From prathibhab.tvm at gmail.com Sun Oct 1 09:25:18 2023 From: prathibhab.tvm at gmail.com (Prathibha B) Date: Sun, 1 Oct 2023 14:55:18 +0530 Subject: [OpenSIPS-Users] TLS connection errors Message-ID: On starting up openips, I get TLS connection to 14.139.183.221:63931 write failed (5:-1:9) ERROR:tls_openssl:openssl_tls_write: TLS write error: ERROR:tls_openssl:openssl_tls_blocking_write: TLS failed to send data ERROR:tls_openssl:openssl_tls_write: TLS connection to 111.92.11.139:54102 write failed (5:-1:9) ERROR:tls_openssl:openssl_tls_write: TLS write error: ERROR:tls_openssl:openssl_tls_blocking_write: TLS failed to send data I am trying for web to web communication using opensips. -- Regards, B.Prathibha -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Mon Oct 2 06:39:27 2023 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Mon, 2 Oct 2023 09:39:27 +0300 Subject: [OpenSIPS-Users] OpenSIPS Bootcamp training Asia-Pacific - 2 weeks to start Message-ID: <5f2c679d-6881-29b9-8205-9434e9ffaa03@opensips.org> Asia-Pacific session 16-27 October, 8am-12pm GMT *Why OpenSIPS eBootcamp training? * * the only official OpenSIPS training * perfect timing for Asia / Pacific - 8am-12pm GMT * covering latest OpenSIPS 3.4 content * 50% hands-on labs * interactive live sessions * ten days / 40 hours of training * two trainers * in-cloud environment * training materials retention * sip, voip, rtc knowledge and more Register Now Any questions? do not hesitate to contact us ! ------------------------------------------------------------------------ You received this email as part of your relationship with the OpenSIPS Project. If you do not want to receive any more news, please email to unsubscribe . -- Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com https://www.siphub.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From razvan at opensips.org Mon Oct 2 06:49:01 2023 From: razvan at opensips.org (=?UTF-8?Q?R=C4=83zvan_Crainea?=) Date: Mon, 2 Oct 2023 09:49:01 +0300 Subject: [OpenSIPS-Users] OpenSIPS Control Panel supported OpenSIPS version question In-Reply-To: References: Message-ID: Hi, Nineto! Although it was not already released, OpenSIPS master branch should be compatible with OpenSIPS 3.4. The compatibility process is not yet complete, therefore a full release (9.3.4) is not available yet for OpenSIPS 3.4. Best regards, Răzvan Crainea OpenSIPS Core Developer / SIPhub CTO http://www.opensips-solutions.com / https://www.siphub.com On 9/29/23 19:15, Nine to one wrote: > Hello OpenSIPS Control Panel developers, > > From website OCP only mentioned support up to OpenSIPS 3.3, I am using > 3.4, so want to know if current OCP already support OpenSIPS 3.4 or not. > > Thanks, > Nineto > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users From razvan at opensips.org Mon Oct 2 06:58:16 2023 From: razvan at opensips.org (=?UTF-8?Q?R=C4=83zvan_Crainea?=) Date: Mon, 2 Oct 2023 09:58:16 +0300 Subject: [OpenSIPS-Users] Can't set TLS ciphers_list to NULL In-Reply-To: References: Message-ID: <86f85043-02c8-4010-be3e-713e5511e906@opensips.org> Hi, Matt! Are you sure that wolfssl supports the NULL cipher list? You can see all the available ciphers when OpenSIPS starts. For example, my setup has the following ciphers: ``` Oct 2 09:56:43 [207525] INFO:tls_wolfssl:_wolfssl_show_ciphers: Ciphers: TLS13-AES128-GCM-SHA256:TLS13-AES256-GCM-SHA384:TLS13-CHACHA20-POLY1305-SHA256:TLS13-AES128-CCM-SHA256:TLS13-AES128-CCM-8-SHA256:TLS13-AES128-CCM8-SHA256:TLS13-SHA256-SHA256:TLS13-SHA384-SHA384:RC4-SHA:RC4-MD5:DES-CBC3-SHA:AES128-SHA:AES256-SHA:NULL-MD5:NULL-SHA:NULL-SHA256:DHE-RSA-AES128-SHA:DHE-RSA-AES256-SHA:DHE-PSK-AES256-GCM-SHA384:DHE-PSK-AES128-GCM-SHA256:DHE-PSK-AES256-CBC-SHA384:DHE-PSK-AES128-CBC-SHA256:DHE-PSK-AES128-CCM:DHE-PSK-AES256-CCM:DHE-PSK-NULL-SHA384:DHE-PSK-NULL-SHA256:AES128-CCM-8:AES128-CCM8:AES256-CCM-8:AES256-CCM8:ECDHE-ECDSA-AES128-CCM:ECDHE-ECDSA-AES128-CCM-8:ECDHE-ECDSA-AES128-CCM8:ECDHE-ECDSA-AES256-CCM-8:ECDHE-ECDSA-AES256-CCM8:ECDHE-RSA-AES128-SHA:ECDHE-RSA-AES256-SHA:ECDHE-ECDSA-AES128-SHA:ECDHE-ECDSA-AES256-SHA:ECDHE-RSA-RC4-SHA:ECDHE-RSA-DES-CBC3-SHA:ECDHE-ECDSA-RC4-SHA:ECDHE-ECDSA-DES-CBC3-SHA:AES128-SHA256:AES256-SHA256:DHE-RSA-AES128-SHA256:DHE-RSA-AES256-SHA256:AES128-GCM-SHA256:AES256-GCM-SHA384:DHE-RSA-AES128-GCM-SHA256:DHE-RSA-AES256-GCM-SHA384:ECDHE-RSA-AES128-GCM-SHA256:ECDHE-RSA-AES256-GCM-SHA384:ECDHE-ECDSA-AES128-GCM-SHA256:ECDHE-ECDSA-AES256-GCM-SHA384:CAMELLIA128-SHA:DHE-RSA-CAMELLIA128-SHA:CAMELLIA256-SHA:DHE-RSA-CAMELLIA256-SHA:CAMELLIA128-SHA256:DHE-RSA-CAMELLIA128-SHA256:CAMELLIA256-SHA256:DHE-RSA-CAMELLIA256-SHA256:ECDHE-RSA-AES128-SHA256:ECDHE-ECDSA-AES128-SHA256:ECDHE-RSA-AES256-SHA384:ECDHE-ECDSA-AES256-SHA384:ECDHE-RSA-CHACHA20-POLY1305:ECDHE-ECDSA-CHACHA20-POLY1305:DHE-RSA-CHACHA20-POLY1305:ECDHE-RSA-CHACHA20-POLY1305-OLD:ECDHE-ECDSA-CHACHA20-POLY1305-OLD:DHE-RSA-CHACHA20-POLY1305-OLD:ADH-AES128-SHA:ADH-AES256-GCM-SHA384:ECDHE-ECDSA-NULL-SHA:ECDHE-PSK-NULL-SHA256:ECDHE-PSK-AES128-CBC-SHA256:ECDHE-PSK-AES128-GCM-SHA256:PSK-CHACHA20-POLY1305:ECDHE-PSK-CHACHA20-POLY1305:DHE-PSK-CHACHA20-POLY1305:EDH-RSA-DES-CBC3-SHA:WDM-NULL-SHA256 ``` And plain NULL cipher is not available, only a set of its other variants. Best regards, Răzvan Crainea OpenSIPS Core Developer / SIPhub CTO http://www.opensips-solutions.com / https://www.siphub.com On 9/30/23 17:16, L S wrote: > Wolfssl gives an error and Opensips doesn't start when trying to set the > ciphers_list to NULL for a client domain in 3.2.13. > > modparam("tls_mgm", "ciphers_list", "[testclient]NULL") > > ERROR:tls_wolfssl:_wolfssl_init_tls_dom: failure to set SSL context > cipher list 'NULL' > > Any suggestions? > > Thanks, > Matt > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users From Johan at democon.be Mon Oct 2 09:51:21 2023 From: Johan at democon.be (Johan De Clercq) Date: Mon, 2 Oct 2023 11:51:21 +0200 Subject: [OpenSIPS-Users] Can't set TLS ciphers_list to NULL In-Reply-To: <86f85043-02c8-4010-be3e-713e5511e906@opensips.org> References: <86f85043-02c8-4010-be3e-713e5511e906@opensips.org> Message-ID: I find it strange to start with the null suite. What version of this is this? On Mon, 2 Oct 2023, 11:36 Răzvan Crainea, wrote: > Hi, Matt! > > Are you sure that wolfssl supports the NULL cipher list? You can see all > the available ciphers when OpenSIPS starts. For example, my setup has > the following ciphers: > > ``` > Oct 2 09:56:43 [207525] INFO:tls_wolfssl:_wolfssl_show_ciphers: > Ciphers: > > TLS13-AES128-GCM-SHA256:TLS13-AES256-GCM-SHA384:TLS13-CHACHA20-POLY1305-SHA256:TLS13-AES128-CCM-SHA256:TLS13-AES128-CCM-8-SHA256:TLS13-AES128-CCM8-SHA256:TLS13-SHA256-SHA256:TLS13-SHA384-SHA384:RC4-SHA:RC4-MD5:DES-CBC3-SHA:AES128-SHA:AES256-SHA:NULL-MD5:NULL-SHA:NULL-SHA256:DHE-RSA-AES128-SHA:DHE-RSA-AES256-SHA:DHE-PSK-AES256-GCM-SHA384:DHE-PSK-AES128-GCM-SHA256:DHE-PSK-AES256-CBC-SHA384:DHE-PSK-AES128-CBC-SHA256:DHE-PSK-AES128-CCM:DHE-PSK-AES256-CCM:DHE-PSK-NULL-SHA384:DHE-PSK-NULL-SHA256:AES128-CCM-8:AES128-CCM8:AES256-CCM-8:AES256-CCM8:ECDHE-ECDSA-AES128-CCM:ECDHE-ECDSA-AES128-CCM-8:ECDHE-ECDSA-AES128-CCM8:ECDHE-ECDSA-AES256-CCM-8:ECDHE-ECDSA-AES256-CCM8:ECDHE-RSA-AES128-SHA:ECDHE-RSA-AES256-SHA:ECDHE-ECDSA-AES128-SHA:ECDHE-ECDSA-AES256-SHA:ECDHE-RSA-RC4-SHA:ECDHE-RSA-DES-CBC3-SHA:ECDHE-ECDSA-RC4-SHA:ECDHE-ECDSA-DES-CBC3-SHA:AES128-SHA256:AES256-SHA256:DHE-RSA-AES128-SHA256:DHE-RSA-AES256-SHA256:AES128-GCM-SHA256:AES256-GCM-SHA384:DHE-RSA-AES128-GCM-SHA256:DHE-RSA-AES256-GCM-SHA384:ECDHE-RSA-AES128-GCM-SHA256:ECDHE-RSA-AES256-GCM-SHA384:ECDHE-ECDSA-AES128-GCM-SHA256:ECDHE-ECDSA-AES256-GCM-SHA384:CAMELLIA128-SHA:DHE-RSA-CAMELLIA128-SHA:CAMELLIA256-SHA:DHE-RSA-CAMELLIA256-SHA:CAMELLIA128-SHA256:DHE-RSA-CAMELLIA128-SHA256:CAMELLIA256-SHA256:DHE-RSA-CAMELLIA256-SHA256:ECDHE-RSA-AES128-SHA256:ECDHE-ECDSA-AES128-SHA256:ECDHE-RSA-AES256-SHA384:ECDHE-ECDSA-AES256-SHA384:ECDHE-RSA-CHACHA20-POLY1305:ECDHE-ECDSA-CHACHA20-POLY1305:DHE-RSA-CHACHA20-POLY1305:ECDHE-RSA-CHACHA20-POLY1305-OLD:ECDHE-ECDSA-CHACHA20-POLY1305-OLD:DHE-RSA-CHACHA20-POLY1305-OLD:ADH-AES128-SHA:ADH-AES256-GCM-SHA384:ECDHE-ECDSA-NULL-SHA:ECDHE-PSK-NULL-SHA256:ECDHE-PSK-AES128-CBC-SHA256:ECDHE-PSK-AES128-GCM-SHA256:PSK-CHACHA20-POLY1305:ECDHE-PSK-CHACHA20-POLY1305:DHE-PSK-CHACHA20-POLY1305:EDH-RSA-DES-CBC3-SHA:WDM-NULL-SHA256 > > ``` > > And plain NULL cipher is not available, only a set of its other variants. > > Best regards, > > Răzvan Crainea > OpenSIPS Core Developer / SIPhub CTO > http://www.opensips-solutions.com / https://www.siphub.com > > On 9/30/23 17:16, L S wrote: > > Wolfssl gives an error and Opensips doesn't start when trying to set the > > ciphers_list to NULL for a client domain in 3.2.13. > > > > modparam("tls_mgm", "ciphers_list", "[testclient]NULL") > > > > ERROR:tls_wolfssl:_wolfssl_init_tls_dom: failure to set SSL context > > cipher list 'NULL' > > > > Any suggestions? > > > > Thanks, > > Matt > > > > _______________________________________________ > > 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 prathibhab.tvm at gmail.com Mon Oct 2 11:46:43 2023 From: prathibhab.tvm at gmail.com (Prathibha B) Date: Mon, 2 Oct 2023 17:16:43 +0530 Subject: [OpenSIPS-Users] not getting audio and video Message-ID: I'm using opensips-2.4. Connection is getting established between two browsers. But there is no audio and video. One user is behind NAT. I'm using rtpengine as media proxy. There is audio and video when both the users are in the same n/w. -- Regards, B.Prathibha -------------- next part -------------- An HTML attachment was scrubbed... URL: From prathibhab.tvm at gmail.com Mon Oct 2 11:49:47 2023 From: prathibhab.tvm at gmail.com (Prathibha B) Date: Mon, 2 Oct 2023 17:19:47 +0530 Subject: [OpenSIPS-Users] not getting audio and video In-Reply-To: References: Message-ID: I'm seeing these errors in syslog: ERROR:core:tcp_connect_blocking_timeout: connect timed out, 99195 us elapsed out of 100000 us 2023-10-02T11:35:39.819075+00:00 bperss opensips[117122]: Oct 2 11:35:39 [117122] ERROR:proto_wss:ws_sync_connect: tcp_blocking_connect failed 2023-10-02T11:35:39.819132+00:00 bperss opensips[117122]: Oct 2 11:35:39 [117122] ERROR:proto_wss:ws_connect: connect failed 2023-10-02T11:35:39.819175+00:00 bperss opensips[117122]: Oct 2 11:35:39 [117122] ERROR:proto_wss:proto_wss_send: connect failed 2023-10-02T11:35:39.819211+00:00 bperss opensips[117122]: Oct 2 11:35:39 [117122] ERROR:core:msg_send: send() to 192.0.2.214:443 for proto wss/6 failed 2023-10-02T11:36:08.229522+00:00 bperss opensips[117114]: Oct 2 11:36:08 [117114] ERROR:proto_wss:tls_write: TLS connection to 14.139.183.221:59248 write failed 2023-10-02T11:36:08.230156+00:00 bperss opensips[117114]: Oct 2 11:36:08 [117114] ERROR:proto_wss:tls_write: TLS write error: 2023-10-02T11:36:08.230218+00:00 bperss opensips[117114]: Oct 2 11:36:08 [117114] ERROR:proto_wss:tls_blocking_write: TLS failed to send data 2023-10-02T11:36:08.230433+00:00 bperss opensips[117114]: Oct 2 11:36:08 [117114] ERROR:proto_wss:tls_write: TLS connection to 111.92.11.139:59416 write failed 2023-10-02T11:36:08.230491+00:00 bperss opensips[117114]: Oct 2 11:36:08 [117114] ERROR:proto_wss:tls_write: TLS write error: 2023-10-02T11:36:08.230543+00:00 bperss opensips[117114]: Oct 2 11:36:08 [117114] ERROR:proto_wss:tls_blocking_write: TLS failed to send data On Mon, 2 Oct 2023 at 17:16, Prathibha B wrote: > I'm using opensips-2.4. Connection is getting established between two > browsers. But there is no audio and video. One user is behind NAT. I'm > using rtpengine as media proxy. > > There is audio and video when both the users are in the same n/w. > > -- > Regards, > B.Prathibha > -- Regards, B.Prathibha -------------- next part -------------- An HTML attachment was scrubbed... URL: From efes99999 at gmail.com Tue Oct 3 10:37:11 2023 From: efes99999 at gmail.com (L S) Date: Tue, 3 Oct 2023 06:37:11 -0400 Subject: [OpenSIPS-Users] Can't set TLS ciphers_list to NULL In-Reply-To: <86f85043-02c8-4010-be3e-713e5511e906@opensips.org> References: <86f85043-02c8-4010-be3e-713e5511e906@opensips.org> Message-ID: Thanks Razvan. I have a similar set of ciphers, I will try one of the variants. I misinterpreted NULL in that context - I didn't think of it as the name of a cipher - more like a generic value that tells Opensips/wolfssl not to encrypt (for debugging). Matt On Mon, Oct 2, 2023, 5:36 AM Răzvan Crainea wrote: > Hi, Matt! > > Are you sure that wolfssl supports the NULL cipher list? You can see all > the available ciphers when OpenSIPS starts. For example, my setup has > the following ciphers: > > ``` > Oct 2 09:56:43 [207525] INFO:tls_wolfssl:_wolfssl_show_ciphers: > Ciphers: > > TLS13-AES128-GCM-SHA256:TLS13-AES256-GCM-SHA384:TLS13-CHACHA20-POLY1305-SHA256:TLS13-AES128-CCM-SHA256:TLS13-AES128-CCM-8-SHA256:TLS13-AES128-CCM8-SHA256:TLS13-SHA256-SHA256:TLS13-SHA384-SHA384:RC4-SHA:RC4-MD5:DES-CBC3-SHA:AES128-SHA:AES256-SHA:NULL-MD5:NULL-SHA:NULL-SHA256:DHE-RSA-AES128-SHA:DHE-RSA-AES256-SHA:DHE-PSK-AES256-GCM-SHA384:DHE-PSK-AES128-GCM-SHA256:DHE-PSK-AES256-CBC-SHA384:DHE-PSK-AES128-CBC-SHA256:DHE-PSK-AES128-CCM:DHE-PSK-AES256-CCM:DHE-PSK-NULL-SHA384:DHE-PSK-NULL-SHA256:AES128-CCM-8:AES128-CCM8:AES256-CCM-8:AES256-CCM8:ECDHE-ECDSA-AES128-CCM:ECDHE-ECDSA-AES128-CCM-8:ECDHE-ECDSA-AES128-CCM8:ECDHE-ECDSA-AES256-CCM-8:ECDHE-ECDSA-AES256-CCM8:ECDHE-RSA-AES128-SHA:ECDHE-RSA-AES256-SHA:ECDHE-ECDSA-AES128-SHA:ECDHE-ECDSA-AES256-SHA:ECDHE-RSA-RC4-SHA:ECDHE-RSA-DES-CBC3-SHA:ECDHE-ECDSA-RC4-SHA:ECDHE-ECDSA-DES-CBC3-SHA:AES128-SHA256:AES256-SHA256:DHE-RSA-AES128-SHA256:DHE-RSA-AES256-SHA256:AES128-GCM-SHA256:AES256-GCM-SHA384:DHE-RSA-AES128-GCM-SHA256:DHE-RSA-AES256-GCM-SHA384:ECDHE-RSA-AES128-GCM-SHA256:ECDHE-RSA-AES256-GCM-SHA384:ECDHE-ECDSA-AES128-GCM-SHA256:ECDHE-ECDSA-AES256-GCM-SHA384:CAMELLIA128-SHA:DHE-RSA-CAMELLIA128-SHA:CAMELLIA256-SHA:DHE-RSA-CAMELLIA256-SHA:CAMELLIA128-SHA256:DHE-RSA-CAMELLIA128-SHA256:CAMELLIA256-SHA256:DHE-RSA-CAMELLIA256-SHA256:ECDHE-RSA-AES128-SHA256:ECDHE-ECDSA-AES128-SHA256:ECDHE-RSA-AES256-SHA384:ECDHE-ECDSA-AES256-SHA384:ECDHE-RSA-CHACHA20-POLY1305:ECDHE-ECDSA-CHACHA20-POLY1305:DHE-RSA-CHACHA20-POLY1305:ECDHE-RSA-CHACHA20-POLY1305-OLD:ECDHE-ECDSA-CHACHA20-POLY1305-OLD:DHE-RSA-CHACHA20-POLY1305-OLD:ADH-AES128-SHA:ADH-AES256-GCM-SHA384:ECDHE-ECDSA-NULL-SHA:ECDHE-PSK-NULL-SHA256:ECDHE-PSK-AES128-CBC-SHA256:ECDHE-PSK-AES128-GCM-SHA256:PSK-CHACHA20-POLY1305:ECDHE-PSK-CHACHA20-POLY1305:DHE-PSK-CHACHA20-POLY1305:EDH-RSA-DES-CBC3-SHA:WDM-NULL-SHA256 > > ``` > > And plain NULL cipher is not available, only a set of its other variants. > > Best regards, > > Răzvan Crainea > OpenSIPS Core Developer / SIPhub CTO > http://www.opensips-solutions.com / https://www.siphub.com > > On 9/30/23 17:16, L S wrote: > > Wolfssl gives an error and Opensips doesn't start when trying to set the > > ciphers_list to NULL for a client domain in 3.2.13. > > > > modparam("tls_mgm", "ciphers_list", "[testclient]NULL") > > > > ERROR:tls_wolfssl:_wolfssl_init_tls_dom: failure to set SSL context > > cipher list 'NULL' > > > > Any suggestions? > > > > Thanks, > > Matt > > > > _______________________________________________ > > 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 9to1url at gmail.com Tue Oct 3 15:00:37 2023 From: 9to1url at gmail.com (Nine to one) Date: Tue, 3 Oct 2023 11:00:37 -0400 Subject: [OpenSIPS-Users] OpenSIPS Control Panel supported OpenSIPS version question In-Reply-To: References: Message-ID: Hi Răzvan, Thanks for advise this. At the point in time I installed OpenSIPS 3.3 in order to install OCP OpenSIPS Control Panel. I am not a PHP developer, but when I run OCP I found many issues with Postgres, I use an external Postgres, so I modified the source in order to let this Postgres version works, here are several fixes : https://github.com/9to1url/opensips-cp/commit/b47702907f166d049641d99cff956e1f25423d94 Since I am not a PHP developer so let it work both for MySQL and Postgres is not easy job for me. Might be you can consider to update your code to make it works for both. :-) Thanks, Nineto Răzvan Crainea 於 2023年10月2日 週一 上午5:36寫道： > Hi, Nineto! > > Although it was not already released, OpenSIPS master branch should be > compatible with OpenSIPS 3.4. The compatibility process is not yet > complete, therefore a full release (9.3.4) is not available yet for > OpenSIPS 3.4. > > Best regards, > > Răzvan Crainea > OpenSIPS Core Developer / SIPhub CTO > http://www.opensips-solutions.com / https://www.siphub.com > > On 9/29/23 19:15, Nine to one wrote: > > Hello OpenSIPS Control Panel developers, > > > > From website OCP only mentioned support up to OpenSIPS 3.3, I am using > > 3.4, so want to know if current OCP already support OpenSIPS 3.4 or not. > > > > Thanks, > > Nineto > > > > _______________________________________________ > > 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 liviu at opensips.org Wed Oct 4 17:16:16 2023 From: liviu at opensips.org (Liviu Chircu) Date: Wed, 4 Oct 2023 20:16:16 +0300 Subject: [OpenSIPS-Users] [Release Freeze] Upcoming OpenSIPS 3.4.2, 3.3.8 and 3.2.15 Minor Releases Message-ID: <4c93c90c-8bba-f08a-16fb-eeee25fe8a40@opensips.org> Hi, everyone! The 3.4.2, 3.3.8 and 3.2.15 OpenSIPS minor versions are scheduled for release on *Wednesday, Oct 18th*. In preparation for the releases, starting *Monday, Oct 9th*, we will impose the usual *freeze* on any significant fixes (as complexity) on these stable branches, in order to ensure a safe window for testing in the days ahead. Happy testing, -- Liviu Chircu www.twitter.com/liviuchircu |www.opensips-solutions.com OpenSIPS eBootcamp, Oct 16-27 |www.opensips.org/training -------------- next part -------------- An HTML attachment was scrubbed... URL: From venefax at gmail.com Wed Oct 4 20:19:05 2023 From: venefax at gmail.com (Saint Michael) Date: Wed, 4 Oct 2023 16:19:05 -0400 Subject: [OpenSIPS-Users] [Release Freeze] Upcoming OpenSIPS 3.4.2, 3.3.8 and 3.2.15 Minor Releases In-Reply-To: <4c93c90c-8bba-f08a-16fb-eeee25fe8a40@opensips.org> References: <4c93c90c-8bba-f08a-16fb-eeee25fe8a40@opensips.org> Message-ID: Question: is there a significant speed difference in servicing events from version 2.4 to 3.4? On Wed, Oct 4, 2023 at 1:19 PM Liviu Chircu wrote: > > Hi, everyone! > > The 3.4.2, 3.3.8 and 3.2.15 OpenSIPS minor versions are scheduled for release on Wednesday, Oct 18th. > > In preparation for the releases, starting Monday, Oct 9th, we will impose the usual freeze on any significant fixes (as complexity) on these stable branches, in order to ensure a safe window for testing in the days ahead. > > Happy testing, > > -- > Liviu Chircu > www.twitter.com/liviuchircu | www.opensips-solutions.com > OpenSIPS eBootcamp, Oct 16-27 | www.opensips.org/training > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users From osas at voipembedded.com Wed Oct 4 20:42:43 2023 From: osas at voipembedded.com (Ovidiu Sas) Date: Wed, 4 Oct 2023 16:42:43 -0400 Subject: [OpenSIPS-Users] b2b_logic - add/modifiy custom header on script_req_route/script_reply_route In-Reply-To: <1a1b1026.b5.186023536dd.Coremail.feiyingcheung@126.com> References: <1a1b1026.b5.186023536dd.Coremail.feiyingcheung@126.com> Message-ID: Take a look at: https://opensips.org/docs/modules/3.4.x/b2b_logic.html#func_b2b_server_new https://opensips.org/docs/modules/3.4.x/b2b_logic.html#func_b2b_client_new Check the extra_hdrs and extra_hdr_bodies params. -ovidiu On Thu, Feb 16, 2023 at 2:00 AM Eagle Cheung wrote: > > Hi, > I want to know how to add/modifiy custom header on script_req_route/script_reply_route. > > cfg file statement > ------------------------------------------ > loadmodule "b2b_logic.so" > modparam("b2b_logic", "script_req_route", "b2b_request") > modparam("b2b_logic", "script_reply_route", "b2b_reply") > modparam("b2b_logic", "custom_headers", "P-Local-Header") > > rotue { > ... > > if (is_method("INVITE") && !has_totag()) { > b2b_server_new("caller"); > b2b_client_new("media", "sip:1234 at 192.168.216.1"); > # !!!!!!!!!!!!!work fine, add header succeed. > append_hf("P-Local-Header: test1\r\n"); > b2b_init_request("base"); > exit; > } > > ... > } > > route[b2b_request] { > # !!!!!!!!!!!!!!!!dosen't work, could not to add. > append_hf("P-Local-Header: test2\r\n"); > b2b_pass_request(); > } > > route[b2b_reply] { > # !!!!!!!!!!!!!!dosen't work, could not to add. > append_hf("P-Local-Header: test2\r\n"); > b2b_handle_reply(); > } > -------------------------------------- > looking forward your reply! > BR > Sam Cheung > > > > > _______________________________________________ > 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 razvan at opensips.org Thu Oct 5 10:59:56 2023 From: razvan at opensips.org (=?UTF-8?Q?R=C4=83zvan_Crainea?=) Date: Thu, 5 Oct 2023 13:59:56 +0300 Subject: [OpenSIPS-Users] [Release Freeze] Upcoming OpenSIPS 3.4.2, 3.3.8 and 3.2.15 Minor Releases In-Reply-To: References: <4c93c90c-8bba-f08a-16fb-eeee25fe8a40@opensips.org> Message-ID: <7a53a162-6318-4a0e-80a1-ad0ffe583fd6@opensips.org> We don't know, as we have never profiled 2.4 to see how it behaves. But this[1] is how OpenSIPS 3.4 performes. [1] https://blog.opensips.org/2023/05/10/stress-testing-opensips-3-4-lts/ Best regards, Răzvan Crainea OpenSIPS Core Developer / SIPhub CTO http://www.opensips-solutions.com / https://www.siphub.com On 10/4/23 23:19, Saint Michael wrote: > Question: > is there a significant speed difference in servicing events from > version 2.4 to 3.4? > > On Wed, Oct 4, 2023 at 1:19 PM Liviu Chircu wrote: >> >> Hi, everyone! >> >> The 3.4.2, 3.3.8 and 3.2.15 OpenSIPS minor versions are scheduled for release on Wednesday, Oct 18th. >> >> In preparation for the releases, starting Monday, Oct 9th, we will impose the usual freeze on any significant fixes (as complexity) on these stable branches, in order to ensure a safe window for testing in the days ahead. >> >> Happy testing, >> >> -- >> Liviu Chircu >> www.twitter.com/liviuchircu | www.opensips-solutions.com >> OpenSIPS eBootcamp, Oct 16-27 | www.opensips.org/training >> >> _______________________________________________ >> 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 liviu at opensips.org Thu Oct 5 15:48:07 2023 From: liviu at opensips.org (Liviu Chircu) Date: Thu, 5 Oct 2023 18:48:07 +0300 Subject: [OpenSIPS-Users] =?utf-8?q?=5BOpenSIPS_Blog=5D_Thoughts_after_Our?= =?utf-8?q?_Participation_in_OpenSIPIt=E2=80=9903?= Message-ID: Hi all, The OpenSIPS project recently participated in the OpenSIPIt online testing event, so we wrote a blog piece summing up how the events unfolded: https://blog.opensips.org/2023/10/05/thoughts-after-our-participation-in-opensipit03/ Please enjoy! -- Liviu Chircu From olle at zaark.com Fri Oct 6 10:10:27 2023 From: olle at zaark.com (olle at zaark.com) Date: Fri, 6 Oct 2023 12:10:27 +0200 Subject: [OpenSIPS-Users] rtpengine and multiple instances Message-ID: <013401d9f83d$54fc61a0$fef524e0$@zaark.com> Hi we have seen an issue with rtpengine module in 2.4. We ran a setup with two opensips edge proxies each having a local rtpengine running on it. When we start using the rtpengines in a cluster, we saw that some commands perhaps in 1-2% of the calls are send to the wrong rtpengine: e.g rtpengine_offer() is send to rtpengine instance 1 but rtpengine_anser() is send to instant 2 both from same opensips instance. I wonder if this is a known issue or what might cause this? Our solution for know is to only use the local rtpengine , but I would like the cluster up for redundancy, and this leads to my next question: Can you configure with fifo commands so a node is enabled but have weight 0 , and is only used in case the primary node fails? BR/Olle -------------- next part -------------- An HTML attachment was scrubbed... URL: From razvan at opensips.org Fri Oct 6 11:13:29 2023 From: razvan at opensips.org (=?UTF-8?Q?R=C4=83zvan_Crainea?=) Date: Fri, 6 Oct 2023 14:13:29 +0300 Subject: [OpenSIPS-Users] rtpengine and multiple instances In-Reply-To: <013401d9f83d$54fc61a0$fef524e0$@zaark.com> References: <013401d9f83d$54fc61a0$fef524e0$@zaark.com> Message-ID: Hi, Olle! Yes, the offer&answer should be taken by one single node. How it internally works is we compute a hash of the callid of the call and based on that hash, and we consider that hash a random value to pick one available server. This means that as long as the available list does not change, the same server will be always picked. If however the list changes, even the order of the elements in the list change, a different node might get chosen. This has been fixed in OpenSIPS 3.2 by the rtp_relay module[1]. Setting a 0 weight for a node will do what you said - will be used only in case all the other non-0 weighted nodes were tried. All you need to do is to provision in the database, or in the script, the weights. [1] https://opensips.org/docs/modules/3.2.x/rtp_relay.html Best regards, Răzvan Crainea OpenSIPS Core Developer / SIPhub CTO http://www.opensips-solutions.com / https://www.siphub.com On 10/6/23 13:10, olle at zaark.com wrote: > Hi we have seen an issue with rtpengine module in 2.4. > > We ran a setup with two  opensips edge proxies  each having a  local > rtpengine running on it. > > When we start using the rtpengines in a cluster, we saw that some > commands perhaps in  1-2% of the calls are send to the wrong rtpengine: > > e.g rtpengine_offer() is send to rtpengine instance 1 but > rtpengine_anser() is send to instant 2 both from same opensips instance. > > I wonder if this is a known issue or what might cause this? > > Our solution for know is to only use the local rtpengine , but I would > like the cluster up for redundancy, and this leads to my next question: > > Can you configure with fifo commands so a node is enabled but have > weight 0 , and is only used in case the primary node fails? > > BR/Olle > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users From olle at zaark.com Fri Oct 6 13:44:34 2023 From: olle at zaark.com (olle at zaark.com) Date: Fri, 6 Oct 2023 15:44:34 +0200 Subject: [OpenSIPS-Users] rtpengine and multiple instances In-Reply-To: References: <013401d9f83d$54fc61a0$fef524e0$@zaark.com> Message-ID: <015501d9f85b$3e118bf0$ba34a3d0$@zaark.com> Thanks, sounds like it's time to upgrade 😊 BR/OLle -----Original Message----- From: Users On Behalf Of Razvan Crainea Sent: den 6 oktober 2023 13:13 To: users at lists.opensips.org Subject: Re: [OpenSIPS-Users] rtpengine and multiple instances Hi, Olle! Yes, the offer&answer should be taken by one single node. How it internally works is we compute a hash of the callid of the call and based on that hash, and we consider that hash a random value to pick one available server. This means that as long as the available list does not change, the same server will be always picked. If however the list changes, even the order of the elements in the list change, a different node might get chosen. This has been fixed in OpenSIPS 3.2 by the rtp_relay module[1]. Setting a 0 weight for a node will do what you said - will be used only in case all the other non-0 weighted nodes were tried. All you need to do is to provision in the database, or in the script, the weights. [1] https://opensips.org/docs/modules/3.2.x/rtp_relay.html Best regards, Răzvan Crainea OpenSIPS Core Developer / SIPhub CTO http://www.opensips-solutions.com / https://www.siphub.com On 10/6/23 13:10, olle at zaark.com wrote: > Hi we have seen an issue with rtpengine module in 2.4. > > We ran a setup with two opensips edge proxies each having a local > rtpengine running on it. > > When we start using the rtpengines in a cluster, we saw that some > commands perhaps in 1-2% of the calls are send to the wrong rtpengine: > > e.g rtpengine_offer() is send to rtpengine instance 1 but > rtpengine_anser() is send to instant 2 both from same opensips instance. > > I wonder if this is a known issue or what might cause this? > > Our solution for know is to only use the local rtpengine , but I would > like the cluster up for redundancy, and this leads to my next question: > > Can you configure with fifo commands so a node is enabled but have > weight 0 , and is only used in case the primary node fails? > > BR/Olle > > > _______________________________________________ > 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 johan at democon.be Fri Oct 6 14:28:46 2023 From: johan at democon.be (johan) Date: Fri, 6 Oct 2023 16:28:46 +0200 Subject: [OpenSIPS-Users] on routing invite's in a trunk context. Message-ID: <1724e402-3a9c-4095-9c53-ef14550e6724@democon.be> This is a general question on routing (to be honest: it is a really strange case). A has a sip trunk to B (opensips), B has a sip trunk to C A sends an invite to B with contact header A'   and record-route header to A'' B string the record_route header , calls topology hiding and droutes the call to C C sends 200 OK back but it needs to be routed on the content of the contact header. Now how does this route to A ?   