I don't know if this is something asterisk can do at the moment but on my
setup, it does not.
What I intend to do is, while a client is in a call, it will send an
in-dialog re-invite to asterisk (after changes on the client i.e. IP
address). Asterisk should handle this and update internal dialog.
- Original Message -
From: John Kiniston johnkinis...@gmail.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, July 28, 2015 12:12:05 PM
Subject: Re: [asterisk-users] Queues don't follow dialplan if no members are
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Thanks for your reply Larry.
Le 27/07/2015 01:22, Larry Moore a écrit :
I think the 488 Not acceptable here is occurring because the channel
connecting through is not T.38 capable, that will be the IAX channel
from iaxmomdem.
This is what
Hello,
I am running Asterisk 11 on CentOS 6.x. I have configured several queues as
follows in extensions.conf:
exten = s,1,Queue(myqueue,rtnC,18)
same = n,Background(user_unavail)
same = n,WaitExten(10)
exten = 1,1,Voicemail(@my-vm,s)
This rings the phones in the queue for 18 seconds. If no
In your queues.conf do you have a leavewhenempty and joinempty set?
in queues.conf
[myqueue]
leavewhenempty = strict
joinempty = strict
strategy = ringall
ringinuse = no
On Tue, Jul 28, 2015 at 9:58 AM, Andrew Martin amar...@xes-inc.com wrote:
Hello,
I am running Asterisk 11 on CentOS 6.x.
I'm planning on upgrading to Asterisk 13.4 soon and am looking for the
corresponding Siren7 codec. Where do I find it?
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