Hello folks,
It seems that i have a unique problem. So, we have distributed some cisco
spa303 phone and connected them to our asterisk box. We have also lots of
cisco 7911 phones connected Cisco CallManager. We integrated whole system.
But there is a problem let me illustrate it. In our campus
Hi,
I would like to edit reports showing how fast operator and users answer
incoming calls.
Users are spread over 6 locations, each with its own asterisk instance.
Operator is on main site.
Users have casual extension but operator logs as queue agent.
I've read or/and tried Star2Billing's
Hello,
how do I set the language for the VoiceMailMain()-command ?
How do I set the language per voicemail-box ?
Thanks,
Jonas.
--
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-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk?
Creating new subscription
Sending to 10.199.74.5:6060 (no NAT)
Found peer '720001' for '720001' from 10.199.74.5:6060
Looking for 720001 in default (domain open-ims.test)
--- Transmitting (no NAT) to 10.199.74.5:6060 ---
SIP/2.0 404 Not Found
These are the errors I am getting on
The problem was incompetible codec, thanks all.
2013/1/11 Onur Cem Çelebi occel...@gmail.com
Hello folks,
It seems that i have a unique problem. So, we have distributed some cisco
spa303 phone and connected them to our asterisk box. We have also lots of
cisco 7911 phones connected Cisco
AFAIK, the ${CHANNEL(language)} is what controls each. If you wanted to
answer the phone in English, then do voicemails in different languages, this
should work:
[default]
Exten = s,1,Answer()
Exten = s,n,Set(CHANNEL(language)=en) - redundant since default is en
Exten =
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier
Sent: Friday, January 11, 2013 4:22 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Which tool to edit custom reports from CDR and
queues
Thanks you for your answer.
There is no language-parameter that can define the language of mailbox
and VoiceMailMain ?
Jonas.
On 01/11/2013 03:33 PM, Danny Nicholas wrote:
AFAIK, the ${CHANNEL(language)} is what controls each. If you wanted
to answer the phone in English, then do
No. It is purposely set from the dialplan. In Asterisk 11.X you have the
[zonemessage] section in voicemail.conf that could probably be tweaked to
change the language without dialplan changes. Also in sip.conf you can set
language by peer so you could have something like
[London]
Type = peer
Hello,
are you sure that the language-parameter of the SIP peer will
influence the language used by VoiceMailMain() ?
Jonas.
On 01/11/2013 04:07 PM, Danny Nicholas wrote:
No. It is purposely set from the dialplan. In Asterisk 11.X you have
the [zonemessage] section in voicemail.conf
Since the peer language sets CHANNEL(language), I can say yes with
reasonable certainly. Like anything else here, you don't really know until
you try it on your box.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens
Sent:
Well, I thought you had tried it and thus could tell it with 100% certainty.
Thanks for your help.
Jonas.
On 01/11/2013 04:16 PM, Danny Nicholas wrote:
Since the peer language sets CHANNEL(language), I can say yes with
reasonable certainly. Like anything else here, you don't really know
Tried it just now and that is indeed the way it works (100% for me).
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens
Sent: Friday, January 11, 2013 9:22 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Hello,
I have a 16 port FXS device for register analog phones. I see TTL (time tol
ive for re-register) option in fxs menu and I have to chose a time between
10 to 7200 second.
All ports going to unregister after the time what I choose. Im getting a
registration failed message. In
quick question that leaves alittle confusion here. Im confused on the
difference or when to use the other if i have 1 = sign or 2 == signs .. so
If i had
exten = _,1,answer()
same= n,Set($[${a}==1]?true:false] --double equal sign
same = n(true),Goto(main,s,1)
same= n(false),
We are just delivering version 2 of our ADTransform data connector.
It would allow your to read in your CDR files, manipulate them, validate
them and put out JasperReports based on the data.
It has a plug-in based workflow engine so that file transfers,
input,transformation, validation,
Hello,
On 11.01.2013 17:34, Emre Özcan (Alfacom) wrote:
I have a 16 port FXS device for register analog phones. I see TTL
(time tol ive for re-register) option in fxs menu and I have to chose
a time between 10 to 7200 second.
All ports going to unregister after the time what I choose. Im
On Friday 11 January 2013, penguin wrote:
quick question that leaves alittle confusion here. Im confused on the
difference or when to use the other if i have 1 = sign or 2 == signs .. so
If i had
exten = _,1,answer()
same= n,Set($[${a}==1]?true:false] --double equal sign
same
On 01/11/2013 12:20 PM, A J Stiles wrote:
I try to write comparisons as != where possible and then there is no
confusion and less mistakes possible.
Most compilers will warn on the example below now.
On Friday 11 January 2013, penguin wrote:
quick question that leaves alittle confusion
On Fri, 11 Jan 2013, jon pounder wrote:
I try to write comparisons as != where possible and then there is no
confusion and less mistakes possible. Most compilers will warn on the
example below now.
Or you can write comparisons as 'constant operator variable' like:
if (0 ==
I only see that message when I have sip debug enabled. It appears harmless.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Emre Özcan
(Alfacom)
Sent: Friday, January 11, 2013 10:34 AM
To:
In Asterisk extensions.conf and extensions.ael inside $[] = and == are the
same comparison operator. I can't quote where I saw this, but it has been
documented somewhere. The == was added to make things more programmer
friendly.
-Original Message-
From:
Had my Asterisk instance stop responding to incoming/outgoing calls today.
Had to kill -9 the asterisk process and restart it to get it back. Not
really looking for help on that as the instance is version 1.6 and is due
to be replaced with an upgraded version shortly.
However, this does make
The general rule seems to be, don't restart it unless there's a problem or
you hear of memory leaks. I had a version of 1.4 that I restarted every
night because I read about memory leaks, but I hear of 1.2 installs that
have been running continuously for 10 years.
From:
On Fri, Jan 11, 2013 at 2:06 PM, Kevin Larsen
kevin.lar...@pioneerballoon.com wrote:
However, this does make me wonder, do you restart periodically to try to
avoid issues or do you just let things run until there is a problem? This
box had 119 days of up time on the Asterisk process. I have a
Hey everyone,
I just put in a fix for the underlying issue that was causing this to
occur. It will be out in a future Asterisk 11 release. If you want the
change now and are comfortable using patch you can retrieve the diff at:
File,
thanks for that quick fix! Using it now.
-- kuj
On Fri, Jan 11, 2013 at 4:09 PM, Joshua Colp jc...@digium.com wrote:
Hey everyone,
I just put in a fix for the underlying issue that was causing this to
occur. It will be out in a future Asterisk 11 release. If you want the
change now
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