Hello Chandrakant,

Problem in your Asterisk configuration of sip.conf, Please check the Trunk creation in sip.conf with Asterisk, Where Kamailio trunk is registered in Asterisk and See whether you getting option Message to Kamailio Server from Asterisk Server.

It should works fine, both scenarios

Thanks &Regards Ravi Prakash Sunkara VoIP Architect & JAVA-SIP Developer +91-9999882776

Bette Davis http://www.brainyquote.com/quotes/authors/b/bette_davis.html - "Brought up to respect the conventions, love had to end in marriage. I'm afraid it did."

2009/5/29 Chandrakant Solanki solanki.chandrakant@gmail.com

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@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

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