So, LOCAL in this context is a 'Technology' or 'Channel Driver' , Instead
of PJSIP, SIP, IAX, it's sending a call to a dialplan target.
Your entry in queues.conf with LOCAL/105@internal would send the call to
the context 'internal' extension '105' and execute whatever that dialplan
does.
The
Got it working! Thanks a lot again. As a bonus, is there a background on why
SIP/ did not work with a sip trunk provider? :)
From: John Kiniston
To: Ivan Demkovitch
Cc: Asterisk Users Mailing List - Non-Commercial Discussion
Sent: Friday, November 16, 2018 3:08 PM
Subject: Re:
John,
Thanks for reply! I use 13.1-cert1, plain vanilla Asterisk. Installed and
configured as per book..
So, from what I understand - LOCAL means I want local extension to be a member
of a queue.
For example, I have this:
[internal]
;Eric on extension 105
exten =>
My settings for the queue.log are in the [general] section of logger.conf
I'm running 13, I didn't see what version you said you were running.
If I wanted to add a LOCAL channel to my queue I'd do it as
member => LOCAL/7124@kiniston-intern,0,John,hint:7124@kiniston-intern
On Thu, Nov 15, 2018
John,
FF1565AABB2D-SLS is probably invalid because it's not registered/lost
registration. This client is connected via VPN to our network, it usually works
when it's "warm". Not concerned about it too much.
155@callcentric OTOH is an actual cell phone that should be dialed out
via
From: John Kiniston
To: idemkovi...@yahoo.com
Sent: Thursday, November 15, 2018 3:17 PM
Subject: Re: [asterisk-users] Queue not dialing out to cell phone for some
reason
OK.
So it looks like asterisk can't ring FF1565AABB2D-SLS because it's invalid.
is the user at
John,
This is output of command below. How do I enable and log queue events?The
1555@callcentric is the one I'm curious about. I just tried calling into
"sales" again and it didn't change this "last was 1219067" output
Sales has 0 calls (max unlimited) in 'ringall' strategy (9s holdtime, 156s
what does the output of 'queue show sales' show?
Do you have queue logging enabled? Have you looked in the queue log to see
what events are firing?
On Thu, Nov 15, 2018 at 9:55 AM Ivan Demkovitch
wrote:
> Hello,
>
> I have queues.conf setup with a group like so:
>
> [Sales](StandardQueue)
>
Aha, I tought you had a SIP client (like MizuDroid or similiar) that registred
via data connection to the asterisk server.
Seems theres a problem with the trunk then.
What does ”sip show registry” tell you?
(asterisk -r in console and then sip show registry)
It should show a status
Sebastian,
I don't think it has to do anything with registration. It is dialing through
the SIP trunk, so it goes out as normal cell phone call.Also, why I don't see
anything in a log? I see only first 2 members being dialed.
From: Sebastian Nielsen
To: 'Ivan Demkovitch' ; 'Asterisk
I would suspect that the cell phone does use battery saving causing the SIP
application to lose registration with the server. Would also suggest using TCP
with a fairly short keepalive to prevent the cellular network from tearing down
the connection to the asterisk server.
You need to go into
Hello,
I have queues.conf setup with a group like so:
[Sales](StandardQueue)
announce = first
member => SIP/FF4C119EEBF8-SLS
member => SIP/FF9EF375CCFC-SLS
member => SIP/1314555@callcentric ;Eric's cell
member => SIP/FF1565AABB2D-SLS ;Eric's Yealink
So, my idea here that it should ring all 4
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