[asterisk-users] dial option 'g' not working
Faisal Hanif
faisal at vopium.com
Wed Feb 9 23:58:14 CST 2011
There are some flags in general settings of dialplan which enable/disable & modify this behaviors of dialplan. Have a look on sample extensions.conf for general tab settings. I will see if I can have time today to tell you exact parameter name.
From: Dovid Bender
Sent: Thursday, February 10, 2011 9:07 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] dial option 'g' not working
Hi,
I had the same issue as well but for some reason I was unable to reproduce. Please have a loo at: https://issues.asterisk.org/view.php?id=18682
Regards,
Dovid
----- Original Message -----
From: M S
To: asterisk-users at lists.digium.com
Sent: Wednesday, February 09, 2011 06:11
Subject: [asterisk-users] dial option 'g' not working
Hi,
I'm trying to get my dialplan to continue executing in the current context after a third-party is called and hangs up. It seems like it should be straightforward but it's not working.
Here's what I have in extensions.conf:
exten => 333,1,Answer()
exten => 333,n,Playback(hello)
exten => 333,n,Dial(SIP/19992223333 at sipcarrier,,g)
exten => 333,n,Playback(hello)
exten => 333,n,Playback(hello)
exten => 333,n,Playback(hello)
exten => 333,n,Hangup()
The 9992223333 number is dialed, but after that party hangs up, there's just dead air. No hello's are played and nothing seems to be happening.
What am I doing wrong?
Thanks,
MS
------------------------------------------------------------------------------
--
_____________________________________________________________________
-- 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
--------------------------------------------------------------------------------
--
_____________________________________________________________________
-- 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
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110210/a6fa4d6f/attachment.htm>
More information about the asterisk-users
mailing list