I am using the Debian package of asterisk which is version
1:1.4.13~dfsg-1. I have created an AGI script which plays files in
response to external events and am using a callfile to connect it to a
meetme conference. The first two or three files generally play
correctly, but after that point almost
Hi all,
I have seen a lot of message talking about asterisk crashed when
using queue and mixmonitor together. I do use both in our system and
also get the crash (segfault) randomly. I don't know it is related to
the reason above as I have no idea about how it happened. I get the
core dump belo
Another quick question (Spending the day trying to get this project sorted
and tucked away) If I am dynamically adding queue members, they will not
abide to settings within agents.conf will they?
Ie. I need the equivalent of Autologoff however want my agents to receive
calls when someone joins the
Hello
Thanks for the reply..
I could use Asterisk as SIP server and establish call using two SIP phones.
But I need H323 support also.
For that I have compiled the files in asterisk/channel/h323 and installed
without problem.
But even after i have started Asterisk,it is not supporting h323 co
On 8/29/07, Steve Underwood <[EMAIL PROTECTED]> wrote:
> Carlos Chavez wrote:
> > On Wed, 2007-08-29 at 00:03 +0300, Tzafrir Cohen wrote:
> >
> >> On Tue, Aug 28, 2007 at 10:11:03PM +0200, Christian Peter wrote:
> >>
> >>> Hi list,
> >>>
> >>> I'm running current SpanDSP
> >>> http://www.soft-switc
The freepbx system has a primary number option in its ring group dialing
which if selected as a ring strategy means it won't ring any further if
the primary number is engaged. This is useful in follow me setups.
I haven't dug into how its implemented but it works for ring groups and
follow me o
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Paul Hales
> Sent: Wednesday, October 24, 2007 9:50 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Compatibility Issues with dell powere
Have you tried the ringinuse option? This will not ring phones if they are
busy.
Sent from my Verizon Wireless BlackBerry
-Original Message-
From: Nick Brown <[EMAIL PROTECTED]>
Date: Mon, 05 Nov 2007 10:26:19
To:"asterisk-users@lists.digium.com"
Subject: [asterisk-users] 7960 Queue I
Your question seems to be two I think so I've covered both options here.
Mediatrix does two different series of boxes - 4 port version ...
4 port to extensions - a 1104 (also in 2 port and higher numbers too)...each is
an analog line to a phone, ie extensions for house, small office, etc
4 port t
Thanks Eric, this is the case. A bit of a shame that it removes the
functionality for the member to see calls that have not come from a queue
however there is not much choice in the matter.
FWIW to get this option a firmware upgrade was required (Now running
POS3-08-8-00).
Cheers.
On 5/11/07 11
On 11/4/07, Nick Brown <[EMAIL PROTECTED]> wrote:
> Morning All,
>
> Quick question that has me stumped. Have a queue with several members
> (Statically defined in queues.conf at this stage for testing) who use Cisco
> 7960's.
>
> The queue is configured to use rrmemory and generally this works cor
Are there ATAs that allow different phone numbers from one network connection?
Such as supporting multiple IP addresses so that each RJ11 has a
different extension or some other way?
___
--Bandwidth and Colocation Provided by http://www.api-digital.com-
My memory tells me that there is a flag (something like 'ringinuse')
which can make sure this sort of thing does not happen.
PaulH
On Mon, 2007-11-05 at 10:26 +1100, Nick Brown wrote:
> Morning All,
>
> Quick question that has me stumped. Have a queue with several members
> (Statically defined
Hi,
I'am comparative newbie to the world of Asterisk. I'd like to
introduce an Asterisk based PBX into my company but need to convince my
executive of it's worthiness. I need some reference sites to quote in my
discussion, preferably well known companies of course. I have surfed the
net but
Morning All,
Quick question that has me stumped. Have a queue with several members
(Statically defined in queues.conf at this stage for testing) who use Cisco
7960's.
The queue is configured to use rrmemory and generally this works correctly.
However if a member is already on a call their phone w
Quoting Doug Lytle <[EMAIL PROTECTED]>:
> Anybody else encountered this?
I have. But i did it manually, not from a cron job...
It didn't restart for me either... I had to resort
to a full restart with the init script...