Is this the default topology_hiding_match() case ? Or do I need to do something special ? From Ben.Newlin at genesys.com Fri Oct 6 15:35:40 2023 From: Ben.Newlin at genesys.com (Ben Newlin) Date: Fri, 6 Oct 2023 15:35:40 +0000 Subject: [OpenSIPS-Users] on routing invite's in a trunk context. In-Reply-To: <1724e402-3a9c-4095-9c53-ef14550e6724@democon.be> References: <1724e402-3a9c-4095-9c53-ef14550e6724@democon.be> Message-ID: Johan, This is actually a pretty standard SIP flow that we use all the time. I recommend this article [1] for an overview of how routing works in SIP. In short, the Contact and Record-Route headers are only used for routing of requests not responses, and mostly only sequential requests which are requests within a dialog. The topology_hiding_match and loose_route functions both operate on requests only, so they can both only be called from a request_route [1] [2]. The 200 OK is a response within the initial INVITE transaction. Responses within a SIP transaction are routed based on the source IP of request and/or the Via headers. OpenSIPS should just handle that for you, unless you have some strange routing. [1] https://kb.smartvox.co.uk/opensips/contact-and-record-route-headers-explained/ [2] https://opensips.org/docs/modules/3.2.x/topology_hiding.html#func_topology_hiding_match [3] https://opensips.org/docs/modules/3.2.x/rr.html#func_loose_route Ben Newlin From: Users on behalf of johan Date: Friday, October 6, 2023 at 10:30 AM To: OpenSIPS users mailling list Subject: [OpenSIPS-Users] on routing invite's in a trunk context. EXTERNAL EMAIL - Please use caution with links and attachments This is a general question on routing (to be honest: it is a really strange case). A has a sip trunk to B (opensips), B has a sip trunk to C A sends an invite to B with contact header A' and record-route header to A'' B string the record_route header , calls topology hiding and droutes the call to C C sends 200 OK back but it needs to be routed on the content of the contact header. Now how does this route to A ? Is this the default topology_hiding_match() case ? Or do I need to do something special ? _______________________________________________ Users mailing 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 Sat Oct 7 05:38:29 2023 From: johan at democon.be (Johan De Clercq) Date: Sat, 7 Oct 2023 05:38:29 +0000 Subject: [OpenSIPS-Users] on routing invite's in a trunk context. In-Reply-To: References: <1724e402-3a9c-4095-9c53-ef14550e6724@democon.be> Message-ID: It’s a very non rfc scenario. Source ip and via are the ip of a loadbalancer that I need to skip …. ;-(. The contact does have the correct ip. Anyway, you gave me a good hint. I will add the contact ip as record route and see what that gives. Verzonden vanuit Outlook voor iOS ________________________________ Van: Users namens Ben Newlin Verzonden: Friday, October 6, 2023 5:35:40 PM Aan: OpenSIPS users mailling list Onderwerp: Re: [OpenSIPS-Users] on routing invite's in a trunk context. Johan, This is actually a pretty standard SIP flow that we use all the time. I recommend this article [1] for an overview of how routing works in SIP. In short, the Contact and Record-Route headers are only used for routing of requests not responses, and mostly only sequential requests which are requests within a dialog. The topology_hiding_match and loose_route functions both operate on requests only, so they can both only be called from a request_route [1] [2]. The 200 OK is a response within the initial INVITE transaction. Responses within a SIP transaction are routed based on the source IP of request and/or the Via headers. OpenSIPS should just handle that for you, unless you have some strange routing. [1] https://kb.smartvox.co.uk/opensips/contact-and-record-route-headers-explained/ [2] https://opensips.org/docs/modules/3.2.x/topology_hiding.html#func_topology_hiding_match [3] https://opensips.org/docs/modules/3.2.x/rr.html#func_loose_route Ben Newlin From: Users on behalf of johan Date: Friday, October 6, 2023 at 10:30 AM To: OpenSIPS users mailling list Subject: [OpenSIPS-Users] on routing invite's in a trunk context. EXTERNAL EMAIL - Please use caution with links and attachments This is a general question on routing (to be honest: it is a really strange case). A has a sip trunk to B (opensips), B has a sip trunk to C A sends an invite to B with contact header A' and record-route header to A'' B string the record_route header , calls topology hiding and droutes the call to C C sends 200 OK back but it needs to be routed on the content of the contact header. Now how does this route to A ? Is this the default topology_hiding_match() case ? Or do I need to do something special ? _______________________________________________ Users mailing list Users at lists.opensips.org https://protect-us.mimecast.com/s/8fzVCG6Q9Ni160o8YuKJzbj?domain=lists.opensips.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From spanda at 3clogic.com Mon Oct 9 12:45:20 2023 From: spanda at 3clogic.com (Sasmita Panda) Date: Mon, 9 Oct 2023 18:15:20 +0530 Subject: [OpenSIPS-Users] Need to understand the seral forking logic of opensips . Message-ID: Hi , I know this is a very basic question . Still I need some clarification . I have 2 contact for same number , When I place a call Openips give that to 1st if 1st contact reject the call with 486 busy , then opensips try the same call to the 2nd contact without rejecting the call . In the next call for same number , opensips tries to 2nd contact if that rejects the call with 486 it wont try to the 1st contact again . Why is this happening ? route{ if (!serialize_branches(1)){ sl_send_reply("500","Unable to load contacts"); exit; }else{ if (next_branches()){ t_on_failure("1"); } } } failure_route[1] { if ( t_check_status("404|477|480|481|408|486|50[234]")){ if (next_branches()) { t_on_failure("1"); route(1); } } } This is in my config file . i have gone through the doc https://www.opensips.org/Documentation/Script-CoreFunctions-1-11#toc44 but not able to understand properly . How can I print the internally stored branches on avp ? *Thanks & Regards* *Sasmita Panda* *Senior Network Testing and Software Engineer* *3CLogic , ph:07827611765* -------------- next part -------------- An HTML attachment was scrubbed... URL: From mickael at winlux.fr Mon Oct 9 13:59:01 2023 From: mickael at winlux.fr (Mickael Hubert) Date: Mon, 9 Oct 2023 15:59:01 +0200 Subject: [OpenSIPS-Users] t_new_request and X-headers Message-ID: Hi all, I need to send an INFO message with X-headers from other dialog. I use t_new_request to send my request (apart from first dialog): OK I use local_route to add fixed X-header: OK But I want to copy some information from the first dialog to the new dialog (created by t_new_request). How can I do that please ? Thanks in advance -------------- next part -------------- An HTML attachment was scrubbed... URL: From Ben.Newlin at genesys.com Mon Oct 9 14:05:23 2023 From: Ben.Newlin at genesys.com (Ben Newlin) Date: Mon, 9 Oct 2023 14:05:23 +0000 Subject: [OpenSIPS-Users] on routing invite's in a trunk context. In-Reply-To: References: <1724e402-3a9c-4095-9c53-ef14550e6724@democon.be> Message-ID: Johan, I think you must have left some pieces out of your initial description then. You stated that node A is sending directly to your OpenSIPS node, so the source IP and Via would both point to node A. As I mentioned, changes to Contact and Record-Route headers will only affect the routing of sequential requests for the dialog. They will not impact the routing of responses. I don’t think that OpenSIPS provides any mechanism to change the destination of responses it proxies, since as you said this is outside of the RFC. Ben Newlin From: Users on behalf of Johan De Clercq Date: Saturday, October 7, 2023 at 1:40 AM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] on routing invite's in a trunk context. EXTERNAL EMAIL - Please use caution with links and attachments ________________________________ It’s a very non rfc scenario. Source ip and via are the ip of a loadbalancer that I need to skip …. ;-(. The contact does have the correct ip. Anyway, you gave me a good hint. I will add the contact ip as record route and see what that gives. Verzonden vanuit Outlook voor iOS ________________________________ Van: Users namens Ben Newlin Verzonden: Friday, October 6, 2023 5:35:40 PM Aan: OpenSIPS users mailling list Onderwerp: Re: [OpenSIPS-Users] on routing invite's in a trunk context. Johan, This is actually a pretty standard SIP flow that we use all the time. I recommend this article [1] for an overview of how routing works in SIP. In short, the Contact and Record-Route headers are only used for routing of requests not responses, and mostly only sequential requests which are requests within a dialog. The topology_hiding_match and loose_route functions both operate on requests only, so they can both only be called from a request_route [1] [2]. The 200 OK is a response within the initial INVITE transaction. Responses within a SIP transaction are routed based on the source IP of request and/or the Via headers. OpenSIPS should just handle that for you, unless you have some strange routing. [1] https://kb.smartvox.co.uk/opensips/contact-and-record-route-headers-explained/ [2] https://opensips.org/docs/modules/3.2.x/topology_hiding.html#func_topology_hiding_match [3] https://opensips.org/docs/modules/3.2.x/rr.html#func_loose_route Ben Newlin From: Users on behalf of johan Date: Friday, October 6, 2023 at 10:30 AM To: OpenSIPS users mailling list Subject: [OpenSIPS-Users] on routing invite's in a trunk context. EXTERNAL EMAIL - Please use caution with links and attachments This is a general question on routing (to be honest: it is a really strange case). A has a sip trunk to B (opensips), B has a sip trunk to C A sends an invite to B with contact header A' and record-route header to A'' B string the record_route header , calls topology hiding and droutes the call to C C sends 200 OK back but it needs to be routed on the content of the contact header. Now how does this route to A ? Is this the default topology_hiding_match() case ? Or do I need to do something special ? _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From Ben.Newlin at genesys.com Mon Oct 9 14:07:58 2023 From: Ben.Newlin at genesys.com (Ben Newlin) Date: Mon, 9 Oct 2023 14:07:58 +0000 Subject: [OpenSIPS-Users] Need to understand the seral forking logic of opensips . In-Reply-To: References: Message-ID: Sasmita, I think the relevant piece here is not the call to “serialize_branches”, but how the branches are being created to begin with. Are you using drouting, dispatcher, registrar, etc? It seems like what is happening is that whatever mechanism you are using to track the contacts is considering the 486 response to mean that URI should not be tried again and it is removed from routing. This behavior is likely configurable in that module. Ben Newlin From: Users on behalf of Sasmita Panda Date: Monday, October 9, 2023 at 8:47 AM To: OpenSIPS users mailling list Subject: [OpenSIPS-Users] Need to understand the seral forking logic of opensips . EXTERNAL EMAIL - Please use caution with links and attachments ________________________________ Hi , I know this is a very basic question . Still I need some clarification . I have 2 contact for same number , When I place a call Openips give that to 1st if 1st contact reject the call with 486 busy , then opensips try the same call to the 2nd contact without rejecting the call . In the next call for same number , opensips tries to 2nd contact if that rejects the call with 486 it wont try to the 1st contact again . Why is this happening ? route{ if (!serialize_branches(1)){ sl_send_reply("500","Unable to load contacts"); exit; }else{ if (next_branches()){ t_on_failure("1"); } } } failure_route[1] { if ( t_check_status("404|477|480|481|408|486|50[234]")){ if (next_branches()) { t_on_failure("1"); route(1); } } } This is in my config file . i have gone through the doc https://www.opensips.org/Documentation/Script-CoreFunctions-1-11#toc44 but not able to understand properly . How can I print the internally stored branches on avp ? Thanks & Regards Sasmita Panda Senior Network Testing and Software Engineer 3CLogic , ph:07827611765 -------------- next part -------------- An HTML attachment was scrubbed... URL: From Ben.Newlin at genesys.com Mon Oct 9 14:11:22 2023 From: Ben.Newlin at genesys.com (Ben Newlin) Date: Mon, 9 Oct 2023 14:11:22 +0000 Subject: [OpenSIPS-Users] t_new_request and X-headers In-Reply-To: References: Message-ID: Mickael, There are a few functions provided by the dialog module that allow you to get values from a dialog that isn’t already loaded, depending on where is the information you need. I recommend load_dialog_ctx as it is the easiest to use. https://opensips.org/docs/modules/3.2.x/dialog.html#func_get_dialog_info https://opensips.org/docs/modules/3.2.x/dialog.html#func_get_dialog_vals https://opensips.org/docs/modules/3.2.x/dialog.html#func_get_dialog_by_val https://opensips.org/docs/modules/3.2.x/dialog.html#func_get_dialog_by_profile https://opensips.org/docs/modules/3.2.x/dialog.html#func_load_dialog_ctx Ben Newlin From: Users on behalf of Mickael Hubert Date: Monday, October 9, 2023 at 10:00 AM To: OpenSIPS users mailling list Subject: [OpenSIPS-Users] t_new_request and X-headers EXTERNAL EMAIL - Please use caution with links and attachments ________________________________ Hi all, I need to send an INFO message with X-headers from other dialog. I use t_new_request to send my request (apart from first dialog): OK I use local_route to add fixed X-header: OK But I want to copy some information from the first dialog to the new dialog (created by t_new_request). How can I do that please ? Thanks in advance -------------- next part -------------- An HTML attachment was scrubbed... URL: From Johan at democon.be Mon Oct 9 14:31:13 2023 From: Johan at democon.be (Johan De Clercq) Date: Mon, 9 Oct 2023 16:31:13 +0200 Subject: [OpenSIPS-Users] t_new_request and X-headers In-Reply-To: References: Message-ID: write the info to a custom table and fetch it from there ? Op ma 9 okt 2023 om 16:03 schreef Mickael Hubert : > Hi all, > I need to send an INFO message with X-headers from other dialog. > I use t_new_request to send my request (apart from first dialog): OK > I use local_route to add fixed X-header: OK > But I want to copy some information from the first dialog to the new > dialog (created by t_new_request). How can I do that please ? > > Thanks in advance > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -------------- next part -------------- An HTML attachment was scrubbed... URL: From Johan at democon.be Mon Oct 9 14:31:34 2023 From: Johan at democon.be (Johan De Clercq) Date: Mon, 9 Oct 2023 16:31:34 +0200 Subject: [OpenSIPS-Users] t_new_request and X-headers In-Reply-To: References: Message-ID: avp_dbquery if I am not wrong. Op ma 9 okt 2023 om 16:31 schreef Johan De Clercq : > write the info to a custom table and fetch it from there ? > > > Op ma 9 okt 2023 om 16:03 schreef Mickael Hubert : > >> Hi all, >> I need to send an INFO message with X-headers from other dialog. >> I use t_new_request to send my request (apart from first dialog): OK >> I use local_route to add fixed X-header: OK >> But I want to copy some information from the first dialog to the new >> dialog (created by t_new_request). How can I do that please ? >> >> Thanks in advance >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: From mickael at winlux.fr Mon Oct 9 14:58:22 2023 From: mickael at winlux.fr (Mickael Hubert) Date: Mon, 9 Oct 2023 16:58:22 +0200 Subject: [OpenSIPS-Users] t_new_request and X-headers In-Reply-To: References: Message-ID: Thanks for your help, I already tested avp, but the main issue now is, how can I catch call_id (or other id) from my second request (created by t_new_request) to get my variables from the first leg. Le lun. 9 oct. 2023 à 16:33, Johan De Clercq a écrit : > avp_dbquery if I am not wrong. > > Op ma 9 okt 2023 om 16:31 schreef Johan De Clercq : > >> write the info to a custom table and fetch it from there ? >> >> >> Op ma 9 okt 2023 om 16:03 schreef Mickael Hubert : >> >>> Hi all, >>> I need to send an INFO message with X-headers from other dialog. >>> I use t_new_request to send my request (apart from first dialog): OK >>> I use local_route to add fixed X-header: OK >>> But I want to copy some information from the first dialog to the new >>> dialog (created by t_new_request). How can I do that please ? >>> >>> Thanks in advance >>> _______________________________________________ >>> Users mailing list >>> Users at lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >> _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -------------- next part -------------- An HTML attachment was scrubbed... URL: From Ben.Newlin at genesys.com Mon Oct 9 15:09:22 2023 From: Ben.Newlin at genesys.com (Ben Newlin) Date: Mon, 9 Oct 2023 15:09:22 +0000 Subject: [OpenSIPS-Users] t_new_request and X-headers In-Reply-To: References: Message-ID: Do you have the Call-ID from the first leg when you create the second leg with t_new_request? If os, you can use the ctx parameter from that function to make it visible in the new transaction. https://opensips.org/docs/modules/3.4.x/tm.html#func_t_new_request Ben Newlin From: Users on behalf of Mickael Hubert Date: Monday, October 9, 2023 at 11:00 AM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] t_new_request and X-headers EXTERNAL EMAIL - Please use caution with links and attachments ________________________________ Thanks for your help, I already tested avp, but the main issue now is, how can I catch call_id (or other id) from my second request (created by t_new_request) to get my variables from the first leg. Le lun. 9 oct. 2023 à 16:33, Johan De Clercq > a écrit : avp_dbquery if I am not wrong. Op ma 9 okt 2023 om 16:31 schreef Johan De Clercq >: write the info to a custom table and fetch it from there ? Op ma 9 okt 2023 om 16:03 schreef Mickael Hubert >: Hi all, I need to send an INFO message with X-headers from other dialog. I use t_new_request to send my request (apart from first dialog): OK I use local_route to add fixed X-header: OK But I want to copy some information from the first dialog to the new dialog (created by t_new_request). How can I do that please ? Thanks in advance _______________________________________________ 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 spanda at 3clogic.com Tue Oct 10 06:07:04 2023 From: spanda at 3clogic.com (Sasmita Panda) Date: Tue, 10 Oct 2023 11:37:04 +0530 Subject: [OpenSIPS-Users] Need to understand the seral forking logic of opensips . In-Reply-To: References: Message-ID: In this case I am using the registrar module . *Thanks & Regards* *Sasmita Panda* *Senior Network Testing and Software Engineer* *3CLogic , ph:07827611765* On Mon, Oct 9, 2023 at 7:40 PM Ben Newlin wrote: > Sasmita, > > > > I think the relevant piece here is not the call to “serialize_branches”, > but how the branches are being created to begin with. Are you using > drouting, dispatcher, registrar, etc? It seems like what is happening is > that whatever mechanism you are using to track the contacts is considering > the 486 response to mean that URI should not be tried again and it is > removed from routing. This behavior is likely configurable in that module. > > > > Ben Newlin > > > > *From: *Users on behalf of Sasmita > Panda > *Date: *Monday, October 9, 2023 at 8:47 AM > *To: *OpenSIPS users mailling list > *Subject: *[OpenSIPS-Users] Need to understand the seral forking logic of > opensips . > > * EXTERNAL EMAIL - Please use caution with links and attachments * > > > ------------------------------ > > Hi , > > > > I know this is a very basic question . Still I need some clarification . > > > > I have 2 contact for same number , When I place a call Openips give that > to 1st if 1st contact reject the call with 486 busy , then opensips try the > same call to the 2nd contact without rejecting the call . > > > > In the next call for same number , opensips tries to 2nd contact if that > rejects the call with 486 it wont try to the 1st contact again . Why is > this happening ? > > > > route{ > > > > if (!serialize_branches(1)){ > sl_send_reply("500","Unable to load > contacts"); > exit; > }else{ > if (next_branches()){ > t_on_failure("1"); > } > } > > } > > > > failure_route[1] { > > if ( t_check_status("404|477|480|481|408|486|50[234]")){ > if (next_branches()) > { > t_on_failure("1"); > route(1); > } > > } > } > > > > This is in my config file . i have gone through the doc > https://www.opensips.org/Documentation/Script-CoreFunctions-1-11#toc44 > > but not able to understand properly . > > > > How can I print the internally stored branches on avp ? > > > *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 mickael at winlux.fr Tue Oct 10 09:34:45 2023 From: mickael at winlux.fr (Mickael Hubert) Date: Tue, 10 Oct 2023 11:34:45 +0200 Subject: [OpenSIPS-Users] t_new_request and X-headers In-Reply-To: References: Message-ID: Hi, it's perfect, thanks a lot ! I use json to catch and send my data to t_new_request t_new_request("INFO","$tu","$fu","$tu", , "$json(data)"); local_route { if (is_method("INFO")) { $json(data) := $avp(uac_ctx); #xlog("$json(data)\n"); append_hf("X-call-id: $json(data/X-call-id)\r\n"); append_hf("X-rtp-a: $json(data/X-rtp-a)\r\n"); append_hf("X-rtp-b: $json(data/X-rtp-b)\r\n"); } } Le lun. 9 oct. 2023 à 17:13, Ben Newlin a écrit : > Do you have the Call-ID from the first leg when you create the second leg > with t_new_request? If os, you can use the ctx parameter from that function > to make it visible in the new transaction. > > > > https://opensips.org/docs/modules/3.4.x/tm.html#func_t_new_request > > > > Ben Newlin > > > > *From: *Users on behalf of Mickael > Hubert > *Date: *Monday, October 9, 2023 at 11:00 AM > *To: *OpenSIPS users mailling list > *Subject: *Re: [OpenSIPS-Users] t_new_request and X-headers > > * EXTERNAL EMAIL - Please use caution with links and attachments * > > > ------------------------------ > > Thanks for your help, I already tested avp, but the main issue now is, how > can I catch call_id (or other id) from my second request (created by > t_new_request) to get my variables from the first leg. > > > > > > Le lun. 9 oct. 2023 à 16:33, Johan De Clercq a écrit : > > avp_dbquery if I am not wrong. > > > > Op ma 9 okt 2023 om 16:31 schreef Johan De Clercq : > > write the info to a custom table and fetch it from there ? > > > > > > Op ma 9 okt 2023 om 16:03 schreef Mickael Hubert : > > Hi all, > > I need to send an INFO message with X-headers from other dialog. > > I use t_new_request to send my request (apart from first dialog): OK > > I use local_route to add fixed X-header: OK > > But I want to copy some information from the first dialog to the new > dialog (created by t_new_request). How can I do that please ? > > > > Thanks in advance > > _______________________________________________ > 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 callum.guy at x-on.co.uk Tue Oct 10 09:50:37 2023 From: callum.guy at x-on.co.uk (Callum Guy) Date: Tue, 10 Oct 2023 10:50:37 +0100 Subject: [OpenSIPS-Users] Reset branch route Message-ID: Hi All, I can't find a function to reset a branch route. Instead I'm just creating a new empty branch route which I set in the failure route but this seems superfluous. Am I missing something or does it simply not exist? Callum -- Voting link  (it takes less than 20 seconds!) *0333 332 0000  |  x-on.co.uk   |   **      **  |   **Practice Index Reviews * *Our new office address: 22 Riduna Park, Melton IP12 1QT.* X-on is a trading name of X-on Health Ltd a limited company registered in England and Wales. Registered Office : Glebe Farm, Down Street, Dummer, Basingstoke, Hampshire, England RG25 2AD. 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 Tue Oct 10 10:34:10 2023 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Tue, 10 Oct 2023 13:34:10 +0300 Subject: [OpenSIPS-Users] Reset branch route In-Reply-To: References: Message-ID: <80eefb76-6527-7b3c-ce8f-6f7cb3f6e575@opensips.org> Hi Callum, In the old, old times, doing a t_on_branch(0) would have reset it. But when moving to named routes, this ability was somehow lost :(...The internal function still allows it, but the parameter fixup does not. The fix will be really small and quite, if you want to try it - it's just about adding the CMD_PARAM_OPT flag to the parameter in the internal definition of the function. Would you be able to run a test if I'll get you a patch ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com https://www.siphub.com On 10/10/23 12:50 PM, Callum Guy wrote: > Hi All, > > I can't find a function to reset a branch route. Instead I'm just > creating a new empty branch route which I set in the failure route but > this seems superfluous. > > Am I missing something or does it simply not exist? > > Callum > > Voting link >  (it takes > less than 20 seconds!) > > > *^0333 332 0000  | x-on.co.uk   | > _**_^ > **^  | > **^Practice Index Reviews * > > *Our new office address: 22 Riduna Park, Melton IP12 1QT.* > > X-on is a trading name of X-on Health Ltd a limited company registered > in England and Wales. > Registered Office : Glebe Farm, Down Street, Dummer, Basingstoke, > Hampshire, England RG25 2AD. Company Registration No. 2578478. > The information in this e-mail is confidential and for use by the > addressee(s) only. If you are not the intended recipient, please > notify X-on immediately on +44(0)333 332 0000 and delete the > message from your computer. If you are not a named addressee you must > not use, disclose, disseminate, distribute, copy, print or reply to > this email. Views or opinions expressed by an individual > within this email may not necessarily reflect the views of X-on or its > associated companies. Although X-on routinely screens for viruses, > addressees should scan this email and any attachments > for viruses. X-on makes no representation or warranty as to the > absence of viruses in this email or any attachments. > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From callum.guy at x-on.co.uk Tue Oct 10 10:59:57 2023 From: callum.guy at x-on.co.uk (Callum Guy) Date: Tue, 10 Oct 2023 11:59:57 +0100 Subject: [OpenSIPS-Users] Reset branch route In-Reply-To: <80eefb76-6527-7b3c-ce8f-6f7cb3f6e575@opensips.org> References: <80eefb76-6527-7b3c-ce8f-6f7cb3f6e575@opensips.org> Message-ID: Hi Bogdan, Thanks for your response! I wasn't clear if named routes were supported for branch/failure routes on 3.2 (I use them fine for reply routes) having recently had issues with a failure route named 1_LB - perhaps that was due to the leading integer? Unfortunately I'm currently heavily invested in a production project so there isn't much time for testing a non-critical patch. I will be upgrading to 3.4 before the end of the year though and will certainly try to implement then. Similarly thank you for resolving #3216 however I'll also have to wait for the next major upgrade before disabling the probes! I do wish I was more available for testing, my apologies. Callum On Tue, 10 Oct 2023 at 11:34, Bogdan-Andrei Iancu wrote: > Hi Callum, > > In the old, old times, doing a t_on_branch(0) would have reset it. But > when moving to named routes, this ability was somehow lost :(...The > internal function still allows it, but the parameter fixup does not. The > fix will be really small and quite, if you want to try it - it's just about > adding the CMD_PARAM_OPT flag to the parameter in the internal definition > of the function. > > Would you be able to run a test if I'll get you a patch ? > > Regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > https://www.siphub.com > > On 10/10/23 12:50 PM, Callum Guy wrote: > > Hi All, > > I can't find a function to reset a branch route. Instead I'm just creating > a new empty branch route which I set in the failure route but this seems > superfluous. > > Am I missing something or does it simply not exist? > > Callum > > Voting link (it > takes less than 20 seconds!) > > > *0333 332 0000 | x-on.co.uk | ** > > ** | **Practice Index Reviews > * > > *Our new office address: 22 Riduna Park, Melton IP12 1QT.* > > X-on is a trading name of X-on Health Ltd a limited company registered in > England and Wales. > Registered Office : Glebe Farm, Down Street, Dummer, Basingstoke, > Hampshire, England RG25 2AD. Company Registration No. 2578478. > The information in this e-mail is confidential and for use by the > addressee(s) only. If you are not the intended recipient, please notify > X-on immediately on +44(0)333 332 0000 and delete the > message from your computer. If you are not a named addressee you must not > use, disclose, disseminate, distribute, copy, print or reply to this email. Views > or opinions expressed by an individual > within this email may not necessarily reflect the views of X-on or its > associated companies. Although X-on routinely screens for viruses, > addressees should scan this email and any attachments > for viruses. X-on makes no representation or warranty as to the absence of > viruses in this email or any attachments. > > > _______________________________________________ > Users mailing listUsers at lists.opensips.orghttp://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > -- Voting link  (it takes less than 20 seconds!) *0333 332 0000  |  x-on.co.uk   |   **      **  |   **Practice Index Reviews * *Our new office address: 22 Riduna Park, Melton IP12 1QT.* X-on is a trading name of X-on Health Ltd a limited company registered in England and Wales. Registered Office : Glebe Farm, Down Street, Dummer, Basingstoke, Hampshire, England RG25 2AD. 