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

[Javier Oviedo]

steve at daviesfam.org 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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