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<DIV><FONT face=Arial size=2>Hi,</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>I'm using IAXphone for remote users which limits me
to the GSM codec. </FONT></DIV>
<DIV><FONT face=Arial size=2>Internally I limit the SIP phones to
iLBC codec (GS 101 1.0.5.0)</FONT></DIV>
<DIV><FONT face=Arial size=2>I also use voiptalk.org for external PSTN access
again using the iLBC codec.</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>The problem I have is that when the IAXphone dials
an internal phone or PSTN number either the line hangs up immediately or there
is only one way audio from IAXphone.</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>It works as soon as I allow GSM codec on the
GS phones.</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>Is there any way I can debug this
issue?</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>Asterisk CVS-04/10/04-15:32:35 built by <A
href="mailto:root@asterisk">root@asterisk</A> on a i686 running
Linux</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>Cheers</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>Giles Scott</FONT></DIV>
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