[Asterisk-Users] Codec issue? Dropping incompatible voice frame ...

[Andy Kuo]

Hi,
 When I call forward on PAP2, the incoming call will right the forwarded
number. However, there is one-way voice problem. The caller can hear the
destination(the forwarded number), but after the called party answers, the
caller can't hear anything. Then the CLI> produce continuous errors as
following:
 Oct 6 10:57:45 NOTICE[11026]: channel.c:1409 ast_read: Dropping
incompatible vo
ice frame on Local/1604xxx8621 at hk-8073,2 of format gsm since our native
format h
as changed to g729
Oct 6 10:57:45 NOTICE[11032]: channel.c:1409 ast_read: Dropping incompatible
vo
ice frame on Local/xxx25837550 at van-c9ae,2 of format ulaw since our native
form
at has changed to slin
 I searched the list and found similar topic on
http://lists.digium.com/pipermail/asterisk-users/2005-May/104942.html, and
used their advice by adding "Answer" before "Dial" in extensions.conf, and
"canreinvite=no" in sip.conf. It worked in the way that I was able to get
2-way communication, but pages and pages of the above messages are still
there.
 Anyone has similar experience?
Please advice.
 Thanks.
AK
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