Hello,
I would like to know whether there is a way to associate a TV media server
with Asterisk. Is it possible to access TV Chanels in the Telephone Sets.
Anybody have any tips or documents related to this please let me know.
Thanks
--
:54 PM, Kyle Kienapfel doctor.w...@gmail.comwrote:
On Mon, Aug 2, 2010 at 5:37 AM, Tino t...@sparksupport.com wrote:
Hello,
I would like to know whether there is a way to associate a TV media
server with Asterisk. Is it possible to access TV Chanels in the Telephone
Sets. Anybody have any
Hello,
In my Asterisk server when i try to set the value for the queue option Skip
Busy Agents in Freepbx GUI it is not being written into the backend file
queues_additional.conf. As a result sometimes agents in queue gets calls
when they are already busy with another call. So i set ringinuse=no
Hello ,
I would like to tweak my Answeing Machine Detection (AMD) in Asterisk. My
current values are
AMD(2500|1500|300|5000|120|50|5|256) and we were able to identify approx
25-30 % of all answering machines.
Anybody have any suggestion to improve the accuracy of AMD.
Thanks
--
this Background
trick as it helped me a lot regarding AMD on SIP.
exten = _X.,n,Background(blank_audio)
exten = _X.,n,AMD
On Wed, Aug 4, 2010 at 5:08 PM, Tino t...@sparksupport.com wrote:
Hello ,
I would like to tweak my Answeing Machine Detection (AMD) in Asterisk. My
current values are
AMD
Thanks Danny, What should be the length of audio file ?
On Wed, Aug 4, 2010 at 9:21 PM, Danny Nicholas da...@debsinc.com wrote:
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *Tino
*Subject:* Re: [asterisk-users] Tweaking AMD
In my Asterisk server following things have been done to detect answering
machines before the answered call connects to the agents in queue.
In extension_additional.conf
==
[ext-queues]
include = ext-queues-custom
exten = 5000,20,Macro(user-callerid,); changed
Hello,
Is there any way to capture the output of the 'System' application in
asterisk dialplan and evaluate it.
For example, i would like to get the output of following System application
and use its value in next line
for decision making
exten = 5000,n,System(command)
--
-boun...@lists.digium.com] *On Behalf Of *Tino
*Subject:* [asterisk-users] 'System' application in asterisk
Hello,
Is there any way to capture the output of the 'System' application in
asterisk dialplan and evaluate it.
For example, i would like to get the output of following System
application
-posting...
On Tue, 10 Aug 2010, Tino wrote:
Is there any way to capture the output of the 'System' application in
asterisk dialplan and evaluate it.
On Mon, Aug 9, 2010 at 11:51 PM, Danny Nicholas da...@debsinc.com
wrote:
I think this answer is no. system only returns
Hello Julian,
I am using Asterisk 1.4.33.1(AsteriskNOW iso) and curl function is not
available in this version.
On Tue, Aug 10, 2010 at 1:21 PM, Julian Lyndon-Smith aster...@dotr.comwrote:
You could always use the CURL function directly in the dialplan
Julian
On 10 August 2010 08:36, Tino
Hello,
Is it possible to install Asterisk on Vmware(centos) from source. Is there
any difference or disadvantage for this compared to asterisk running on
physical machine.
--
_
-- Bandwidth and Colocation Provided by
Thanks Gareth for your quick reply.
It is the lateset version and i think i need access to Dahdi interface. Is
there any disadvantages other than this.
On Wed, Aug 11, 2010 at 2:11 PM, Gareth Blades
list-aster...@skycomuk.comwrote:
Tino wrote:
Hello,
Is it possible to install Asterisk
Hello,
How to take the values of channel variables like 'agi_uniqueid' and
'agi_callerid' in agi script.
For example
#!/bin/bash -x
T=$agi_uniqueid
I want to save value of 'agi_uniqueid' channel variable into a variable
called 'T' in my script
--
Hello,
Can antbody recommend devices that can be used along with my Asterisk
server
Paging Amplifier
SIP enabled Paging Gateway
VOIP SIP loudspeaker
Also , please recommend video phone sets that suppot paging, intercom
(autoanswer)
Thanks
--
Hello,
I would like to send sms to some external phone numbers from my asterisk
server. Is it possible to send sms via softphones like X-Lite ? . Any tips
regarding this will be helpful
thanks
--
_
-- Bandwidth and Colocation
Hello Johann,
Thanks for your advice in this matter. But i am not sure how to pass the
numbers to be sent sms in the dialplan.
On Wed, Aug 18, 2010 at 3:13 AM, Johann Hoehn johann.ho...@ecommerce.comwrote:
On 08/17/2010 09:00 AM, Tino wrote:
Hello,
I would like to send sms to some
Hello,
Is there a way to capture the answering machine message when the dialer
detects the answering machine.
Thanks
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live
Hello,
Is it possible to set a variable in dialpan when the someone calls a
particular DID number so that i can use that variable for calls coming to
that number only.
--
_
-- Bandwidth and Colocation Provided by
But when i call my DID number following dialplans are being executed. What
i need is to set a variable with one value for one DID number and set the
same variable with another value for another DID number. Also any contexts
should be able to use this variable.
-
NoOp(SIP/5070-5407,
Yes, we need to record the message
On Wed, Aug 25, 2010 at 12:35 PM, Matt Riddell li...@venturevoip.comwrote:
On 20/08/10 1:52 AM, Tino wrote:
Hello,
Is there a way to capture the answering machine message when the dialer
detects the answering machine.
Record?
--
Cheers,
Matt
Hello,
I planning to use a web interface to send sms through Asterisk server.
I am planning to use php code which will interact with Asterisk Manager
Interface(AMI) and use Sms() application to send sms.
I am not sure whether it is the write way to do this. Anybody have any
suggestions or tips,
Hello,
Is it possible to avoid playing music on hold during a blind transfer ?
Thanks
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every
Hello,
When an agent does a blind transfer the call hangups for him but shows as
In use in queue in my CRM (used for auto dialing). As a result the agent
have to wait until the transfered call completes. Is there any way to change
this behaviour ?
--
Hello,
Is there any option to set the maximum number of seconds a caller can wait
in a queue before being pulled out ?
Thanks
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us
Thanks Warren for your help
On Mon, Aug 30, 2010 at 9:21 PM, Warren Selby wcse...@selbytech.com wrote:
On Mon, Aug 30, 2010 at 10:31 AM, Tino t...@sparksupport.com wrote:
Hello,
Is there any option to set the maximum number of seconds a caller can wait
in a queue before being pulled out
26 matches
Mail list logo