[asterisk-users] Strange RTP problem...

[Carlos Chavez]

	I have a new installation where an Asterisk server is connected to an
Avaya PBX via a PRI E1.  We are having a problem that I attribute to
their firewall but I just want to make sure.

	When we make a call from the Avaya to a SIP extension there is only
sound on the receiving end.  From what I can see in the CLI is that the
moment the SIP endpoint answers RTP packets stop reaching the Asterisk
server.  The SIP endpoint can hear the other person but nothing reaches
back to Asterisk.  On this same machine I can make calls from one SIP
extension to another (both external to the network) and sound goes both
ways.  The problem only seems to happen when the call originates from
the Avaya, but only if the SIP phone is outside the local network.

	The Asterisk server is behind NAT and I have setup the "externip" and
"localnet" to reflect the proper values.  The firewall has been
configured to forward ports 5060, 19000-20000 (range also in rtp.conf)
to the Asterisk server.  I have "nat=yes" and "canreinvite=no" in
sip.conf for all SIP endpoints.

	Why would the firewall stop rtp coming into the server only when the
call was originated from the PRI and not on SIP to SIP calls.

	Anyone had a similar experience?

-- 
Telecomunicaciones Abiertas de México S.A. de C.V.
Carlos Chávez Prats
Director de Tecnología
+52-55-91169161 ext 2001
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