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@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@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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Andrew Thompson
2004-Aug-25 06:14 UTC
[Asterisk-Users] Voicepulse incoming / dial extension
Anil Purohit wrote:> 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<snip>> 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.Are you referring to two separate inbound calls, where the first one works and the second one does not? Or, are you saying that they can dial an extension, no one answers so they dial another, but nothing happens? A little more info, please. ----- Andrew Thompson http://aktzero.com/
Andrew: - first call comes in, hears my welcome message and the caller dials extension 2001 and it rings my phone. In this instance, I see all the "Executing ...." messages on my console - I answer the call and there is voice path between the two parties - the same caller calls again and he hears the welcome message but this time I do not see any of the "Executing ....." messages on the Asterisk console. Also, the welcome message completes and Asterisk ignores all of the callers dtmf commands to go to an extension. I can now restart Asterisk and it works just the first time. I am using the Asterisk 1.0 RC1 version that I had downloaded on Aug 4th. Thanks for your time. Regards, Anil Anil Purohit wrote:> 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<snip>> 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.Are you referring to two separate inbound calls, where the first one works and the second one does not? Or, are you saying that they can dial an extension, no one answers so they dial another, but nothing happens? A little more info, please. ----- Andrew Thompson http://aktzero.com/