Displaying 20 results from an estimated 102 matches for "astersik".
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asterisk
2004 Jul 19
2
codec translate
HI ALL;
Is astersik enable to translate between different codecs.
I have couple of SIP-UA , one with (a-law) and the other with (g729), registered with my astersik box.Can astersik translate between alaw-g729 and vice varsa.
Regards
mohammad
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2004 Jul 22
0
Re: h323ep----gnugk-----astersik------h323ext
...handle it with asterisk native h323 channel???????/
Regards;
mohammad
----- Original Message -----
From: "administrator tootai" <admin@tootai.net>
To: "mohammad mirzaee" <agent@rasatech.com>
Sent: Wednesday, July 21, 2004 5:59 PM
Subject: Re: h323ep----gnugk-----astersik------h323ext
mohammad mirzaee a ?crit :
> HI DANIEL;
>
>
> I found you expert on asterisk call routing, will you plz help me for
> the following problem::
>
>
> I have an ATA phone registered with GUNGK.I want to send a call to
> another ATA with has an extention in m...
2005 May 23
1
Astersik vs. Pingtel
Slash-dot is pointing to this article on Asterisk and Pingtel.
http://www.theregister.co.uk/2005/05/22/pingtel_voip/
Paul
Paul Mahler
www.signate.com
2006 Jan 16
0
How to put someone on hold with Astersik Manager
Hello,
I am writing a program based on Astersik Manager which needs to put
calls on hold and to redirect them to others extensions.
I haven't funded any action able to do this.
Is there a way to place calls on hold using Asterisk Manager Actions?
Amaury
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2004 May 27
1
Astersik and PostgreSQL
Hi to all!!
I'm successful to connect Asterisk to PostgreSQL database...
If it's possible, can anyone learn me how to store sip user in
PostgreSQL database and how to configure voicemail??
Thanks for all!!!
2004 Sep 20
0
Error compiling astersik-oh323
Dear Sirs,
I had compiled PWlib and OpenH323 correctly in my Fedora Core 2.
But when I try to compile asterisk-oh323 I get the following error:
`__use_AST_MUTEX_DEFINE_STATIC_rather_than_AST_MUTEX_INITIALIZER__'
How can I solve it?
Thank you for your help.
Juanjo
2004 Dec 14
1
Astersik with ISDN up0
Hi,
I am new to the Asterisk world. I don't know much about the
architecture, but I am involved in installing and configuring the VoIP
system.
My requirement is to build a VoIP system using the 4 input lines (ISDN
up0 telephone lines), it must be possible to receive calls from outside
through the 4 ISDN up0 input lines, and also possible for outgoing
calls, conferencing .etc.
I
2005 Aug 09
1
voip solution with SER, ASTERSIK and CCM
We are planning to install a voip system based on asterisk for 2000-3000 retail locations and up to 6000-8000 sip accounts/users.
Instead of setting up a new, centralized PSTN gateway, we are intend to use a CISCO gateway/router of an existing CISCO voip solution in the headquarter and we must able to call all CISCO based voip phones in the headquarter running together with a CCM.
SIP-Phones
2006 Mar 07
1
Help! Connecting two Astersik via SIP channels
Hi everyone,
I want to call from one Asterisk to another Asterisk via SIP, but i dn't
know how. I have found out something in these links:
http://www.voip-info.org/wiki/index.php?page=Asterisk+config+sip.conf
http://www.voip-info.org/tiki-index.php?page=Asterisk%20SIP%20Channels
but I don't understand them very well.
At first, I tried simply doing this:
In SIP Client:
2008 Feb 01
1
Astersik Transcoder support
Hello All:
Does the Asterisk support to insert an off the board transcoder for a call?
Thanks,
Charles
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An
2010 Jun 24
1
Astersik can not detect DTMF key
Hi all,
I'm building a karaoke service. Asterisk will play a music file, people can detect the point when they want to sing and record by press * key during the music is playing, and press # key to stop recording.
I use 2 functions: ast_streamfile and ast_seekstream to play audio file, and function ast_waitstream_fr to detect whenever people press DTMF key.
The problems is that, Asterisk
2010 Aug 07
2
AMD setup in Astersik
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 the priority to 20
...............
==============================
In extension_custom.conf
2005 Feb 22
1
Astersik CVS HEAD + T1 e&m wink + IAX client doesnt detect call answered on Zap channel
Hello,
I've got very annoying behaviour from our asterisk PBX.
We have 12 channels T1 e&m wink start for TDM and using iax softphones
internally (iaxcomm, but tried firefly-thirdparty and discarded for
bad sound quality).
Slackware 9.1 w/ kernel 2.4.26+ digium TE110P card.
In some cases when call is placed from softphone to TDM, system does
not detect call answered on Zap channel and
2010 Apr 28
6
Dial plan question.
...to use Asterisk as my server.
How should I have the dial plans as there are no numbers involved .
so How can I make the configuration to work ( with numbers I can get this done using extensions.conf)
my expected result is :
alice at pbx.com should be able to call bob at pbx.com
where pbx.com is astersik.
Can you pl let me know how I can achieve this?
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2007 Jul 03
6
Need Advice/Suggestion
Hi all,
As we know we can configure in astersik like before 5:00pm calls go to reception and after 5:00
pm calls go to some mobile no. One of my client requested that he wants to manually shift the dial
plan like above as he has flexiable timing sometime he finishes at 3:00pm some time 8pm. I can
not give him freepbx access.
Any idea or sol...
2010 Feb 24
2
Problems in Asterisk Real Time (Urgent help )
Hello,
Asterisk Real time database worked on astersik 1.6.2.0 but now i am working
on Asterisk to latest version which is 1.6.2.2 ,there is a a warning
[Feb 24 16:26:14] WARNING[4053]: config.c:2025 find_engine: Realtime mapping
for 'sippeers' found to engine 'mysql', but the engine is not available
[Feb 24 16:26:14] NOTICE[4053]: chan...
2010 Mar 16
3
Asterisk 1.4.24 DUNDi CLI commands not found
Are there DUNDi CLI commands for Asterisk 1.4? I have searched google
and I only see the dundi commands in Asterisk 1.6, although I see
reference to them in older version's of Asterisk such as Asterisk 1.4
here: http://www.asteriskguru.com/tutorials/cli_cmd_14.html. When I
view the CLI commands through help I don't see any of the dundi
commands and there are errors when I run a command
2005 May 31
2
handytone 486
Hi ;
Have two handytone 486 and want to use them as digium TDM400 fxo-fxs card...
I mean is it possible to direct pstn calls from astersik (extensions) to handytone line port directly and
vice versa ?...
Thanks in advance
Betul
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2011 May 30
3
please help
Hello list
i have configured astersik 1.4 with sip i have a question
when i put in dial plan.conf
exten => _0678922645.,1,Set(CALLERID(number)=520460587)
exten => _0678922645
.,n,MixMonitor(zap_g1_${EXTEN}_${UNIQUEID}.wav|av(0}V(0))
exten => _0678922645
.,n,Dial(Zap/g1/${EXTEN},30,A(this-call-may-be-monitored-or-recorded))...
2007 Sep 06
2
asterisk voicemail to email and relaying
Hi list,
I'm trying to get some ideas on this subject.
Normally astersik sends emails with voicemail attached trough local MTA.
As far as i know there is no way for asterisk to authenticate to an external
mailserver to relay these emails.
Well, these days every provider has some sort of spam blocking, to add to
that usually users of asterisk are behid a dynamic IP with...