Hi All;
To simplify the the login and logout for the agent, I am looking for the
variable that can be used for the AddQueueMember (in the place of the ?? as
following:
exten => 100,1,AddQueueMember(CustomerSupport,${},1)
exten => 100,2,Playback(agent-loginok)
exten => 101,1,RemoveQu
Could it be a weather or other telco problem?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tim Nelson
Sent: Thursday, November 10, 2011 2:45 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subj
On Thursday 10 November 2011 2:09:09 pm Danny Nicholas wrote:
> You might want to see if it is
> 1. a phone or an asterisk transfer - phone transfer hits button on phone
> and does attended/blind transfer that way; asterisk transfer initiated
> with *1 or #2 (whatever value is specified in features
You might want to see if it is
1. a phone or an asterisk transfer - phone transfer hits button on phone and
does attended/blind transfer that way; asterisk transfer initiated with *1
or #2 (whatever value is specified in features.conf)
2. attended or blind transfer.
-Original Message-
From
I'll explore the options outlined in the document below, later tonight.
However, I've been able to reproduce the problem! It seems that when one of
my users, at a particular site, transfers a call to another extension,
asterisk bounces.
They're using Polycom 301's and 501's with SIP version
Greetings-
On occasion, I'm seeing the following in syslog on some systems using analog
cards with FXS modules:
[ 1664.861183] Power alarm on module 1, resetting!
These are typically cleared by restarting asterisk/dahdi, or power cycling the
system. However, I'm wondering if anyone can explain
Unfortunately, I didn't compile with DON'T_OPTIMIZE. Would this render my
backtrace.txt completely useless or should I still submit?
Thanks,
--E
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Leif Madsen
Sen
On 11/10/2011 12:33 PM, Russell Brown wrote:
Yes it was to /etc/asterisk/chan_dahdi.conf (oh how I wish the
problem was being caused by something that simple!).
It was worth a shot. I have wasted time looking for solutions that were
that simple.
I am curious about the D-Channel not being
On 11-11-10 01:15 PM, Eric Wieling wrote:
> The Asterisk source tree has a document with instructions on getting a
> backtrace from the segfaults so you can report it on the issue tracker.
Most up to date documentation is on the Asterisk wiki:
https://wiki.asterisk.org/wiki/display/AST/Debugging
>
>On 11/10/2011 09:38 AM, Russell Brown wrote:
>> I copied /etc/asterisk/zapata.conf to /etc/chan_dahdi.conf
>
>Did you really copy it to /etc/chan_dahdi.conf or to
>/etc/asterisk/chan_dahdi.conf ?
Yes it was to /etc/asterisk/chan_dahdi.conf (oh how I wish the
problem was being caused by somethi
The Asterisk source tree has a document with instructions on getting a
backtrace from the segfaults so you can report it on the issue tracker.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of eherr
Sent: Thursd
I am having similar issues with Asterisk 1.4.26
It happens at random times; could be once a day or a few hours in between up to
a month or so.
Haven't found a solution to my problem yet either.
--E
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-user
Hi all,
I've recently started experiencing frequent * restarts, usually caused by a
segfault. So, I moved my users to a different server and the same thing
happened. It's a fairly busy box, so I considered memory exhaustion. Nope,
not even swapping.
Now I'm thinking file handles, but I won'
On Thu, Nov 10, 2011 at 12:24 PM, Leif Madsen
wrote:
> On 11-11-10 12:12 PM, Danny Nicholas wrote:
>> Yeah! My boss will be much happier having a system that doesn't have the
>> -tail on it.
>
> I hear this kind of statement every once in a while, which makes absolutely no
> sense to me. If you'r
Could we have more hours in the day to play with all the goodiness ? cannot
keep up with everything at the moment :)
--
Thanks, Phil
- Original Message -
> I know what you mean - I'd rather have a working x-beta1 that a
> failing x.0
>
> -Original Message-
> From: asterisk-users
I know what you mean - I'd rather have a working x-beta1 that a failing x.0
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Leif Madsen
Sent: Thursday, November 10, 2011 11:24 AM
To: asterisk-users@lists.digium.
