On Sun, Dec 19, 2010 at 12:14:11AM -0500, Stephen Reese wrote:
Thanks for the heads up, I have been setting the caller-ID but the
trouble I'm running into is specifying the which number to call out
as. How can an extension specify a different number? See below for my
current extension.conf,
Hi Stephen,
Thanks for the heads up, I have been setting the caller-ID but the
trouble I'm running into is specifying the which number to call out
as. How can an extension specify a different number? See below for my
current extension.conf, thanks.
You can check the channel-name to see which
Hey Guys,
In which Version of Asterisk is EventFilter: in manager.conf working?
Higher than 1.6.2.10 or from the 1.8.0 Version?
Thank for your answer
Daniel
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When I call from a mobile to mobile (both registered on OPENBTS) the correct
caller ID is passed. That is the callerid that I set in the callerid=
field.
When calling from openbts to the PSTN the config header is passed.
Thanks,
Dave
-Original Message-
From:
Hi,
I have 2 phones AASTRA 57i with Asterisk 1.6.
When the internet
connection for some reason fall down the 2 phones go to NO SERVICE,
searching on internet i found that this is due to DNS service.
Has
someone solve this problem? or suggestions?
Thanks in advance
Antonio
Supera i limiti:
antse...@tiscali.it wrote:
When the internet
connection for some reason fall down the 2 phones go to NO SERVICE,
searching on internet i found that this is due to DNS service.
Has
someone solve this problem? or suggestions?
Yes,
Install Bind either on the Asterisk server or someplace on
On Wed, Dec 15, 2010 at 12:37:05PM +0530, DHAVAL INDRODIYA wrote:
Guys,
I have rebooted system, and also same issue i have found that DAHDI module
is not found
i am stuck in what to do for loading DAHDI onto EC2
*/etc/init.d/dahdi restart
Unloading DAHDI hardware modules: done
Loading
You can check the channel-name to see which extension is making the
call and set the CallerID accordingly. The channel-name will be
something like SIP/201-abc23ef34 or SIP/User1-def34abc51. The 201
or User1 part depends on how you put the username in sip.conf You can
use the CUT function to
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Stephen Reese
Sent: Sunday, December 19, 2010 12:49 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Specifying DID
On Sun, Dec 19, 2010 at 1:52 PM, William Stillwell
will...@stillwellsoft.com wrote:
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Stephen Reese
Sent: Sunday, December 19, 2010 12:49 PM
To: Asterisk
- Original Message -
The host I am working with has two accounts from the same DID
provider. Incoming calls work correctly and dial the appropriate
extensions. This also allows incoming calls to be billed appropriately
to the
individual DID accounts.
Outgoing calls from either
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joshua Colp
Sent: Sunday, December 19, 2010 2:40 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Specifying DID for
- Original Message -
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joshua
Colp
Sent: Sunday, December 19, 2010 2:40 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
- Original Message -
On Sun, Dec 19, 2010 at 2:40 PM, Joshua Colp jc...@digium.com wrote:
I'm surprised nobody has suggested using the setvar functionality.
It's extremely
useful for stuff like this and would allow you to keep all CallerID
information
with the actual
First, when using multiple accounts from the same DID provider, is it
ideal to use IP based routing using one context as I currently am or
have a separate contexts for each account in the sip.conf?
That's really the only way to do it presently.
So I should have multiple incoming and outgoing
- Original Message -
First, when using multiple accounts from the same DID provider, is
it
ideal to use IP based routing using one context as I currently am
or
have a separate contexts for each account in the sip.conf?
That's really the only way to do it presently.
So I
So I should have multiple incoming and outgoing contexts? Vitelity
will allow me to use IP routing or user/pass auth, the latter would
allow me to specify the outgoing context, this would also guarantee
the correct account is billed and not alone rely on caller-ID.
Let me clarify further.
On Sun, Dec 19, 2010 at 4:36 AM, Jeroen Eeuwes jeroeneeu...@gmail.com wrote:
Hi Stephen,
Thanks for the heads up, I have been setting the caller-ID but the
trouble I'm running into is specifying the which number to call out
as. How can an extension specify a different number? See below for my
On Sun, Dec 19, 2010 at 2:57 PM, Stephen Reese rsre...@gmail.com wrote:
I believe I have made a little headway. I have two outgoing DID
contexts and have changed the GotoIf statement to the extension name.
User One acts as expected and User two now displays unknown when
calling so I believe it
I believe I have made a little headway. I have two outgoing DID
contexts and have changed the GotoIf statement to the extension name.
User One acts as expected and User two now displays unknown when
calling so I believe it is trying to to goto 20 but it's not quite
making it. Any tips? Thanks
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reply please..
On 12/17/2010 03:51 PM, Nikhil wrote:
Hi
Does anyone ported Asterisk to Android OS .please give details
Thanks
Nikhil
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On Sun, Dec 19, 2010 at 10:50 PM, Outback Dingo outbackdi...@gmail.comwrote:
check the serval phone project
On Fri, Dec 17, 2010 at 5:21 AM, Nikhil d.nik...@cem-solutions.netwrote:
Hi
Does anyone ported Asterisk to
check the serval phone project
On Fri, Dec 17, 2010 at 5:21 AM, Nikhil d.nik...@cem-solutions.net wrote:
Hi
Does anyone ported Asterisk to Android OS .please give details
Thanks
Nikhil
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Let me explain this with some more details.
I have 2 members logged into Queue 'retailBanking' using AddQueueMember
application on 2 different softphones.
The softphones from which these 2 members were added, later unregistered
from Asterisk.
I then fired below mentioned AMI actions and observed
Hello all,
I have a perl script that updates a M$ SQL DB based on an ivr that is
run on asterisk.
When it runs as a normal agi, it works great.
when run as a DeadAGI it does not work.
When i execute the script from h channel withDeadAGI and agi debug on i get:
[2010-12-20 01:08:54] --
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