Do you have canreinvite=no in the sip client configuration? If not then
the two sip phones are probably issuing a reinvite command and taking
asterisk out of the call path. If that happens and the phones can't
reach consensus on a codec then you run into audio problems. If you're
not a provider and just using asterisk as a PBX then it's probably
better to set the phones up with a matching codec set and allow them to
establish a direct connection between each other to keep load off the
Asterisk server. Otherwise set canreinvite=no and Asterisk should
transcode correctly.
Good luck,
-Brent
Gonzalo Servat wrote:
Hi All,
I have 2 SIP clients configured and connected to Asterisk. When I
place a call from SIP1 to SIP2, if both codecs are the same then
everything works as expected. I then allowed one of the clients to use
alaw instead of ulaw and there were audio problems (couldn't hear the
other end, etc). Same thing happened when I tried to use gsm<->alaw/ulaw.
Any ideas? I'm using 1.6.0-beta4.
Thanks!
Gonzalo
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