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 Ben.Newlin at genesys.com Tue Oct 10 12:59:16 2023 From: Ben.Newlin at genesys.com (Ben Newlin) Date: Tue, 10 Oct 2023 12:59:16 +0000 Subject: [OpenSIPS-Users] t_new_request and X-headers In-Reply-To: References: Message-ID: Ah I hadn’t even thought of using the context in that way. That’s an excellent solution! Ben Newlin From: Users on behalf of Mickael Hubert Date: Tuesday, October 10, 2023 at 5:37 AM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] t_new_request and X-headers EXTERNAL EMAIL - Please use caution with links and attachments ________________________________ Hi, it's perfect, thanks a lot ! I use json to catch and send my data to t_new_request t_new_request("INFO","$tu","$fu","$tu", , "$json(data)"); local_route { if (is_method("INFO")) { $json(data) := $avp(uac_ctx); #xlog("$json(data)\n"); append_hf("X-call-id: $json(data/X-call-id)\r\n"); append_hf("X-rtp-a: $json(data/X-rtp-a)\r\n"); append_hf("X-rtp-b: $json(data/X-rtp-b)\r\n"); } } Le lun. 9 oct. 2023 à 17:13, Ben Newlin > a écrit : Do you have the Call-ID from the first leg when you create the second leg with t_new_request? If os, you can use the ctx parameter from that function to make it visible in the new transaction. https://opensips.org/docs/modules/3.4.x/tm.html#func_t_new_request Ben Newlin From: Users > on behalf of Mickael Hubert > Date: Monday, October 9, 2023 at 11:00 AM To: OpenSIPS users mailling list > Subject: Re: [OpenSIPS-Users] t_new_request and X-headers EXTERNAL EMAIL - Please use caution with links and attachments ________________________________ Thanks for your help, I already tested avp, but the main issue now is, how can I catch call_id (or other id) from my second request (created by t_new_request) to get my variables from the first leg. Le lun. 9 oct. 2023 à 16:33, Johan De Clercq > a écrit : avp_dbquery if I am not wrong. Op ma 9 okt 2023 om 16:31 schreef Johan De Clercq >: write the info to a custom table and fetch it from there ? Op ma 9 okt 2023 om 16:03 schreef Mickael Hubert >: Hi all, I need to send an INFO message with X-headers from other dialog. I use t_new_request to send my request (apart from first dialog): OK I use local_route to add fixed X-header: OK But I want to copy some information from the first dialog to the new dialog (created by t_new_request). How can I do that please ? Thanks in advance _______________________________________________ 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 darencrew at hotmail.com Tue Oct 10 14:19:32 2023 From: darencrew at hotmail.com (Daren FERREIRA) Date: Tue, 10 Oct 2023 16:19:32 +0200 Subject: [OpenSIPS-Users] Dispatcher ds_next_dst, Call-ID/Tags uniqueness and RFC Message-ID: Hello, I successfully implemented OpenSIPS with dispatcher and failover routes but an upstream provider complains about duplicates requests they receive. After investigations, when a gateway reports an error, like 402, the failure route is triggered and ds_next_dst is used in order to find another gateway and try it. Doing this work as expected, but, for the provider, because none of Call-ID/To Tag/From Tag change, they consider the requests as dupes. My reading of RFC3261 is that it is compliant as the call is always the same and so, there is no need to change anything in none of these parameters. RFC says the cases (401, 413, 415, 416 and 420) where these 3 parameters MUST not change, but nothing tells they have to change in any other 4XX cases. I made many searches on this subject, nothing helped me on this issue. I made searches in OpenSIPS code to find what can eventually trigger changes on any of these parameters, without success. I found some other implementations of failover routes: - using AVP containing several gateways, and calls to append_branch() and serialize_branches() - carrier route module request append_branch but according to my readings, dispatcher already internally implements append_branch and other requested calls and so, they shouldn’t be called manually in script What are your thinkings about the RFC rules about failover requests and Call-ID/To tag/From tag uniqueness requirement? What would you do in order requests not to be identical when trying upstream routes with dispatcher module? Thank you for your comments. Regards From spanda at 3clogic.com Thu Oct 12 11:43:37 2023 From: spanda at 3clogic.com (Sasmita Panda) Date: Thu, 12 Oct 2023 17:13:37 +0530 Subject: [OpenSIPS-Users] Need to understand the seral forking logic of opensips . In-Reply-To: References: Message-ID: Is there any update on this ? How can I make sure each time every contact will get tried ? *Thanks & Regards* *Sasmita Panda* *Senior Network Testing and Software Engineer* *3CLogic , ph:07827611765* On Tue, Oct 10, 2023 at 11:37 AM Sasmita Panda wrote: > In this case I am using the registrar module . > > > *Thanks & Regards* > *Sasmita Panda* > *Senior Network Testing and Software Engineer* > *3CLogic , ph:07827611765* > > > On Mon, Oct 9, 2023 at 7:40 PM Ben Newlin wrote: > >> Sasmita, >> >> >> >> I think the relevant piece here is not the call to “serialize_branches”, >> but how the branches are being created to begin with. Are you using >> drouting, dispatcher, registrar, etc? It seems like what is happening is >> that whatever mechanism you are using to track the contacts is considering >> the 486 response to mean that URI should not be tried again and it is >> removed from routing. This behavior is likely configurable in that module. >> >> >> >> Ben Newlin >> >> >> >> *From: *Users on behalf of Sasmita >> Panda >> *Date: *Monday, October 9, 2023 at 8:47 AM >> *To: *OpenSIPS users mailling list >> *Subject: *[OpenSIPS-Users] Need to understand the seral forking logic >> of opensips . >> >> * EXTERNAL EMAIL - Please use caution with links and attachments * >> >> >> ------------------------------ >> >> Hi , >> >> >> >> I know this is a very basic question . Still I need some clarification . >> >> >> >> I have 2 contact for same number , When I place a call Openips give that >> to 1st if 1st contact reject the call with 486 busy , then opensips try the >> same call to the 2nd contact without rejecting the call . >> >> >> >> In the next call for same number , opensips tries to 2nd contact if that >> rejects the call with 486 it wont try to the 1st contact again . Why is >> this happening ? >> >> >> >> route{ >> >> >> >> if (!serialize_branches(1)){ >> sl_send_reply("500","Unable to load >> contacts"); >> exit; >> }else{ >> if (next_branches()){ >> t_on_failure("1"); >> } >> } >> >> } >> >> >> >> failure_route[1] { >> >> if ( t_check_status("404|477|480|481|408|486|50[234]")){ >> if (next_branches()) >> { >> t_on_failure("1"); >> route(1); >> } >> >> } >> } >> >> >> >> This is in my config file . i have gone through the doc >> https://www.opensips.org/Documentation/Script-CoreFunctions-1-11#toc44 >> >> but not able to understand properly . >> >> >> >> How can I print the internally stored branches on avp ? >> >> >> *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 simon at softnet.si Fri Oct 13 08:43:05 2023 From: simon at softnet.si (Simon Gajski) Date: Fri, 13 Oct 2023 10:43:05 +0200 Subject: [OpenSIPS-Users] Opensips on vmware and multiple CPUs Message-ID: <1d832362-c540-4aff-aaa9-e02e0a396296@softnet.si> Hi I am running one older instance of OpenSIPS (2.1.5 (x86_64/linux)) on VMware 1CPU/1core Lately we have some performance issues with the server so I tried to increase number of  CPU to 2 and 2 cores per CPU In opensips script is set to worker=4 Opensips starts normally after change on vmware, however it doesn't respond to any SIP request anymore. Once I set back to 1CPU/1core it starts responding back to all SIP requests. Not sure where to start digging; vmware, linux or opensips? Gratefull for any tip Thank you BR Simon From bogdan at opensips.org Fri Oct 13 09:07:14 2023 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Fri, 13 Oct 2023 12:07:14 +0300 Subject: [OpenSIPS-Users] Opensips on vmware and multiple CPUs In-Reply-To: <1d832362-c540-4aff-aaa9-e02e0a396296@softnet.si> References: <1d832362-c540-4aff-aaa9-e02e0a396296@softnet.si> Message-ID: <1ffc901f-9cdf-d69c-f2b6-dbea53437886@opensips.org> Hi Simon, You can try to run an "opensipsctl trap" to see what are the opensips processes doing - just to be sure opensips as application is actually listening and ready to handle packages. Best regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com https://www.siphub.com On 10/13/23 11:43 AM, Simon Gajski via Users wrote: > Hi > > I am running one older instance of OpenSIPS (2.1.5 (x86_64/linux)) on > VMware 1CPU/1core > Lately we have some performance issues with the server so I tried to > increase number of  CPU to 2 and 2 cores per CPU > > In opensips script is set to worker=4 > Opensips starts normally after change on vmware, however it doesn't > respond to any SIP request anymore. > Once I set back to 1CPU/1core it starts responding back to all SIP > requests. > > Not sure where to start digging; vmware, linux or opensips? > Gratefull for any tip > > Thank you > > BR > Simon > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users From nutxase at proton.me Mon Oct 16 14:49:55 2023 From: nutxase at proton.me (nutxase) Date: Mon, 16 Oct 2023 14:49:55 +0000 Subject: [OpenSIPS-Users] Handling REFER Message-ID: <9w116U_H-oxo2shVeDScc7Bx551CFRVPsyLSV6aBGA710QAixEpdNCXsBQ2AI51SmsXQPIFUytIKcNIkRzjFKhRbARzcHZGWjQquA5C-uqo=@proton.me> Hi All I am using opensips as a mid_registrar for webrtc and everything is working fine except call transfers, as i understand they use refer, is there anything specific i need to change to get them to work? if you can point me to a module id appreciate it Sent with [Proton Mail](https://proton.me/) secure email. -------------- next part -------------- An HTML attachment was scrubbed... URL: From junkmail at djrance.com Mon Oct 16 21:44:37 2023 From: junkmail at djrance.com (junkmail) Date: Mon, 16 Oct 2023 15:44:37 -0600 Subject: [OpenSIPS-Users] OpenSiips Docker/kubernates configuration Message-ID: Hello. I was working on getting some of my openSips systems running in docker/kubernates. I think I am good except to the point of binding to IP addresses. I am trying to find a good way to have the pod either be passed the Public IP via an Environment variable or using a simple call to get the IP address from within the starting pod. This I can do, but I am not sure how I can get it to dynamically be usable for the Opensips Socket binding or advertised_address configuration variable. I see there is the -l option for the binding but it doesn't seem to allow you to provide the "as" line to specify the public IP address. I tried searching the documentation on how I might be able to use an Environment variable, but didn't find anything that seemed like it would work. Please let me know if there is something I over looked or another way to accomplish getting the advertised address configured. thank you. From Ben.Newlin at genesys.com Mon Oct 16 21:56:43 2023 From: Ben.Newlin at genesys.com (Ben Newlin) Date: Mon, 16 Oct 2023 21:56:43 +0000 Subject: [OpenSIPS-Users] OpenSiips Docker/kubernates configuration In-Reply-To: References: Message-ID: Any sort of templating framework will work for this. We use m4 and do the templating ourselves using Ansible, but as of 3.1 OpenSIPS supports doing the templating for you, with a number of templating engines [1]. [1] - https://www.opensips.org/Documentation/Templating-Config-Files-3-4 Ben Newlin From: Users on behalf of junkmail Date: Monday, October 16, 2023 at 5:46 PM To: users at lists.opensips.org Subject: [OpenSIPS-Users] OpenSiips Docker/kubernates configuration EXTERNAL EMAIL - Please use caution with links and attachments Hello. I was working on getting some of my openSips systems running in docker/kubernates. I think I am good except to the point of binding to IP addresses. I am trying to find a good way to have the pod either be passed the Public IP via an Environment variable or using a simple call to get the IP address from within the starting pod. This I can do, but I am not sure how I can get it to dynamically be usable for the Opensips Socket binding or advertised_address configuration variable. I see there is the -l option for the binding but it doesn't seem to allow you to provide the "as" line to specify the public IP address. I tried searching the documentation on how I might be able to use an Environment variable, but didn't find anything that seemed like it would work. Please let me know if there is something I over looked or another way to accomplish getting the advertised address configured. 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 junkmail at djrance.com Mon Oct 16 22:33:33 2023 From: junkmail at djrance.com (junkmail) Date: Mon, 16 Oct 2023 16:33:33 -0600 Subject: [OpenSIPS-Users] OpenSiips Docker/kubernates configuration In-Reply-To: References: Message-ID: Perfect. Thank you for the pointer not sure why I didn't see it before. 16.10.2023 15:56 に Ben Newlin さんは書きました: > Any sort of templating framework will work for this. We use m4 and do > the templating ourselves using Ansible, but as of 3.1 OpenSIPS > supports doing the templating for you, with a number of templating > engines [1]. > > [1] - > https://www.opensips.org/Documentation/Templating-Config-Files-3-4 [1] > > > Ben Newlin > > From: Users on behalf of junkmail > > Date: Monday, October 16, 2023 at 5:46 PM > To: users at lists.opensips.org > Subject: [OpenSIPS-Users] OpenSiips Docker/kubernates configuration > > EXTERNAL EMAIL - Please use caution with links and attachments > > Hello. I was working on getting some of my openSips systems running > in > docker/kubernates. I think I am good except to the point of binding > to > IP addresses. > > I am trying to find a good way to have the pod either be passed the > Public IP via an Environment variable or using a simple call to get > the > IP address from within the starting pod. This I can do, but I am not > sure how I can get it to dynamically be usable for the Opensips Socket > > binding or advertised_address configuration variable. I see there > is > the -l option for the binding but it doesn't seem to allow you to > provide the "as" line to specify the public IP address. I tried > searching the documentation on how I might be able to use an > Environment > variable, but didn't find anything that seemed like it would work. > > Please let me know if there is something I over looked or another way > to > accomplish getting the advertised address configured. > > thank you. > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > Links: > ------ > [1] https://www.opensips.org/Documentation/Templating-Config-Files-3-4 > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users From spanda at 3clogic.com Tue Oct 17 14:24:14 2023 From: spanda at 3clogic.com (Sasmita Panda) Date: Tue, 17 Oct 2023 19:54:14 +0530 Subject: [OpenSIPS-Users] I need some help in attr matching while forming the Branch . Message-ID: Hi All , My requirements seem complicated to me. I dont know whether I am right or wrong . I am using opensips 1.11 (I have opensips 3.2 as well ) *scenario 1: * 989898 - is registered from 3 different contact with different call-info header . and I saved the call-info header in attr column while saving data . 1 contact : call-info : x=hr,network;y=sales,z=marketing 2nd contact : call-info : x=hr;y=sales,z=marketing 3rd contact : call-info : x=hr,network;y=sales,z=marketing INVITE : call-info : x=hr;y=sales Now while forming a branch for this INVITE , how do I compare this call-info with attr to get all the contact and then do serial forking on this ? *scenario 2: * 989898 - is registered from 3 different contact with different call-info header . and I saved the call-info header in attr column while saving data . 1 contact : call-info : x=hr,network;y=sales,z=marketing 2nd contact : call-info : x=hr;y=sales,z=marketing 3rd contact : call-info : x=hr,network;y=sales,z=marketing INVITE : call-info : call-info : x=hr,network;y=sales,z=marketing In this I am getting only 1 contact where there is exact match . And I am able to gove call to that contact . I am stuck on the 1st scenario . I am thinking of exec module to use on this . But not getting any better examples of how I do this . Please anybody help me out . Any kind of help is appreciated . 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 spanda at 3clogic.com Wed Oct 18 10:31:59 2023 From: spanda at 3clogic.com (Sasmita Panda) Date: Wed, 18 Oct 2023 16:01:59 +0530 Subject: [OpenSIPS-Users] I need some help in attr matching while forming the Branch . In-Reply-To: References: Message-ID: Hi All , Any help on this ? *Thanks & Regards* *Sasmita Panda* *Senior Network Testing and Software Engineer* *3CLogic , ph:07827611765* On Tue, Oct 17, 2023 at 7:54 PM Sasmita Panda wrote: > Hi All , > > My requirements seem complicated to me. I dont know whether I am right or > wrong . > I am using opensips 1.11 (I have opensips 3.2 as well ) > > *scenario 1: * > 989898 - is registered from 3 different contact with different call-info > header . and I saved the call-info header in attr column while saving data > . > > 1 contact : call-info : x=hr,network;y=sales,z=marketing > 2nd contact : call-info : x=hr;y=sales,z=marketing > 3rd contact : call-info : x=hr,network;y=sales,z=marketing > > INVITE : > call-info : x=hr;y=sales > > Now while forming a branch for this INVITE , how do I compare this > call-info with attr to get all the contact and then do serial forking on > this ? > > *scenario 2: * > 989898 - is registered from 3 different contact with different call-info > header . and I saved the call-info header in attr column while saving data > . > > 1 contact : call-info : x=hr,network;y=sales,z=marketing > 2nd contact : call-info : x=hr;y=sales,z=marketing > 3rd contact : call-info : x=hr,network;y=sales,z=marketing > > INVITE : > call-info : call-info : x=hr,network;y=sales,z=marketing > > In this I am getting only 1 contact where there is exact match . And I am > able to gove call to that contact . > > I am stuck on the 1st scenario . I am thinking of exec module to use on > this . But not getting any better examples of how I do this . Please > anybody help me out . > > Any kind of help is appreciated . 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 denys.pozniak at gmail.com Wed Oct 18 12:14:16 2023 From: denys.pozniak at gmail.com (Denys Pozniak) Date: Wed, 18 Oct 2023 14:14:16 +0200 Subject: [OpenSIPS-Users] Setting up the $socket_out and transport for b2b_client_new Message-ID: Hello! My OpenSIPS proxy has several internal interfaces with different transports, how can I specify the protocol type and outgoing socket for a function b2b_client_new? Tried classic methods but it doesn't work: ... $socket_out = "tcp:192.168.100.10:5060"; $avp(b2b_du) = "sip:192.168.100.20:5060;transport=tcp"; b2b_client_new("uac", $ru, $avp(b2b_du)); b2b_server_new("uas"); b2b_init_request("bridging"); ... -- BR, Denys Pozniak -------------- next part -------------- An HTML attachment was scrubbed... URL: From denys.pozniak at gmail.com Wed Oct 18 13:15:22 2023 From: denys.pozniak at gmail.com (Denys Pozniak) Date: Wed, 18 Oct 2023 15:15:22 +0200 Subject: [OpenSIPS-Users] Questions about best practice b2b_logic implementation Message-ID: Hello! I'm trying to implement topology hiding (b2b_logic) in an existing config, but there are points that are not clear to me: - Can I somehow use dialog variables in b2b mode? if not, any alternative? I see they are created for the initial leg (with state=3), but not available in the b2b mode; - How to configure a tracer (proto_hep) in b2b mode? - How to convert transport and select outgoing socket (the question has already been asked in a separate mail)? - How can I add the necessary custom X-headers also for in-dialog requests (for example, for re-INVITE)? It does not work via insert_hf in [script_req_route] neither via function b2b_client_new() as well; - If nodes operate in anycast + clusterer mode, then in what route should function t_anycast_replicate() be called? -- BR, Denys Pozniak -------------- next part -------------- An HTML attachment was scrubbed... URL: From spanda at 3clogic.com Wed Oct 18 13:34:34 2023 From: spanda at 3clogic.com (Sasmita Panda) Date: Wed, 18 Oct 2023 19:04:34 +0530 Subject: [OpenSIPS-Users] I need some information on lookup with user-agent filtering . Message-ID: Hi All , lookup("location","u/phone/i"); # lookup with user-agent filtering The user-agent filtering flag , compares the regular expression given with which column of location table ? Will that only compare the values of User-Agent column ? If I have some data in attr header and I wanted to compare this regular expression with attr column is that possible ? *Thanks & Regards* *Sasmita Panda* *Senior Network Testing and Software Engineer* *3CLogic , ph:07827611765* -------------- next part -------------- An HTML attachment was scrubbed... URL: From Ben.Newlin at genesys.com Wed Oct 18 13:37:43 2023 From: Ben.Newlin at genesys.com (Ben Newlin) Date: Wed, 18 Oct 2023 13:37:43 +0000 Subject: [OpenSIPS-Users] I need some help in attr matching while forming the Branch . In-Reply-To: References: Message-ID: Sasmita, Your examples all contain mixed use of semicolons and commas. Without any description or context, it’s very difficult to tell what or how you expect these to match. Having said that, I recommend making use of OpenSIPS transformations [1]. The parameters list transformations [2] specifically, as they operate on semicolon separated “key=value” pairs, just like you have. Depending on what operation you are expecting with respect to the commas, you may be able to combine that with the string select transformation [3] to get your desired outcome. [1] https://www.opensips.org/Documentation/Script-Tran-3-2 [2] https://www.opensips.org/Documentation/Script-Tran-3-2#toc60 [3] https://www.opensips.org/Documentation/Script-Tran-3-2#toc7 Ben Newlin From: Users on behalf of Sasmita Panda Date: Wednesday, October 18, 2023 at 6:34 AM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] I need some help in attr matching while forming the Branch . EXTERNAL EMAIL - Please use caution with links and attachments ________________________________ Hi All , Any help on this ? Thanks & Regards Sasmita Panda Senior Network Testing and Software Engineer 3CLogic , ph:07827611765 On Tue, Oct 17, 2023 at 7:54 PM Sasmita Panda > wrote: Hi All , My requirements seem complicated to me. I dont know whether I am right or wrong . I am using opensips 1.11 (I have opensips 3.2 as well ) scenario 1: 989898 - is registered from 3 different contact with different call-info header . and I saved the call-info header in attr column while saving data . 1 contact : call-info : x=hr,network;y=sales,z=marketing 2nd contact : call-info : x=hr;y=sales,z=marketing 3rd contact : call-info : x=hr,network;y=sales,z=marketing INVITE : call-info : x=hr;y=sales Now while forming a branch for this INVITE , how do I compare this call-info with attr to get all the contact and then do serial forking on this ? scenario 2: 989898 - is registered from 3 different contact with different call-info header . and I saved the call-info header in attr column while saving data . 1 contact : call-info : x=hr,network;y=sales,z=marketing 2nd contact : call-info : x=hr;y=sales,z=marketing 3rd contact : call-info : x=hr,network;y=sales,z=marketing INVITE : call-info : call-info : x=hr,network;y=sales,z=marketing In this I am getting only 1 contact where there is exact match . And I am able to gove call to that contact . I am stuck on the 1st scenario . I am thinking of exec module to use on this . But not getting any better examples of how I do this . Please anybody help me out . Any kind of help is appreciated . 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 osas at voipembedded.com Wed Oct 18 13:55:56 2023 From: osas at voipembedded.com (Ovidiu Sas) Date: Wed, 18 Oct 2023 09:55:56 -0400 Subject: [OpenSIPS-Users] Questions about best practice b2b_logic implementation In-Reply-To: References: Message-ID: Hello Denys, Dialog module should *never* be mixed with the b2b module. For b2b calls one can use the $b2b_logic.ctx vars [1]. For b2b tracing use the 'b' scope in trace() [2]. There are some issues as not all provisional replies are traced [3]. Custom headers can be set for each b2b_entity during the entity setup via the extra_hdrs and extra_hdrs_bodies [4] [5]. For now, the outgoing socket is set by opensips internally. There were some fixes related to this, make sure that you are using the latest opensips version. -ovidiu [1] https://opensips.org/docs/modules/3.4.x/b2b_logic.html#b2b_logic.ctx [2] https://opensips.org/docs/modules/3.4.x/tracer.html#func_trace [3] https://github.com/OpenSIPS/opensips/issues/3194 [4] https://opensips.org/docs/modules/3.4.x/b2b_logic.html#func_b2b_server_new [5] https://opensips.org/docs/modules/3.4.x/b2b_logic.html#func_b2b_client_new On Wed, Oct 18, 2023 at 9:16 AM Denys Pozniak wrote: > > Hello! > > I'm trying to implement topology hiding (b2b_logic) in an existing config, but there are points that are not clear to me: > - Can I somehow use dialog variables in b2b mode? if not, any alternative? I see they are created for the initial leg (with state=3), but not available in the b2b mode; > - How to configure a tracer (proto_hep) in b2b mode? > - How to convert transport and select outgoing socket (the question has already been asked in a separate mail)? > - How can I add the necessary custom X-headers also for in-dialog requests (for example, for re-INVITE)? It does not work via insert_hf in [script_req_route] neither via function b2b_client_new() as well; > - If nodes operate in anycast + clusterer mode, then in what route should function t_anycast_replicate() be called? > > -- > > BR, > Denys Pozniak > > > _______________________________________________ > 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 denys.pozniak at gmail.com Wed Oct 18 14:49:46 2023 From: denys.pozniak at gmail.com (Denys Pozniak) Date: Wed, 18 Oct 2023 16:49:46 +0200 Subject: [OpenSIPS-Users] Questions about best practice b2b_logic implementation In-Reply-To: References: Message-ID: Thank you for such a cool and detailed answer! But I would like to clarify the following: >Custom headers can be set for each b2b_entity during the entity setup via the extra_hdrs and extra_hdrs_bodies [4] [5]. Do I understand correctly that this only works for initial invites? >For now, the outgoing socket is set by opensips internally. There were some fixes related to this, make sure that you are using the latest opensips version. But I have several sockets and how can I influence the choice? ср, 18 окт. 2023 г. в 15:59, Ovidiu Sas : > Hello Denys, > > Dialog module should *never* be mixed with the b2b module. > For b2b calls one can use the $b2b_logic.ctx vars [1]. > For b2b tracing use the 'b' scope in trace() [2]. > There are some issues as not all provisional replies are traced [3]. > Custom headers can be set for each b2b_entity during the entity setup > via the extra_hdrs and extra_hdrs_bodies [4] [5]. > For now, the outgoing socket is set by opensips internally. There were > some fixes related to this, make sure that you are using the latest > opensips version. > > -ovidiu > > [1] https://opensips.org/docs/modules/3.4.x/b2b_logic.html#b2b_logic.ctx > [2] https://opensips.org/docs/modules/3.4.x/tracer.html#func_trace > [3] https://github.com/OpenSIPS/opensips/issues/3194 > [4] > https://opensips.org/docs/modules/3.4.x/b2b_logic.html#func_b2b_server_new > [5] > https://opensips.org/docs/modules/3.4.x/b2b_logic.html#func_b2b_client_new > > > On Wed, Oct 18, 2023 at 9:16 AM Denys Pozniak > wrote: > > > > Hello! > > > > I'm trying to implement topology hiding (b2b_logic) in an existing > config, but there are points that are not clear to me: > > - Can I somehow use dialog variables in b2b mode? if not, any > alternative? I see they are created for the initial leg (with state=3), but > not available in the b2b mode; > > - How to configure a tracer (proto_hep) in b2b mode? > > - How to convert transport and select outgoing socket (the question has > already been asked in a separate mail)? > > - How can I add the necessary custom X-headers also for in-dialog > requests (for example, for re-INVITE)? It does not work via insert_hf in > [script_req_route] neither via function b2b_client_new() as well; > > - If nodes operate in anycast + clusterer mode, then in what route > should function t_anycast_replicate() be called? > > > > -- > > > > BR, > > Denys Pozniak > > > > > > _______________________________________________ > > Users mailing list > > Users at lists.opensips.org > > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > -- > VoIP Embedded, Inc. > http://www.voipembedded.com > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -- BR, Denys Pozniak -------------- next part -------------- An HTML attachment was scrubbed... URL: From denys.pozniak at gmail.com Wed Oct 18 15:12:04 2023 From: denys.pozniak at gmail.com (Denys Pozniak) Date: Wed, 18 Oct 2023 17:12:04 +0200 Subject: [OpenSIPS-Users] Questions about best practice b2b_logic implementation In-Reply-To: References: Message-ID: and comment please this question if possible: - If nodes