On 04/20/2011 10:02 AM, Oguzhan Kayhan wrote:
Hello,
We have a sip trunk between our voip operator and our asterisk 1.6.2.9
We have no problem during voice communications.
But we can not send any t38 fax via this gateway.
We tried to trace the error made some tests..
There are 2 main tests we tried to do.
As i learned their voip path is like .. we connect to session border
controller..then it routes the call to a cisco media gateway if the call is
originated thru a pstn/telco line.
First test is to send the fax to a client in their SBC device.it was a direct
sip2sip fax call. And it succeeded.
Then when we tried to fax to a pstn number fax hung up because of
communication error.
The only error i received was
Unknown RTP codec 100 received from xxx.xxx.xxx.xx
If i got it right, they say for normal calls they use 99 as dtmf payload. But
for fax they use 100. And they asked me if there is a way that i can change
dtmf payloads on asterisk or not??
So.. what can i do..or what should i try??
There isn't enough information here for us to be able to help you;
you're going to need to provide a packet capture, or 'sip set debug on'
console capture (at least) for anyone to be able to determine what might
be happening.
The quick answer, though, is that Asterisk will use whatever payload
number for RFC2833 DTMF that the other end requests. The message you are
seeing has nothing to do with DTMF.
--
Kevin P. Fleming
Digium, Inc. | Director of Software Technologies
Jabber: [email protected] | SIP: [email protected] | Skype: kpfleming
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at www.digium.com & www.asterisk.org
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