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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.<br>
<br>
Good luck,<br>
-Brent<br>
<br>
Gonzalo Servat wrote:
<blockquote
cite="mid:dcc007e10803111718q795fd156u2d82c60c579b9c4a@mail.gmail.com"
type="cite">Hi All,<br>
<br>
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.<br>
<br>
Any ideas? I'm using 1.6.0-beta4.<br>
<br>
Thanks!<br>
Gonzalo<br>
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