Re: [SR-Users] Problem routing to voicemail

:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/TCP 192.168.14.25;branch=z9hG4bK5c88.bc649332.0;i=c;received=192.168.14.25 Via: SIP/2.0/TCP 192.168.14.188:5060;rport=3075;branch=z9hG4bK1571 From: ;tag=19669 To: "Alexis" ;tag=as73c46699 Call-ID: 8250 CSeq: 21 BYE Server: Asterisk PBX 11.0.1 Allow: INV

Re: [SR-Users] Problem routing to voicemail

2012/11/15 Olle E. Johansson > > > 15 nov 2012 kl. 11:58 skrev Christophe ROY : > > Hi everyone > > I'm trying to integrate Asterisk with Kamailio for voicemail. > I tried to follow this tutorial: > http://kb.asipto.com/asterisk:realtime:kamailio-3.3.x-ast

[SR-Users] Get sip clients to work with sip:user@domain and ldap

Hello I'm quite new to SIP and TOIP so I may not understand the concepts correctly. I'm supposed to make work SIP clients (for now, linphone on android and windows) with a Kamailio 3.2 server and ldap authentication. Ldap authentication seems to be working fine, as the login/pass is validated aga