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
--------
Cheers,
- Michael
Michael Furdyk
Director of Technology, TakingITGlobal
http://www.takingitglobal.org/