[asterisk-users] call is not going to Voicemail with "1,n"
William Stillwell
william at stillwellsoft.com
Thu Dec 30 06:38:43 UTC 2010
The n = prev + 1
So you dialplan technically looks like this:
exten => 1,1,Playback(transfer)
exten => 1,2,Dial(${sales_support}&IAX2/iaxy-322,20,jrw)
exten => 1,103,Voicemail(11,b)
exten => 1,104,Hangup()
exten => 1,105,Voicemail(11,b) ; Right to voicemail
exten => 1,106,Hangup()
which in the result, there is no 1,3 which goes auto fallthru.
Try this instead:
exten => 1,1,Playback(transfer)
exten => 1,n,Dial(${sales_support}&IAX2/iaxy-322,20,jrw)
exten => 1,n,Voicemail(11,b)
exten => 1,n,Hangup()
exten => 1,n+101,Voicemail(11,b)
exten => 1,n,Hangup()
which will result in a DP looking like this:
exten => 1,1,Playback(transfer)
exten => 1,2,Dial(${sales_support}&IAX2/iaxy-322,20,jrw)
exten => 1,3,Voicemail(11,b)
exten => 1,4,Hangup()
exten => 1,105,Voicemail(11,b)
exten => 1,106,Hangup()
-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Joseph
Sent: Wednesday, December 29, 2010 11:56 PM
To: asterisk-users at lists.digium.com
Subject: [asterisk-users] call is not going to Voicemail with "1,n"
I've tried to simplified the dial plan and use "n" instead of numbers but
I've noticed it is not executing my voicemail if I substitute number with
"n"
In the example below when the call is not answered, it does not go to
voicemail; call just hangup.
exten => 1,1,Playback(transfer)
exten => 1,n,Dial(${sales_support}&IAX2/iaxy-322,20,jrw)
exten => 1,103,Voicemail(11,b)
exten => 1,104,Hangup()
exten => 1,n,Voicemail(11,b) ; Right to voicemail
exten => 1,n,Hangup()
Here is the transcript:
-- Executing [1 at office-open:1] Playback("SIP/pstn-5665-000000be",
"transfer") in new stack
-- <SIP/pstn-5665-000000be> Playing 'transfer' (language 'en')
-- Executing [1 at office-open:2] Dial("SIP/pstn-5665-000000be",
"SIP/11&IAX2/iaxy-322|20|jrw") in new stack
-- Called 11
-- Called iaxy-322
-- Call accepted by 10.0.0.108 (format ulaw)
-- Format for call is ulaw
-- IAX2/iaxy-322-8406 is busy
-- Hungup 'IAX2/iaxy-322-8406'
-- SIP/11-000000bf is ringing
-- Nobody picked up in 20000 ms
== Auto fallthrough, channel 'SIP/pstn-5665-000000be' status is
'NOANSWER'
However, if I number the dial plan in the old fashion way and don't answer
the phone it goes to voicemail just fine:
exten => 1,1,Playback(transfer)
exten => 1,2,Dial(${sales_support}&IAX2/iaxy-322,20,jrw)
exten => 1,103,Voicemail(11,b)
exten => 1,104,Hangup()
exten => 1,3,Voicemail(11,b) ; Right to voicemail
exten => 1,4,Hangup()
--
Joseph
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
More information about the asterisk-users
mailing list