[Asterisk-Users] SIP to SIP no audio help

[Michael Furdyk]

Hi everyone!

I've been working on setting up an Asterisk server and my two Digium
cards I ordered will arrive tommorow, so I'm excited to plug some 'real'
old-school lines into it.

But tonight I've been testing with some of our staff around the world,
and while handing off 'real' (PSTN - over VoIP using Voicepulse) calls
to SIP and SIP calls to VoIP PSTN works fine, SIP to SIP calls provide
no audio, and just this message on console:

    -- Executing NoOp("SIP/322-3edd", ""call for "331") in new stack
    -- Executing Dial("SIP/322-3edd", "SIP/331|60|tr") in new stack
    -- Called 331
    -- SIP/331-fd48 is ringing
    -- SIP/331-fd48 answered SIP/322-3edd
    -- Attempting native bridge of SIP/322-3edd and SIP/331-fd48

My sip.conf has nat on and canreinvite=no, and those were the only
suggestions I could find. Help would be greatly appreciated! We are
really excited about the potential of Asterisk.

Sip.conf:

[322]
type=friend
context=softphone ; match with the outgoing context in extensions.conf
host=dynamic ; This device needs to register
callerid="Michael Furdyk" <XXXX>
nat=yes ; X-Lite is behind a NAT router
canreinvite=no ; Typically set to NO if behind NAT
allow=all ; codec choice: GSM consumes far less bandwidth than ulaw
dtmfmode=rfc2833 ; Choices are inband, rfc2833, or info