[Asterisk-Users] Voicepulse incoming / dial extension

[Anil Purohit]

All:
 
I am trying to use Voicepulse as my incoming line and want the caller to
simply dial the extension of the party they want to reach.
 
Here is my problem:
 
-          the first time they dial it works fine and  I see the
following on my console
 
 
Aug 24 23:14:31 DEBUG[-1126876240]: chan_sip.c:4408 build_route:
build_route: Contact hop: <sip:6035057098 at 66.234.228.137>
    -- Executing Wait("SIP/s00227156-a5ef", "2") in new stack
    -- Executing Answer("SIP/s00227156-a5ef", "") in new stack
    -- Executing DigitTimeout("SIP/s00227156-a5ef", "10") in new stack
    -- Set Digit Timeout to 10
    -- Executing ResponseTimeout("SIP/s00227156-a5ef", "30") in new
stack
    -- Set Response Timeout to 30
    -- Executing BackGround("SIP/s00227156-a5ef", "welcome-mainmenu") in
new stack
Aug 24 23:14:33 DEBUG[-1221325904]: rtp.c:1146 ast_rtp_write: Ooh,
format changed from UNKN to ULAW
Aug 24 23:14:33 DEBUG[-1221325904]: channel.c:1101 ast_settimeout:
Scheduling timer at 160 sample intervals
    -- Playing 'welcome-mainmenu' (language 'en')
Aug 24 23:14:33 DEBUG[-1126876240]: chan_sip.c:817 __sip_ack: Stopping
retransmission on '5bd8e9e869d4a8306d256fa02f387e09 at 66.234.228.137' of
Response 103: Found
Aug 24 23:14:37 DEBUG[-1221325904]: rtp.c:186 send_dtmf: Sending dtmf:
50 (2), at 66.234.228.137
Aug 24 23:14:37 DEBUG[-1221325904]: channel.c:1101 ast_settimeout:
Scheduling timer at 0 sample intervals
Aug 24 23:14:37 DEBUG[-1221325904]: pbx.c:1801 ast_pbx_run: Oooh, got
something to jump out with ('2')!
Aug 24 23:14:38 DEBUG[-1221325904]: rtp.c:186 send_dtmf: Sending dtmf:
48 (0), at 66.234.228.137
Aug 24 23:14:38 DEBUG[-1221325904]: rtp.c:186 send_dtmf: Sending dtmf:
48 (0), at 66.234.228.137
Aug 24 23:14:39 DEBUG[-1221325904]: rtp.c:186 send_dtmf: Sending dtmf:
49 (1), at 66.234.228.137
  == CDR updated on SIP/s00227156-a5ef

    -- Executing Dial("SIP/s00227156-a5ef", "SIP/2001|60|tr") in new
stack
Aug 24 23:14:39 DEBUG[-1221325904]: app_dial.c:468 dial_exec: SIMPLE
DIAL (NO URL)
 
 
The problem is that the second time the caller dials my voicepulse
number, I do not see the "Executing ... " debug statements in my console
and it ignores my dtmf and the call is NOT transferred to the extension.
The call is just dropped after the timeout of 30 secs. Nothing on my
console.
 
Any ideas would be greatly appreciated. Also, please do a reply-all so
that I can receive e-mail directly. 
 
Thanks,
 
Anil
 
 
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