On 11-11-10 12:12 PM, Danny Nicholas wrote:
> Yeah! My boss will be much happier having a system that doesn't have the
> -tail on it.
I hear this kind of statement every once in a while, which makes absolutely no
sense to me. If you're blindly running a version of any software in production
(rega
On 11-11-09 04:30 PM, Danny Nicholas wrote:
> If you have an "ancient" version of Asterisk you want to stick with, you can
> do this with asterisk -rx "sip set debug on" and asterisk -rx "agi set debug
> on" in your safe_asterisk script.
Not sure about AGI, but pretty sure the sipdebug=yes option
On 11-11-07 08:38 AM, Bryant Zimmerman wrote:
> I have a test box that has been running asterisk without issue. I updated
> it to 1.8.8.0-rc2 and now I am getting some wierd issues I have never seen
> before.
>
>
> All the modules seem to have compiled without issue.
>
>
> Asterisk starts up
Yeah! My boss will be much happier having a system that doesn't have the
-tail on it.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Leif Madsen
Sent: Thursday, November 10, 2011 11:06 AM
To: asterisk-users@li
> I recently turned up some 1.8.6.0 call servers in productions, SIP trunks in
> routing calls to upstream carrier via SIP trunks out. I spent a lot of time
> in the lab testing 1.8 which included heavily testing DTMF with no issues
> that came up. It all just seemed to work fine. But then again
On 11-11-10 11:57 AM, Danny Nicholas wrote:
> Misspoke - when should we expect 10.0 that is not -rc or -beta?
Well Asterisk 10.0.0 is now in release candidate status, which means pending any
major issues or regressions, a full Asterisk 10.0.0 is inevitably due in the
near future.
Leif.
--
__
Misspoke - when should we expect 10.0 that is not -rc or -beta?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Leif Madsen
Sent: Thursday, November 10, 2011 10:55 AM
To: asterisk-users@lists.digium.com
Subject
> Il 09/11/2011 17.37, Richard Mudgett ha scritto:
> > You would then use
> > the DAHDISendCallreroutingFacility application*before* you answer
> > the call to forward/deflect the incoming call back to the network.
>
> I think "Answer" makes no sense at all because the network will
> redirect
> th
On 11-11-10 11:43 AM, Danny Nicholas wrote:
> Does this mean a "non-beta" labeled Asterisk 10.0 is due out shortly?
I'm confused by shortly the announcement means it is out now. That was the
purpose of the announcement...
http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-10.0.0-rc
Does this mean a "non-beta" labeled Asterisk 10.0 is due out shortly?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Asterisk
Development Team
Sent: Thursday, November 10, 2011 10:39 AM
To: Asterisk Development
The Asterisk Development Team is pleased to announce the first release candidate
of Asterisk 10.0.0. This release candidate is available for immediate download
at http://downloads.asterisk.org/pub/telephony/asterisk/
All Asterisk users are encouraged to participate in the Asterisk 10 testing
proce
On 11/10/2011 09:38 AM, Russell Brown wrote:
I copied /etc/asterisk/zapata.conf to /etc/chan_dahdi.conf
Did you really copy it to /etc/chan_dahdi.conf or to
/etc/asterisk/chan_dahdi.conf ?
Also,
What is the output of 'module reload chan_dahdi'?
Dale
--
"The truth speaks for itself. I'
I bit the bullet last night and upgraded from Asterisk 1.4.42 with
Zaptel 1.4.12.1 to Asterisk 1.8.7.1 and DAHDI 2.5.0.2 (libpri is 1.4.12
for both configs).
Everything went reasonably well except for calls to and from an ISDN PBX
on span 2. which now just won't work :-(
There's obviously s
ChanisAvail?
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jaap Winius
Sent: Wednesday, November 09, 2011 9:29 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] IAX2 avai
Il 09/11/2011 17.37, Richard Mudgett ha scritto:
You would then use
the DAHDISendCallreroutingFacility application*before* you answer
the call to forward/deflect the incoming call back to the network.
I think "Answer" makes no sense at all because the network will redirect
then continue to ca
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