asterisk users - Jan 2006 - Disconnecting call 'SIP/X.X.X.X-085340d0' for lack of RTP activity in 11 seconds

Hi all!
This is my VoIP network scheme

 H323EndPoint -----                               --- GW H323/SIP-IN
--                -- SIP Phone
                                  |                             |      
     (Sipquest)           |              |
                                  |                            
|                                          |             |
                                  |                            
|                                          |             |
H323EndPoint --------- GK1 ---- GK2-|          
                                |-- SER ---- SIP Phone
                                  |                             |      
                                    |            |
                                  |                             |       
                                   |            |
                                  |                            
|                                           |            |
H323EndPoint -----                                --- GW
H323/SIP-OUT--              -- Asterisk as Voicemail
                                                                       
    (Sipquest)      

In calls between SIP to H323 endpoints it works fine . I have a problem
in calls between H323 endpoints with asterisk voicemail functionality.
In case of not response, the call is forwarded to asterisk voicemail by
SER Router but I obtain the following error:

-- Executing Set("SIP/X.X.X.X-085340d0", "LANGUAGE()=es") in
new stack
    -- Executing SetCallerID("SIP/X.X.X.X-085340d0",
"331223") in new stack
    -- Executing VoiceMail("SIP/X.X.X.X-085340d0",
"u331222@default") 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/2' (language 'es')
    -- Playing 'vm-isunavail' (language 'es')
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

Someone has an idea to arrange this problem

Thanks in advance!




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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