[Asterisk-Users] Voicepulse incoming / dial extension

Anil Purohit Tue, 24 Aug 2004 20:25:21 -0700

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:[EMAIL PROTECTED]>

-- 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 '[EMAIL PROTECTED]' 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 23:14:39 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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[Asterisk-Users] Voicepulse incoming / dial extension

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