have you installed the odbc devel packages? if not, try that.
On 31 July 2013 08:19, Prashant A. wrote:
> Hi,
>
> I am using ubuntu-12.04 and installed asterisk from repository (apt-get
> install asterisk).
>
> I have configured it to work with odbc,
>
> *CLI> odbc show
>
> ODBC DSN Settings
>
On Tue, Jul 30, 2013 at 3:55 PM, Shaun Ruffell wrote:
> On Tue, Jul 30, 2013 at 11:46:04AM -0400, Andre Goree wrote:
>>
>> Thanks I'll definitely be contacting Digium support shortly. For the
>> hell of it, here's my dmesg output -- I'm seemingly safe from an IRQ
>> issue, thankfully, ha:
>>
>> [
Is there an easy way to have app_meetme create the recording in a temp
location and move it once the conference is over?
or should I just have a perl script run every minute to check for no users
in the conference room and then move it?
Asterisk 11
Thanks in advance!
--
_
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On 8/1/13 9:17 PM, Michael L. Young wrote:
> - Original Message -
>> From: "Carlos Chavez" To:
>> asterisk-users@lists.digium.com Sent: Thursday, August 1, 2013
>> 8:41:19 PM Subject: [asterisk-users] External sip phones register
>> with the s
On Fri, Aug 02, 2013 at 10:12:54AM -0400, Andre Goree wrote:
>
> Ran through a loopback test with Digium which seemingly proved that
> there was no issue with their card. I've contacted my telco for
> assistance but so far have been unable to come up with
> anything...though there may be a timing
The last two companies I have worked for have both had this problem with
a BT ISDN30 line at some point. I also manage SS7 interconnects and its
not unusual for there to be issues with them either. So dont assume its
probably at your end :P
On 02/08/13 15:12, Andre Goree wrote:
I've contacted
On 02/08/13 17:10, John Doe wrote:
Is there an easy way to have app_meetme create the recording in a temp
location and move it once the conference is over?
or should I just have a perl script run every minute to check for no
users in the conference room and then move it?
Asterisk 11
Thanks
Please post one of your sip.conf phone configs, so we can have a look.
Alyed
2013/8/2 Carlos Chavez
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> Hash: SHA1
>
> On 8/1/13 9:17 PM, Michael L. Young wrote:
> > - Original Message -
> >> From: "Carlos Chavez" To:
> >> asterisk-users@lists.digium
On Fri, Aug 2, 2013 at 12:29 PM, Shaun Ruffell wrote:
> On Fri, Aug 02, 2013 at 10:12:54AM -0400, Andre Goree wrote:
>>
>> Ran through a loopback test with Digium which seemingly proved that
>> there was no issue with their card. I've contacted my telco for
>> assistance but so far have been unab
Asterisk 11.1.0
I'm trying to use the "b" subroutine of the Dial application so that I
can do some stuff with our internal applications that need to have
access to the called channel information. I can see that the subroutine
is being executed, but the arguments I pass don't see to make it to
On Fri, Aug 2, 2013 at 12:57 PM, Mitch Claborn wrote:
> Asterisk 11.1.0
>
> I'm trying to use the "b" subroutine of the Dial application so that I can
> do some stuff with our internal applications that need to have access to
> the called channel information. I can see that the subroutine is be
On 08/02/2013 01:28 PM, Matthew Jordan wrote:
On Fri, Aug 2, 2013 at 12:57 PM, Mitch Claborn mailto:mitch...@claborn.net>> wrote:
Asterisk 11.1.0
I'm trying to use the "b" subroutine of the Dial application so that
I can do some stuff with our internal applications that need to hav
Can you create dynamic bridges like in meemet?
On Aug 2, 2013 12:34 PM, "Gareth Blades"
wrote:
> On 02/08/13 17:10, John Doe wrote:
>
>> Is there an easy way to have app_meetme create the recording in a temp
>> location and move it once the conference is over?
>>
>> or should I just have a perl s
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