operate in anycast + clusterer mode, then in what route should function t_anycast_replicate() be called? ср, 18 окт. 2023 г. в 16:49, Denys Pozniak : > Thank you for such a cool and detailed answer! > But I would like to clarify the following: > > >Custom headers can be set for each b2b_entity during the entity setup via > the extra_hdrs and extra_hdrs_bodies [4] [5]. > Do I understand correctly that this only works for initial invites? > > >For now, the outgoing socket is set by opensips internally. There were > some fixes related to this, make sure that you are using the latest > opensips version. > But I have several sockets and how can I influence the choice? > > ср, 18 окт. 2023 г. в 15:59, Ovidiu Sas : > >> Hello Denys, >> >> Dialog module should *never* be mixed with the b2b module. >> For b2b calls one can use the $b2b_logic.ctx vars [1]. >> For b2b tracing use the 'b' scope in trace() [2]. >> There are some issues as not all provisional replies are traced [3]. >> Custom headers can be set for each b2b_entity during the entity setup >> via the extra_hdrs and extra_hdrs_bodies [4] [5]. >> For now, the outgoing socket is set by opensips internally. There were >> some fixes related to this, make sure that you are using the latest >> opensips version. >> >> -ovidiu >> >> [1] https://opensips.org/docs/modules/3.4.x/b2b_logic.html#b2b_logic.ctx >> [2] https://opensips.org/docs/modules/3.4.x/tracer.html#func_trace >> [3] https://github.com/OpenSIPS/opensips/issues/3194 >> [4] >> https://opensips.org/docs/modules/3.4.x/b2b_logic.html#func_b2b_server_new >> [5] >> https://opensips.org/docs/modules/3.4.x/b2b_logic.html#func_b2b_client_new >> >> >> On Wed, Oct 18, 2023 at 9:16 AM Denys Pozniak >> wrote: >> > >> > Hello! >> > >> > I'm trying to implement topology hiding (b2b_logic) in an existing >> config, but there are points that are not clear to me: >> > - Can I somehow use dialog variables in b2b mode? if not, any >> alternative? I see they are created for the initial leg (with state=3), but >> not available in the b2b mode; >> > - How to configure a tracer (proto_hep) in b2b mode? >> > - How to convert transport and select outgoing socket (the question has >> already been asked in a separate mail)? >> > - How can I add the necessary custom X-headers also for in-dialog >> requests (for example, for re-INVITE)? It does not work via insert_hf in >> [script_req_route] neither via function b2b_client_new() as well; >> > - If nodes operate in anycast + clusterer mode, then in what route >> should function t_anycast_replicate() be called? >> > >> > -- >> > >> > BR, >> > Denys Pozniak >> > >> > >> > _______________________________________________ >> > Users mailing list >> > Users at lists.opensips.org >> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> >> >> -- >> VoIP Embedded, Inc. >> http://www.voipembedded.com >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> > > > -- > > BR, > Denys Pozniak > > > -- BR, Denys Pozniak -------------- next part -------------- An HTML attachment was scrubbed... URL: From rodrigopassos at gmail.com Wed Oct 18 16:40:53 2023 From: rodrigopassos at gmail.com (Rodrigo R Passos) Date: Wed, 18 Oct 2023 13:40:53 -0300 Subject: [OpenSIPS-Users] RLC in 200 OK after BYE Message-ID: Hi, How can i capture 200 OK after BYE (REL) from Interconnection to send RLC on reply message? I'm trying to get the message on loose_route, but without success. Anyone could help me or share with us an example of a route script? BR, Rodrigo Passos -------------- next part -------------- An HTML attachment was scrubbed... URL: From osas at voipembedded.com Wed Oct 18 17:30:35 2023 From: osas at voipembedded.com (Ovidiu Sas) Date: Wed, 18 Oct 2023 13:30:35 -0400 Subject: [OpenSIPS-Users] Questions about best practice b2b_logic implementation In-Reply-To: References: Message-ID: Extra headers are for the initial invite. If you have one socket per transport, then the automatic selection should work ok. If you have multiple sockets per transport, the first one will be used when routing. Hope this helps, Ovidiu On Wed, Oct 18, 2023 at 10:53 Denys Pozniak wrote: > Thank you for such a cool and detailed answer! > But I would like to clarify the following: > > >Custom headers can be set for each b2b_entity during the entity setup via > the extra_hdrs and extra_hdrs_bodies [4] [5]. > Do I understand correctly that this only works for initial invites? > > >For now, the outgoing socket is set by opensips internally. There were > some fixes related to this, make sure that you are using the latest > opensips version. > But I have several sockets and how can I influence the choice? > > ср, 18 окт. 2023 г. в 15:59, Ovidiu Sas : > >> Hello Denys, >> >> Dialog module should *never* be mixed with the b2b module. >> For b2b calls one can use the $b2b_logic.ctx vars [1]. >> For b2b tracing use the 'b' scope in trace() [2]. >> There are some issues as not all provisional replies are traced [3]. >> Custom headers can be set for each b2b_entity during the entity setup >> via the extra_hdrs and extra_hdrs_bodies [4] [5]. >> For now, the outgoing socket is set by opensips internally. There were >> some fixes related to this, make sure that you are using the latest >> opensips version. >> >> -ovidiu >> >> [1] https://opensips.org/docs/modules/3.4.x/b2b_logic.html#b2b_logic.ctx >> [2] https://opensips.org/docs/modules/3.4.x/tracer.html#func_trace >> [3] https://github.com/OpenSIPS/opensips/issues/3194 >> [4] >> https://opensips.org/docs/modules/3.4.x/b2b_logic.html#func_b2b_server_new >> [5] >> https://opensips.org/docs/modules/3.4.x/b2b_logic.html#func_b2b_client_new >> >> >> On Wed, Oct 18, 2023 at 9:16 AM Denys Pozniak >> wrote: >> > >> > Hello! >> > >> > I'm trying to implement topology hiding (b2b_logic) in an existing >> config, but there are points that are not clear to me: >> > - Can I somehow use dialog variables in b2b mode? if not, any >> alternative? I see they are created for the initial leg (with state=3), but >> not available in the b2b mode; >> > - How to configure a tracer (proto_hep) in b2b mode? >> > - How to convert transport and select outgoing socket (the question has >> already been asked in a separate mail)? >> > - How can I add the necessary custom X-headers also for in-dialog >> requests (for example, for re-INVITE)? It does not work via insert_hf in >> [script_req_route] neither via function b2b_client_new() as well; >> > - If nodes operate in anycast + clusterer mode, then in what route >> should function t_anycast_replicate() be called? >> > >> > -- >> > >> > BR, >> > Denys Pozniak >> > >> > >> > _______________________________________________ >> > Users mailing list >> > Users at lists.opensips.org >> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> >> >> -- >> VoIP Embedded, Inc. >> http://www.voipembedded.com >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> > > > -- > > BR, > Denys Pozniak > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -------------- next part -------------- An HTML attachment was scrubbed... URL: From osas at voipembedded.com Wed Oct 18 17:31:42 2023 From: osas at voipembedded.com (Ovidiu Sas) Date: Wed, 18 Oct 2023 13:31:42 -0400 Subject: [OpenSIPS-Users] Questions about best practice b2b_logic implementation In-Reply-To: References: Message-ID: I haven’t experimented with this … so I can’t comment. -ovidiu On Wed, Oct 18, 2023 at 11:13 Denys Pozniak wrote: > and comment please this question if possible: > - If nodes operate in anycast + clusterer mode, then in what route should > function t_anycast_replicate() be called? > > > ср, 18 окт. 2023 г. в 16:49, Denys Pozniak : > >> Thank you for such a cool and detailed answer! >> But I would like to clarify the following: >> >> >Custom headers can be set for each b2b_entity during the entity setup >> via the extra_hdrs and extra_hdrs_bodies [4] [5]. >> Do I understand correctly that this only works for initial invites? >> >> >For now, the outgoing socket is set by opensips internally. There were >> some fixes related to this, make sure that you are using the latest >> opensips version. >> But I have several sockets and how can I influence the choice? >> >> ср, 18 окт. 2023 г. в 15:59, Ovidiu Sas : >> >>> Hello Denys, >>> >>> Dialog module should *never* be mixed with the b2b module. >>> For b2b calls one can use the $b2b_logic.ctx vars [1]. >>> For b2b tracing use the 'b' scope in trace() [2]. >>> There are some issues as not all provisional replies are traced [3]. >>> Custom headers can be set for each b2b_entity during the entity setup >>> via the extra_hdrs and extra_hdrs_bodies [4] [5]. >>> For now, the outgoing socket is set by opensips internally. There were >>> some fixes related to this, make sure that you are using the latest >>> opensips version. >>> >>> -ovidiu >>> >>> [1] https://opensips.org/docs/modules/3.4.x/b2b_logic.html#b2b_logic.ctx >>> [2] https://opensips.org/docs/modules/3.4.x/tracer.html#func_trace >>> [3] https://github.com/OpenSIPS/opensips/issues/3194 >>> [4] >>> https://opensips.org/docs/modules/3.4.x/b2b_logic.html#func_b2b_server_new >>> [5] >>> https://opensips.org/docs/modules/3.4.x/b2b_logic.html#func_b2b_client_new >>> >>> >>> On Wed, Oct 18, 2023 at 9:16 AM Denys Pozniak >>> wrote: >>> > >>> > Hello! >>> > >>> > I'm trying to implement topology hiding (b2b_logic) in an existing >>> config, but there are points that are not clear to me: >>> > - Can I somehow use dialog variables in b2b mode? if not, any >>> alternative? I see they are created for the initial leg (with state=3), but >>> not available in the b2b mode; >>> > - How to configure a tracer (proto_hep) in b2b mode? >>> > - How to convert transport and select outgoing socket (the question >>> has already been asked in a separate mail)? >>> > - How can I add the necessary custom X-headers also for in-dialog >>> requests (for example, for re-INVITE)? It does not work via insert_hf in >>> [script_req_route] neither via function b2b_client_new() as well; >>> > - If nodes operate in anycast + clusterer mode, then in what route >>> should function t_anycast_replicate() be called? >>> > >>> > -- >>> > >>> > BR, >>> > Denys Pozniak >>> > >>> > >>> > _______________________________________________ >>> > Users mailing list >>> > Users at lists.opensips.org >>> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >>> >>> >>> -- >>> VoIP Embedded, Inc. >>> http://www.voipembedded.com >>> >>> _______________________________________________ >>> Users mailing list >>> Users at lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >> >> >> -- >> >> BR, >> Denys Pozniak >> >> >> > > -- > > BR, > Denys Pozniak > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -------------- next part -------------- An HTML attachment was scrubbed... URL: From kwem at gmx.de Wed Oct 18 17:38:57 2023 From: kwem at gmx.de (Karsten Wemheuer) Date: Wed, 18 Oct 2023 19:38:57 +0200 Subject: [OpenSIPS-Users] RLC in 200 OK after BYE In-Reply-To: References: Message-ID: Hi Rodrigo, Am Mittwoch, dem 18.10.2023 um 13:40 -0300 schrieb Rodrigo R Passos: > Hi, > > How can i capture 200 OK after BYE (REL) from Interconnection to send > RLC on reply message? > > I'm trying to get the message on loose_route, but without success. > Anyone could help me or share with us an example of a route script? I think, You should use on_reply_route to get response to the BYE request. Than check status and method with equal to "BYE". HTH, Karsten From rodrigopassos at gmail.com Wed Oct 18 19:54:39 2023 From: rodrigopassos at gmail.com (Rodrigo R Passos) Date: Wed, 18 Oct 2023 16:54:39 -0300 Subject: [OpenSIPS-Users] RLC in 200 OK after BYE In-Reply-To: References: Message-ID: Hi Karsten, We couldn't see 200 OK after BYE in onreply_route. 😭😭 BR, Rodrigo Em qua., 18 de out. de 2023 às 14:41, Karsten Wemheuer escreveu: > Hi Rodrigo, > > Am Mittwoch, dem 18.10.2023 um 13:40 -0300 schrieb Rodrigo R Passos: > > Hi, > > > > How can i capture 200 OK after BYE (REL) from Interconnection to send > > RLC on reply message? > > > > I'm trying to get the message on loose_route, but without success. > > Anyone could help me or share with us an example of a route script? > > I think, You should use on_reply_route to get response to the BYE > request. Than check status and method with equal to "BYE". > > HTH, > > Karsten > > > _______________________________________________ > Users mailing 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 Oct 18 20:04:45 2023 From: Ben.Newlin at genesys.com (Ben Newlin) Date: Wed, 18 Oct 2023 20:04:45 +0000 Subject: [OpenSIPS-Users] RLC in 200 OK after BYE In-Reply-To: References: Message-ID: Are you using the global onreply_route? All replies should end up there. If you are using a named reply route, you will need to “arm” the route during processing of the BYE using t_on_reply. https://www.opensips.org/Documentation/Script-Routes-3-2#toc4 https://opensips.org/docs/modules/3.2.x/tm.html#func_t_on_reply Ben Newlin From: Users on behalf of Rodrigo R Passos Date: Wednesday, October 18, 2023 at 3:56 PM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] RLC in 200 OK after BYE EXTERNAL EMAIL - Please use caution with links and attachments ________________________________ Hi Karsten, We couldn't see 200 OK after BYE in onreply_route. 😭😭 BR, Rodrigo Em qua., 18 de out. de 2023 às 14:41, Karsten Wemheuer > escreveu: Hi Rodrigo, Am Mittwoch, dem 18.10.2023 um 13:40 -0300 schrieb Rodrigo R Passos: > Hi, > > How can i capture 200 OK after BYE (REL) from Interconnection to send > RLC on reply message? > > I'm trying to get the message on loose_route, but without success. > Anyone could help me or share with us an example of a route script? I think, You should use on_reply_route to get response to the BYE request. Than check status and method with equal to "BYE". HTH, Karsten _______________________________________________ Users mailing list Users at lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From rodrigopassos at gmail.com Wed Oct 18 22:27:01 2023 From: rodrigopassos at gmail.com (Rodrigo R Passos) Date: Wed, 18 Oct 2023 19:27:01 -0300 Subject: [OpenSIPS-Users] RLC in 200 OK after BYE In-Reply-To: References: Message-ID: Ben, Thank you. Now, we put RLC The code: onreply_route[global] { if (t_check_status("200")) { xlog("L_INFO","[$ci] - Reply Global - $fU - O: $si - D: $rr - RM: $rm - Code: $T_reply_code - RS: $rs\n"); if ($rm == "BYE") { if ($si == "xxx.xxx.xxx.xxx") { add_isup_part("Release complete"); } } } } SIP/2.0 200 OK CSeq: 342021 BYE P-RTP-Stat: PS=4611,OS=737760,PR=4618,OR=736108,PL=0 Allow: ACK, INVITE, BYE, CANCEL, REGISTER, REFER, OPTIONS, INFO Content-Length: 2 Content-Type: application/ISUP;base=itu-t92+;version=itu-t Content-Disposition:signal;handling=optional Em qua., 18 de out. de 2023 às 17:08, Ben Newlin escreveu: > Are you using the global onreply_route? All replies should end up there. > If you are using a named reply route, you will need to “arm” the route > during processing of the BYE using t_on_reply. > > > > https://www.opensips.org/Documentation/Script-Routes-3-2#toc4 > > https://opensips.org/docs/modules/3.2.x/tm.html#func_t_on_reply > > > > Ben Newlin > > > > *From: *Users on behalf of Rodrigo R > Passos > *Date: *Wednesday, October 18, 2023 at 3:56 PM > *To: *OpenSIPS users mailling list > *Subject: *Re: [OpenSIPS-Users] RLC in 200 OK after BYE > > * EXTERNAL EMAIL - Please use caution with links and attachments * > > > ------------------------------ > > Hi Karsten, > > > > We couldn't see 200 OK after BYE in onreply_route. 😭😭 > > > > BR, > > Rodrigo > > > > Em qua., 18 de out. de 2023 às 14:41, Karsten Wemheuer > escreveu: > > Hi Rodrigo, > > Am Mittwoch, dem 18.10.2023 um 13:40 -0300 schrieb Rodrigo R Passos: > > Hi, > > > > How can i capture 200 OK after BYE (REL) from Interconnection to send > > RLC on reply message? > > > > I'm trying to get the message on loose_route, but without success. > > Anyone could help me or share with us an example of a route script? > > I think, You should use on_reply_route to get response to the BYE > request. Than check status and method with equal to "BYE". > > HTH, > > Karsten > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -------------- next part -------------- An HTML attachment was scrubbed... URL: From Ben.Newlin at genesys.com Thu Oct 19 02:07:36 2023 From: Ben.Newlin at genesys.com (Ben Newlin) Date: Thu, 19 Oct 2023 02:07:36 +0000 Subject: [OpenSIPS-Users] RLC in 200 OK after BYE In-Reply-To: References: Message-ID: Just to clarify, that is not a global reply route. It is a named reply route that just happens to be named “global”. This would still need to be armed by calling t_on_reply(“global”) during processing of the BYE request. A true global onreply_route does not have a name at all. It is defined like: onreply_route { # do some stuff here } This type of reply route will be called for every reply received in the system and does not require any special arming during request processing. Ben Newlin From: Users on behalf of Rodrigo R Passos Date: Wednesday, October 18, 2023 at 6:28 PM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] RLC in 200 OK after BYE EXTERNAL EMAIL - Please use caution with links and attachments ________________________________ Ben, Thank you. Now, we put RLC The code: onreply_route[global] { if (t_check_status("200")) { xlog("L_INFO","[$ci] - Reply Global - $fU - O: $si - D: $rr - RM: $rm - Code: $T_reply_code - RS: $rs\n"); if ($rm == "BYE") { if ($si == "xxx.xxx.xxx.xxx") { add_isup_part("Release complete"); } } } } SIP/2.0 200 OK CSeq: 342021 BYE P-RTP-Stat: PS=4611,OS=737760,PR=4618,OR=736108,PL=0 Allow: ACK, INVITE, BYE, CANCEL, REGISTER, REFER, OPTIONS, INFO Content-Length: 2 Content-Type: application/ISUP;base=itu-t92+;version=itu-t Content-Disposition:signal;handling=optional Em qua., 18 de out. de 2023 às 17:08, Ben Newlin > escreveu: Are you using the global onreply_route? All replies should end up there. If you are using a named reply route, you will need to “arm” the route during processing of the BYE using t_on_reply. https://www.opensips.org/Documentation/Script-Routes-3-2#toc4 https://opensips.org/docs/modules/3.2.x/tm.html#func_t_on_reply Ben Newlin From: Users > on behalf of Rodrigo R Passos > Date: Wednesday, October 18, 2023 at 3:56 PM To: OpenSIPS users mailling list > Subject: Re: [OpenSIPS-Users] RLC in 200 OK after BYE EXTERNAL EMAIL - Please use caution with links and attachments ________________________________ Hi Karsten, We couldn't see 200 OK after BYE in onreply_route. 😭😭 BR, Rodrigo Em qua., 18 de out. de 2023 às 14:41, Karsten Wemheuer > escreveu: Hi Rodrigo, Am Mittwoch, dem 18.10.2023 um 13:40 -0300 schrieb Rodrigo R Passos: > Hi, > > How can i capture 200 OK after BYE (REL) from Interconnection to send > RLC on reply message? > > I'm trying to get the message on loose_route, but without success. > Anyone could help me or share with us an example of a route script? I think, You should use on_reply_route to get response to the BYE request. Than check status and method with equal to "BYE". HTH, Karsten _______________________________________________ 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 denys.pozniak at gmail.com Thu Oct 19 07:11:37 2023 From: denys.pozniak at gmail.com (Denys Pozniak) Date: Thu, 19 Oct 2023 09:11:37 +0200 Subject: [OpenSIPS-Users] Questions about best practice b2b_logic implementation In-Reply-To: References: Message-ID: > If you have one socket per transport, then the automatic selection should work ok. For example, I have 3 external interfaces (sip/udp, sip/tcp, sip/tls) and 1 internal one (sip/udp). If a request comes for an internal one, how will the outgoing routing proceed as per needed transport (eg, this is sip/tls) in b2b? What will OpenSIPS look at ($ru/Route/...)? ср, 18 окт. 2023 г. в 19:34, Ovidiu Sas : > I haven’t experimented with this … so I can’t comment. > > -ovidiu > > On Wed, Oct 18, 2023 at 11:13 Denys Pozniak > wrote: > >> and comment please this question if possible: >> - If nodes operate in anycast + clusterer mode, then in what route should >> function t_anycast_replicate() be called? >> >> >> ср, 18 окт. 2023 г. в 16:49, Denys Pozniak : >> >>> Thank you for such a cool and detailed answer! >>> But I would like to clarify the following: >>> >>> >Custom headers can be set for each b2b_entity during the entity setup >>> via the extra_hdrs and extra_hdrs_bodies [4] [5]. >>> Do I understand correctly that this only works for initial invites? >>> >>> >For now, the outgoing socket is set by opensips internally. There were >>> some fixes related to this, make sure that you are using the latest >>> opensips version. >>> But I have several sockets and how can I influence the choice? >>> >>> ср, 18 окт. 2023 г. в 15:59, Ovidiu Sas : >>> >>>> Hello Denys, >>>> >>>> Dialog module should *never* be mixed with the b2b module. >>>> For b2b calls one can use the $b2b_logic.ctx vars [1]. >>>> For b2b tracing use the 'b' scope in trace() [2]. >>>> There are some issues as not all provisional replies are traced [3]. >>>> Custom headers can be set for each b2b_entity during the entity setup >>>> via the extra_hdrs and extra_hdrs_bodies [4] [5]. >>>> For now, the outgoing socket is set by opensips internally. There were >>>> some fixes related to this, make sure that you are using the latest >>>> opensips version. >>>> >>>> -ovidiu >>>> >>>> [1] >>>> https://opensips.org/docs/modules/3.4.x/b2b_logic.html#b2b_logic.ctx >>>> [2] https://opensips.org/docs/modules/3.4.x/tracer.html#func_trace >>>> [3] https://github.com/OpenSIPS/opensips/issues/3194 >>>> [4] >>>> https://opensips.org/docs/modules/3.4.x/b2b_logic.html#func_b2b_server_new >>>> [5] >>>> https://opensips.org/docs/modules/3.4.x/b2b_logic.html#func_b2b_client_new >>>> >>>> >>>> On Wed, Oct 18, 2023 at 9:16 AM Denys Pozniak >>>> wrote: >>>> > >>>> > Hello! >>>> > >>>> > I'm trying to implement topology hiding (b2b_logic) in an existing >>>> config, but there are points that are not clear to me: >>>> > - Can I somehow use dialog variables in b2b mode? if not, any >>>> alternative? I see they are created for the initial leg (with state=3), but >>>> not available in the b2b mode; >>>> > - How to configure a tracer (proto_hep) in b2b mode? >>>> > - How to convert transport and select outgoing socket (the question >>>> has already been asked in a separate mail)? >>>> > - How can I add the necessary custom X-headers also for in-dialog >>>> requests (for example, for re-INVITE)? It does not work via insert_hf in >>>> [script_req_route] neither via function b2b_client_new() as well; >>>> > - If nodes operate in anycast + clusterer mode, then in what route >>>> should function t_anycast_replicate() be called? >>>> > >>>> > -- >>>> > >>>> > BR, >>>> > Denys Pozniak >>>> > >>>> > >>>> > _______________________________________________ >>>> > Users mailing list >>>> > Users at lists.opensips.org >>>> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>> >>>> >>>> >>>> -- >>>> VoIP Embedded, Inc. >>>> http://www.voipembedded.com >>>> >>>> _______________________________________________ >>>> Users mailing list >>>> Users at lists.opensips.org >>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>> >>> >>> >>> -- >>> >>> BR, >>> Denys Pozniak >>> >>> >>> >> >> -- >> >> BR, >> Denys Pozniak >> >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -- BR, Denys Pozniak -------------- next part -------------- An HTML attachment was scrubbed... URL: From liviu at opensips.org Thu Oct 19 16:11:56 2023 From: liviu at opensips.org (Liviu Chircu) Date: Thu, 19 Oct 2023 19:11:56 +0300 Subject: [OpenSIPS-Users] [Release Freeze] Upcoming OpenSIPS 3.4.2, 3.3.8 and 3.2.15 Minor Releases In-Reply-To: <4c93c90c-8bba-f08a-16fb-eeee25fe8a40@opensips.org> References: <4c93c90c-8bba-f08a-16fb-eeee25fe8a40@opensips.org> Message-ID: Hi, all! A new round of stable minor releases is now available: *3.4.2*, *3.3.8 *and *3.2.15*! Do make sure to schedule an update, as they contain important fixes. Full changelogs: https://opensips.org/pub/opensips/3.4.2/ChangeLog https://opensips.org/pub/opensips/3.3.8/ChangeLog https://opensips.org/pub/opensips/3.2.15/ChangeLog Enjoy! -- Liviu Chircu www.twitter.com/liviuchircu |www.opensips-solutions.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From akogan at 5gfuture.com Thu Oct 19 17:47:55 2023 From: akogan at 5gfuture.com (Alexander Kogan) Date: Thu, 19 Oct 2023 21:47:55 +0400 Subject: [OpenSIPS-Users] [Release Freeze] Upcoming OpenSIPS 3.4.2, 3.3.8 and 3.2.15 Minor Releases In-Reply-To: References: <4c93c90c-8bba-f08a-16fb-eeee25fe8a40@opensips.org> Message-ID: <392b452d-09a3-4dbf-a77d-e24418739266@5gfuture.com> Hi! Failed to build 3.2.15 from opensips-3.2.15.tar.gz: make[3]: *** No rule to make target 'lib/patches/wolfssl-internal-memleak-fix.patch', needed by 'lib/patches/wolfssl-internal-memleak-fix.patched'.  Stop. Best regards, Alexander Kogan, Director of R&D 5g Future http://5gfuture.com On 19.10.2023 20:11, Liviu Chircu wrote: > > Hi, all! > > A new round of stable minor releases is now available: *3.4.2*, *3.3.8 > *and *3.2.15*! > > Do make sure to schedule an update, as they contain important fixes. > > Full changelogs: > > https://opensips.org/pub/opensips/3.4.2/ChangeLog > https://opensips.org/pub/opensips/3.3.8/ChangeLog > https://opensips.org/pub/opensips/3.2.15/ChangeLog > > Enjoy! > > -- > Liviu Chircu > www.twitter.com/liviuchircu |www.opensips-solutions.com > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From osas at voipembedded.com Thu Oct 19 19:52:16 2023 From: osas at voipembedded.com (Ovidiu Sas) Date: Thu, 19 Oct 2023 15:52:16 -0400 Subject: [OpenSIPS-Users] Questions about best practice b2b_logic implementation In-Reply-To: References: Message-ID: If the internal has a private ip and the external has a public ip, then it should work ok. On Thu, Oct 19, 2023 at 03:14 Denys Pozniak wrote: > > If you have one socket per transport, then the automatic selection > should work ok. > For example, I have 3 external interfaces (sip/udp, sip/tcp, sip/tls) and > 1 internal one (sip/udp). > If a request comes for an internal one, how will the outgoing routing > proceed as per needed transport (eg, this is sip/tls) in b2b? What will > OpenSIPS look at ($ru/Route/...)? > > ср, 18 окт. 2023 г. в 19:34, Ovidiu Sas : > >> I haven’t experimented with this … so I can’t comment. >> >> -ovidiu >> >> On Wed, Oct 18, 2023 at 11:13 Denys Pozniak >> wrote: >> >>> and comment please this question if possible: >>> - If nodes operate in anycast + clusterer mode, then in what route >>> should function t_anycast_replicate() be called? >>> >>> >>> ср, 18 окт. 2023 г. в 16:49, Denys Pozniak : >>> >>>> Thank you for such a cool and detailed answer! >>>> But I would like to clarify the following: >>>> >>>> >Custom headers can be set for each b2b_entity during the entity setup >>>> via the extra_hdrs and extra_hdrs_bodies [4] [5]. >>>> Do I understand correctly that this only works for initial invites? >>>> >>>> >For now, the outgoing socket is set by opensips internally. There were >>>> some fixes related to this, make sure that you are using the latest >>>> opensips version. >>>> But I have several sockets and how can I influence the choice? >>>> >>>> ср, 18 окт. 2023 г. в 15:59, Ovidiu Sas : >>>> >>>>> Hello Denys, >>>>> >>>>> Dialog module should *never* be mixed with the b2b module. >>>>> For b2b calls one can use the $b2b_logic.ctx vars [1]. >>>>> For b2b tracing use the 'b' scope in trace() [2]. >>>>> There are some issues as not all provisional replies are traced [3]. >>>>> Custom headers can be set for each b2b_entity during the entity setup >>>>> via the extra_hdrs and extra_hdrs_bodies [4] [5]. >>>>> For now, the outgoing socket is set by opensips internally. There were >>>>> some fixes related to this, make sure that you are using the latest >>>>> opensips version. >>>>> >>>>> -ovidiu >>>>> >>>>> [1] >>>>> https://opensips.org/docs/modules/3.4.x/b2b_logic.html#b2b_logic.ctx >>>>> [2] https://opensips.org/docs/modules/3.4.x/tracer.html#func_trace >>>>> [3] https://github.com/OpenSIPS/opensips/issues/3194 >>>>> [4] >>>>> https://opensips.org/docs/modules/3.4.x/b2b_logic.html#func_b2b_server_new >>>>> [5] >>>>> https://opensips.org/docs/modules/3.4.x/b2b_logic.html#func_b2b_client_new >>>>> >>>>> >>>>> On Wed, Oct 18, 2023 at 9:16 AM Denys Pozniak >>>>> wrote: >>>>> > >>>>> > Hello! >>>>> > >>>>> > I'm trying to implement topology hiding (b2b_logic) in an existing >>>>> config, but there are points that are not clear to me: >>>>> > - Can I somehow use dialog variables in b2b mode? if not, any >>>>> alternative? I see they are created for the initial leg (with state=3), but >>>>> not available in the b2b mode; >>>>> > - How to configure a tracer (proto_hep) in b2b mode? >>>>> > - How to convert transport and select outgoing socket (the question >>>>> has already been asked in a separate mail)? >>>>> > - How can I add the necessary custom X-headers also for in-dialog >>>>> requests (for example, for re-INVITE)? It does not work via insert_hf in >>>>> [script_req_route] neither via function b2b_client_new() as well; >>>>> > - If nodes operate in anycast + clusterer mode, then in what route >>>>> should function t_anycast_replicate() be called? >>>>> > >>>>> > -- >>>>> > >>>>> > BR, >>>>> > Denys Pozniak >>>>> > >>>>> > >>>>> > _______________________________________________ >>>>> > Users mailing list >>>>> > Users at lists.opensips.org >>>>> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>>> >>>>> >>>>> >>>>> -- >>>>> VoIP Embedded, Inc. >>>>> http://www.voipembedded.com >>>>> >>>>> _______________________________________________ >>>>> Users mailing list >>>>> Users at lists.opensips.org >>>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>>> >>>> >>>> >>>> -- >>>> >>>> BR, >>>> Denys Pozniak >>>> >>>> >>>> >>> >>> -- >>> >>> BR, >>> Denys Pozniak >>> >>> >>> _______________________________________________ >>> Users mailing list >>> Users at lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> > > > -- > > BR, > Denys Pozniak > > > _______________________________________________ > Users mailing 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 Fri Oct 20 13:22:36 2023 From: liviu at opensips.org (Liviu Chircu) Date: Fri, 20 Oct 2023 16:22:36 +0300 Subject: [OpenSIPS-Users] [Release Freeze] Upcoming OpenSIPS 3.4.2, 3.3.8 and 3.2.15 Minor Releases In-Reply-To: <392b452d-09a3-4dbf-a77d-e24418739266@5gfuture.com> References: <4c93c90c-8bba-f08a-16fb-eeee25fe8a40@opensips.org> <392b452d-09a3-4dbf-a77d-e24418739266@5gfuture.com> Message-ID: On 19.10.2023 20:47, Alexander Kogan wrote: > > Hi! > > Failed to build 3.2.15 from opensips-3.2.15.tar.gz: > > make[3]: *** No rule to make target > 'lib/patches/wolfssl-internal-memleak-fix.patch', needed by > 'lib/patches/wolfssl-internal-memleak-fix.patched'.  