Hello All,
We have been experiencing some ongoing reliability problems with
Asterisk for quite some time, and I am trying to find out if anyone else
has experienced the same problems.
We are running asterisk 1.4.17~dfsg-2+b1 on Debian Lenny, with a Digium
PRI card, and have approximately 120 sip
Hi All,
Thanks for all the replies. Here are my responses to the responses:
On Tue, 2008-03-18 at 06:13 -0400, Al Baker wrote:
Curious, you mention a number of problems that have gone on for months
Question: Have you reported ANY or ALL of them to DIGIUM and if so
what has
Jared Smith wrote:
No, unfortunately this was done under NDA, but the general gist goes like
this:
As it happens, I deployed my solution to this on our live PBX today, which I
wrote with some help from another asterisk-users user. Here is what I
came up with:
Firstly, in features.conf I
Adam Moffett wrote:
So you want a device that will answer a SIP call, and play the audio out
to a speaker?
You're looking to build a PA system then?
We achieved this using a Grandstream Budgetone configured to
auto-answer, and just soldered a pair of wires across its speaker
terminals
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kev S
Sent: Tuesday, January 29, 2008 9:53 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] SIP GSM
With that sort of set up, If for example i get a 8
Jared Smith wrote:
I've always done this by setting the MONITOR_EXEC channel variable to
point to an external program that takes care of moving the recording to
the proper location so that it can be accessed by the user who made the
recording. I'll bet if you search for MONITOR_EXEC and
Hello All,
Our old Lucent Argent system had a feature whereby when you initiate
recording during a call, it would afterwards send the recording as a
voicemail message to the user who initiated the recording.
We use the automon *1 recording function in asterisk, which allows users
to record a
Khaled Chehab wrote:
Hi All
I am using asterisk 1.2.4
Please see the results when I execute Sip show channels
*569 *active SIP channels
What phones are you using? We had a similar problem with Snom 360 phones
with firmware version 6.2.2 and asterisk 1.2, whereby channels would
Mark Greene wrote:
I could do that. The only issue is that I don't understand why others
with my setup have not had to do the same. What's unique about my TDMoE
setup that makes it intolerant to channel restarts? I did everything by
the book.
We had a similar problem, and although your
Olivier wrote:
At the opposite, I think it could be useful for an Asterisk server to
act as XMPP User Activity provider (ie update XEP-0108 field with
on-the-phone value).
Do you agree ?
Is there any XMPP client supporting User Activity ?
Is Asterisk capable of getting or sending such User
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