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
> AddQueueMembe
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 &
now of that fills our needs without
deadlocking and causing unnecessary downtime.
I hope 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
&g
blems. 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 Gr
need 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 PROTE
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, t
ura's 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?
, here.
> >
> > Note that the agent hears agentcallbacklogin's "the agent is logged off"
> > message, except the initial portion is cut off briefly.
> >
> > [macro-agent_logout]
> >
> > exten => s, 1, setglobalvar(agent=${ARG1})
> > exte
=> 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]
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erate DTMF INFO packets prior to the call
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 NoO
nsit
latency measurements taken at the phone were 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 ph
the whole configuration in there. My only problem with this is that I
can't specify per-MAC configuration elements easily this way, because
I'm provisioning most of the configuration via an HTTP cgi-bin script.
Tell me if you have any luck,
Alan Ferrency
pair Networks, Inc.
[EMAIL PROTECTE
But a more 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) sol
a? Do you have a lot of ethernet traffic? We found
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.
C
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