Hello, When I receive a call to my CounterPath Bria from Asterisk 1.4.18.1 and I place the call on hold, the call is dropped after 30 seconds. 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: NOTICE[24194] rtp.c: Unknown RTP codec 126 received from '0.0.0.0' In sip.conf I have rtpkeepalive=15 but that does not seem to help. Does anyone know what I can do to fix this, other than increase the timeout on Bria? Thanks, Adrian -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080408/0f205c55/attachment.htm
On Tue, 8 Apr 2008, Adrian A wrote:> Hello, > > When I receive a call to my CounterPath Bria from Asterisk 1.4.18.1 and I > place the call on hold, the call is dropped after 30 seconds. > 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: > > NOTICE[24194] rtp.c: Unknown RTP codec 126 received from '0.0.0.0' > > In sip.conf I have rtpkeepalive=15 but that does not seem to help. > > Does anyone know what I can do to fix this, other than increase the timeout > on Bria?Are you also recording the call? I had to put this: [options] transmit_silence_during_record = yes into asterisk.conf to stop hangups after 30 seconds ... Gordon
Adrian A wrote:> Hello, > > When I receive a call to my CounterPath Bria from Asterisk 1.4.18.1 > <http://1.4.18.1> and I place the call on hold, the call is dropped > after 30 seconds. > 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: > > NOTICE[24194] rtp.c: Unknown RTP codec 126 received from '0.0.0.0 > <http://0.0.0.0>' > > In sip.conf I have rtpkeepalive=15 but that does not seem to help. > > Does anyone know what I can do to fix this, other than increase the > timeout on Bria? > > Thanks, > AdrianIs it not up to the phone to send the keep-alive packets? Sounds like Asterisk does not understand the keep-alive packets coming from the phone. Try setting "rtptimeout=300" in sip.conf to test this. It should now hangup after 5 minutes. regards, Drew regards, Drew -- Drew Gibson Systems Administrator OANDA Corporation www.oanda.com
It would be interesting to see a wireshark trace of the SIP and RTP traffic during call setup and hold, to see: a) what codec 126 has been negotiated as and b) who is sourcing the unknown RTP datagram. ________________________________ From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Adrian A Sent: 09 April 2008 00:55 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] RTCP not being sent when on hold Hello, When I receive a call to my CounterPath Bria from Asterisk 1.4.18.1 and I place the call on hold, the call is dropped after 30 seconds. 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: NOTICE[24194] rtp.c: Unknown RTP codec 126 received from '0.0.0.0' In sip.conf I have rtpkeepalive=15 but that does not seem to help. Does anyone know what I can do to fix this, other than increase the timeout on Bria? Thanks, Adrian -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080409/7a03a042/attachment.htm