hi
I am using..
Asterisk : 1.6.0.5
Kamailio : 1.5.0
Here, is network diagram ...
172.18.100.10/20/30 ============
192.168.1.68 ========================== 192.168.1.70
(SIP Phone Register on this
IP) (Kamailio
IP)
(Asterisk Server)
And here link for kamailio file
I have registered 111 and 222 user on asterisk (192.168.1.70)... and call to
kamailio user (1212(a)domain.com)... call established successfully.. but sip
phone is not hangup..
As well as i call from kamailio user like 1212 to 2121 ... call established
.. but sip phone not hangup..
Help me out....
Thanks in advance
--
Regards,
Chandrakant Solanki