I am trying to make inter-asterisk communication through h323 & oh323 protocol using SIP phones as the end devices
But the thing is when i make calls the calls get established but there is no audio transfer
I think the RTP transfer is not getting established…
People please help in this
Thanks in advance
Port configuration issues come up all the time. I aill have to refer you to te following thread to check on your various port configs and forwardings including rtp.conf (also reaad the bottom NOTE on the page):
forums.digium.com/viewtopic.php?t=7854
Thanks man
Your info was really very helpful
But the actual thing is i dint mention the codecs to be spefied in sip.conf for the sip phones i was using to transmit calls via the h323 channel
added the lines
disallow=all
allow=alaw
allow=ulaw
allow=gsm
for each sip phone
this solved the problem