I am trying to host a conference meeting using Asterisk's ConfBridge. How to handle nat for a confBridge application like it is handled in sip.conf by specifying
nat=yes
Is there a way to configure something similar for confBridge. I went through confBridge.conf comments. But couldnt find any help.
The issue I am facing is that I am receving only one directional audio when I try to join 2 participants. What could be the possible reason?
EDIT:
Configuaration of the sip peers from sip.conf I have the sip peers configured correctly I guess. Here is my sip peers configuration.
[5001]
type=friend
username=aki
secret=password
callerid=aki<5001>
host=dynamic
context=t***e
nat=yes
[5002]
type=friend
username=apu
secret=password
callerid=apu<5002>
host=dynamic
context=t***e
nat=yes
This might help to get a better perspective.
I am writing my own sip client using jain-sip. The same configuration works well(bi-directional audio) when my sipclient and a 3rd party sip softphone are communicating directly through Asterisk. It is only when I use ConfBridge that the audio from a 3rd party sip softphone to my sipclient is not audible. I have tried different sip softphones and still the result is the same.