--
PLO toluene SEAL Team 6 supercomputer president DES Waco, Texas
Cocaine N
Hi
On 11/5/07, Tzafrir Cohen <[EMAIL PROTECTED]> wrote:
> astdatadir ?
What is the default location in asterisk?
Why have this hen you have astvarlibdir ?
Jean-Yves
___
--Bandwidth and Colocation Provided by http://www.api-digital.com--
asterisk-use
Eric "ManxPower" Wieling wrote:
> Doug Lytle wrote:
>> Eric "ManxPower" Wieling wrote:
>>> Doug Lytle wrote:
>>>
>>> document was called UPGRADE.txt. You might want to look at that, in
>>> addition to the 1.4 UPGRADE.txt, as there are many features that were
>>>
>>
>> The document was revi
On Mon, Nov 05, 2007 at 04:52:14AM +1100, Jean-Yves Avenard wrote:
> Hi
>
>
> On 11/5/07, Eric ManxPower Wieling <[EMAIL PROTECTED]> wrote:
> > Look at the section starting on line 100 in
> > /path/to/src/asterisk-1.4.13/UPGRADE.txt
> >
> > You should have read this file before upgrading to 1.4.
Doug Lytle wrote:
> Eric "ManxPower" Wieling wrote:
>> Doug Lytle wrote:
>>
>> document was called UPGRADE.txt. You might want to look at that, in
>> addition to the 1.4 UPGRADE.txt, as there are many features that were
>>
>
>
> The document was reviewed and the appropriate changes to th
Jean-Yves Avenard wrote:
> Hi
>
>
> On 11/5/07, Eric ManxPower Wieling <[EMAIL PROTECTED]> wrote:
>> Look at the section starting on line 100 in
>> /path/to/src/asterisk-1.4.13/UPGRADE.txt
>>
>> You should have read this file before upgrading to 1.4.
>>
>
> Excellent. Thank you!
>
> I've added
Hi
On 11/5/07, Eric ManxPower Wieling <[EMAIL PROTECTED]> wrote:
> Look at the section starting on line 100 in
> /path/to/src/asterisk-1.4.13/UPGRADE.txt
>
> You should have read this file before upgrading to 1.4.
>
Excellent. Thank you!
I've added a WaitExten() just after and now everything wo
Eric "ManxPower" Wieling wrote:
> Doug Lytle wrote:
>
> document was called UPGRADE.txt. You might want to look at that, in
> addition to the 1.4 UPGRADE.txt, as there are many features that were
>
The document was reviewed and the appropriate changes to the dial plan
were made.
I foun
Look at the section starting on line 100 in
/path/to/src/asterisk-1.4.13/UPGRADE.txt
You should have read this file before upgrading to 1.4.
Jean-Yves Avenard wrote:
> Dear all
>
> I am trying to upgrade our asterisk from 1.2 to 1.4.x
>
> There is something that now fails to work, reading the
Dear all
I am trying to upgrade our asterisk from 1.2 to 1.4.x
There is something that now fails to work, reading the various
documentations, I can not explain why.
Here is an extract of my extensions.conf
[welcome]
exten => 299,1,Answer ; Answer the line
exten => 299,2,Set(TI
Doug Lytle wrote:
> I've moved 1 of our facilities over from 1.2 to 1.4 two weeks back. So
> far, the only issue that I've encounted is.
>
> I have a scheduled CRON job that runs at 3am every Sunday, that issues a:
>
> asterisk -rx 'restart when convenient'
>
> The first Sunday that it ran, As
Latest version of X-lite does not have GSM codec . Downgrde your versiona dn
you will get gsm codec . I read on their forums that next version will
again be including GSM codec .
On 03/11/2007, Julio Tejera <[EMAIL PROTECTED]> wrote:
>
> Latest version of X-Lite does not
> support GSM codecs any
I've moved 1 of our facilities over from 1.2 to 1.4 two weeks back. So
far, the only issue that I've encounted is.
I have a scheduled CRON job that runs at 3am every Sunday, that issues a:
asterisk -rx 'restart when convenient'
The first Sunday that it ran, Asterisk never restarted. The CRON
Hi,
I would like to achive such thing:
Asterisk is gateway between two sip domain domain.a and domain.b.
There are two users registered in domain A: bob1 and bob2.
When they are making outbound calls the calls should go through the asterisk
gateway.
Asterisk has two users registered in domain B s
29 matches
Mail list logo
