[Asterisk-Dev] Re: [Asterisk-Users] Disconnecting call 'SIP/X.X.X.X-085340d0' for lack of RTP activity in 11 seconds

[steve at daviesfam.org]

On Wed, 18 Jan 2006, Javier Oviedo wrote:

> Jan 18 18:06:17 NOTICE[16386]: chan_sip.c:11213 do_monitor:
> Disconnecting call 'SIP/X.X.X.X-085340d0' for lack of RTP activity in 11
> seconds
> Jan 18 18:06:17 WARNING[17340]: file.c:583 ast_readaudio_callback:
> Failed to write frame
>   == Spawn extension (default, 331222, 3) exited non-zero on
> 'SIP/172.25.92.153-085340d0'
> 
> The channels has RTP activity because I hear the voicemail message
> 

The problem is that no RTP is coming from the other side (ie towards 
Asterisk).  This check is in case the other side has disappeared 
suddenly.  It doesn't help Asterisk to know that its transmitting.  It 
could transmit for hours and hours to nowhere and never know the other 
side is gone.  (that's UDP for you).

Best is to fix the original source so as to not do silence suppression. If
you can't do that, you can remove or lengthen the rtp timeout by adjusting
rtptimeout= and rtpholdtimeout= in the sip.conf file.

Regards,
Steve