[asterisk-users] g729 on 1.4.10.1

Scott Moseman scmoseman at gmail.com
Tue Sep 18 15:19:39 CDT 2007


Follow me on this, it seems odd (or maybe I don't undertand)...

Test #1

[src_phone]
disallow=all
allow=g729

[dest_phone]
disallow=all
allow=g729

I can make the call (src to dest) and it will work using g729.
Both the call handling and media are going through Asterisk.

Test #2

[src_phone]
disallow=all
allow=g729
allow=ulaw

[dest_phone]
disallow=all
allow=g729

I can make the call (src to dest) and it will work using g729.
Both the call handling and media are going through Asterisk.

Test #3

[src_phone]
disallow=all
allow=ulaw
allow=g729

[dest_phone]
disallow=all
allow=g729

The above call attempt will fail, and this is what I'm seeing:
chan_sip.c:2944 sip_call: No audio format found to offer.

In every test, the source INVITE includes ulaw, alaw and 729.
That is the codecs that I configured on the phone themselves.

However, in Test #3 the call will fail.  Why?

This does not necessarily have to do with my g729 gateway,
but I'm curious what's wrong with this scenario, maybe using
this situation to understand will help me with my gateway...
(Although I tried setting only g729 on the gateway and the
gateway's peer in the Asterisk and it did not appear to help.)

Thanks,
Scott



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