Hi, i see this tutorial http://kb.asipto.com/asterisk:realtime:kamailio-3.3.x-asterisk-10.7.0-astdb, I've done step by step as described in the tutorial, but i have a problem, when A extension dial to B extension, B extension doesn't ringing...
I send my sip.conf and kamailio.cfg
Addittional, asterisk and kamailio are installed on same server, which have one private IP (192.168.50.217) and public IP 200.41.110.94
Hello,
can you get the sip trace with ngrep of such call on kamailio server port 5060?
Cheers, Daniel
On 16/04/14 19:50, William Baylón Huerta wrote:
Hi, i see this tutorial http://kb.asipto.com/asterisk:realtime:kamailio-3.3.x-asterisk-10.7.0-astdb, I've done step by step as described in the tutorial, but i have a problem, when A extension dial to B extension, B extension doesn't ringing...
I send my sip.conf and kamailio.cfg
Addittional, asterisk and kamailio are installed on same server, which have one private IP (192.168.50.217) and public IP 200.41.110.94
-- *Kevin W. Baylón Huerta* *Departamento Sistemas - Sitatel.
*Teléfono**: (51) 17073501 Anexo 208* *Celular: (51) 989715598*
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
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Daniel-Constantin Mierla -
William Baylón Huerta