Your log shows rtp streams being allocated.
did you look at at the packets on the wire with a packet capture program?
You are better off using proper jingle and component mode. What you are describing sounds like
a workaround to avoid doing it right.
I modified mod_dingaling.c so I can intercept google talk chat messages
via socket - nothing fancy.
Then I wrote a java app that connects to freeswitch socket and in case of
a proper message (trigger) it sends a command to freeswitch, in my case:
api originate sofia/default/1001@10.99.8.221
&bridge(dingaling/gmail.com/my_mail@gmail.com)
Dingaling is logged in as another user which I have added as buddy, chat
messages go trough, however when a call is started
between SIP and Gtalk client, we cannot hear each other at all.
Using freeswitch revision: 10866
Here is the log:
http://pastebin.com/m1eba2cb8
What can be the problem? First I thought it is because running sip client
+ gtalk and freeswitch on one host, but then I
moved SIP phone and Gtalk to 2 different workstations, using the third
only for freeswitch. Also calls from "call" example program
from google lib works fine with same setup - something must be problematic
with freeswitch, however cannot see what.
Thank you!
--
kriko
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