. Mine don't work in 3.0.2. Although they were good in 3.0 On Thu, 30 Apr 2020, 15:31 Alex A, wrote: > 3.0.2 > On Apr 30, 2020, at 9:12 AM, Johan De Clercq wrote: >> >> on what version is this ? >> >> Op do 30 apr. 2020 om 15:09 schreef Alex A < >> alex.a at gtanetworkconsulting.com>: >> >>> Setting the "First Only" flag on the carrier seem to be done the trick >>> for me. >>> It round-robins, while failing over to another carrier directly. >>> >>> >>> Thank you for your help. >>> >>> >>> >>> >>> ---- On Thu, 30 Apr 2020 07:12:04 -0400 *Alex A >>> >* >>> wrote ---- >>> >>> Thank you for the tip, Ben. >>> >>> Do you by chance have a script snipped for this scenario? >>> >>> For some reason, the drouting part: >>> if (!do_routing(0,"F")) { >>> xlog("DRoute GATEWAY: Failed. source:$si -$fU -> $rU\n"); >>> send_reply(500, "No Gateway to Route found"); >>> exit; >>> } >>> xlog("Before Entering Relay\n CarrierID_avp: $avp(carrier_id), >>> GW ID avp: $avp(gw_id)"); >>> route(relay); >>> >>> Both AVPs return a single value instead of the a list for me, so I must >>> be missing something. >>> >>> >>> Thanks. >>> >>> >>> >>> >>> ---- On Wed, 29 Apr 2020 14:50:38 -0400 *Ben Newlin >>> >* wrote ---- >>> >>> We also had a need to do this and did not find a way to do it via >>> drouting directly, but it will work with drouting with a little help. >>> >>> >>> >>> First, we specify the carrier_id_avp param [1]. After we call >>> do_routing, we copy out the carrier IDs from that avp into our own AVP. >>> Then we use route_to_carrier [2] for each carrier ID in the list. To >>> continue routing within the carrier, you can still do use_next_gw. When >>> that returns false (no more gateways), or if you want to skip to the next >>> carrier you just call route_to_carrier again with the next ID in your list. >>> >>> >>> >>> This solved two problem for us: >>> >>> * allows us to fail over by carrier instead of just by gateway >>> >>> * allows us to call do_routing multiple times for a call with different >>> groups and aggregate the results >>> >>> >>> >>> [1] - >>> https://opensips.org/docs/modules/3.0.x/drouting.html#param_carrier_id_avp >>> >>> [2] - >>> https://opensips.org/docs/modules/3.0.x/drouting.html#func_route_to_carrier >>> >>> >>> >>> >>> >>> Ben Newlin >>> >>> >>> >>> *From: *Users on behalf of Alex A < >>> alex.a at gtanetworkconsulting.com> >>> *Reply-To: *OpenSIPS users mailling list >>> *Date: *Wednesday, April 29, 2020 at 2:40 PM >>> *To: *users >>> *Subject: *[OpenSIPS-Users] Drouting failover by carrier only >>> >>> >>> >>> Hi Everyone, >>> >>> >>> >>> Is it possible to failover to next carrier (instead next gateway) while >>> using drouting? >>> >>> >>> >>> >>> >>> I got the below to work; however currently, use_next_gw gets the next >>> gateway in the list, so >>> >>> if gwlist= #0,#3 >>> >>> >>> >>> and one of the carriers has multiple gateway IPs, the retry happens many >>> times to the same carrier: >>> >>> >>> >>> >>> >>> route[droute] { >>> >>> >>> >>> xlog("DRoute GATEWAY: source:$si - $fU -> $rU\n"); >>> >>> if (!do_routing(0,"F")) { >>> >>> xlog("DRoute GATEWAY: Failed. source:$si -$fU -> $rU\n"); >>> >>> send_reply(500, "No Gateway to Route found"); >>> >>> exit; >>> >>> } >>> >>> route(relay); >>> >>> exit; >>> >>> >>> >>> } >>> >>> >>> >>> route[relay] { >>> >>> >>> >>> if (is_method("INVITE")) { >>> >>> t_on_failure("missed_call"); >>> >>> } >>> >>> >>> >>> if (!t_relay()) { >>> >>> if (use_next_gw()) { >>> >>> xlog("L_INFO","Next Gateway: From=$fu, >>> To=$tu,RU=$ru, CI=$ci IP=$si\n"); >>> >>> t_on_failure("missed_call"); >>> >>> route(relay); >>> >>> exit; >>> >>> } >>> >>> else { >>> >>> send_reply(503, "Service not available, no more >>> gws"); >>> >>> exit; >>> >>> } >>> >>> }; >>> >>> exit; >>> >>> } >>> >>> >>> >>> >>> >>> failure_route[missed_call] { >>> >>> >>> >>> #if (use_next_gw(, , $var(carrier_attrs))) { >>> >>> if (use_next_gw(, $avp(gw_id),$avp(carrier_id))) { >>> >>> xlog("MissedCall--Next Gateway: From=$fu, >>> To=$tu,RU=$ru, CI=$ci IP=$si\n"); >>> >>> xlog("Carrier attributes of current gateway: >>> $avp(gw_id). carrier: $avp(carrier_id)\n"); >>> >>> >>> >>> t_on_failure("missed_call"); >>> >>> route(relay); >>> >>> exit; >>> >>> } >>> >>> else { >>> >>> send_reply(503, "Service not available, no more >>> gws"); >>> >>> exit; >>> >>> } >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> 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 >>> >>> >>> >>> _______________________________________________ >>> Users mailing list >>> Users at lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >> ------------------------------ >> >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Thu Apr 30 17:11:52 2020 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Thu, 30 Apr 2020 20:11:52 +0300 Subject: [OpenSIPS-Users] [Blog] Back-to-Back & Clustering, a love story in OpenSIPS 3.1 Message-ID: The need for the clustering support in Back-2-Back is even stronger as coming from the High-Availability direction. Yes, in order to build a highly available system for OpenSIPS, with an hot (fully sync’ed) stand-by, you need clustering support, mainly the ability of the back-2-back engine to replicate its internal data in realtime, with other OpenSIPS instances. https://blog.opensips.org/2020/04/30/back-to-back-clustering-a-love-story-in-opensips-3-1/ Enjoy, -- Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com