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

[Javier Oviedo]

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 at 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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