Hi,
thanks a lot. That helped me going most of my intended way. The only
thing is it still calls even busy lines (shown in use by show
queues) with either roundrobin (which is marked deprecated) or rrmemory
method. Did I miss something while reading howtos?
Thanks in advance. Regards, Christophorus
How about a call queue using the roundrobin strategy?
http://www.voip-info.org/wiki/view/Asterisk+call+queues
Dave
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Christophorus
Laube
Sent: Friday, November 14, 2008 11:29 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] no dial to busy sip line
Hi list,
is it possible to get in the running dialplan the status of (SIP) lines
without using AGI or anything like that? What I want is a stepwise
calling: I have several SIP lines (let's say they are three) which I
want to dial to alternatingly. But I do not want to dial to a already
busy line and catch the busy. Instead I do not want to dial to that peer
but to the next one. I want to have a kind of a adaptive dialplan.
Using AGI and such things just makes it slower in my opinion (if I call
an AGI script that does an asterisk -rx 'sip show channels' |gawk -F
{' print $1 '}, for example). Does anyone of you have an idea of how
to do that?
Thanks in advance. Best regards,
Christophorus Laube
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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 --
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 --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users