Stop. > Hi, Alexander! Good catch!  I just rebuilt the tarball, as the faulty file (added by accident) has been removed in the meantime. Best regards, -- Liviu Chircu www.twitter.com/liviuchircu | www.opensips-solutions.com From akogan at 5gfuture.com Fri Oct 20 13:23:45 2023 From: akogan at 5gfuture.com (Alexander Kogan) Date: Fri, 20 Oct 2023 17:23:45 +0400 Subject: [OpenSIPS-Users] [Release Freeze] Upcoming OpenSIPS 3.4.2, 3.3.8 and 3.2.15 Minor Releases In-Reply-To: References: <4c93c90c-8bba-f08a-16fb-eeee25fe8a40@opensips.org> <392b452d-09a3-4dbf-a77d-e24418739266@5gfuture.com> Message-ID: <3d88e03e-1d0c-49a4-ad11-83bb698164e0@5gfuture.com> Thank you! Best regards, Alexander Kogan, Director of R&D 5g Future http://5gfuture.com On 20.10.2023 17:22, Liviu Chircu wrote: > On 19.10.2023 20:47, Alexander Kogan wrote: >> >> Hi! >> >> Failed to build 3.2.15 from opensips-3.2.15.tar.gz: >> >> make[3]: *** No rule to make target >> 'lib/patches/wolfssl-internal-memleak-fix.patch', needed by >> 'lib/patches/wolfssl-internal-memleak-fix.patched'.  Stop. >> > Hi, Alexander! > > Good catch!  I just rebuilt the tarball, as the faulty file (added by > accident) has been removed in the meantime. > > Best regards, > From efes99999 at gmail.com Sat Oct 21 14:12:25 2023 From: efes99999 at gmail.com (L S) Date: Sat, 21 Oct 2023 10:12:25 -0400 Subject: [OpenSIPS-Users] [Release Freeze] Upcoming OpenSIPS 3.4.2, 3.3.8 and 3.2.15 Minor Releases In-Reply-To: References: <4c93c90c-8bba-f08a-16fb-eeee25fe8a40@opensips.org> <392b452d-09a3-4dbf-a77d-e24418739266@5gfuture.com> Message-ID: Hi Liviu, Just download the latest tarball. Seems like this issue is still happening. Thanks, Matt On Fri, Oct 20, 2023, 9:25 AM Liviu Chircu wrote: > On 19.10.2023 20:47, Alexander Kogan wrote: > > > > Hi! > > > > Failed to build 3.2.15 from opensips-3.2.15.tar.gz: > > > > make[3]: *** No rule to make target > > 'lib/patches/wolfssl-internal-memleak-fix.patch', needed by > > 'lib/patches/wolfssl-internal-memleak-fix.patched'. Stop. > > > Hi, Alexander! > > Good catch! I just rebuilt the tarball, as the faulty file (added by > accident) has been removed in the meantime. > > Best regards, > > -- > Liviu Chircu > www.twitter.com/liviuchircu | www.opensips-solutions.com > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -------------- next part -------------- An HTML attachment was scrubbed... URL: From efes99999 at gmail.com Sat Oct 21 14:14:01 2023 From: efes99999 at gmail.com (L S) Date: Sat, 21 Oct 2023 10:14:01 -0400 Subject: [OpenSIPS-Users] [Release Freeze] Upcoming OpenSIPS 3.4.2, 3.3.8 and 3.2.15 Minor Releases In-Reply-To: References: <4c93c90c-8bba-f08a-16fb-eeee25fe8a40@opensips.org> <392b452d-09a3-4dbf-a77d-e24418739266@5gfuture.com> Message-ID: Hi Liviu, Just download the latest tarball. Seems like this issue is still happening. Thanks, Matt On Fri, Oct 20, 2023, 9:25 AM Liviu Chircu wrote: > On 19.10.2023 20:47, Alexander Kogan wrote: > > > > Hi! > > > > Failed to build 3.2.15 from opensips-3.2.15.tar.gz: > > > > make[3]: *** No rule to make target > > 'lib/patches/wolfssl-internal-memleak-fix.patch', needed by > > 'lib/patches/wolfssl-internal-memleak-fix.patched'. Stop. > > > Hi, Alexander! > > Good catch! I just rebuilt the tarball, as the faulty file (added by > accident) has been removed in the meantime. > > Best regards, > > -- > Liviu Chircu > www.twitter.com/liviuchircu | www.opensips-solutions.com > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -------------- next part -------------- An HTML attachment was scrubbed... URL: From efes99999 at gmail.com Sun Oct 22 14:03:06 2023 From: efes99999 at gmail.com (L S) Date: Sun, 22 Oct 2023 10:03:06 -0400 Subject: [OpenSIPS-Users] [Release Freeze] Upcoming OpenSIPS 3.4.2, 3.3.8 and 3.2.15 Minor Releases In-Reply-To: References: <4c93c90c-8bba-f08a-16fb-eeee25fe8a40@opensips.org> <392b452d-09a3-4dbf-a77d-e24418739266@5gfuture.com> Message-ID: Hi Liviu, Doesn't that patch (lib/patches/wolfssl-internal-memleak-fix.patch) need to be in 3.2.15? You said you removed it. Thanks, Matt On Fri, Oct 20, 2023, 9:25 AM Liviu Chircu wrote: > On 19.10.2023 20:47, Alexander Kogan wrote: > > > > Hi! > > > > Failed to build 3.2.15 from opensips-3.2.15.tar.gz: > > > > make[3]: *** No rule to make target > > 'lib/patches/wolfssl-internal-memleak-fix.patch', needed by > > 'lib/patches/wolfssl-internal-memleak-fix.patched'. Stop. > > > Hi, Alexander! > > Good catch! I just rebuilt the tarball, as the faulty file (added by > accident) has been removed in the meantime. > > Best regards, > > -- > Liviu Chircu > www.twitter.com/liviuchircu | www.opensips-solutions.com > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -------------- next part -------------- An HTML attachment was scrubbed... URL: From spanda at 3clogic.com Mon Oct 23 06:11:37 2023 From: spanda at 3clogic.com (Sasmita Panda) Date: Mon, 23 Oct 2023 11:41:37 +0530 Subject: [OpenSIPS-Users] I need some help in attr matching while forming the Branch . In-Reply-To: References: Message-ID: There are some changes in the requirements now . Although I appreciate your suggestion, I will go with script transformation . But not in a different way . * In the register I will get set of comma separated values in Call-Info * *header and will save that in attr column . * Agent1: call-info : en,galileo,level1,true Agent2 : call-info : en,spanish,lenovo,true Agent3: call-info : en,galileo,sales,level1,true 1st Invite : *call-info : en,galileo,true* (I have to match the contact in which this the substring of Registered contacts Attr . ex : Agent1 and Agent3 will the matching agents here ) 2nd Invite : *call-info : en,true* (All there agents are matching this critaria ) 3rd Invite : *call-info : spanish,lenovo,true *(Agent2 will match the criteria ) This is kind of substring matching . Here the challenge for me is , the number of values in the Invite header is not fixed , That can be 2 , that can be 4 that can be 7 . If every comma separated value is present in the Attr field then that contact will get call else not . *#Test for existence of string in anotherif ($(var(strtosearch){s.index, $var(str)}) != NULL) xlog("found $var(sstr) in $var(strtosearch)\n");* *So will this help me in this case ? * *Thanks & Regards* *Sasmita Panda* *Senior Network Testing and Software Engineer* *3CLogic , ph:07827611765* On Wed, Oct 18, 2023 at 7:10 PM Ben Newlin wrote: > Sasmita, > > > > Your examples all contain mixed use of semicolons and commas. Without any > description or context, it’s very difficult to tell what or how you expect > these to match. > > > > Having said that, I recommend making use of OpenSIPS transformations [1]. > The parameters list transformations [2] specifically, as they operate on > semicolon separated “key=value” pairs, just like you have. > > > > Depending on what operation you are expecting with respect to the commas, > you may be able to combine that with the string select transformation [3] > to get your desired outcome. > > > > [1] https://www.opensips.org/Documentation/Script-Tran-3-2 > > [2] https://www.opensips.org/Documentation/Script-Tran-3-2#toc60 > > [3] https://www.opensips.org/Documentation/Script-Tran-3-2#toc7 > > > > Ben Newlin > > > > *From: *Users on behalf of Sasmita > Panda > *Date: *Wednesday, October 18, 2023 at 6:34 AM > *To: *OpenSIPS users mailling list > *Subject: *Re: [OpenSIPS-Users] I need some help in attr matching while > forming the Branch . > > * EXTERNAL EMAIL - Please use caution with links and attachments * > > > ------------------------------ > > Hi All , > > > > Any help on this ? > > > > > *Thanks & Regards* > > *Sasmita Panda* > > *Senior Network Testing and Software Engineer* > > *3CLogic , ph:07827611765* > > > > > > On Tue, Oct 17, 2023 at 7:54 PM Sasmita Panda wrote: > > Hi All , > > > > My requirements seem complicated to me. I dont know whether I am right or > wrong . > > I am using opensips 1.11 (I have opensips 3.2 as well ) > > > > *scenario 1: * > > 989898 - is registered from 3 different contact with different call-info > header . and I saved the call-info header in attr column while saving data > . > > > > 1 contact : call-info : x=hr,network;y=sales,z=marketing > > 2nd contact : call-info : x=hr;y=sales,z=marketing > > 3rd contact : call-info : x=hr,network;y=sales,z=marketing > > > > INVITE : > > call-info : x=hr;y=sales > > > > Now while forming a branch for this INVITE , how do I compare this > call-info with attr to get all the contact and then do serial forking on > this ? > > > > *scenario 2: * > > 989898 - is registered from 3 different contact with different call-info > header . and I saved the call-info header in attr column while saving data > . > > > > 1 contact : call-info : x=hr,network;y=sales,z=marketing > > 2nd contact : call-info : x=hr;y=sales,z=marketing > > 3rd contact : call-info : x=hr,network;y=sales,z=marketing > > > > INVITE : > > call-info : call-info : x=hr,network;y=sales,z=marketing > > > > In this I am getting only 1 contact where there is exact match . And I am > able to gove call to that contact . > > > > I am stuck on the 1st scenario . I am thinking of exec module to use on > this . But not getting any better examples of how I do this . Please > anybody help me out . > > > > Any kind of help is appreciated . 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 Ben.Newlin at genesys.com Mon Oct 23 13:28:09 2023 From: Ben.Newlin at genesys.com (Ben Newlin) Date: Mon, 23 Oct 2023 13:28:09 +0000 Subject: [OpenSIPS-Users] I need some help in attr matching while forming the Branch . In-Reply-To: References: Message-ID: Yes, based on your description of the problem that transformation should work. We use that transformation to test a comma-separated list for the existence of a value in the same way. Ben Newlin From: Users on behalf of Sasmita Panda Date: Monday, October 23, 2023 at 2:13 AM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] I need some help in attr matching while forming the Branch . EXTERNAL EMAIL - Please use caution with links and attachments ________________________________ There are some changes in the requirements now . Although I appreciate your suggestion, I will go with script transformation . But not in a different way . In the register I will get set of comma separated values in Call-Info header and will save that in attr column . Agent1: call-info : en,galileo,level1,true Agent2 : call-info : en,spanish,lenovo,true Agent3: call-info : en,galileo,sales,level1,true 1st Invite : call-info : en,galileo,true (I have to match the contact in which this the substring of Registered contacts Attr . ex : Agent1 and Agent3 will the matching agents here ) 2nd Invite : call-info : en,true (All there agents are matching this critaria ) 3rd Invite : call-info : spanish,lenovo,true (Agent2 will match the criteria ) This is kind of substring matching . Here the challenge for me is , the number of values in the Invite header is not fixed , That can be 2 , that can be 4 that can be 7 . If every comma separated value is present in the Attr field then that contact will get call else not . #Test for existence of string in another if ($(var(strtosearch){s.index, $var(str)}) != NULL) xlog("found $var(sstr) in $var(strtosearch)\n"); So will this help me in this case ? Thanks & Regards Sasmita Panda Senior Network Testing and Software Engineer 3CLogic , ph:07827611765 On Wed, Oct 18, 2023 at 7:10 PM Ben Newlin > wrote: Sasmita, Your examples all contain mixed use of semicolons and commas. Without any description or context, it’s very difficult to tell what or how you expect these to match. Having said that, I recommend making use of OpenSIPS transformations [1]. The parameters list transformations [2] specifically, as they operate on semicolon separated “key=value” pairs, just like you have. Depending on what operation you are expecting with respect to the commas, you may be able to combine that with the string select transformation [3] to get your desired outcome. [1] https://www.opensips.org/Documentation/Script-Tran-3-2 [2] https://www.opensips.org/Documentation/Script-Tran-3-2#toc60 [3] https://www.opensips.org/Documentation/Script-Tran-3-2#toc7 Ben Newlin From: Users > on behalf of Sasmita Panda > Date: Wednesday, October 18, 2023 at 6:34 AM To: OpenSIPS users mailling list > Subject: Re: [OpenSIPS-Users] I need some help in attr matching while forming the Branch . EXTERNAL EMAIL - Please use caution with links and attachments ________________________________ Hi All , Any help on this ? Thanks & Regards Sasmita Panda Senior Network Testing and Software Engineer 3CLogic , ph:07827611765 On Tue, Oct 17, 2023 at 7:54 PM Sasmita Panda > wrote: Hi All , My requirements seem complicated to me. I dont know whether I am right or wrong . I am using opensips 1.11 (I have opensips 3.2 as well ) scenario 1: 989898 - is registered from 3 different contact with different call-info header . and I saved the call-info header in attr column while saving data . 1 contact : call-info : x=hr,network;y=sales,z=marketing 2nd contact : call-info : x=hr;y=sales,z=marketing 3rd contact : call-info : x=hr,network;y=sales,z=marketing INVITE : call-info : x=hr;y=sales Now while forming a branch for this INVITE , how do I compare this call-info with attr to get all the contact and then do serial forking on this ? scenario 2: 989898 - is registered from 3 different contact with different call-info header . and I saved the call-info header in attr column while saving data . 1 contact : call-info : x=hr,network;y=sales,z=marketing 2nd contact : call-info : x=hr;y=sales,z=marketing 3rd contact : call-info : x=hr,network;y=sales,z=marketing INVITE : call-info : call-info : x=hr,network;y=sales,z=marketing In this I am getting only 1 contact where there is exact match . And I am able to gove call to that contact . I am stuck on the 1st scenario . I am thinking of exec module to use on this . But not getting any better examples of how I do this . Please anybody help me out . Any kind of help is appreciated . 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 bogdan at opensips.org Mon Oct 23 14:29:02 2023 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Mon, 23 Oct 2023 17:29:02 +0300 Subject: [OpenSIPS-Users] Dispatcher ds_next_dst, Call-ID/Tags uniqueness and RFC In-Reply-To: References: Message-ID: <358e23d7-67c1-41e7-c9db-e865ae3892a9@opensips.org> HI Daren, In SIP there are transactional and dialog layers. Indeed, from dialog level perspective, the Call-ID and the To/FROM tags are uniquely identifying the dialog. But before having a dialog, you need to have a transaction for an initial request creating the dialog. This is the INVITE transaction (when comes to calls). And at the transactional layer, the uniqueness is given by the VIA-branch parameter (see the RFC when comes to transaction matching). Bottom line, during SIP forking, OpenSIPS creates different VIA-branches for each branch, so they further (on the next hop) create different transactions. So there should be no conflict at all - a new incoming branch will create a new transaction (initial one), creating a new dialog. Nevertheless, there is a lot of confusion in this grey area when comes to the SIP stack. To try to address this you may try to change the cseq number for each branch, so they will have different values and not confuse the next hop, like this https://opensips.org/html/docs/modules/3.5.x/uac.html#func_uac_inc_cseq Best regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com https://www.siphub.com On 10/10/23 5:19 PM, Daren FERREIRA wrote: > Hello, > > I successfully implemented OpenSIPS with dispatcher and failover routes but an upstream provider complains about duplicates requests they receive. > > After investigations, when a gateway reports an error, like 402, the failure route is triggered and ds_next_dst is used in order to find another gateway and try it. > > Doing this work as expected, but, for the provider, because none of Call-ID/To Tag/From Tag change, they consider the requests as dupes. > > My reading of RFC3261 is that it is compliant as the call is always the same and so, there is no need to change anything in none of these parameters. > > RFC says the cases (401, 413, 415, 416 and 420) where these 3 parameters MUST not change, but nothing tells they have to change in any other 4XX cases. > > I made many searches on this subject, nothing helped me on this issue. > > I made searches in OpenSIPS code to find what can eventually trigger changes on any of these parameters, without success. > > I found some other implementations of failover routes: > - using AVP containing several gateways, and calls to append_branch() and serialize_branches() > - carrier route module request append_branch > but according to my readings, dispatcher already internally implements append_branch and other requested calls and so, they shouldn’t be called manually in script > > > What are your thinkings about the RFC rules about failover requests and Call-ID/To tag/From tag uniqueness requirement? > > What would you do in order requests not to be identical when trying upstream routes with dispatcher module? > > > Thank you for your comments. > > > Regards > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users From bogdan at opensips.org Mon Oct 23 14:33:47 2023 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Mon, 23 Oct 2023 17:33:47 +0300 Subject: [OpenSIPS-Users] Need to understand the seral forking logic of opensips . In-Reply-To: References: Message-ID: <18d273bf-63fa-111f-3b14-6d55af3550d4@opensips.org> Hi, Could you post a ladder diagram showing what is the problem you have with the 2 contacts? For me it is not clear from your initial description. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com https://www.siphub.com On 10/12/23 2:43 PM, Sasmita Panda wrote: > Is there any update on this ? How can I make sure each time every > contact will get tried ? > > */Thanks & Regards/* > /Sasmita Panda/ > /Senior Network Testing and Software Engineer/ > /3CLogic , ph:07827611765/ > > > On Tue, Oct 10, 2023 at 11:37 AM Sasmita Panda > wrote: > > In this case I am using the registrar module . > > > */Thanks & Regards/* > /Sasmita Panda/ > /Senior Network Testing and Software Engineer/ > /3CLogic , ph:07827611765/ > > > On Mon, Oct 9, 2023 at 7:40 PM Ben Newlin > wrote: > > Sasmita, > > I think the relevant piece here is not the call to > “serialize_branches”, but how the branches are being created > to begin with. Are you using drouting, dispatcher, registrar, > etc? It seems like what is happening is that whatever > mechanism you are using to track the contacts is considering > the 486 response to mean that URI should not be tried again > and it is removed from routing. This behavior is likely > configurable in that module. > > Ben Newlin > > *From: *Users > on behalf of > Sasmita Panda > > *Date: *Monday, October 9, 2023 at 8:47 AM > *To: *OpenSIPS users mailling list > > *Subject: *[OpenSIPS-Users] Need to understand the seral > forking logic of opensips . > > * EXTERNAL EMAIL - Please use caution with links and attachments * > > ------------------------------------------------------------------------ > > Hi , > > I know this is a very basic question . Still I need some > clarification . > > I have 2 contact for same number , When I place a call Openips > give that to 1st if 1st contact reject the call with 486 busy > , then opensips try the same call to the 2nd contact without > rejecting the call . > > In the next call for same number , opensips tries to 2nd > contact if that rejects the call with 486 it wont try to the > 1st contact again . Why is this happening ? > > route{ > >       if (!serialize_branches(1)){ > sl_send_reply("500","Unable to load contacts"); >                                 exit; >                         }else{ >                                 if (next_branches()){ > t_on_failure("1"); >                                 } >                         } > > } > > failure_route[1] { > >    if ( t_check_status("404|477|480|481|408|486|50[234]")){ >          if (next_branches()) >          { >                 t_on_failure("1"); >                 route(1); >          } > >     } > } > > This is in my config file . i have gone through the doc > https://www.opensips.org/Documentation/Script-CoreFunctions-1-11#toc44 > > > but not able to understand properly . > > How can I print the internally stored branches on avp ? > > > */Thanks & Regards/* > > /Sasmita Panda/ > > /Senior Network Testing and Software Engineer/ > > /3CLogic , ph:07827611765/ > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Mon Oct 23 14:39:01 2023 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Mon, 23 Oct 2023 17:39:01 +0300 Subject: [OpenSIPS-Users] Handling REFER In-Reply-To: <9w116U_H-oxo2shVeDScc7Bx551CFRVPsyLSV6aBGA710QAixEpdNCXsBQ2AI51SmsXQPIFUytIKcNIkRzjFKhRbARzcHZGWjQquA5C-uqo=@proton.me> References: <9w116U_H-oxo2shVeDScc7Bx551CFRVPsyLSV6aBGA710QAixEpdNCXsBQ2AI51SmsXQPIFUytIKcNIkRzjFKhRbARzcHZGWjQquA5C-uqo=@proton.me> Message-ID: <89fb98c3-4bd3-1547-026c-d399fd8cdb05@opensips.org> Hi, The REFER is an in-dialog request like any other (re-INVITE and BYE), so no special handling. What transfer scenario are you currently failing ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com https://www.siphub.com On 10/16/23 5:49 PM, nutxase via Users wrote: > Hi All > > I am using opensips as a mid_registrar for webrtc and everything is > working fine except call transfers, as i understand they use refer, is > there anything specific i need to change to get them to work? > if you can point me to a module id appreciate it > > > > Sent with Proton Mail secure email. > > _______________________________________________ > Users mailing 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 Oct 23 14:41:20 2023 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Mon, 23 Oct 2023 17:41:20 +0300 Subject: [OpenSIPS-Users] Setting up the $socket_out and transport for b2b_client_new In-Reply-To: References: Message-ID: <7d468e60-c908-3f58-42e2-432ac9338928@opensips.org> Hi, It looks right what you are doing, so...what is failing? is going via wrong IP? wrong proto? I assume the incoming INVITE is via UDP, right ? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com https://www.siphub.com On 10/18/23 3:14 PM, Denys Pozniak wrote: > Hello! > > My OpenSIPS proxy has several internal interfaces with different > transports, how can I specify the protocol type and outgoing socket > for a function b2b_client_new? > Tried classic methods but it doesn't work: > ... > $socket_out = "tcp:192.168.100.10:5060 "; > $avp(b2b_du) = "sip:192.168.100.20:5060;transport=tcp"; > b2b_client_new("uac", $ru, $avp(b2b_du)); > b2b_server_new("uas"); > b2b_init_request("bridging"); > ... > > -- > BR, > Denys Pozniak > > > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Mon Oct 23 14:44:46 2023 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Mon, 23 Oct 2023 17:44:46 +0300 Subject: [OpenSIPS-Users] I need some information on lookup with user-agent filtering . In-Reply-To: References: Message-ID: <3ec9295f-0ef1-f492-9e7c-8756f6a779f8@opensips.org> Hi, YEs, it will check it against the user-agent value from the registration. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com https://www.siphub.com On 10/18/23 4:34 PM, Sasmita Panda wrote: > Hi All , > > lookup("location","u/phone/i"); # lookup with user-agent filtering > The user-agent filtering flag , compares the regular expression given with which > column of location table ? Will that only compare the values of User-Agent > column ? > If I have some data in attr header and I wanted to compare this regular expression > with attr column is that possible ? > */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 nutxase at proton.me Mon Oct 23 14:58:54 2023 From: nutxase at proton.me (nutxase) Date: Mon, 23 Oct 2023 14:58:54 +0000 Subject: [OpenSIPS-Users] Handling REFER In-Reply-To: <89fb98c3-4bd3-1547-026c-d399fd8cdb05@opensips.org> References: <9w116U_H-oxo2shVeDScc7Bx551CFRVPsyLSV6aBGA710QAixEpdNCXsBQ2AI51SmsXQPIFUytIKcNIkRzjFKhRbARzcHZGWjQquA5C-uqo=@proton.me> <89fb98c3-4bd3-1547-026c-d399fd8cdb05@opensips.org> Message-ID: So when using a webrtc client with mid_registrar it seems the transfer does nothing but if i put something like this if ( has_totag() && is_method("REFER") ) { mid_registrar_lookup("location","i","$tu:5060"); t_relay(); exit; }then a call transfers but doesnt drop the transferer's call Sent with [Proton Mail](https://proton.me/) secure email. ------- Original Message ------- On Monday, October 23rd, 2023 at 3:39 PM, Bogdan-Andrei Iancu wrote: > Hi, > > The REFER is an in-dialog request like any other (re-INVITE and BYE), so no special handling. What transfer scenario are you currently failing ? > > Regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > > https://www.siphub.com > > On 10/16/23 5:49 PM, nutxase via Users wrote: > >> Hi All >> >> I am using opensips as a mid_registrar for webrtc and everything is working fine except call transfers, as i understand they use refer, is there anything specific i need to change to get them to work? >> if you can point me to a module id appreciate it >> >> Sent with [Proton Mail](https://proton.me/) secure email. >> >> _______________________________________________ >> Users mailing 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 Mon Oct 23 15:24:01 2023 From: spanda at 3clogic.com (Sasmita Panda) Date: Mon, 23 Oct 2023 20:54:01 +0530 Subject: [OpenSIPS-Users] I need some help in attr matching while forming the Branch . In-Reply-To: References: Message-ID: Thank u so much for the confirmation. Although I have tested this in the same way and it's working . Still after you confirmation I am double sure that I am in the right track . Thank u once again . On Mon, 23 Oct 2023, 19:00 Ben Newlin, wrote: > Yes, based on your description of the problem that transformation should > work. We use that transformation to test a comma-separated list for the > existence of a value in the same way. > > > > Ben Newlin > > > > *From: *Users on behalf of Sasmita > Panda > *Date: *Monday, October 23, 2023 at 2:13 AM > *To: *OpenSIPS users mailling list > *Subject: *Re: [OpenSIPS-Users] I need some help in attr matching while > forming the Branch . > > * EXTERNAL EMAIL - Please use caution with links and attachments * > > > ------------------------------ > > There are some changes in the requirements now . > > > > Although I appreciate your suggestion, I will go with > script transformation . But not in a different way . * In the register I > will get set of comma separated values in Call-Info * > > *header and will save that in attr column . * > > > > Agent1: call-info : en,galileo,level1,true > > Agent2 : call-info : en,spanish,lenovo,true > > Agent3: call-info : en,galileo,sales,level1,true > > > > 1st Invite : > > *call-info : en,galileo,true* (I have to match the contact in which this > the substring of Registered contacts Attr . ex : Agent1 and Agent3 will the > matching agents here ) > > > > 2nd Invite : > > *call-info : en,true* (All there agents are matching this critaria ) > > > > 3rd Invite : > > *call-info : spanish,lenovo,true *(Agent2 will match the criteria ) > > > > This is kind of substring matching . Here the challenge for me is , the > number of values in the Invite header is not fixed , That can be 2 , that > can be 4 that can be 7 . If every comma separated value is present in the > Attr field then that contact will get call else not . > > > > > > *#Test for existence of string in another if ($(var(strtosearch){s.index, > $var(str)}) != NULL) xlog("found $var(sstr) in $var(strtosearch)\n");* > > > > *So will this help me in this case ? * > > > > > *Thanks & Regards* > > *Sasmita Panda* > > *Senior Network Testing and Software Engineer* > > *3CLogic , ph:07827611765* > > > > > > On Wed, Oct 18, 2023 at 7:10 PM Ben Newlin wrote: > > Sasmita, > > > > Your examples all contain mixed use of semicolons and commas. Without any > description or context, it’s very difficult to tell what or how you expect > these to match. > > > > Having said that, I recommend making use of OpenSIPS transformations [1]. > The parameters list transformations [2] specifically, as they operate on > semicolon separated “key=value” pairs, just like you have. > > > > Depending on what operation you are expecting with respect to the commas, > you may be able to combine that with the string select transformation [3] > to get your desired outcome. > > > > [1] https://www.opensips.org/Documentation/Script-Tran-3-2 > > [2] https://www.opensips.org/Documentation/Script-Tran-3-2#toc60 > > [3] https://www.opensips.org/Documentation/Script-Tran-3-2#toc7 > > > > Ben Newlin > > > > *From: *Users on behalf of Sasmita > Panda > *Date: *Wednesday, October 18, 2023 at 6:34 AM > *To: *OpenSIPS users mailling list > *Subject: *Re: [OpenSIPS-Users] I need some help in attr matching while > forming the Branch . > > * EXTERNAL EMAIL - Please use caution with links and attachments * > > > ------------------------------ > > Hi All , > > > > Any help on this ? > > > > > *Thanks & Regards* > > *Sasmita Panda* > > *Senior Network Testing and Software Engineer* > > *3CLogic , ph:07827611765* > > > > > > On Tue, Oct 17, 2023 at 7:54 PM Sasmita Panda wrote: > > Hi All , > > > > My requirements seem complicated to me. I dont know whether I am right or > wrong . > > I am using opensips 1.11 (I have opensips 3.2 as well ) > > > > *scenario 1: * > > 989898 - is registered from 3 different contact with different call-info > header . and I saved the call-info header in attr column while saving data > . > > > > 1 contact : call-info : x=hr,network;y=sales,z=marketing > > 2nd contact : call-info : x=hr;y=sales,z=marketing > > 3rd contact : call-info : x=hr,network;y=sales,z=marketing > > > > INVITE : > > call-info : x=hr;y=sales > > > > Now while forming a branch for this INVITE , how do I compare this > call-info with attr to get all the contact and then do serial forking on > this ? > > > > *scenario 2: * > > 989898 - is registered from 3 different contact with different call-info > header . and I saved the call-info header in attr column while saving data > . > > > > 1 contact : call-info : x=hr,network;y=sales,z=marketing > > 2nd contact : call-info : x=hr;y=sales,z=marketing > > 3rd contact : call-info : x=hr,network;y=sales,z=marketing > > > > INVITE : > > call-info : call-info : x=hr,network;y=sales,z=marketing > > > > In this I am getting only 1 contact where there is exact match . And I am > able to gove call to that contact . > > > > I am stuck on the 1st scenario . I am thinking of exec module to use on > this . But not getting any better examples of how I do this . Please > anybody help me out . > > > > Any kind of help is appreciated . 