On Wed, 11 Apr 2007, Kevin P. Fleming wrote:
Alan Ferrency wrote:
This means that all queue activity is associated with a SIP channel
in the logs, which is not acceptable.
This is why we added the 'membername' argument to the
AddQueueMember application, so that queue logs can reflect
I apologize for not responding sooner, I obviously don't read this
mailing list regularly.
Alan Ferrency wrote:
In our investigation of the AddQueueMember vs.
AgentCallbackLogin situation, the major loss with using the
published AddQueueMember replacement is that it assumes each agent
. The only thing I find
slightly less than optimal is that for major configuration changes,
the phones seem to need a factory reset to pick up the changes in a
timely manner.
Alan Ferrency
On Mon, 12 Feb 2007, George Pajari wrote:
Aastra are a delight -- no need for a compiler (like the Grandstream
this helps,
Alan Ferrency
On Wed, 14 Feb 2007, gc wrote:
So you have to hard code the each queue name in the dialplan for an
agent to login. What about hundreds of agents login 30-40 different
queues? If this is the only way to do it, I will not use
AddQueueMember at all. I do not know
access to the AMI as we do.
But yeah, I'd expect Asterisk has diverged a lot since rami was last
updated. I did a round of refactoring at the time we were initially
developing our screen pop app, but none of it has had to change in
over a year.
Alan Ferrency
pair Networks, Inc.
[EMAIL PROTECTED
site are still valuable
resources for the Linksys phones. Only specific configuration elements
have changed.
Alan Ferrency
pair Networks, Inc.
[EMAIL PROTECTED]
On Mon, 10 Apr 2006, Kerry Garrison wrote:
Has anyone got any information on bulk provisioning of Linksys SPA-941/94s
phones calling their Asterisk box, and failing to do
DTMF correctly.
Alan Ferrency
pair Networks, Inc.
[EMAIL PROTECTED]
On Thu, 13 Apr 2006, Aaron Daniel wrote:
Anyone have any ideas why DTMF would not work on only one number? Looking
through the logs, anytime a button is pressed
= logout, 1, noop(agent logout ${agent})
exten = logout, 2, wait(1)
exten = logout, 3, agentcallbacklogin(${agent},,@shared_phones)
I hope this helps.
Alan Ferrency
pair Networks, Inc.
[EMAIL PROTECTED]
___
--Bandwidth and Colocation
=${ARG1})
exten = s, 2, noop(agent ${agent})
exten = s, 3, dial(local/[EMAIL PROTECTED]/n,,D(w#)o)
exten = logout, 1, noop(agent logout ${agent})
exten = logout, 2, wait(1)
exten = logout, 3, agentcallbacklogin(${agent},,@shared_phones)
I hope this helps.
Alan Ferrency
pair Networks, Inc.
[EMAIL
being answered.
A complete SIP debug log (with some console verbosity as well) follows.
Thank you very much for your help.
Alan Ferrency
pair Networks, Inc.
[EMAIL PROTECTED]
--- SIP DEBUG LOG -
-- Executing NoOp(Zap/3-1, Dialing staff extension sip
not problems at all.
Alan Ferrency
pair Networks, Inc.
[EMAIL PROTECTED]
On Wed, 1 Mar 2006, mustardman29 wrote:
Would QoS on a managed switch solve the ARP problem?
Regarding sound quality issues with Sipura SPA-841 phones...
snip
After that: Are your phones sharing the same network segments
that even on a fully switched network, if the SPA-841's received
excessive ARP traffic (which is broadcast to all switch segments, even
though most other network packets are suppressed), we had periodic
robot voice sound issues.
Check this out, and see if it helps.
Thanks!
Alan Ferrency
pair Networks
generic goto context would be more
useful I'd expect.
I know this is possible by having the internal extension simply transfer
the call to a different extension; however, this is not a suitable
solution in this case. I need a one button press (dtmf tone) solution.
Thanks,
Alan Ferrency
pair Networks
of the configuration via an HTTP cgi-bin script.
Tell me if you have any luck,
Alan Ferrency
pair Networks, Inc.
[EMAIL PROTECTED]
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http
14 matches
Mail list logo
