Hi again,

So, It was a very minor change in configuration and it was working. Basically FreeSwicth was bridging the two legs BUT there was a codec issue.
All I had to do was in Asterisk (serving as my gateway) to allow only ulaw and alaw
i.e

disallow=all
allow=ulaw
allow=alaw

What I really really wish to know is that why there was no indication of codec mismatch or ptime mismatch or sample rate mismatch while transcoding or anything.

It will be fine if  none replies but it will be great to know the real reason behind this and from where in logs can I verify this !!

Thanks
Sammy


On Sat, Jul 28, 2012 at 8:18 PM, SamyGo <govoiper@gmail.com> wrote:
Here are the FS console logs: 
http://pastebin.freeswitch.org/19595 

Please suggest what am I missing here.


On Sat, Jul 28, 2012 at 7:59 PM, SamyGo <govoiper@gmail.com> wrote:
Hello,
I wanted to make a lua script which just dials out two different numbers via some external gateway and when both calls are answered they are just bridged. For this a very impressive Lua example http://wiki.freeswitch.org/wiki/Mod_lua#Example:_Call_Control is copied and all I had to do was change the dialA and dialB strings and its working great as far as the SIP signalling is concerned.

execute this string and I get calls on two different number but things get interesting when Freeswitch bridge() the two legs. No AUDIO..not even one-way. I could see on my own gateway  that RTPs for both the legs are actually forwarded to Freeswitch !

On my sip pcap traces analyzing on wireshark I could actually hear the two persons saying Hello but neither could hear anything.

The above example lua call_control script is used as it is. 
Please suggest.

Regards
Sammy Go.