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 > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > -------------- next part -------------- An HTML attachment was scrubbed... URL: From darencrew at hotmail.com Mon Oct 23 15:34:56 2023 From: darencrew at hotmail.com (Daren FERREIRA) Date: Mon, 23 Oct 2023 17:34:56 +0200 Subject: [OpenSIPS-Users] Dispatcher ds_next_dst, Call-ID/Tags uniqueness and RFC In-Reply-To: <358e23d7-67c1-41e7-c9db-e865ae3892a9@opensips.org> References: <358e23d7-67c1-41e7-c9db-e865ae3892a9@opensips.org> Message-ID: Hello Bogdan, Thank you for your clear reply. I’m happy to see you and I have the same understanding of the RFC. Since my previous message, I made a point with the upstream provider (Orange) and they consider since the transaction has failed, that’s another dialog and so, Call-ID « MUST » change (like if a transaction failure was a call failure). I argued that a transaction failure was not a call failure, and, as the caller and callee have no idea / nothing to do with intermediate tries and failures, the call is the same and so Call-ID / To / From, have no reasons to change. They don’t mind… they stay on their position. As they don’t mind of branches tag that are RFC compliant in the perspective of uniqueness, I’m pretty sure they won’t mind of CSEQ. They wan’t the Call-ID to change… and « that’s it ». We didn’t succeed to agree on RFC contents and rules but we found a kind of « workaround », by stopping to make tries on 5XX. That’s not really RFC compliant, as 5XX are not definitive failures and allows retries, but these days: - if a failure happens on an operator gateway, there is very few luck it will succeed at another gateway from the same operator - making multiple tries with the same operator on failure can lead to a snowball effect / amplification effect when network failures occur, and increase their consequences (already happened in the past) so main French operators (and the French Telecom Federation) consider as a good practice to stop trying on 5XX replies (and BTW they also advise to rewrite 480 replies to 603 on high-load periods). Anyway, thank you for your advices and your help. Best regards > Le 23 oct. 2023 à 16:29, Bogdan-Andrei Iancu a écrit : > > HI Daren, > > In SIP there are transactional and dialog layers. Indeed, from dialog level perspective, the Call-ID and the To/FROM tags are uniquely identifying the dialog. But before having a dialog, you need to have a transaction for an initial request creating the dialog. This is the INVITE transaction (when comes to calls). > And at the transactional layer, the uniqueness is given by the VIA-branch parameter (see the RFC when comes to transaction matching). > > Bottom line, during SIP forking, OpenSIPS creates different VIA-branches for each branch, so they further (on the next hop) create different transactions. So there should be no conflict at all - a new incoming branch will create a new transaction (initial one), creating a new dialog. > > Nevertheless, there is a lot of confusion in this grey area when comes to the SIP stack. To try to address this you may try to change the cseq number for each branch, so they will have different values and not confuse the next hop, like this https://opensips.org/html/docs/modules/3.5.x/uac.html#func_uac_inc_cseq > > Best regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > https://www.siphub.com > > On 10/10/23 5:19 PM, Daren FERREIRA wrote: >> Hello, >> >> I successfully implemented OpenSIPS with dispatcher and failover routes but an upstream provider complains about duplicates requests they receive. >> >> After investigations, when a gateway reports an error, like 402, the failure route is triggered and ds_next_dst is used in order to find another gateway and try it. >> >> Doing this work as expected, but, for the provider, because none of Call-ID/To Tag/From Tag change, they consider the requests as dupes. >> >> My reading of RFC3261 is that it is compliant as the call is always the same and so, there is no need to change anything in none of these parameters. >> >> RFC says the cases (401, 413, 415, 416 and 420) where these 3 parameters MUST not change, but nothing tells they have to change in any other 4XX cases. >> >> I made many searches on this subject, nothing helped me on this issue. >> >> I made searches in OpenSIPS code to find what can eventually trigger changes on any of these parameters, without success. >> >> I found some other implementations of failover routes: >> - using AVP containing several gateways, and calls to append_branch() and serialize_branches() >> - carrier route module request append_branch >> but according to my readings, dispatcher already internally implements append_branch and other requested calls and so, they shouldn’t be called manually in script >> >> >> What are your thinkings about the RFC rules about failover requests and Call-ID/To tag/From tag uniqueness requirement? >> >> What would you do in order requests not to be identical when trying upstream routes with dispatcher module? >> >> >> Thank you for your comments. >> >> >> Regards >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users > From kaduww at gmail.com Tue Oct 24 04:02:00 2023 From: kaduww at gmail.com (Carlos Eduardo) Date: Tue, 24 Oct 2023 01:02:00 -0300 Subject: [OpenSIPS-Users] Handling REFER In-Reply-To: References: <9w116U_H-oxo2shVeDScc7Bx551CFRVPsyLSV6aBGA710QAixEpdNCXsBQ2AI51SmsXQPIFUytIKcNIkRzjFKhRbARzcHZGWjQquA5C-uqo=@proton.me> <89fb98c3-4bd3-1547-026c-d399fd8cdb05@opensips.org> Message-ID: It fails because you're sending a sequential request to another endpoint. As it doesn't have the dialog there, it will fail. You should route the REFER as any other sequential request and then the other UA will handle it and transfer. Em seg., 23 de out. de 2023 às 12:02, nutxase via Users < users at lists.opensips.org> escreveu: > So when using a webrtc client with mid_registrar it seems the transfer > does nothing > but if i put something like this > > if ( has_totag() && is_method("REFER") ) { > mid_registrar_lookup("location","i","$tu:5060"); > t_relay(); > exit; > } > then a call transfers but doesnt drop the transferer's call > > > Sent with Proton Mail secure email. > > ------- Original Message ------- > On Monday, October 23rd, 2023 at 3:39 PM, Bogdan-Andrei Iancu < > bogdan at opensips.org> wrote: > > Hi, > > The REFER is an in-dialog request like any other (re-INVITE and BYE), so > no special handling. What transfer scenario are you currently failing ? > > Regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > https://www.siphub.com > > On 10/16/23 5:49 PM, nutxase via Users wrote: > > Hi All > > I am using opensips as a mid_registrar for webrtc and everything is > working fine except call transfers, as i understand they use refer, is > there anything specific i need to change to get them to work? > if you can point me to a module id appreciate it > > > > Sent with Proton Mail secure email. > > _______________________________________________ > 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 > -- *Carlos E. Wagner* *Tecnólogo em Telecomunicações, Opensips Certified Professional* *Fone: +55 48 99981-0894* *E-mail:* kaduww at gmail.com *LinkedIn:* https://www.linkedin.com/in/carlos-eduardo-wagner-96bbb433/ -------------- next part -------------- An HTML attachment was scrubbed... URL: From venefax at gmail.com Tue Oct 24 05:17:37 2023 From: venefax at gmail.com (Saint Michael) Date: Tue, 24 Oct 2023 01:17:37 -0400 Subject: [OpenSIPS-Users] fifo not found Message-ID: I am using opensips 3.4 my /etc/opensips/opensips.cfg has loadmodule "mi_fifo.so" modparam("mi_fifo", "fifo_name", "/home/opensips/opensips_fifo") modparam("mi_fifo", "fifo_mode", 0666) and it does exists: file /home/opensips/opensips_fifo /home/opensips/opensips_fifo: fifo (named pipe) stat /home/opensips/opensips_fifo File: /home/opensips/opensips_fifo Size: 0 Blocks: 0 IO Block: 4096 fifo Device: fe00h/65024d Inode: 21530043 Links: 1 Access: (0666/prw-rw-rw-) Uid: ( 1001/opensips) Gid: ( 1001/opensips) Access: 2023-10-24 01:13:01.766009199 -0400 Modify: 2023-10-24 01:13:01.766009199 -0400 Change: 2023-10-24 01:13:01.766009199 -0400 opensips is running ps ax | grep opensips /usr/local/sbin/opensips -P /run/opensips/opensips.pid -f /etc/opensips/opensips.cfg -m 5096 -M .... but /usr/local/bin/opensips-cli -x mi address_reload ERROR: fifo file /tmp/opensips_fifo does not exist! ERROR: Is OpenSIPS running? it is trying to find opensips_fifo at the default place, how do make it look in the place indicated by opensips.cfg? From bogdan at opensips.org Tue Oct 24 05:59:35 2023 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Tue, 24 Oct 2023 08:59:35 +0300 Subject: [OpenSIPS-Users] Handling REFER In-Reply-To: References: <9w116U_H-oxo2shVeDScc7Bx551CFRVPsyLSV6aBGA710QAixEpdNCXsBQ2AI51SmsXQPIFUytIKcNIkRzjFKhRbARzcHZGWjQquA5C-uqo=@proton.me> <89fb98c3-4bd3-1547-026c-d399fd8cdb05@opensips.org> Message-ID: <3a694642-6f14-e1a2-eb01-fb48c5a3dcfe@opensips.org> :+1: no lookup for sequential, just loose_route().  Again, you should do nothing special for REFER. If the BYE's work for you, the REFER should also. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com https://www.siphub.com On 10/24/23 7:02 AM, Carlos Eduardo wrote: > It fails because you're sending a sequential request to another > endpoint. As it doesn't have the dialog there, it will fail. > > You should route the REFER as any other sequential request and then > the other UA will handle it and transfer. > > Em seg., 23 de out. de 2023 às 12:02, nutxase via Users > > escreveu: > > So when using a webrtc client with mid_registrar it seems the > transfer does nothing > but if i put something like this > > if ( has_totag() && is_method("REFER") ) { > mid_registrar_lookup("location","i","$tu:5060"); > t_relay(); >                 exit; >         } > then a call transfers but doesnt drop the transferer's call > > > Sent with Proton Mail secure email. > > ------- Original Message ------- > On Monday, October 23rd, 2023 at 3:39 PM, Bogdan-Andrei Iancu > > wrote: > >> Hi, >> >> The REFER is an in-dialog request like any other (re-INVITE and >> BYE), so no special handling. What transfer scenario are you >> currently failing ? >> >> Regards, >> Bogdan-Andrei Iancu >> >> OpenSIPS Founder and Developer >> https://www.opensips-solutions.com >> https://www.siphub.com >> On 10/16/23 5:49 PM, nutxase via Users wrote: >>> Hi All >>> >>> I am using opensips as a mid_registrar for webrtc and everything >>> is working fine except call transfers, as i understand they use >>> refer, is there anything specific i need to change to get them >>> to work? >>> if you can point me to a module id appreciate it >>> >>> >>> >>> Sent with Proton Mail secure email. >>> >>> _______________________________________________ >>> 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 > > > > > -- > *Carlos E. Wagner* > /Tecnólogo em Telecomunicações, Opensips Certified Professional/**** > > * > *Fone:*+55 48 99981-0894 > * > *E-mail:*kaduww at gmail.com _ > _ > *LinkedIn:* > https://www.linkedin.com/in/carlos-eduardo-wagner-96bbb433/ > > > _______________________________________________ > Users mailing 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 Oct 24 06:24:36 2023 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Tue, 24 Oct 2023 09:24:36 +0300 Subject: [OpenSIPS-Users] Dispatcher ds_next_dst, Call-ID/Tags uniqueness and RFC In-Reply-To: References: <358e23d7-67c1-41e7-c9db-e865ae3892a9@opensips.org> Message-ID: Hi Daren, I hear you, dinosaurs are .... just dinosaurs :) Now, the 5xx indicates a error on the upper server level, but IMHO the RFC does not enforce anything in terms of behavior, or what how to react on it. The best practice here is to consider the replying entity (server) as faulty and to skip it. So do not try it again on the spot. In your case, moving to another carrier (or other server of the same carrier) may be the right thing to do. But do not retry to the same server for the same call. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com https://www.siphub.com On 10/23/23 6:34 PM, Daren FERREIRA wrote: > Hello Bogdan, > > Thank you for your clear reply. > > I’m happy to see you and I have the same understanding of the RFC. > > Since my previous message, I made a point with the upstream provider (Orange) and they consider since the transaction has failed, that’s another dialog and so, Call-ID « MUST » change (like if a transaction failure was a call failure). > I argued that a transaction failure was not a call failure, and, as the caller and callee have no idea / nothing to do with intermediate tries and failures, the call is the same and so Call-ID / To / From, have no reasons to change. > They don’t mind… they stay on their position. > > As they don’t mind of branches tag that are RFC compliant in the perspective of uniqueness, I’m pretty sure they won’t mind of CSEQ. > They wan’t the Call-ID to change… and « that’s it ». > > We didn’t succeed to agree on RFC contents and rules but we found a kind of « workaround », by stopping to make tries on 5XX. > That’s not really RFC compliant, as 5XX are not definitive failures and allows retries, but these days: > - if a failure happens on an operator gateway, there is very few luck it will succeed at another gateway from the same operator > - making multiple tries with the same operator on failure can lead to a snowball effect / amplification effect when network failures occur, and increase their consequences (already happened in the past) > so main French operators (and the French Telecom Federation) consider as a good practice to stop trying on 5XX replies (and BTW they also advise to rewrite 480 replies to 603 on high-load periods). > > Anyway, thank you for your advices and your help. > > Best regards > > >> Le 23 oct. 2023 à 16:29, Bogdan-Andrei Iancu a écrit : >> >> HI Daren, >> >> In SIP there are transactional and dialog layers. Indeed, from dialog level perspective, the Call-ID and the To/FROM tags are uniquely identifying the dialog. But before having a dialog, you need to have a transaction for an initial request creating the dialog. This is the INVITE transaction (when comes to calls). >> And at the transactional layer, the uniqueness is given by the VIA-branch parameter (see the RFC when comes to transaction matching). >> >> Bottom line, during SIP forking, OpenSIPS creates different VIA-branches for each branch, so they further (on the next hop) create different transactions. So there should be no conflict at all - a new incoming branch will create a new transaction (initial one), creating a new dialog. >> >> Nevertheless, there is a lot of confusion in this grey area when comes to the SIP stack. To try to address this you may try to change the cseq number for each branch, so they will have different values and not confuse the next hop, like this https://opensips.org/html/docs/modules/3.5.x/uac.html#func_uac_inc_cseq >> >> Best regards, >> >> Bogdan-Andrei Iancu >> >> OpenSIPS Founder and Developer >> https://www.opensips-solutions.com >> https://www.siphub.com >> >> On 10/10/23 5:19 PM, Daren FERREIRA wrote: >>> Hello, >>> >>> I successfully implemented OpenSIPS with dispatcher and failover routes but an upstream provider complains about duplicates requests they receive. >>> >>> After investigations, when a gateway reports an error, like 402, the failure route is triggered and ds_next_dst is used in order to find another gateway and try it. >>> >>> Doing this work as expected, but, for the provider, because none of Call-ID/To Tag/From Tag change, they consider the requests as dupes. >>> >>> My reading of RFC3261 is that it is compliant as the call is always the same and so, there is no need to change anything in none of these parameters. >>> >>> RFC says the cases (401, 413, 415, 416 and 420) where these 3 parameters MUST not change, but nothing tells they have to change in any other 4XX cases. >>> >>> I made many searches on this subject, nothing helped me on this issue. >>> >>> I made searches in OpenSIPS code to find what can eventually trigger changes on any of these parameters, without success. >>> >>> I found some other implementations of failover routes: >>> - using AVP containing several gateways, and calls to append_branch() and serialize_branches() >>> - carrier route module request append_branch >>> but according to my readings, dispatcher already internally implements append_branch and other requested calls and so, they shouldn’t be called manually in script >>> >>> >>> What are your thinkings about the RFC rules about failover requests and Call-ID/To tag/From tag uniqueness requirement? >>> >>> What would you do in order requests not to be identical when trying upstream routes with dispatcher module? >>> >>> >>> Thank you for your comments. >>> >>> >>> Regards >>> _______________________________________________ >>> Users mailing list >>> Users at lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users From bogdan at opensips.org Tue Oct 24 06:26:57 2023 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Tue, 24 Oct 2023 09:26:57 +0300 Subject: [OpenSIPS-Users] fifo not found In-Reply-To: References: Message-ID: <21e4f134-f1ac-43d5-ab18-818f992c2f9c@opensips.org> Hi, See the opensips-cli documentation     https://github.com/OpenSIPS/opensips-cli#configuration In the CLI cfg file you can set a custom `|fifo_file` value, so that it will align with the value you have in the opensips cfg file. Regards, | Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com https://www.siphub.com On 10/24/23 8:17 AM, Saint Michael wrote: > I am using opensips 3.4 > my /etc/opensips/opensips.cfg has > > loadmodule "mi_fifo.so" > modparam("mi_fifo", "fifo_name", "/home/opensips/opensips_fifo") > modparam("mi_fifo", "fifo_mode", 0666) > > and it does exists: > file /home/opensips/opensips_fifo > /home/opensips/opensips_fifo: fifo (named pipe) > > stat /home/opensips/opensips_fifo > File: /home/opensips/opensips_fifo > Size: 0 Blocks: 0 IO Block: 4096 fifo > Device: fe00h/65024d Inode: 21530043 Links: 1 > Access: (0666/prw-rw-rw-) Uid: ( 1001/opensips) Gid: ( 1001/opensips) > Access: 2023-10-24 01:13:01.766009199 -0400 > Modify: 2023-10-24 01:13:01.766009199 -0400 > Change: 2023-10-24 01:13:01.766009199 -0400 > > opensips is running > ps ax | grep opensips > /usr/local/sbin/opensips -P /run/opensips/opensips.pid -f > /etc/opensips/opensips.cfg -m 5096 -M > .... > > but > /usr/local/bin/opensips-cli -x mi address_reload > > ERROR: fifo file /tmp/opensips_fifo does not exist! > ERROR: Is OpenSIPS running? > > it is trying to find opensips_fifo at the default place, how do make > it look in the place indicated by opensips.cfg? > > _______________________________________________ > Users mailing 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 Oct 24 14:28:47 2023 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Tue, 24 Oct 2023 17:28:47 +0300 Subject: [OpenSIPS-Users] [Events] VON Builder Message-ID: VON was one of the first events OpenSIPS project was part of and we owe it so much. We are now so happy to see it emerging again. Welcome 🎉 *VON Builder*, 31st of October, NYC, an one-day developers conference      https://www.vonevolution.com/von-builder Where @OpenSIPS  is a Community Sponsor 😊     www.eventbrite.com/e/707729657457/?discount=OPENSIPS Big round of applause!! -- Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com https://www.siphub.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From nutxase at proton.me Tue Oct 24 15:01:52 2023 From: nutxase at proton.me (nutxase) Date: Tue, 24 Oct 2023 15:01:52 +0000 Subject: [OpenSIPS-Users] Handling REFER In-Reply-To: <3a694642-6f14-e1a2-eb01-fb48c5a3dcfe@opensips.org> References: <9w116U_H-oxo2shVeDScc7Bx551CFRVPsyLSV6aBGA710QAixEpdNCXsBQ2AI51SmsXQPIFUytIKcNIkRzjFKhRbARzcHZGWjQquA5C-uqo=@proton.me> <89fb98c3-4bd3-1547-026c-d399fd8cdb05@opensips.org> <3a694642-6f14-e1a2-eb01-fb48c5a3dcfe@opensips.org> Message-ID: <6eK22Wq6obs-_ocahY75s1IpsIukpjEnEG3CPXdo1Ixj2hi8EYr60QfmAK_Icjm40vtiD6qlSOrMuTnu2lIrqCdqX2eFsTMEwYRi_qh2zBM=@proton.me> Strangely, when i put a loose_route() or record_route() then it does not even try transfer This is a webrtc client going from opensips to asterisk/freeswitch with mid_registrar Sent with [Proton Mail](https://proton.me/) secure email. ------- Original Message ------- On Tuesday, October 24th, 2023 at 6:59 AM, Bogdan-Andrei Iancu wrote: > :+1: > > no lookup for sequential, just loose_route(). Again, you should do nothing special for REFER. If the BYE's work for you, the REFER should also. > > Regards, > > Bogdan-Andrei Iancu > > OpenSIPS Founder and Developer > https://www.opensips-solutions.com > > https://www.siphub.com > > On 10/24/23 7:02 AM, Carlos Eduardo wrote: > >> It fails because you're sending a sequential request to another endpoint. As it doesn't have the dialog there, it will fail. >> >> You should route the REFER as any other sequential request and then the other UA will handle it and transfer. >> >> Em seg., 23 de out. de 2023 às 12:02, nutxase via Users escreveu: >> >>> So when using a webrtc client with mid_registrar it seems the transfer does nothing >>> but if i put something like this >>> >>> if ( has_totag() && is_method("REFER") ) { >>> mid_registrar_lookup("location","i","$tu:5060"); >>> t_relay(); >>> exit; >>> } then a call transfers but doesnt drop the transferer's call >>> >>> Sent with [Proton Mail](https://proton.me/) secure email. >>> >>> ------- Original Message ------- >>> On Monday, October 23rd, 2023 at 3:39 PM, Bogdan-Andrei Iancu wrote: >>> >>>> Hi, >>>> >>>> The REFER is an in-dialog request like any other (re-INVITE and BYE), so no special handling. What transfer scenario are you currently failing ? >>>> >>>> Regards, >>>> >>>> Bogdan-Andrei Iancu >>>> >>>> OpenSIPS Founder and Developer >>>> https://www.opensips-solutions.com >>>> >>>> https://www.siphub.com >>>> >>>> On 10/16/23 5:49 PM, nutxase via Users wrote: >>>> >>>>> Hi All >>>>> >>>>> I am using opensips as a mid_registrar for webrtc and everything is working fine except call transfers, as i understand they use refer, is there anything specific i need to change to get them to work? >>>>> if you can point me to a module id appreciate it >>>>> >>>>> Sent with [Proton Mail](https://proton.me/) secure email. >>>>> >>>>> _______________________________________________ >>>>> 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 >> >> -- >> >> Carlos E. Wagner >> Tecnólogo em Telecomunicações, Opensips Certified Professional >> >> Fone: +55 48 99981-0894 >> E-mail: kaduww at gmail.com >> >> LinkedIn: https://www.linkedin.com/in/carlos-eduardo-wagner-96bbb433/ >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From efes99999 at gmail.com Wed Oct 25 05:46:56 2023 From: efes99999 at gmail.com (L S) Date: Wed, 25 Oct 2023 01:46:56 -0400 Subject: [OpenSIPS-Users] 3.2.15 installation Message-ID: Has anyone successfully installed 3.2.15 (revised on Oct 20th)? Thanks, Matt -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Wed Oct 25 09:14:45 2023 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Wed, 25 Oct 2023 12:14:45 +0300 Subject: [OpenSIPS-Users] 3.2.15 installation In-Reply-To: References: Message-ID: Hi, Have you came across any issue with it? Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com https://www.siphub.com On 10/25/23 8:46 AM, L S wrote: > Has anyone successfully installed 3.2.15 (revised on Oct 20th)? > > Thanks, > Matt > > _______________________________________________ > Users mailing 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 Oct 25 13:57:43 2023 From: spanda at 3clogic.com (Sasmita Panda) Date: Wed, 25 Oct 2023 19:27:43 +0530 Subject: [OpenSIPS-Users] I need some help in attr matching while forming the Branch . In-Reply-To: References: Message-ID: Hi Ben , Please correct me if I am wrong . *$avp(attr) : sales,level20,en,true* *$hdr(Call-Info) : sales,en* * if ($(avp(attr){s.index, $hdr(Call-Info)}) != NULL) { xlog("found $hdr(Call-Info) in $avp(attr) \n"); if (next_branches()){ t_on_failure("1"); } } else { sl_send_reply(500,"Unable to load contacts"); exit; }* In this case , although $hdr(Call-Info) is a substring of $avp(attr) still it's going in the else block and giving me *500 Unable to load contacts* . Does the substring must contain continuous values ? *Thanks & Regards* *Sasmita Panda* *Senior Network Testing and Software Engineer* *3CLogic , ph:07827611765* On Mon, Oct 23, 2023 at 8:54 PM Sasmita Panda wrote: > Thank u so much for the confirmation. > > Although I have tested this in the same way and it's working . Still after > you confirmation I am double sure that I am in the right track . > > Thank u once again . > > On Mon, 23 Oct 2023, 19:00 Ben Newlin, wrote: > >> Yes, based on your description of the problem that transformation should >> work. We use that transformation to test a comma-separated list for the >> existence of a value in the same way. >> >> >> >> Ben Newlin >> >> >> >> *From: *Users on behalf of Sasmita >> Panda >> *Date: *Monday, October 23, 2023 at 2:13 AM >> *To: *OpenSIPS users mailling list >> *Subject: *Re: [OpenSIPS-Users] I need some help in attr matching while >> forming the Branch . >> >> * EXTERNAL EMAIL - Please use caution with links and attachments * >> >> >> ------------------------------ >> >> There are some changes in the requirements now . >> >> >> >> Although I appreciate your suggestion, I will go with >> script transformation . But not in a different way . * In the register I >> will get set of comma separated values in Call-Info * >> >> *header and will save that in attr column . * >> >> >> >> Agent1: call-info : en,galileo,level1,true >> >> Agent2 : call-info : en,spanish,lenovo,true >> >> Agent3: call-info : en,galileo,sales,level1,true >> >> >> >> 1st Invite : >> >> *call-info : en,galileo,true* (I have to match the contact in which >> this the substring of Registered contacts Attr . ex : Agent1 and Agent3 >> will the matching agents here ) >> >> >> >> 2nd Invite : >> >> *call-info : en,true* (All there agents are matching this critaria ) >> >> >> >> 3rd Invite : >> >> *call-info : spanish,lenovo,true *(Agent2 will match the criteria ) >> >> >> >> This is kind of substring matching . Here the challenge for me is , the >> number of values in the Invite header is not fixed , That can be 2 , that >> can be 4 that can be 7 . If every comma separated value is present in the >> Attr field then that contact will get call else not . >> >> >> >> >> >> *#Test for existence of string in another if ($(var(strtosearch){s.index, >> $var(str)}) != NULL) xlog("found $var(sstr) in $var(strtosearch)\n");* >> >> >> >> *So will this help me in this case ? * >> >> >> >> >> *Thanks & Regards* >> >> *Sasmita Panda* >> >> *Senior Network Testing and Software Engineer* >> >> *3CLogic , ph:07827611765* >> >> >> >> >> >> On Wed, Oct 18, 2023 at 7:10 PM Ben Newlin >> wrote: >> >> Sasmita, >> >> >> >> Your examples all contain mixed use of semicolons and commas. Without any >> description or context, it’s very difficult to tell what or how you expect >> these to match. >> >> >> >> Having said that, I recommend making use of OpenSIPS transformations [1]. >> The parameters list transformations [2] specifically, as they operate on >> semicolon separated “key=value” pairs, just like you have. >> >> >> >> Depending on what operation you are expecting with respect to the commas, >> you may be able to combine that with the string select transformation [3] >> to get your desired outcome. >> >> >> >> [1] https://www.opensips.org/Documentation/Script-Tran-3-2 >> >> [2] https://www.opensips.org/Documentation/Script-Tran-3-2#toc60 >> >> [3] https://www.opensips.org/Documentation/Script-Tran-3-2#toc7 >> >> >> >> Ben Newlin >> >> >> >> *From: *Users on behalf of Sasmita >> Panda >> *Date: *Wednesday, October 18, 2023 at 6:34 AM >> *To: *OpenSIPS users mailling list >> *Subject: *Re: [OpenSIPS-Users] I need some help in attr matching while >> forming the Branch . >> >> * EXTERNAL EMAIL - Please use caution with links and attachments * >> >> >> ------------------------------ >> >> Hi All , >> >> >> >> Any help on this ? >> >> >> >> >> *Thanks & Regards* >> >> *Sasmita Panda* >> >> *Senior Network Testing and Software Engineer* >> >> *3CLogic , ph:07827611765* >> >> >> >> >> >> On Tue, Oct 17, 2023 at 7:54 PM Sasmita Panda wrote: >> >> Hi All , >> >> >> >> My requirements seem complicated to me. I dont know whether I am right or >> wrong . >> >> I am using opensips 1.11 (I have opensips 3.2 as well ) >> >> >> >> *scenario 1: * >> >> 989898 - is registered from 3 different contact with different call-info >> header . and I saved the call-info header in attr column while saving data >> . >> >> >> >> 1 contact : call-info : x=hr,network;y=sales,z=marketing >> >> 2nd contact : call-info : x=hr;y=sales,z=marketing >> >> 3rd contact : call-info : x=hr,network;y=sales,z=marketing >> >> >> >> INVITE : >> >> call-info : x=hr;y=sales >> >> >> >> Now while forming a branch for this INVITE , how do I compare this >> call-info with attr to get all the contact and then do serial forking on >> this ? >> >> >> >> *scenario 2: * >> >> 989898 - is registered from 3 different contact with different call-info >> header . and I saved the call-info header in attr column while saving data >> . >> >> >> >> 1 contact : call-info : x=hr,network;y=sales,z=marketing >> >> 2nd contact : call-info : x=hr;y=sales,z=marketing >> >> 3rd contact : call-info : x=hr,network;y=sales,z=marketing >> >> >> >> INVITE : >> >> call-info : call-info : x=hr,network;y=sales,z=marketing >> >> >> >> In this I am getting only 1 contact where there is exact match . And I am >> able to gove call to that contact . >> >> >> >> I am stuck on the 1st scenario . I am thinking of exec module to use on >> this . But not getting any better examples of how I do this . Please >> anybody help me out . >> >> >> >> Any kind of help is appreciated . 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 >> >> _______________________________________________ >> Users mailing list >> Users at lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: From jjackson at aninetworks.net Wed Oct 25 20:32:36 2023 From: jjackson at aninetworks.net (Joseph Jackson) Date: Wed, 25 Oct 2023 20:32:36 +0000 Subject: [OpenSIPS-Users] 3.2.15 installation In-Reply-To: References: Message-ID: We have but we use the debian packages and we installed it on release day. ________________________________ From: Users on behalf of L S Sent: Wednesday, October 25, 2023 12:46 AM To: OpenSIPS users mailling list Subject: [OpenSIPS-Users] 3.2.15 installation Has anyone successfully installed 3.2.15 (revised on Oct 20th)? Thanks, Matt ________________________________ From: Users on behalf of L S Sent: Wednesday, October 25, 2023 12:46 AM To: OpenSIPS users mailling list Subject: [OpenSIPS-Users] 3.2.15 installation Has anyone successfully installed 3.2.15 (revised on Oct 20th)? Thanks, Matt -------------- next part -------------- An HTML attachment was scrubbed... URL: From kneeoh at yahoo.com Wed Oct 25 20:38:11 2023 From: kneeoh at yahoo.com (Knee Oh) Date: Wed, 25 Oct 2023 16:38:11 -0400 Subject: [OpenSIPS-Users] 3.2.15 installation In-Reply-To: References: Message-ID: Yes, compiled from source on Ubuntu 22.04. About to move to production. > On Oct 25, 2023, at 4:35 PM, Joseph Jackson wrote: > >  > We have but we use the debian packages and we installed it on release day. > > > > From: Users on behalf of L S > Sent: Wednesday, October 25, 2023 12:46 AM > To: OpenSIPS users mailling list > Subject: [OpenSIPS-Users] 3.2.15 installation > > Has anyone successfully installed 3.2.15 (revised on Oct 20th)? > > Thanks, > Matt > > From: Users on behalf of L S > Sent: Wednesday, October 25, 2023 12:46 AM > To: OpenSIPS users mailling list > Subject: [OpenSIPS-Users] 3.2.15 installation > > Has anyone successfully installed 3.2.15 (revised on Oct 20th)? > > Thanks, > Matt > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users -------------- next part -------------- An HTML attachment was scrubbed... URL: From efes99999 at gmail.com Thu Oct 26 06:53:50 2023 From: efes99999 at gmail.com (L S) Date: Thu, 26 Oct 2023 02:53:50 -0400 Subject: [OpenSIPS-Users] 3.2.15 installation In-Reply-To: References: Message-ID: Thanks. We are having issues compiling from source on Centos (error because of a patch related to wolfssl). Will try again. On Wed, Oct 25, 2023, 4:41 PM Knee Oh via Users wrote: > Yes, compiled from source on Ubuntu 22.04. About to move to production. > > > On Oct 25, 2023, at 4:35 PM, Joseph Jackson > wrote: > >  > We have but we use the debian packages and we installed it on release day. > > > > ------------------------------ > *From:* Users on behalf of L S < > efes99999 at gmail.com> > *Sent:* Wednesday, October 25, 2023 12:46 AM > *To:* OpenSIPS users mailling list > *Subject:* [OpenSIPS-Users] 3.2.15 installation > > Has anyone successfully installed 3.2.15 (revised on Oct 20th)? > > Thanks, > Matt > > ------------------------------ > *From:* Users on behalf of L S < > efes99999 at gmail.com> > *Sent:* Wednesday, October 25, 2023 12:46 AM > *To:* OpenSIPS users mailling list > *Subject:* [OpenSIPS-Users] 3.2.15 installation > > Has anyone successfully installed 3.2.15 (revised on Oct 20th)? > > Thanks, > Matt > _______________________________________________ > 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 spanda at 3clogic.com Thu Oct 26 12:43:51 2023 From: spanda at 3clogic.com (Sasmita Panda) Date: Thu, 26 Oct 2023 18:13:51 +0530 Subject: [OpenSIPS-Users] I need some help in attr matching while forming the Branch . In-Reply-To: References: Message-ID: Is there any update on this ? On Wed, 25 Oct 2023, 19:27 Sasmita Panda, wrote: > Hi Ben , > > Please correct me if I am wrong . > > *$avp(attr) : sales,level20,en,true* > *$hdr(Call-Info) : sales,en* > > > > > > > > > > > > > * if ($(avp(attr){s.index, $hdr(Call-Info)}) != NULL) > { xlog("found $hdr(Call-Info) in > $avp(attr) \n"); if (next_branches()){ > t_on_failure("1"); > } } else { > sl_send_reply(500,"Unable to load contacts"); > exit; }* > > In this case , although $hdr(Call-Info) is a substring of $avp(attr) still > it's going in the else block and giving me *500 Unable to load contacts* > . > > Does the substring must contain continuous values ? > > > > > *Thanks & Regards* > *Sasmita Panda* > *Senior Network Testing and Software Engineer* > *3CLogic , ph:07827611765* > > > On Mon, Oct 23, 2023 at 8:54 PM Sasmita Panda wrote: > >> Thank u so much for the confirmation. >> >> Although I have tested this in the same way and it's working . Still >> after you confirmation I am double sure that I am in the right track . >> >> Thank u once again . >> >> On Mon, 23 Oct 2023, 19:00 Ben Newlin, wrote: >> >>> Yes, based on your description of the problem that transformation should >>> work. We use that transformation to test a comma-separated list for the >>> existence of a value in the same way. >>> >>> >>> >>> Ben Newlin >>> >>> >>> >>> *From: *Users on behalf of Sasmita >>> Panda >>> *Date: *Monday, October 23, 2023 at 2:13 AM >>> *To: *OpenSIPS users mailling list >>> *Subject: *Re: [OpenSIPS-Users] I need some help in attr matching while >>> forming the Branch . >>> >>> * EXTERNAL EMAIL - Please use caution with links and attachments * >>> >>> >>> ------------------------------ >>> >>> There are some changes in the requirements now . >>> >>> >>> >>> Although I appreciate your suggestion, I will go with >>> script transformation . But not in a different way . * In the register >>> I will get set of comma separated values in Call-Info * >>> >>> *header and will save that in attr column . * >>> >>> >>> >>> Agent1: call-info : en,galileo,level1,true >>> >>> Agent2 : call-info : en,spanish,lenovo,true >>> >>> Agent3: call-info : en,galileo,sales,level1,true >>> >>> >>> >>> 1st Invite : >>> >>> *call-info : en,galileo,true* (I have to match the contact in which >>> this the substring of Registered contacts Attr . ex : Agent1 and Agent3 >>> will the matching agents here ) >>> >>> >>> >>> 2nd Invite : >>> >>> *call-info : en,true* (All there agents are matching this critaria ) >>> >>> >>> >>> 3rd Invite : >>> >>> *call-info : spanish,lenovo,true *(Agent2 will match the criteria ) >>> >>> >>> >>> This is kind of substring matching . Here the challenge for me is , the >>> number of values in the Invite header is not fixed , That can be 2 , that >>> can be 4 that can be 7 . If every comma separated value is present in the >>> Attr field then that contact will get call else not . >>> >>> >>> >>> >>> >>> *#Test for existence of string in another if >>> ($(var(strtosearch){s.index, $var(str)}) != NULL) xlog("found >>> $var(sstr) in $var(strtosearch)\n");* >>> >>> >>> >>> *So will this help me in this case ? * >>> >>> >>> >>> >>> *Thanks & Regards* >>> >>> *Sasmita Panda* >>> >>> *Senior Network Testing and Software Engineer* >>> >>> *3CLogic , ph:07827611765* >>> >>> >>> >>> >>> >>> On Wed, Oct 18, 2023 at 7:10 PM Ben Newlin >>> wrote: >>> >>> Sasmita, >>> >>> >>> >>> Your examples all contain mixed use of semicolons and commas. Without >>> any description or context, it’s very difficult to tell what or how you >>> expect these to match. >>> >>> >>> >>> Having said that, I recommend making use of OpenSIPS transformations >>> [1]. The parameters list transformations [2] specifically, as they operate >>> on semicolon separated “key=value” pairs, just like you have. >>> >>> >>> >>> Depending on what operation you are expecting with respect to the >>> commas, you may be able to combine that with the string select >>> transformation [3] to get your desired outcome. >>> >>> >>> >>> [1] https://www.opensips.org/Documentation/Script-Tran-3-2 >>> >>> [2] https://www.opensips.org/Documentation/Script-Tran-3-2#toc60 >>> >>> [3] https://www.opensips.org/Documentation/Script-Tran-3-2#toc7 >>> >>> >>> >>> Ben Newlin >>> >>> >>> >>> *From: *Users on behalf of Sasmita >>> Panda >>> *Date: *Wednesday, October 18, 2023 at 6:34 AM >>> *To: *OpenSIPS users mailling list >>> *Subject: *Re: [OpenSIPS-Users] I need some help in attr matching while >>> forming the Branch . >>> >>> * EXTERNAL EMAIL - Please use caution with links and attachments * >>> >>> >>> ------------------------------ >>> >>> Hi All , >>> >>> >>> >>> Any help on this ? >>> >>> >>> >>> >>> *Thanks & Regards* >>> >>> *Sasmita Panda* >>> >>> *Senior Network Testing and Software Engineer* >>> >>> *3CLogic , ph:07827611765* >>> >>> >>> >>> >>> >>> On Tue, Oct 17, 2023 at 7:54 PM Sasmita Panda >>> wrote: >>> >>> Hi All , >>> >>> >>> >>> My requirements seem complicated to me. I dont know whether I am right >>> or wrong . >>> >>> I am using opensips 1.11 (I have opensips 3.2 as well ) >>> >>> >>> >>> *scenario 1: * >>> >>> 989898 - is registered from 3 different contact with different >>> call-info header . and I saved the call-info header in attr column while >>> saving data . >>> >>> >>> >>> 1 contact : call-info : x=hr,network;y=sales,z=marketing >>> >>> 2nd contact : call-info : x=hr;y=sales,z=marketing >>> >>> 3rd contact : call-info : x=hr,network;y=sales,z=marketing >>> >>> >>> >>> INVITE : >>> >>> call-info : x=hr;y=sales >>> >>> >>> >>> Now while forming a branch for this INVITE , how do I compare this >>> call-info with attr to get all the contact and then do serial forking on >>> this ? >>> >>> >>> >>> *scenario 2: * >>> >>> 989898 - is registered from 3 different contact with different >>> call-info header . and I saved the call-info header in attr column while >>> saving data . >>> >>> >>> >>> 1 contact : call-info : x=hr,network;y=sales,z=marketing >>> >>> 2nd contact : call-info : x=hr;y=sales,z=marketing >>> >>> 3rd contact : call-info : x=hr,network;y=sales,z=marketing >>> >>> >>> >>> INVITE : >>> >>> call-info : call-info : x=hr,network;y=sales,z=marketing >>> >>> >>> >>> In this I am getting only 1 contact where there is exact match . And I >>> am able to gove call to that contact . >>> >>> >>> >>> I am stuck on the 1st scenario . I am thinking of exec module to use on >>> this . But not getting any better examples of how I do this . Please >>> anybody help me out . >>> >>> >>> >>> Any kind of help is appreciated . 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 >>> >>> _______________________________________________ >>> Users mailing 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 Oct 26 13:46:00 2023 From: Ben.Newlin at genesys.com (Ben Newlin) Date: Thu, 26 Oct 2023 13:46:00 +0000 Subject: [OpenSIPS-Users] I need some help in attr matching while forming the Branch . In-Reply-To: References: Message-ID: Sasmita, Apologies, I replied yesterday but the message is being held by the list as the quoted replies have made it too large. I’ve removed some of the quoted replies and I’m copying my response below: Yes, a substring match means the exact complete string exists somewhere in the string being searched . In your example, the $avp(attr) does not contain any substring that matches $hdr(Call-Info), so it is correctly failing. If you want to check for the presence of each element, you need to loop through the elements in $hdr(Call-Info) and check for each one in the $avp(attr) using the s.index mechanism. The best options for looping on the header are probably s.select [1] or the csv tranformations [2]. [1] https://www.opensips.org/Documentation/Script-Tran-3-2#toc7 [2] https://www.opensips.org/Documentation/Script-Tran-3-2#toc82 Ben Newlin From: Users on behalf of Sasmita Panda Date: Thursday, October 26, 2023 at 8:46 AM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] I need some help in attr matching while forming the Branch . EXTERNAL EMAIL - Please use caution with links and attachments ________________________________ Is there any update on this ? On Wed, 25 Oct 2023, 19:27 Sasmita Panda, > wrote: Hi Ben , Please correct me if I am wrong . $avp(attr) : sales,level20,en,true $hdr(Call-Info) : sales,en if ($(avp(attr){s.index, $hdr(Call-Info)}) != NULL) { xlog("found $hdr(Call-Info) in $avp(attr) \n"); if (next_branches()){ t_on_failure("1"); } } else { sl_send_reply(500,"Unable to load contacts"); exit; } In this case , although $hdr(Call-Info) is a substring of $avp(attr) still it's going in the else block and giving me 500 Unable to load contacts . Does the substring must contain continuous values ? Thanks & Regards Sasmita Panda Senior Network Testing and Software Engineer 3CLogic , ph:07827611765 -------------- next part -------------- An HTML attachment was scrubbed... URL: From bogdan at opensips.org Thu Oct 26 14:22:15 2023 From: bogdan at opensips.org (Bogdan-Andrei Iancu) Date: Thu, 26 Oct 2023 17:22:15 +0300 Subject: [OpenSIPS-Users] Handling REFER In-Reply-To: <6eK22Wq6obs-_ocahY75s1IpsIukpjEnEG3CPXdo1Ixj2hi8EYr60QfmAK_Icjm40vtiD6qlSOrMuTnu2lIrqCdqX2eFsTMEwYRi_qh2zBM=@proton.me> References: <9w116U_H-oxo2shVeDScc7Bx551CFRVPsyLSV6aBGA710QAixEpdNCXsBQ2AI51SmsXQPIFUytIKcNIkRzjFKhRbARzcHZGWjQquA5C-uqo=@proton.me> <89fb98c3-4bd3-1547-026c-d399fd8cdb05@opensips.org> <3a694642-6f14-e1a2-eb01-fb48c5a3dcfe@opensips.org> <6eK22Wq6obs-_ocahY75s1IpsIukpjEnEG3CPXdo1Ixj2hi8EYr60QfmAK_Icjm40vtiD6qlSOrMuTnu2lIrqCdqX2eFsTMEwYRi_qh2zBM=@proton.me> Message-ID: <83539ac2-6343-7ad5-8519-308620c4afc7@opensips.org> Well, maybe posting some network capture  + script snippet for the sequential requests will help here. Regards, Bogdan-Andrei Iancu OpenSIPS Founder and Developer https://www.opensips-solutions.com https://www.siphub.com On 10/24/23 6:01 PM, nutxase wrote: > Strangely, when i put a loose_route() or record_route() then it does > not even try transfer > > This is a webrtc client going from opensips to asterisk/freeswitch > with mid_registrar > > > Sent with Proton Mail secure email. > > ------- Original Message ------- > On Tuesday, October 24th, 2023 at 6:59 AM, Bogdan-Andrei Iancu > wrote: > >> :+1: >> >> no lookup for sequential, just loose_route(). Again, you should do >> nothing special for REFER. If the BYE's work for you, the REFER >> should also. >> >> Regards, >> Bogdan-Andrei Iancu >> >> OpenSIPS Founder and Developer >> https://www.opensips-solutions.com >> https://www.siphub.com >> On 10/24/23 7:02 AM, Carlos Eduardo wrote: >>> It fails because you're sending a sequential request to another >>> endpoint. As it doesn't have the dialog there, it will fail. >>> >>> You should route the REFER as any other sequential request and then >>> the other UA will handle it and transfer. >>> >>> Em seg., 23 de out. de 2023 às 12:02, nutxase via Users >>> > escreveu: >>> >>> So when using a webrtc client with mid_registrar it seems the >>> transfer does nothing >>> but if i put something like this >>> >>> if ( has_totag() && is_method("REFER") ) { >>> mid_registrar_lookup("location","i","$tu:5060"); >>> t_relay(); >>> exit; >>> } >>> then a call transfers but doesnt drop the transferer's call >>> >>> >>> Sent with Proton Mail secure email. >>> >>> ------- Original Message ------- >>> On Monday, October 23rd, 2023 at 3:39 PM, Bogdan-Andrei Iancu >>> > wrote: >>> >>>> Hi, >>>> >>>> The REFER is an in-dialog request like any other (re-INVITE and >>>> BYE), so no special handling. What transfer scenario are you >>>> currently failing ? >>>> >>>> Regards, >>>> Bogdan-Andrei Iancu >>>> >>>> OpenSIPS Founder and Developer >>>> https://www.opensips-solutions.com >>>> https://www.siphub.com >>>> On 10/16/23 5:49 PM, nutxase via Users wrote: >>>>> Hi All >>>>> >>>>> I am using opensips as a mid_registrar for webrtc and >>>>> everything is working fine except call transfers, as i >>>>> understand they use refer, is there anything specific i need >>>>> to change to get them to work? >>>>> if you can point me to a module id appreciate it >>>>> >>>>> >>>>> >>>>> Sent with Proton Mail secure email. >>>>> >>>>> _______________________________________________ >>>>> 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 >>> >>> >>> >>> >>> -- >>> *Carlos E. Wagner* >>> /Tecnólogo em Telecomunicações, Opensips Certified Professional/**** >>> >>> * >>> *Fone:*+55 48 99981-0894 >>> * >>> *E-mail:*kaduww at gmail.com _ >>> _ >>> *LinkedIn:* >>> https://www.linkedin.com/in/carlos-eduardo-wagner-96bbb433/ >>> >>> >>> _______________________________________________ >>> Users mailing list >>> Users at lists.opensips.org >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: From greg at switchtel.co.za Thu Oct 26 15:03:57 2023 From: greg at switchtel.co.za (Gregory Massel) Date: Thu, 26 Oct 2023 17:03:57 +0200 Subject: [OpenSIPS-Users] tls_peer_subject_cn not set on wolfssl? Message-ID: <2156886a-4e38-42c1-b777-cb89baddc3c2@switchtel.co.za> Hello I'm using OpenSIPS 3.4.2. When using tls_wolfssl.so, $tls_peer_subject_cn appears to always be . However, when changing to tls_openssl.so, $tls_peer_subject_cn is then set correctly. Is this an issue within tls_wolfssl, or, is $tls_peer_subject_cn only meant to work with OpenSSL? -- Regards *Gregory Massel* *T* +27 87 550 0000 *F* +27 11 783 4877 *W* www.switchtel.co.za -------------- next part -------------- An HTML attachment was scrubbed... URL: From spanda at 3clogic.com Fri Oct 27 10:15:34 2023 From: spanda at 3clogic.com (Sasmita Panda) Date: Fri, 27 Oct 2023 15:45:34 +0530 Subject: [OpenSIPS-Users] I need some help in attr matching while forming the Branch . In-Reply-To: References: Message-ID: Ahhh , Ok . Now it's very complicated . When you said a loop that's looping in my mind . Sorry for the bad joke . In case , the number of elements in each Invite won't be fixed . So the number of loops will vary from one Invite to another . I was thinking of counting the number of elements first , then as the index starts from 0 I will loop till *count-1* to fetch every element properly . Example : $var(count) = $(hdr(Call-Info){csv.count}) ## if the number is 3 then loop will be for 3 times starting from 0 to 2 $var(i) = 0; while($var(i) < $var(count) ) { xlog("counter: $var(i)\n"); $var(i) = $var(i) + 1; } This was my initial thought . But while finding the count it gave me an error . * $var(count) = $(hdr(Call-Info){csv.count}) Is this not in the correct format ? * *parse error in /usr/local/etc/opensips/opensips-p2p.cfg:267:26-55: unknown script variable* *As I have earlier mentioned my header will look like . * *Call-Info: sales,en,level20,en (this can be anything but format will be like this . ) . How do I count the number of values ?* *2. As I am doing this matching to filter out contacts, where should I do this ? If I am doing this while giving a call to the contacts , for the 1st transaction it is doing the comparison , after that for the next branch it processes the call without matching . Which is not right . For every contact this comparison should loop * *Thanks & Regards* *Sasmita Panda* *Senior Network Testing and Software Engineer* *3CLogic , ph:07827611765* On Thu, Oct 26, 2023 at 7:19 PM Ben Newlin wrote: > Sasmita, > > > > Apologies, I replied yesterday but the message is being held by the list > as the quoted replies have made it too large. I’ve removed some of the > quoted replies and I’m copying my response below: > > > > Yes, a substring match means the exact complete string exists somewhere in > the string being searched . In your example, the $avp(attr) does not > contain any substring that matches $hdr(Call-Info), so it is correctly > failing. > > > > If you want to check for the presence of each element, you need to loop > through the elements in $hdr(Call-Info) and check for each one in the > $avp(attr) using the s.index mechanism. > > > > The best options for looping on the header are probably s.select [1] or > the csv tranformations [2]. > > > > [1] https://www.opensips.org/Documentation/Script-Tran-3-2#toc7 > > [2] https://www.opensips.org/Documentation/Script-Tran-3-2#toc82 > > > > Ben Newlin > > > > *From: *Users on behalf of Sasmita > Panda > *Date: *Thursday, October 26, 2023 at 8:46 AM > *To: *OpenSIPS users mailling list > *Subject: *Re: [OpenSIPS-Users] I need some help in attr matching while > forming the Branch . > > * EXTERNAL EMAIL - Please use caution with links and attachments * > > > ------------------------------ > > Is there any update on this ? > > > > On Wed, 25 Oct 2023, 19:27 Sasmita Panda, wrote: > > Hi Ben , > > > > Please correct me if I am wrong . > > > > *$avp(attr) : sales,level20,en,true* > > *$hdr(Call-Info) : sales,en* > > > > > > > > > > > > > > > * if ($(avp(attr){s.index, $hdr(Call-Info)}) != NULL) > { xlog("found $hdr(Call-Info) in > $avp(attr) \n"); if (next_branches()){ > t_on_failure("1"); > } } else { > sl_send_reply(500,"Unable to load contacts"); > exit; }* > > > > In this case , although $hdr(Call-Info) is a substring of $avp(attr) still > it's going in the else block and giving me *500 Unable to load contacts* > . > > > > Does the substring must contain continuous values ? > > > > > > > > > > *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 Ben.Newlin at genesys.com Fri Oct 27 13:32:28 2023 From: Ben.Newlin at genesys.com (Ben Newlin) Date: Fri, 27 Oct 2023 13:32:28 +0000 Subject: [OpenSIPS-Users] I need some help in attr matching while forming the Branch . In-Reply-To: References: Message-ID: Without seeing the specific exact code, I can’t say what is causing that error. It seems like it may be a syntax issue. For example, in your email you are missing the semicolon after the line setting the count. I don’t know if it is that way in your config script. I recommend double checking all your syntax, and if you still get the error please provide the exact code from your script for review. For #2, I can’t really be very specific there because I don’t know all of your logic, nor am I very knowledgeable about the registrar module. My first thought is to not do any filtering/checking in the request route. Just allow the registrar module to load all contacts in branches, but arm a branch_route. Then perform the check in the branch route for each branch. If the Contact for the branch doesn’t match what you want, you can drop the branch [1]. [1] https://www.opensips.org/Documentation/Script-Routes-3-2#toc2 Ben Newlin From: Users on behalf of Sasmita Panda Date: Friday, October 27, 2023 at 6:17 AM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] I need some help in attr matching while forming the Branch . EXTERNAL EMAIL - Please use caution with links and attachments ________________________________ Ahhh , Ok . Now it's very complicated . When you said a loop that's looping in my mind . Sorry for the bad joke . In case , the number of elements in each Invite won't be fixed . So the number of loops will vary from one Invite to another . I was thinking of counting the number of elements first , then as the index starts from 0 I will loop till count-1 to fetch every element properly . Example : $var(count) = $(hdr(Call-Info){csv.count}) ## if the number is 3 then loop will be for 3 times starting from 0 to 2 $var(i) = 0; while($var(i) < $var(count) ) { xlog("counter: $var(i)\n"); $var(i) = $var(i) + 1; } This was my initial thought . But while finding the count it gave me an error . $var(count) = $(hdr(Call-Info){csv.count}) Is this not in the correct format ? parse error in /usr/local/etc/opensips/opensips-p2p.cfg:267:26-55: unknown script variable As I have earlier mentioned my header will look like . Call-Info: sales,en,level20,en (this can be anything but format will be like this . ) . How do I count the number of values ? 