Eric Wieling wrote:
You are doing it wrong. I know 50 bazillion Asterisk dialplan examples on the
internet do it the same way. It is still wrong.
When you do a Dial on the dialplan you need check the value of DIALSTATUS or
HANGUPCAUSE before dialing again. Both variables will give you some indication
of why the first call ended. Then your dialplan logic can decide how to
proceed.
Thanks for your help.
In previous versions of asterisk it worked, and iirc after the called
party hung up, the dialplan only progressed if there was a particular
flag used with Dial (g?).
It's going to cause a heck of a headache but I'll look into doing this
properly in the week.
-----Original Message-----
From: [email protected]
[mailto:[email protected]] On Behalf Of Thomas Kenyon
Sent: Monday, September 24, 2012 7:00 AM
To: [email protected]
Subject: [asterisk-users] Peculiar problem with failover provision.
I have noticed a peculiar problem recently with the way that the failover
operates in my dialplan.
I normally have:
1,Dial(SIP/<provider-1>/extension)
n,Dial(SIP/<provider-2>/extension)
(or something similar).
This has up until now worked flawlessly.
If there is an error with the first provider, the call is completed with the
second one.
Now, what is happening is, if the remote party hags up first, then the call
progresses to the next priority and re-dials them.
Is this a change in default behaviour?
Do I need to add a particular flag / config directive to my dialplan
I am running Asterisk 10.6.0.
Thanks for any help in solving this.
--
_____________________________________________________________________
-- 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
--
_____________________________________________________________________
-- 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
