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ansi
2007 Oct 01
1
Odd one way RTP on SIP to SIP calls
Hi everyone,
I'm having an odd problem with one way RTP on SIP to SIP calls.
I have two SIP servers, one is an Asterisk and the remote SIP server
is a Nortel SIP server.
When a call comes to the Nortel server through the PSTN and is routed
to the Asterisk, audio is fine. Two way RTP and no problems. When a
SIP client registered on the Nortel server calls the Asterisk, the
Asterisk