2. As I am doing this matching to filter out contacts, where should I do this ? If I am doing this while giving a call to the contacts , for the 1st transaction it is doing the comparison , after that for the next branch it processes the call without matching . Which is not right . For every contact this comparison should loop Thanks & Regards Sasmita Panda Senior Network Testing and Software Engineer 3CLogic , ph:07827611765 On Thu, Oct 26, 2023 at 7:19 PM Ben Newlin > wrote: Sasmita, Apologies, I replied yesterday but the message is being held by the list as the quoted replies have made it too large. I’ve removed some of the quoted replies and I’m copying my response below: Yes, a substring match means the exact complete string exists somewhere in the string being searched . In your example, the $avp(attr) does not contain any substring that matches $hdr(Call-Info), so it is correctly failing. If you want to check for the presence of each element, you need to loop through the elements in $hdr(Call-Info) and check for each one in the $avp(attr) using the s.index mechanism. The best options for looping on the header are probably s.select [1] or the csv tranformations [2]. [1] https://www.opensips.org/Documentation/Script-Tran-3-2#toc7 [2] https://www.opensips.org/Documentation/Script-Tran-3-2#toc82 Ben Newlin -------------- next part -------------- An HTML attachment was scrubbed... URL: From Ben.Newlin at genesys.com Fri Oct 27 13:38:03 2023 From: Ben.Newlin at genesys.com (Ben Newlin) Date: Fri, 27 Oct 2023 13:38:03 +0000 Subject: [OpenSIPS-Users] I need some help in attr matching while forming the Branch . In-Reply-To: References: Message-ID: Just wanted to add that with the solution I recommended below, you would want to make sure you properly handle the case where all branches were dropped because none matched. The proper handling in that case for your system would be defined by your requirements. Ben Newlin From: Users on behalf of Ben Newlin Date: Friday, October 27, 2023 at 9:33 AM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] I need some help in attr matching while forming the Branch . ________________________________ Without seeing the specific exact code, I can’t say what is causing that error. It seems like it may be a syntax issue. For example, in your email you are missing the semicolon after the line setting the count. I don’t know if it is that way in your config script. I recommend double checking all your syntax, and if you still get the error please provide the exact code from your script for review. For #2, I can’t really be very specific there because I don’t know all of your logic, nor am I very knowledgeable about the registrar module. My first thought is to not do any filtering/checking in the request route. Just allow the registrar module to load all contacts in branches, but arm a branch_route. Then perform the check in the branch route for each branch. If the Contact for the branch doesn’t match what you want, you can drop the branch [1]. [1] https://www.opensips.org/Documentation/Script-Routes-3-2#toc2 Ben Newlin From: Users on behalf of Sasmita Panda Date: Friday, October 27, 2023 at 6:17 AM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] I need some help in attr matching while forming the Branch . ________________________________ Ahhh , Ok . Now it's very complicated . When you said a loop that's looping in my mind . Sorry for the bad joke . In case , the number of elements in each Invite won't be fixed . So the number of loops will vary from one Invite to another . I was thinking of counting the number of elements first , then as the index starts from 0 I will loop till count-1 to fetch every element properly . Example : $var(count) = $(hdr(Call-Info){csv.count}) ## if the number is 3 then loop will be for 3 times starting from 0 to 2 $var(i) = 0; while($var(i) < $var(count) ) { xlog("counter: $var(i)\n"); $var(i) = $var(i) + 1; } This was my initial thought . But while finding the count it gave me an error . $var(count) = $(hdr(Call-Info){csv.count}) Is this not in the correct format ? parse error in /usr/local/etc/opensips/opensips-p2p.cfg:267:26-55: unknown script variable As I have earlier mentioned my header will look like . Call-Info: sales,en,level20,en (this can be anything but format will be like this . ) . How do I count the number of values ? 2. As I am doing this matching to filter out contacts, where should I do this ? If I am doing this while giving a call to the contacts , for the 1st transaction it is doing the comparison , after that for the next branch it processes the call without matching . Which is not right . For every contact this comparison should loop Thanks & Regards Sasmita Panda Senior Network Testing and Software Engineer 3CLogic , ph:07827611765 On Thu, Oct 26, 2023 at 7:19 PM Ben Newlin > wrote: Sasmita, Apologies, I replied yesterday but the message is being held by the list as the quoted replies have made it too large. I’ve removed some of the quoted replies and I’m copying my response below: Yes, a substring match means the exact complete string exists somewhere in the string being searched . In your example, the $avp(attr) does not contain any substring that matches $hdr(Call-Info), so it is correctly failing. If you want to check for the presence of each element, you need to loop through the elements in $hdr(Call-Info) and check for each one in the $avp(attr) using the s.index mechanism. The best options for looping on the header are probably s.select [1] or the csv tranformations [2]. [1] https://www.opensips.org/Documentation/Script-Tran-3-2#toc7 [2] https://www.opensips.org/Documentation/Script-Tran-3-2#toc82 Ben Newlin -------------- next part -------------- An HTML attachment was scrubbed... URL: From adrien.lacomme at appliwave.com Fri Oct 27 15:06:26 2023 From: adrien.lacomme at appliwave.com (Adrien LACOMME) Date: Fri, 27 Oct 2023 15:06:26 +0000 Subject: [OpenSIPS-Users] Ratelimit usage Message-ID: Hi, I'm trying to implement CPS limitation on OpenSIPS and I got some difficulties to understand how ratelimit module works. Here is the setup (I want to limit cps to 1) : if (!rl_check("test1", 1, "RED")) {       send_reply(487, "Request Terminated: CPS limit exceeded"); } I tried to make few calls simultaneously but rl_check return is always FALSE. Here is the log I got : 2023-10-27T16:52:19.792337 SBC /usr/sbin/opensips[137782]: DBG:ratelimit:w_rl_check: Pipe test1 counter:1 load:0 limit:1 should NOT be blocked (0x7fcae1504888) 2023-10-27T16:52:19.804064 SBC /usr/sbin/opensips[137781]: DBG:ratelimit:w_rl_check: Pipe test1 counter:2 load:0 limit:1 should NOT be blocked (0x7fcae1504888) 2023-10-27T16:52:22.872991 SBC /usr/sbin/opensips[137782]: DBG:ratelimit:w_rl_check: Pipe test1 counter:3 load:0 limit:1 should NOT be blocked (0x7fcae1504888) 2023-10-27T16:52:22.918123 SBC /usr/sbin/opensips[137781]: DBG:ratelimit:w_rl_check: Pipe test1 counter:4 load:0 limit:1 should NOT be blocked (0x7fcae1504888) Shouldn't the module block calls when the counter goes to 2 ? Or is he checking the load value to block ? If yes, why the load value isn't incremented ? Regards, Adrien LACOMME -------------- next part -------------- An HTML attachment was scrubbed... URL: From wadii at evenmedia.fr Fri Oct 27 15:39:02 2023 From: wadii at evenmedia.fr (Wadii ELMAJDI | Evenmedia) Date: Fri, 27 Oct 2023 15:39:02 +0000 Subject: [OpenSIPS-Users] Ratelimit usage In-Reply-To: References: Message-ID: Hello Adrien Try : if (!rl_check("test1", 1, "RED")) { rl_dec_count("test1") send_reply(487, "Request Terminated: CPS limit exceeded"); exit; } * If you are using ratelimit to limit only successful traffic, you need to explicitely decrease the counter for the declined calls using the rl_dec_count() function also I’m not sure RED is the best algorithm for your usecase. De : Users De la part de Adrien LACOMME Envoyé : vendredi 27 octobre 2023 17:06 À : users at lists.opensips.org Objet : [OpenSIPS-Users] Ratelimit usage Hi, I'm trying to implement CPS limitation on OpenSIPS and I got some difficulties to understand how ratelimit module works. Here is the setup (I want to limit cps to 1) : if (!rl_check("test1", 1, "RED")) {       send_reply(487, "Request Terminated: CPS limit exceeded"); } I tried to make few calls simultaneously but rl_check return is always FALSE. Here is the log I got : 2023-10-27T16:52:19.792337 SBC /usr/sbin/opensips[137782]: DBG:ratelimit:w_rl_check: Pipe test1 counter:1 load:0 limit:1 should NOT be blocked (0x7fcae1504888) 2023-10-27T16:52:19.804064 SBC /usr/sbin/opensips[137781]: DBG:ratelimit:w_rl_check: Pipe test1 counter:2 load:0 limit:1 should NOT be blocked (0x7fcae1504888) 2023-10-27T16:52:22.872991 SBC /usr/sbin/opensips[137782]: DBG:ratelimit:w_rl_check: Pipe test1 counter:3 load:0 limit:1 should NOT be blocked (0x7fcae1504888) 2023-10-27T16:52:22.918123 SBC /usr/sbin/opensips[137781]: DBG:ratelimit:w_rl_check: Pipe test1 counter:4 load:0 limit:1 should NOT be blocked (0x7fcae1504888) Shouldn't the module block calls when the counter goes to 2 ? Or is he checking the load value to block ? If yes, why the load value isn't incremented ? Regards, Adrien LACOMME -------------- next part -------------- An HTML attachment was scrubbed... URL: From adrien.lacomme at appliwave.com Fri Oct 27 16:01:48 2023 From: adrien.lacomme at appliwave.com (Adrien LACOMME) Date: Fri, 27 Oct 2023 16:01:48 +0000 Subject: [OpenSIPS-Users] Ratelimit usage In-Reply-To: References: Message-ID: Hello, I added this line to my code, in fact it's better to decrease the counter if I drop the call. But my problem is that the rl_check function never drop calls, I do not enter in the IF condition. For the algorithm, I'm actually just trying the module, I will probably not stay with RED. ________________________________ De : Users de la part de Wadii ELMAJDI | Evenmedia Envoyé : vendredi 27 octobre 2023 17:39 À : OpenSIPS users mailling list Objet : Re: [OpenSIPS-Users] Ratelimit usage Hello Adrien Try : if (!rl_check("test1", 1, "RED")) { rl_dec_count("test1") send_reply(487, "Request Terminated: CPS limit exceeded"); exit; } * If you are using ratelimit to limit only successful traffic, you need to explicitely decrease the counter for the declined calls using the rl_dec_count() function also I’m not sure RED is the best algorithm for your usecase. De : Users De la part de Adrien LACOMME Envoyé : vendredi 27 octobre 2023 17:06 À : users at lists.opensips.org Objet : [OpenSIPS-Users] Ratelimit usage Hi, I'm trying to implement CPS limitation on OpenSIPS and I got some difficulties to understand how ratelimit module works. Here is the setup (I want to limit cps to 1) : if (!rl_check("test1", 1, "RED")) {       send_reply(487, "Request Terminated: CPS limit exceeded"); } I tried to make few calls simultaneously but rl_check return is always FALSE. Here is the log I got : 2023-10-27T16:52:19.792337 SBC /usr/sbin/opensips[137782]: DBG:ratelimit:w_rl_check: Pipe test1 counter:1 load:0 limit:1 should NOT be blocked (0x7fcae1504888) 2023-10-27T16:52:19.804064 SBC /usr/sbin/opensips[137781]: DBG:ratelimit:w_rl_check: Pipe test1 counter:2 load:0 limit:1 should NOT be blocked (0x7fcae1504888) 2023-10-27T16:52:22.872991 SBC /usr/sbin/opensips[137782]: DBG:ratelimit:w_rl_check: Pipe test1 counter:3 load:0 limit:1 should NOT be blocked (0x7fcae1504888) 2023-10-27T16:52:22.918123 SBC /usr/sbin/opensips[137781]: DBG:ratelimit:w_rl_check: Pipe test1 counter:4 load:0 limit:1 should NOT be blocked (0x7fcae1504888) Shouldn't the module block calls when the counter goes to 2 ? Or is he checking the load value to block ? If yes, why the load value isn't incremented ? Regards, Adrien LACOMME -------------- next part -------------- An HTML attachment was scrubbed... URL: From Ben.Newlin at genesys.com Fri Oct 27 16:13:01 2023 From: Ben.Newlin at genesys.com (Ben Newlin) Date: Fri, 27 Oct 2023 16:13:01 +0000 Subject: [OpenSIPS-Users] Ratelimit usage In-Reply-To: References: Message-ID: Have you read the description of the RED algorithm? Are you sure you are generating appropriate traffic to cause a drop? That algorithm seems like it would be difficult to control/predict the drops accurately for a test. If you are sending a constant traffic volume to it, it never will drop. If you are just testing, you may want to go with a simpler algorithm where you can more easily control when/how many are being dropped. The standard way to use the module for CPS is to use the SBT algorithm with a window_size of 1. [1] [1] https://opensips.org/docs/modules/3.2.x/ratelimit.html#param_window_size Ben Newlin From: Users on behalf of Adrien LACOMME Date: Friday, October 27, 2023 at 12:03 PM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] Ratelimit usage EXTERNAL EMAIL - Please use caution with links and attachments ________________________________ Hello, I added this line to my code, in fact it's better to decrease the counter if I drop the call. But my problem is that the rl_check function never drop calls, I do not enter in the IF condition. For the algorithm, I'm actually just trying the module, I will probably not stay with RED. ________________________________ De : Users de la part de Wadii ELMAJDI | Evenmedia Envoyé : vendredi 27 octobre 2023 17:39 À : OpenSIPS users mailling list Objet : Re: [OpenSIPS-Users] Ratelimit usage Hello Adrien Try : if (!rl_check("test1", 1, "RED")) { rl_dec_count("test1") send_reply(487, "Request Terminated: CPS limit exceeded"); exit; } * If you are using ratelimit to limit only successful traffic, you need to explicitely decrease the counter for the declined calls using the rl_dec_count() function also I’m not sure RED is the best algorithm for your usecase. De : Users De la part de Adrien LACOMME Envoyé : vendredi 27 octobre 2023 17:06 À : users at lists.opensips.org Objet : [OpenSIPS-Users] Ratelimit usage Hi, I'm trying to implement CPS limitation on OpenSIPS and I got some difficulties to understand how ratelimit module works. Here is the setup (I want to limit cps to 1) : if (!rl_check("test1", 1, "RED")) {       send_reply(487, "Request Terminated: CPS limit exceeded"); } I tried to make few calls simultaneously but rl_check return is always FALSE. Here is the log I got : 2023-10-27T16:52:19.792337 SBC /usr/sbin/opensips[137782]: DBG:ratelimit:w_rl_check: Pipe test1 counter:1 load:0 limit:1 should NOT be blocked (0x7fcae1504888) 2023-10-27T16:52:19.804064 SBC /usr/sbin/opensips[137781]: DBG:ratelimit:w_rl_check: Pipe test1 counter:2 load:0 limit:1 should NOT be blocked (0x7fcae1504888) 2023-10-27T16:52:22.872991 SBC /usr/sbin/opensips[137782]: DBG:ratelimit:w_rl_check: Pipe test1 counter:3 load:0 limit:1 should NOT be blocked (0x7fcae1504888) 2023-10-27T16:52:22.918123 SBC /usr/sbin/opensips[137781]: DBG:ratelimit:w_rl_check: Pipe test1 counter:4 load:0 limit:1 should NOT be blocked (0x7fcae1504888) Shouldn't the module block calls when the counter goes to 2 ? Or is he checking the load value to block ? If yes, why the load value isn't incremented ? Regards, Adrien LACOMME -------------- next part -------------- An HTML attachment was scrubbed... URL: From adrien.lacomme at appliwave.com Fri Oct 27 19:31:41 2023 From: adrien.lacomme at appliwave.com (Adrien LACOMME) Date: Fri, 27 Oct 2023 19:31:41 +0000 Subject: [OpenSIPS-Users] Ratelimit usage In-Reply-To: References: Message-ID: I tried with SBT and it is way better, gonna stay with this one for CPS. I did my tests by trying RED and TAILDROP algorithm, wasn't probably the good one to try for cps and with my type of traffic. Thanks for your response and your time Adrien ________________________________ De : Users de la part de Ben Newlin Envoyé : vendredi 27 octobre 2023 18:13 À : OpenSIPS users mailling list Objet : Re: [OpenSIPS-Users] Ratelimit usage Have you read the description of the RED algorithm? Are you sure you are generating appropriate traffic to cause a drop? That algorithm seems like it would be difficult to control/predict the drops accurately for a test. If you are sending a constant traffic volume to it, it never will drop. If you are just testing, you may want to go with a simpler algorithm where you can more easily control when/how many are being dropped. The standard way to use the module for CPS is to use the SBT algorithm with a window_size of 1. [1] [1] https://opensips.org/docs/modules/3.2.x/ratelimit.html#param_window_size Ben Newlin From: Users on behalf of Adrien LACOMME Date: Friday, October 27, 2023 at 12:03 PM To: OpenSIPS users mailling list Subject: Re: [OpenSIPS-Users] Ratelimit usage EXTERNAL EMAIL - Please use caution with links and attachments ________________________________ Hello, I added this line to my code, in fact it's better to decrease the counter if I drop the call. But my problem is that the rl_check function never drop calls, I do not enter in the IF condition. For the algorithm, I'm actually just trying the module, I will probably not stay with RED. ________________________________ De : Users de la part de Wadii ELMAJDI | Evenmedia Envoyé : vendredi 27 octobre 2023 17:39 À : OpenSIPS users mailling list Objet : Re: [OpenSIPS-Users] Ratelimit usage Hello Adrien Try : if (!rl_check("test1", 1, "RED")) { rl_dec_count("test1") send_reply(487, "Request Terminated: CPS limit exceeded"); exit; } * If you are using ratelimit to limit only successful traffic, you need to explicitely decrease the counter for the declined calls using the rl_dec_count() function also I’m not sure RED is the best algorithm for your usecase. De : Users De la part de Adrien LACOMME Envoyé : vendredi 27 octobre 2023 17:06 À : users at lists.opensips.org Objet : [OpenSIPS-Users] Ratelimit usage Hi, I'm trying to implement CPS limitation on OpenSIPS and I got some difficulties to understand how ratelimit module works. Here is the setup (I want to limit cps to 1) : if (!rl_check("test1", 1, "RED")) {       send_reply(487, "Request Terminated: CPS limit exceeded"); } I tried to make few calls simultaneously but rl_check return is always FALSE. Here is the log I got : 2023-10-27T16:52:19.792337 SBC /usr/sbin/opensips[137782]: DBG:ratelimit:w_rl_check: Pipe test1 counter:1 load:0 limit:1 should NOT be blocked (0x7fcae1504888) 2023-10-27T16:52:19.804064 SBC /usr/sbin/opensips[137781]: DBG:ratelimit:w_rl_check: Pipe test1 counter:2 load:0 limit:1 should NOT be blocked (0x7fcae1504888) 2023-10-27T16:52:22.872991 SBC /usr/sbin/opensips[137782]: DBG:ratelimit:w_rl_check: Pipe test1 counter:3 load:0 limit:1 should NOT be blocked (0x7fcae1504888) 2023-10-27T16:52:22.918123 SBC /usr/sbin/opensips[137781]: DBG:ratelimit:w_rl_check: Pipe test1 counter:4 load:0 limit:1 should NOT be blocked (0x7fcae1504888) Shouldn't the module block calls when the counter goes to 2 ? Or is he checking the load value to block ? If yes, why the load value isn't incremented ? Regards, Adrien LACOMME -------------- next part -------------- An HTML attachment was scrubbed... URL: From razvan at opensips.org Tue Oct 31 07:22:29 2023 From: razvan at opensips.org (=?UTF-8?Q?R=C4=83zvan_Crainea?=) Date: Tue, 31 Oct 2023 09:22:29 +0200 Subject: [OpenSIPS-Users] tls_peer_subject_cn not set on wolfssl? In-Reply-To: <2156886a-4e38-42c1-b777-cb89baddc3c2@switchtel.co.za> References: <2156886a-4e38-42c1-b777-cb89baddc3c2@switchtel.co.za> Message-ID: Hi, Gregory! This looks like a bug - please open a ticket on our GitHub tracker[1] to keep track of this issue. [1] https://github.com/OpenSIPS/opensips/issues Best regards, Răzvan Crainea OpenSIPS Core Developer / SIPhub CTO http://www.opensips-solutions.com / https://www.siphub.com On 10/26/23 18:03, Gregory Massel via Users wrote: > Hello > > I'm using OpenSIPS 3.4.2. > > When using tls_wolfssl.so, $tls_peer_subject_cn appears to always be . > > However, when changing to tls_openssl.so, $tls_peer_subject_cn is then > set correctly. > > Is this an issue within tls_wolfssl, or, is $tls_peer_subject_cn only > meant to work with OpenSSL? > > -- > Regards > *Gregory Massel* > *T* +27 87 550 0000 > *F* +27 11 783 4877 > *W* www.switchtel.co.za > > _______________________________________________ > Users mailing list > Users at lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users From spanda at 3clogic.com Tue Oct 31 08:19:53 2023 From: spanda at 3clogic.com (Sasmita Panda) Date: Tue, 31 Oct 2023 13:49:53 +0530 Subject: [OpenSIPS-Users] Need some information regarding accounting module . Message-ID: Hi All , I just got an wired call example on which the ACC information is not as for my expectation . I have Opensips 1.11 version . Example : Call started at : *Oct 17 19:24:40:512 * Re-Invite on this session : *Oct 17 19:24:40:810* For the Re-Invite The client got " 500 Server Internal Error " which Opensips gave . After that there was no transaction in the session . The call got disconnected due to the error message . But The ACC info for that call : *Oct 18 07:33:32 (After 12 hr )* Although Opensips created the 500 error message still the session did not get cleaned and Acc was printed after so long with a bigger talktime : duration=43729 . Which is totally beyond my expectation . I have attached the opensips config file as well . I am not sure . Do I need to call drop_accounting while in an error scenario ? *Thanks & Regards* *Sasmita Panda* *Senior Network Testing and Software Engineer* *3CLogic , ph:07827611765* -------------- next part -------------- An HTML attachment was scrubbed... URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: carrier-config.cfg Type: application/octet-stream Size: 14709 bytes Desc: not available URL: From iamhalje at gmail.com Tue Oct 31 09:37:00 2023 From: iamhalje at gmail.com (Dmitry Ponomaryov) Date: Tue, 31 Oct 2023 14:37:00 +0500 Subject: [OpenSIPS-Users] Correct syntax with statement if Message-ID: Have a nice day, everyone, there are cases in routing when in if there is only one command, for example: ### example 1 if ( has_body("application/sdp") ) { rtpengine_offer(); } In this case, how different will this expression be from: ### example 2 if ( has_body("application/sdp") ) rtpengine_offer(); To what extent is it correct to use one or another type of if statement? I understand that if we have to use else, then only the first option will suit us, but here it’s more about a single if.. thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: From iamhalje at gmail.com Tue Oct 31 10:09:00 2023 From: iamhalje at gmail.com (Dmitry Ponomaryov) Date: Tue, 31 Oct 2023 15:09:00 +0500 Subject: [OpenSIPS-Users] fix scrubbed text Message-ID: <7400c756-40a0-e8c7-c530-5ea548fb6480@gmail.com> ### example 1 if ( has_body("application/sdp") ) { rtpengine_offer(); } ### example 2 if ( has_body("application/sdp") ) rtpengine_offer(); -------------- next part -------------- An HTML attachment was scrubbed... URL: From liviu at opensips.org Tue Oct 31 10:13:37 2023 From: liviu at opensips.org (Liviu Chircu) Date: Tue, 31 Oct 2023 12:13:37 +0200 Subject: [OpenSIPS-Users] Correct syntax with statement if In-Reply-To: References: Message-ID: <2a00fc09-5a11-4b98-8c7e-acfc06fc7230@opensips.org> Hi, Dmitry! > how different will this expression be from: ### example 2 They are identical, regardless of the presence/absence of any code below. > To what extent is it correct to use one or another type of if statement? > It's not about correctness, rather readability.  Example 1 uses 3 LoC, while example 2 uses only LoC and it more readable, thanks to the absence of the braces. > I understand that if we have to use else, then only the first option > will suit us, but here it’s more about a single if.. thanks > I recommend going through the Linux kernel coding style docs, there is a "placing braces and spaces" section, which includes several if-statement examples, with braces, "else" blocks, etc.  Why not borrow the same ideas for our own /opensips.cfg/ coding style? :-) Best, -- Liviu Chircu www.twitter.com/liviuchircu |www.opensips-solutions.com OpenSIPS eBootcamp, Nov 6-17 |www.opensips.org/training -------------- next part -------------- An HTML attachment was scrubbed... URL: From iamhalje at gmail.com Tue Oct 31 10:30:48 2023 From: iamhalje at gmail.com (Dmitry Ponomaryov) Date: Tue, 31 Oct 2023 15:30:48 +0500 Subject: [OpenSIPS-Users] Correct syntax with statement if Message-ID: Hello Liviu I created a separate topic where I indicated the condensed syntax of my lines, but a separate question was created there... I’m still learning how to respond to a message in the mail list, sorry for this :) Initially I thought that the brackets, on the contrary, indicate that the text is easy to read, I will definitely look at your link to the code styling, thank you. From slackway2me at gmail.com Tue Oct 31 11:17:19 2023 From: slackway2me at gmail.com (A A) Date: Tue, 31 Oct 2023 16:17:19 +0500 Subject: [OpenSIPS-Users] fifo not found Message-ID: Sometimes a .fifo file is removed by Operating System from /tmp. This manual [1] may be helpful to you. [1]. https://alexeyka.zantsev.com/?p=1252 -- BR, Alexey From liviu at opensips.org Tue Oct 31 13:20:55 2023 From: liviu at opensips.org (Liviu Chircu) Date: Tue, 31 Oct 2023 15:20:55 +0200 Subject: [OpenSIPS-Users] 3.2.15 installation In-Reply-To: References: Message-ID: <734f9afa-3f91-4612-9927-880734fba9c7@opensips.org> Hi! It seems there was an issue related to the tarball packing introduced in the source tree a couple weeks *before* release day which confused me, as I thought I was just building the tarball incorrectly on release day (e.g. maybe due to a dirty directory or a bad script, etc.). A fix is now available and all opensips.org tarballs have been rebuilt.  Still, if you were to download the latest stable git tag and run "make tar" yourself, of course you'd run into the same bug again (/no change there, the git tag hasn't been moved/)... but that will also get resolved on the next stable release round in a month or so. Best regards, On 26.10.2023 09:53, L S wrote: > Thanks. We are having issues compiling from source on Centos (error > because of a patch related to wolfssl). Will try again. > > > > On Wed, Oct 25, 2023, 4:41 PM Knee Oh via Users > wrote: > > Yes, compiled from source on Ubuntu 22.04. About to move to > production. > > >> On Oct 25, 2023, at 4:35 PM, Joseph Jackson >> wrote: >> >>  >> We have but we use the debian packages and we installed it on >> release day. >> >> >> >> ------------------------------------------------------------------------ >> *From:* Users on behalf of L S >> >> *Sent:* Wednesday, October 25, 2023 12:46 AM >> *To:* OpenSIPS users mailling list >> *Subject:* [OpenSIPS-Users] 3.2.15 installation >> Has anyone successfully installed 3.2.15 (revised on Oct 20th)? >> >> Thanks, >> Matt >> >> ------------------------------------------------------------------------ >> *From:* Users on behalf of L S >> >> *Sent:* Wednesday, October 25, 2023 12:46 AM >> *To:* OpenSIPS users mailling list >> *Subject:* [OpenSIPS-Users] 3.2.15 installation >> Has anyone successfully installed 3.2.15 (revised on Oct 20th)? >> >> Thanks, >> Matt >> _______________________________________________ >> 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 -- Liviu Chircu www.twitter.com/liviuchircu |www.opensips-solutions.com OpenSIPS eBootcamp, Nov 6-17 |www.opensips.org/training -------------- next part -------------- An HTML attachment was scrubbed... URL: From Ben.Newlin at genesys.com Tue Oct 31 14:31:26 2023 From: Ben.Newlin at genesys.com (Ben Newlin) Date: Tue, 31 Oct 2023 14:31:26 +0000 Subject: [OpenSIPS-Users] Need some information regarding accounting module . In-Reply-To: References: Message-ID: Sasmita, A negative response to an in-dialog request does not automatically terminate the call. If the rejection of the re-INVITE is not acceptable to either party, they must still send a BYE if they desire to disconnect the call. After the 500 response, it is correct that there are no transactions as the re-INVITE transaction has been responded and closed. But the lack of active transactions does not mean the dialog/call is not still active. Based on your description the accounting seems correct. Any UAs involved that believed the call was disconnected based on the 500 response are not operating correctly. Ben Newlin From: Users on behalf of Sasmita Panda Date: Tuesday, October 31, 2023 at 4:21 AM To: OpenSIPS users mailling list Subject: [OpenSIPS-Users] Need some information regarding accounting module . EXTERNAL EMAIL - Please use caution with links and attachments ________________________________ Hi All , I just got an wired call example on which the ACC information is not as for my expectation . I have Opensips 1.11 version . Example : Call started at : Oct 17 19:24:40:512 Re-Invite on this session : Oct 17 19:24:40:810 For the Re-Invite The client got " 500 Server Internal Error " which Opensips gave . After that there was no transaction in the session . The call got disconnected due to the error message . But The ACC info for that call : Oct 18 07:33:32 (After 12 hr ) Although Opensips created the 500 error message still the session did not get cleaned and Acc was printed after so long with a bigger talktime : duration=43729 . Which is totally beyond my expectation . I have attached the opensips config file as well . I am not sure . Do I need to call drop_accounting while in an error scenario ? Thanks & Regards Sasmita Panda Senior Network Testing and Software Engineer 3CLogic , ph:07827611765 -------------- next part -------------- An HTML attachment was scrubbed... URL: From callum.guy at x-on.co.uk Tue Oct 31 16:21:43 2023 From: callum.guy at x-on.co.uk (Callum Guy) Date: Tue, 31 Oct 2023 16:21:43 +0000 Subject: [OpenSIPS-Users] Async rest_client timeout Message-ID: Hi All, I'm seeing a small number of sessions reporting timeouts for async rest_client post requests. These occur at peak times for system load and present the following error: ERROR:rest_client:_resume_async_http_req: connected, but transfer timed out (5s) The return code is -3 and HTTP response arrives as 0. My web server itself shows that the requests complete in <200ms. The _resume_async_http_req errors arrive ~400ms after the request was issued. I suspect that something is exhausted (i.e. too many async or curl sessions) however I am not aware of any configuration options that would help. The following settings are employed: modparam("rest_client", "connection_timeout", 4) modparam("rest_client", "curl_timeout", 5) modparam("rest_client", "max_async_transfers", 250) modparam("rest_client", "ssl_verifyhost", 0) Can anyone offer advice on where I should be looking to resolve this issue? version: opensips 3.2.10 (x86_64/linux) libcurl-7.76.1-23.el9_2.1.x86_64 Many thanks, Callum -- Voting link  (it takes less than 20 seconds!) *0333 332 0000  |  x-on.co.uk   |   **      **  |   **Practice Index Reviews * *Our new office address: 22 Riduna Park, Melton IP12 1QT.* X-on is a trading name of X-on Health Ltd a limited company registered in England and Wales. Registered Office : Glebe Farm, Down Street, Dummer, Basingstoke, Hampshire, England RG25 2AD. 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 slackway2me at gmail.com Tue Oct 31 11:12:59 2023 From: slackway2me at gmail.com (A A) Date: Tue, 31 Oct 2023 16:12:59 +0500 Subject: [OpenSIPS-Users] fifo not found Message-ID: Sometimes a .fifo file is removed by Operating System from /tmp. This manual [1] may be helpful to you. [1]. https://alexeyka.zantsev.com/?p=1252 -- BR, Alexey