Re: [SR-Users] just a quick glance at my kamailio.cfg (suspect a rtpproxy_manage behavior)

Hi Daniel, debbuger module does not come handy, in fact it told me that yes rtpproxy_manage is executed two times in a raw (look at the xlog tag AFTER RTPPROXY_MANAGE) rtpproxy[16]: DBUG:GLOBAL:rtpc_doreply: sending reply "1\n" {1 2 INVITE 79961MDUxZmM4YWEyNThlMjUxM2FkNTA3M2Q3NWYwNTZjZTk} 8(30)

[SR-Users] just a quick glance at my kamailio.cfg (suspect a rtpproxy_manage behavior)

Hi guys i am bridging call on a private ip asterisk behind kamailio/rtpproxy i am calling rtpproxy with the route [FROMASTERISK] route[FROMASTERISK] { if(ds_is_from_list()){ xlog("L_INFO","$dd Call from Media-Server Cluster\n"); rtpproxy_manage("cawei"); SIP INVITE with sdp always

[SR-Users] 404", "Not Found location"

Hi guys when i try to register i have route[LOCATION] that returns "404 Not Found location" due to this statement: $avp(oexten) = $rU; if (!lookup("location")) { $var(rc) = $rc; route(TOVOICEMAIL); t_newtran(); swit

Re: [SR-Users] advertise xxx.xxx.xxx.xxx:5060 is weird

Hi i am using Kamailio 4.1.x for centos - Mail original - De: "Daniel-Constantin Mierla" À: "Sébastien Brice" Cc: "sr-users" Envoyé: Mercredi 1 Juin 2016 09:05:35 Objet: Re: [SR-Users] advertise xxx.xxx.xxx.xxx:5060 is weird Hello, On 30/05/16 20:2

[SR-Users] advertise xxx.xxx.xxx.xxx:5060 is weird

Hi guys, I have a local proxy running on private ip 192.168.3.97 and advertising the wan ip of my box (THEPUBLICIPOFMYBOX) this local Kamailio is a stateful proxy that TM REGISTERS and INVITES to a Kamailio registrar(PUBLICIPKAMAILIOREGISTRAR) (please have a look at the kamailio.cfg attached to

[SR-Users] T_RELAY_TO distribute local proxy to a central one

Hi guys, i have a kamailio running on the wan (my central kamailio) and i have phones on a private network range 172.20.0.0/16 behind a NAT. i also run another kamailio locally to collect all sip phones on my network and this one use the DB of the remote central kamailio. the local kamailio doe

Re: [SR-Users] DISPATCHER module No destinations availables even if my Asterisk Boxes are up and running

:: sip:172.16.0.202:5060 flags=IP priority=0 attrs=weight=50 URI:: sip:172.16.0.203:5060 flags=IP priority=0 attrs=weight=50 No ACTIVE gateway has been elected, i don't know what's wrong here. - Mail original - De: "Daniel-Constantin Mierla" À: "Séb

Re: [SR-Users] DISPATCHER module No destinations availables even if my Asterisk Boxes are up and running

? Thanks. - Mail original - De: "Marrold" À: "Sébastien Brice" , "sr-users" Envoyé: Lundi 18 Avril 2016 01:16:05 Objet: Re: [SR-Users] DISPATCHER module No destinations availables even if my Asterisk Boxes are up and running First of all I would check the result of

[SR-Users] DISPATCHER module No destinations availables even if my Asterisk Boxes are up and running

Hello everyone. I'm using kamailio as a dispatcher in front of asterisk boxes and i use a failure route if asterisk box does not respond or send 503 error (please look at my attached kamailio.cfg) I use "ds_probing_mode", 1 and "ds_ping_reply_codes", "class=2;class=3;class=4" to OPTIONS ping th

[SR-Users] Kamailio local and hosted

C) i did setup kamailio/asterisk in a real-time configuration. Sébastien BRICE VoIP, Support et Intégration ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

[SR-Users] Local Kamailio share a hosted Kamailio/Asterisk sip server

did setup kamailio/asterisk in a real-time configuration. Sébastien BRICE VoIP, Support et Intégration ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

[SR-Users] want to go mobile, common issues ?

Hi everyone. I have enjoyed talk about realtime of last FOSDEM, talk about K. was good. I want to know how do you solve sip dialogs with mobile? actually I know REGISTER is important to authenticate and this request generally last 1hour until then you have to register again. How to solve that fo

[SR-Users] ovh trunk

Hi everyone I followed this guide http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb and got it working (101, 102 and 103) can call eachother. But now i am trying to figure Asterisk's role out. I am more an ipbx person and i am used to register providers trunk in asterisk

[SR-Users] kamailio in a multi-site configuration

to maintain internal call if my internet provider get down by exemple. thx you for your idea. Sébastien BRICE VoIP, Support et Intégration ___ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http

[SR-Users] kamailio 4 with asterisk

ailio. By the way i took a good start with kamailio as it seems to work flawlessly on my system. thx you. On Mon, Feb 15, 2016 at 12:26:06PM +0100, Sébastien Brice wrote: > Hi Everyone, i like the way this tutorial explain asterisk and kamailio > integration. Which tutorial? > the on

[SR-Users] kamailio 4 with asterisk

Hi Everyone, i like the way this tutorial explain asterisk and kamailio integration. the only thing i missed is asterisk behaviors'r regarding sip registration ? I tryed to place a call between 102 and 103 extensions and experimenting an issue Asterisk tells me that the subscriber is absent an

[SR-Users] Kamailio 4 && Asterisk. Getting started

Hi everyone I did follow the tutorial Kamailio 4.0.x and asterisk written by Daniel-C and I need a hand :-) after a successful install (kamailio and asterisk on the same server), i did setup 2 sip extensions (102 and 103) and tryed to place a call between them and experimenting an issue Aste