Sorry for the top-post...
If you do a core show application AddQueueMember from the cli, you'll see the
option I was referring to.
You'll also need to make sure you're properly reporting device state to
asterisk. I think this means you need to set a call-limit for each sip peer
that you want
You'll also need to make sure you're properly reporting device state to
asterisk. I think this means you need to set a call-limit for each sip peer
that you want to monitor in sip.conf (we use 25 so there are no accidental
limits actually applied), and setup hints in your extensions.conf
15.10.2010 9:40, Warren Selby пишет:
I think this means you need to set a call-limit for each sip peer
Is there any alternative for obsolete call-limit option in 1.6/1.8?
Thanks,
--Warren Selby
On Oct 14, 2010, at 11:36 PM, Matt Darnellmattdarn...@gmail.com wrote:
Warren,
I tried using
On 10-10-15 04:10 AM, Сикорский Сергей wrote:
15.10.2010 9:40, Warren Selby пишет:
I think this means you need to set a call-limit for each sip peer
Is there any alternative for obsolete call-limit option in 1.6/1.8?
The correct answer is to use ringinuse=no in queues.conf and callcounter=yes
On Fri, Oct 15, 2010 at 1:21 AM, Leif Madsen
leif.mad...@asteriskdocs.org wrote:
On 10-10-15 04:10 AM, Сикорский Сергей wrote:
15.10.2010 9:40, Warren Selby пишет:
I think this means you need to set a call-limit for each sip peer
Is there any alternative for obsolete call-limit option in
2010/10/15 Matt Darnell mattdarn...@gmail.com:
On Fri, Oct 15, 2010 at 1:21 AM, Leif Madsen
leif.mad...@asteriskdocs.org wrote:
On 10-10-15 04:10 AM, Сикорский Сергей wrote:
15.10.2010 9:40, Warren Selby пишет:
I think this means you need to set a call-limit for each sip peer
Is there any
We have a queue that agents log into through the dial plan. Extension
Sip/101 logs in as Agent/101
We have 'ringinuse = no' in the queues.conf file.
The issue is that when Ext 101 is on a 'non queue' call (they placed a
call, someone called their DID, etc) they still receive queue calls.
Is
What version of asterisk are you using and method are you using to login your
agents? I recently had this issue with a 1.4.33 install where the agents
logged in with agentcallbacklogin. In the end I had to move them away from
chan_agent altogether, using dynamic agents and AddQueueMember,
Warren,
I tried using AddQueueMember to add agents.
If they a user is on a call asterisk shows:
Members:
SIP/101 (dynamic) (Not in use) has taken no calls yet
No Callers
We are using 1.4.36.
What did you use to keep track of the extension state? Didn't see any
option for that at
On Thu, Oct 14, 2010 at 6:04 PM, Warren Selby wcse...@selbytech.com wrote:
What version of asterisk are you using and method are you using to login your
agents? I recently had this issue with a 1.4.33 install where the agents
logged in with agentcallbacklogin. In the end I had to move them
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