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<DIV dir=ltr align=left><SPAN class=036331214-09042008><FONT face=Arial
color=#0000ff size=2>It would be interesting to see a wireshark trace of
the SIP and RTP traffic during call setup and hold, to
see:</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=036331214-09042008><FONT face=Arial
color=#0000ff size=2>a) what codec 126 has been negotiated as
and</FONT></SPAN></DIV>
<DIV dir=ltr align=left><SPAN class=036331214-09042008><FONT face=Arial
color=#0000ff size=2>b) who is sourcing the unknown RTP
datagram.</FONT></SPAN></DIV><BR>
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<FONT face=Tahoma size=2><B>From:</B> asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] <B>On Behalf Of </B>Adrian
A<BR><B>Sent:</B> 09 April 2008 00:55<BR><B>To:</B> Asterisk Users Mailing
List - Non-Commercial Discussion<BR><B>Subject:</B> [asterisk-users] RTCP not
being sent when on hold<BR></FONT><BR></DIV>
<DIV></DIV>Hello,<BR><BR>When I receive a call to my CounterPath Bria from
Asterisk <A href="http://1.4.18.1">1.4.18.1</A> and I place the call on hold,
the call is dropped after 30 seconds.<BR>It looks like there is no RTCP/RTP
sent to the client from Asterisk while on hold (music on hold playing to
caller) thus client disconnects the call. During this time, I get the
following messages in the CLI:<BR><BR>NOTICE[24194] rtp.c: Unknown RTP codec
126 received from '<A href="http://0.0.0.0">0.0.0.0</A>'<BR><BR>In sip.conf I
have rtpkeepalive=15 but that does not seem to help.<BR><BR>Does anyone know
what I can do to fix this, other than increase the timeout on
Bria?<BR><BR>Thanks,<BR>Adrian<BR></BLOCKQUOTE></BODY></HTML>