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<DIV><!--StartFragment --> <SPAN class=postbody>Hi everyone <BR><BR>I am
having trouble with codec negotiation. I have Asterisk running at the office and
a SIP phone at home. In my sip.conf, I have allow ordered as follows: <BR>alaw
ulaw g729 and gsm <BR><BR>On all my office extensions, I have no allow, or
disallow entries. My Cisco gateway is setup to do alaw ulaw g729 and gsm
<BR><BR>My home phone does g729 alaw and ulaw. In sip.conf, I have disallow all
<BR>and allow g729. In all my extensions and cisco gateway I have canreinvite
set to yes and my dial commands don't have the t option, so all sip endpoints
can talk directly to each other (rtp). <BR><BR>If I call from home to the
office, calls go through fine. SIP show channels, shows that the call is g729 as
one would expect. <BR><BR>If I get a call from Office to home, or from PSTN (via
Cisco) to home, the phone rings, but as soon as I answer it hangs up.
<BR><BR>Asterisk says: <BR><BR>May 29 05:45:49 WARNING[7514]: channel.c:2115
ast_channel_make_compatible: No path to translate from SIP/390-8a3b(256) to
SIP/192.168.44.23-08acccf0(<IMG alt=Cool
src="http://forums.digium.com/images/smiles/icon_cool.gif" border=0> <BR>--
SIP/390-8a3b is ringing <BR>-- SIP/390-8a3b answered SIP/192.168.44.23-08acccf0
<BR>May 29 05:45:55 WARNING[7514]: channel.c:2115 ast_channel_make_compatible:
No path to translate from SIP/192.168.44.23-08acccf0(<IMG alt=Cool
src="http://forums.digium.com/images/smiles/icon_cool.gif" border=0> to
SIP/390-8a3b(256) <BR>May 29 05:45:55 WARNING[7514]: app_dial.c:1006 dial_exec:
Had to drop call because I couldn't make SIP/192.168.44.23-08acccf0 compatible
with SIP/390-8a3b <BR>== Spawn extension (default, 390, 1) exited non-zero on
'SIP/192.168.44.23-08acccf0' <BR><BR><SPAN class=861523110-01062005><FONT
face=Arial size=2>SNIP.....</FONT></SPAN></SPAN></DIV>
<DIV><SPAN class=postbody><SPAN class=861523110-01062005><FONT face=Arial
size=2></FONT></SPAN></SPAN> </DIV>
<DIV><SPAN class=postbody><SPAN class=861523110-01062005><FONT face=Arial
size=2>One week later. I have now purchased two 9.729 licences as I suspected
Asterisk was not allowing direct endpoint negotiation. Now my home phone
answers, but it is receiving on 9.729 and sending on
g.711a</FONT></SPAN></SPAN></DIV>
<DIV><SPAN class=postbody><SPAN class=861523110-01062005><FONT face=Arial
size=2></FONT></SPAN></SPAN> </DIV>
<DIV><SPAN class=postbody><SPAN class=861523110-01062005><FONT face=Arial
size=2>This leads to delays building up on the g711a
side.</FONT></SPAN></SPAN></DIV>
<DIV><SPAN class=postbody><SPAN class=861523110-01062005><FONT face=Arial
size=2></FONT></SPAN></SPAN> </DIV>
<DIV><SPAN class=postbody><SPAN class=861523110-01062005><FONT face=Arial
size=2>I want to calls coming to my office phones to use alaw as the prefered
codec in sip.conf, but I want calls to my home and remote sites to be g.729.
Asterisk seems to ignore the codec negotiation phase and insists on running two
different codecs in two directions. Most sip servers will always use the same
codec in both directions based on the first agreed codec. As my home phone is
set in sip.conf to only allow g.729, then it should do g.729 in both directions.
I see this as a bug. Anyone know how to make it work
properly?</FONT></SPAN></SPAN></DIV>
<DIV><SPAN class=postbody><FONT face=Arial size=2></FONT></SPAN> </DIV>
<DIV><SPAN class=postbody>Thanks to the gurus who might no the answer to this
one. <BR><BR>Cheers <BR><BR>Mark</SPAN> </DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV>
<P><SPAN class=861523110-01062005><FONT face=Arial size=2>P.S. Why do the real
experts not use the users web forum? Much easier to manage than a mailing
list.</FONT></SPAN></P></DIV></BODY></HTML>