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<p class=MsoNormal><span lang=EN-US>Hi all,<o:p></o:p></span></p>
<p class=MsoNormal><span lang=EN-US>I have this installation:<o:p></o:p></span></p>
<p class=MsoNormal><span lang=EN-US>Asterisk 1.6.1.1 with h323 support, pwlib_v1_10_3
and openh323_v1_18_0. <o:p></o:p></span></p>
<p class=MsoNormal><span lang=EN-US>I have a problem that is, when a call
comes from H323 and goes to a Sip phone the asterisk sends two rtp streams to
the sip. I checked this with tcpdump, save the payload (voice is in G711u), one
is the ringing indication and the other is the voice coming from the user in
h323 side. And worst they go to the same port… This causes that in the
sip phone there are problems, when the call is answered sometimes we get the
riging indication, others a mix of the two with very bad sound quality and
others(few) a god audio call. <o:p></o:p></span></p>
<p class=MsoNormal><span lang=EN-US>The outgoing calls from sip to H323 are ok.<o:p></o:p></span></p>
<p class=MsoNormal><span lang=EN-US>I also tested an incoming call from a dahdi
channel and from here everything is ok, only one rtp stream and a good call.<o:p></o:p></span></p>
<p class=MsoNormal><span lang=EN-US><o:p> </o:p></span></p>
<p class=MsoNormal><span lang=EN-US>By the way I had other problem that I
fixed, but don’t know if it was in the best way. <o:p></o:p></span></p>
<p class=MsoNormal><span lang=EN-US>The h323 box is a Cisco AS5300 (or 5350?)
and when I was making outgoing calls the AS disconnected all of them after 10
sec. <o:p></o:p></span></p>
<p class=MsoNormal><span lang=EN-US> I investigated I noticed that the AS as
a limitation to the G711 payload to 20 ms, and asterisk was using 150 ms. I
resolve this changing in frame.c the codec value and recompile asterisk. There
is simpler way to do this? Like changing values in codec.conf?...<o:p></o:p></span></p>
<p class=MsoNormal><span lang=EN-US><o:p> </o:p></span></p>
<p class=MsoNormal><span lang=EN-US>Regards<o:p></o:p></span></p>
<p class=MsoNormal><span lang=EN-US>LS <o:p></o:p></span></p>
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