<div>Hi,</div>
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<div>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:
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<div>Oct 6 10:57:45 NOTICE[11026]: channel.c:1409 ast_read: Dropping incompatible vo<br>ice frame on <a href="mailto:Local/1604xxx8621@hk-8073,2">Local/1604xxx8621@hk-8073,2</a> of format gsm since our native format h<br>
as changed to g729<br>Oct 6 10:57:45 NOTICE[11032]: channel.c:1409 ast_read: Dropping incompatible vo<br>ice frame on <a href="mailto:Local/xxx25837550@van-c9ae,2">Local/xxx25837550@van-c9ae,2</a> of format ulaw since our native form
<br>at has changed to slin</div>
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<div>I searched the list and found similar topic on <a href="http://lists.digium.com/pipermail/asterisk-users/2005-May/104942.html">http://lists.digium.com/pipermail/asterisk-users/2005-May/104942.html</a>, 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.</div>
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<div>Anyone has similar experience?</div>
<div>Please advice.</div>
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<div>Thanks.</div>
<div>AK</div>