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

Thu, 19 Jan 2006 03:06:27 -0800

[EMAIL PROTECTED] wrote: 
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

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Hi Steve, thanks for your response, my h323 endpoints have the silence suppression option set a off. I remove the rtptimeout and rtoholdtimeout options in the sip.conf file and now I obtain the following error:


*CLI>     -- Executing Set("SIP/X.X.X.X-09f3ebf8", "LANGUAGE()=es") in new stack
    -- Executing SetCallerID("SIP/X.X.X.X-09f3ebf8", "331222") in new stack
    -- Executing VoiceMail("SIP/X.X.X.X-09f3ebf8", "u331223") in new stack
    -- Playing 'vm-theperson' (language 'es')
    -- Playing 'digits/3' (language 'es')
    -- Playing 'digits/3' (language 'es')
    -- Playing 'digits/1' (language 'es')
    -- Playing 'digits/2' (language 'es')
    -- Playing 'digits/2' (language 'es')
    -- Playing 'digits/3' (language 'es')
    -- Playing 'vm-isunavail' (language 'es')
    -- Playing 'vm-intro' (language 'es')
    -- Playing 'beep' (language 'es')
    -- Recording the message
    -- x=0, open writing:  /var/spool/asterisk/voicemail/default/331223/INBOX/msg0011 format: wav49, 0x9ef4f60
Jan 19 11:51:11 WARNING[19282]: app.c:653 ast_play_and_record: No audio available on SIP/X.X.X.X-09f3ebf8??
    -- User hung up

I think that it's a rare behavior of asterisk because the problem only ocurs in "Not Response" case study but not in "Busy" or "Unavailable" responses.

Thanks in advance!

Regards

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