You could start with a standard installation. Once things are running
you can specify unneeded modules in modules.conf.
http://www.asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/asterisk-CHP-5-SECT-1.html
explains how things are related.
jg
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Hi David ,
Thank you for the reply .
I need to use only the IVR functions for mobile users and they are landing to
the Asterisk server via SIP.
Hence , please help me to understand , I really do not need the other modules
other than Asterisk core ?
I am so sorry , As I am new to Asterisk , I
You do not require DAHDI Linux or Tools if you do not have any TDM devices
unless you want to use MeetMe instead of ConfBridge.
Regards
David.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of luke devon
Sent: Saturday, 8 June 201
Hi
If I do not use any DAHDI - hardware , can I ignore the DAHDI linux and tools
installation ?
Can I use just asterisk ? is there any dependencies while executing Asterisk
with DAHDI modules ?
Thank you
Luke
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-- Bandwidt
Hello,
I'm trying to create a H.323 trunk between Asterisk 11 and Avaya. I have
done this before between Asterisk 1.6 and Avaya but had some issues placing
external calls from the Asterisk to the Public network which is connected
to Avaya. I'm trying to create that trunk on Asterisk 11 because
The Asterisk Development Team has announced the releases of:
DAHDI-Linux-v2.7.0
DAHDI-Tools-v2.7.0
dahdi-linux-complete-2.7.0+2.7.0
This release is available for immediate download at:
http://downloads.asterisk.org/pub/telephony/dahdi-linux
http://downloads.asterisk.org/pub/telephony/dahdi-tools
h
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Hash: SHA1
The best guide is the Asterisk Fefinitive guide and a virtual machine
so you can install several Asterisk servers and make them talk to each
other.
On 6/7/13 1:20 PM, Michael Gilleran wrote:
> Greetings. Anyone have any recommendations for studying fo
Hi,
If you read, O'Reilly - Asterisk - The Definitive Guide - 3rd Edition, you
should be ready for take the test.
Of course, you must read voip-info too.
Carlos Rojas
Dcap 2171
On Fri, Jun 7, 2013 at 2:20 PM, Michael Gilleran wrote:
> Greetings. Anyone have any recommendations for studying f
Greetings. Anyone have any recommendations for studying for the dCAP
Certification? Other than the expensive Digium courses, there doesn't seem to
be anything online.
Thanks,
Michael Gilleran
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-- Bandwidth and Colocation
hi.
check here for agi
http://forum.voxilla.com/threads/introducing-waits-w-in-dial-destination-number-variable.14628/
On Fri, Jun 7, 2013 at 7:50 PM, Sean Darcy wrote:
> On 06/07/2013 01:17 PM, Yves A. wrote:
>
>> This would be possible with an agi...
>> the agi can wait for silence or 10 seco
On 06/07/2013 01:17 PM, Yves A. wrote:
This would be possible with an agi...
the agi can wait for silence or 10 seconds, as u like and then play the
dtmf tones and bridge the call to your extension afterwards.
yves
Am 07.06.2013 17:51, schrieb Sean Darcy:
I'm trying to call a conference servi
hi,
you can add more w (ww1234#) for more delay.
On Fri, Jun 7, 2013 at 7:17 PM, Yves A. wrote:
> This would be possible with an agi...
> the agi can wait for silence or 10 seconds, as u like and then play the
> dtmf tones and bridge the call to your extension afterwards.
>
> yves
>
>
This would be possible with an agi...
the agi can wait for silence or 10 seconds, as u like and then play the
dtmf tones and bridge the call to your extension afterwards.
yves
Am 07.06.2013 17:51, schrieb Sean Darcy:
I'm trying to call a conference service, wait 10 seconds, then send
the pa
I'm trying to call a conference service, wait 10 seconds, then send the
passcode.
I've tried ww:
Dial(SIP/18005551212ww12345#@sip.com,60,r)
The sip channel didn't like that. Added 'p' , still no help.
I tried D:
Dial(SIP/18005551...@sip.com,60,rD(12345#)
The dtmf is sent too soon. I tried
ICTDialer http://www.ictdialer.org is free and open source dialer suitable
for mentioned requirments
Regards
*Tahir Almas*
Managing Partner
ICT Innovations
http://www.ictinnovations.com
Leveraging open source in ICT
On Fri, May 24, 2013 at 1:26 AM, Ron Wheeler wrote:
> One of the tricks
The sample config files in the Asterisk distribution and packages are
really good for getting the demo up and running quickly, for example, to
extend the demo to run behind a WebRTC proxy only required about 6 lines
of extra code to define a peer in sip.conf and enable TCP
However, I'm not sure t
Thank you so much , This would be a great help . Thank you once again .
From: Justin Killen
To: luke devon ; Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Thursday, 6 June 2013, 23:50
Subject: RE: [asterisk-users] Minimum requirement for Aster
6 jun 2013 kl. 17:41 skrev Daniel Pocock :
> On 06/06/13 15:51, Daniel Pocock wrote:
>> Is the template capability in sip.conf compatible with realtime sip.conf
>> entries such as users in a database?
>>
>> I notice that contrib/realtime/mysql/sippeers.sql and the wiki page
>> don't mention a te
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