Sip trunk configuration between Cisco and Yate

[samir.sakhare]

Hello,
Our server has two Ethernet ports ( eth0 and eth1) .
ETH0 is configured to my local network ( 192.168.x.x) for sip client .where the sip clients are working on UDP port 5060.
ETH1 is configured for cisco sip trunk ( 10.10.x.x.).Cisco works on TCP port.
On this server Yate application is running.

In ysipchan.conf file I configured TCP listener. In regexroute.conf file I created routing from Cisco to sip clients.

Outgoing Calls can be  made  from my sip client to Cisco .
When  I Try incoming call came from Cisco to my sip client via yate, the call rings on sip clients but when I  try  to  access the same call  the call is dropping.

Please help me with this issues.

Thanks

[marian]

Hi,
Can you post a yate log with dopped call?
SIP messages + message sniffer?

[samir.sakhare]

Hi,
  This is my incoming call yate log.

Thanks

[marian]

Hi,
Log is incomplete (missing call end).
Log is missing SIP signalling.

[samir.sakhare]

Hi,
     I  am attaching log file again. Please  tell me Is is right .


Thanks

[marian]

Seems to be related to original incoming call starting without media offer.
The answer is returned with no media also.
You may try to configure the sender to offer some media.

[samir.sakhare]

Hi,

  When I stopped yate and started Twinkle ( sip client ) on the server.  I gave incoming  call from sip trunk to the server where twinkle running. Twinkle started ringing and answered the call . The call got established with  proper voice.
  No call drop.
   But when I starts yate and incoming call  is routed through the Yate then incoming call drops.
     

[samir.sakhare]

Hi ,

    My problem has been solved.
    I did rtp_start=enable in Ysipchan.conf.


Thanks