Hi Bruce,
On 01/03/2011 06:03 AM, Bruce B wrote:
Thanks for the input. The errors I pointed out should be available in
the /var/log/messages or messages.1 .2 .3 etc...or in dmesg each time
you reboot the machine.
I have just rebooted the server and can confirm that neither dmesg,
/var/log/mes
Hi,
On 01/01/2011 05:43 PM, bilal ghayyad wrote:
Dear List;
For each call (in specific case), I need to do a record and save in a spearated
file, so I am thinking the best thing is to save based on the time.
Monitor(wav,Record1,m)
So, how can I make the file name to be based on the current t
Le 01/01/2011 18:32, Gilles a écrit :
On Wed, 29 Dec 2010 16:55:46 +0100, Administrator TOOTAI
wrote:
I wouldn't be one of your friend: when I'm calling you I call a landline
but finally will be charged for a mobile call (imagine I have free calls
to landlines from my ISP). I give you an i
Hi. I am using Digium TDM2400PFXO with Asterisk1.6. When call sets in the
box , it answers the call even the phone is not picked. ideally it should
answer the call when the phone is picked up. Its charging the clients.
Please let me know how can I cover this ? Thanks in advance.
--
On 01/03/2011 06:16 AM, Muhammad Usman wrote:
> Hi. I am using Digium TDM2400PFXO with Asterisk1.6. When call sets in
> the box , it answers the call even the phone is not picked. ideally it
> should answer the call when the phone is picked up. Its charging the
> clients. Please let me know how can
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Randy R
Sent: Sunday, January 02, 2011 3:05 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Callback form to place on s
I am trying to configure mysql to use a different datadir than default in
order to move this to a larger volume. I have copied all mysql data from
/var/lib/mysql to my new volume and ran both chown -R mysql:mysql * and
chmod -R 660 * in order to setup correct ownership and rights for the
data. I
Sorry, sent this to wrong group list.
On Mon, Jan 3, 2011 at 10:13 AM, Nicholas Hart wrote:
> I am trying to configure mysql to use a different datadir than default in
> order to move this to a larger volume. I have copied all mysql data from
> /var/lib/mysql to my new volume and ran both chown
Greetings,
I mailed the list regarding an intercom system some months ago and we
came to the conclusion that I should purchase a Polycom 501 phone.
I'm now considering the purchase for this year, and I'm now wondering
between the Polycom 501 and the 320 for the intercom.
I don't need the spa
On Sun, Jan 02, 2011 at 12:04:07AM +, JP CR wrote:
>What I want is when a potential client submits his number... the PBX dials the
>number makes an announcement and dials an extension (which is actually a
>cellhopne dahdi member) and makes the connection.
You might try something based on th
Any idea ??
Em 28/12/2010 11:32, Eduardo Lobo Blanco escreveu:
Hi list,
I´m using Asterisk 1.6.2.14.
I have some queues configured in my solution, using dynamics members.
All my members are IAX2 clients.
Each member will be at least in two queues at same time.
So i´m trying to use the announc
Thanks a lot for that Sebastian. I will report back my findings when I find
the resolution on this.
Regards,
On Mon, Jan 3, 2011 at 3:28 AM, Sebastian wrote:
> Hi Bruce,
>
>
> On 01/03/2011 06:03 AM, Bruce B wrote:
>
>> Thanks for the input. The errors I pointed out should be available in
>> th
On Mon, 03 Jan 2011 12:27:56 +0100, Administrator TOOTAI
wrote:
>As you are a Free Telecom customer, why not using your freephonie
>account to forward incoming calls to your mobile?
Thanks for the tip, but experience shows that their SIP access sucks
(not reliable, quality NOK). That's why I got
Hi folks,
I was hoping that someone might be able to help clarify some confusion I
have on DAHDI Fax detection after spending some time searching. My
understanding is this:
1.) Echo cancellation is automatically disabled upon recognition of a
CNG tone, regardless of the faxdetect setting. Th
On 01/03/2011 11:26 AM, Tom Rymes wrote:
Hi folks,
I was hoping that someone might be able to help clarify some confusion I
have on DAHDI Fax detection after spending some time searching. My
understanding is this:
I'll try.
1.) Echo cancellation is automatically disabled upon recognition of
On 01/03/2011 11:20 AM, Bruce B wrote:
Thanks a lot for that Sebastian. I will report back my findings when I
find the resolution on this.
I'm a bit late here, but I can say that Sangoma support has always been
extremely helpful when I call them. If you haven't already, definitely
give them a
On Jan 3, 2011, at 3:22 PM, Kevin P. Fleming wrote:
> On 01/03/2011 11:26 AM, Tom Rymes wrote:
[snip]
>> 1.) Echo cancellation is automatically disabled upon recognition of a
>> CNG tone, regardless of the faxdetect setting. This can only be disabled
>> at compile time.
>
> No. CNG tone is neve
Andy,
The 501 and 320 are EOL. I'd go for the IP335 and a 2626-PWR, since the
2626 can support vlans you can isolate data and voice. Make sure to spec a
UPS on the PoE switch.
On Mon, Jan 3, 2011 at 8:30 AM, Andy Graybeal
wrote:
> Greetings,
> I mailed the list regarding an intercom system som
On 01/04/2011 04:22 AM, Kevin P. Fleming wrote:
On 01/03/2011 11:26 AM, Tom Rymes wrote:
Hi folks,
I was hoping that someone might be able to help clarify some confusion I
have on DAHDI Fax detection after spending some time searching. My
understanding is this:
I'll try.
1.) Echo cancellat
I don't imagine this would be too complicated - don't have any experience
with AsteriskNOW - but on a 'vanilla' linux distro it would just be a matter
of making sure dahdi is loading the correct drivers and doing a couple of
minor config file updates.
On Tue, Dec 28, 2010 at 3:01 PM, Tyler Davis
Cannot find asterisk-mib.txt and digium-mib.txt anywhere. Were they dropped?
-kkm
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My company is building a VOIP application, and initially were just using a
barebones OpenSIPS implementation to host one-on-one calls; however, we want
to expand the functionality to conferencing (which, of course, OpenSIPS
doesn't handle) and was looking into Asterisk (the other option being
Frees
My company is building a VOIP application, and initially were just using a
barebones OpenSIPS implementation to host one-on-one calls; however, we want
to expand the functionality to conferencing (which, of course, OpenSIPS
doesn't handle) and was looking into Asterisk (the other option being
Frees
On Mon, Jan 03, 2011 at 08:04:48PM -0800, Kirill Katsnelson wrote:
> Cannot find asterisk-mib.txt and digium-mib.txt anywhere. Were they dropped?
They're now part of the wiki.
https://wiki.asterisk.org/wiki/display/AST/Asterisk+MIB+Definitions
https://wiki.asterisk.org/wiki/display/AST/Di
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