Hi,
I want to reconfigure my asterisk dialplan.I have a problem.I have 4 agents
in a queue.How is the configuration for the asterisk dialplan if I want to
have only 4 agents maximum who can receive the phone,so if the fifth caller
try to entering the queue they will be noted by my IVR that all our
Thx, that did the trick!
On Sat, 12 Dec 2009 17:34:19 +0100, Philipp Kempgen wrote:
Magnus Benngård schrieb:
I am trying to figure out how DEVICE_STATE is working, no luck so far.
sip.conf
[0317998975]
Set
call-limit=10
(or any other value 0)
extensions.conf
exten =
Hi!
Have a weired problem with Avaya 9650 phones:
extensions.conf
exten = 0317998975,hint,SIP/0317998975
exten = 0317998975,1,Goto(0317998975-${DEVICE_STATE(SIP/0317998975)},1)
exten = 0317998975,2,Hangup()
exten = 0317998975-INUSE,1,VoiceMail(0317998...@inputinterior.se,bs)
exten =
Hi
Got a new iphone, want to know about peoples experience with any apps
that work well with asterisk and run on a iphone
Alex
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On Sun, Dec 13, 2009 at 11:15 AM, Alex Samad a...@samad.com.au wrote:
Got a new iphone, want to know about peoples experience with any apps
that work well with asterisk and run on a iphone
http://www.voipusersconference.org/2009/sip-for-apple-iphone/
I have not done any Asterisk-specific
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On 13/12/2009 10:24, Randy R wrote:
On Sun, Dec 13, 2009 at 11:15 AM, Alex Samad a...@samad.com.au wrote:
Got a new iphone, want to know about peoples experience with any apps
that work well with asterisk and run on a iphone
Removing the spaces did it. I works now. I used the space for clarity but,
Asterisk didn't like it.
Thanks for your time.
--- On Sat, 12/12/09, Warren Selby wcse...@selbytech.com wrote:
From: Warren Selby wcse...@selbytech.com
Subject: Re: [asterisk-users] Unable to open file...
To:
On Sun, Dec 13, 2009 at 11:24 AM, Randy R randulo2...@gmail.com wrote:
On Sun, Dec 13, 2009 at 11:15 AM, Alex Samad a...@samad.com.au wrote:
Got a new iphone, want to know about peoples experience with any apps
that work well with asterisk and run on a iphone
Hi!
Trying to figure out what I am doing wrong...
1 std SIP phone (0317998975) registered to an Asterisk (SVN-trunk-r234256)
1 Cell phone 00733025975 attached through H323.
extensions.conf
exten = 975,1,Goto(975-${DEVICE_STATE(SIP/0317998975)},1)
exten =
I have tried every sip phone offered on Apple's APP store (as of three weeks
ago) and the only one that worked well for me was iPico and I think I paid
$10.00 or so for it.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On
Siax is working great for me and as far as I know/remember well, you
can get it from the app store for a reasonable price. It supports SIP
and IAX2 and works easy with Asterisk.
\erik
On 13 dec 2009, at 15:32, John Regal wrote:
I have tried every sip phone offered on Apple's APP store (as
Thank you, very interesting!
As I understand the Digium card is used as a interrupt source for Asterisk?
Is there a diagnostic tool available ?
Anybody else experienced a simmialr problem?
Thank you!
HB
From:
cov...@ccs.covici.com
Date:
Sat, 12 Dec 2009 19:04:23 -0500
To:
Asterisk
Philipp Kempgen wrote:
Magnus Benngård schrieb:
Set
call-limit=10
(or any other value 0)
Actually, I believe call-limit is deprecated, and you can instead use
callcounter=yes
Leif Madsen.
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I have been looking for a way from the dialplan to inquire if there are
any members in a queue.
So what I want to do is if no users are members of a queue then I can
send the call to a given extention.
I have the queue setup all that is working. Just need to be able to send
the call to a
On Dec 13, 2009, at 7:20 PM, Jerry Geis wrote:
I have been looking for a way from the dialplan to inquire if there are
any members in a queue.
So what I want to do is if no users are members of a queue then I can
send the call to a given extention.
I have the queue setup all that is
On Tue, 08 Dec 2009 10:42:58 -0800, Dave Platt dpl...@radagast.org
wrote:
[Dec 8 18:24:48] ERROR[10571]: chan_alsa.c:693 alsa_read: Read error:
Resource temporarily unavailable
I agree, this looks like some form of conflict for the sound device.
i've got so far that i can use pulseaudio
Hi All,
I'm having problems getting results from a PHP file. This is what the CLI is
showing.
-- Executing [...@internal:1] AGI(Console/dsp, GoTalk.php) in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/GoTalk.php
[Dec 14 11:57:25] ERROR[20260]: utils.c:1019
On Mon, 14 Dec 2009, David Klaverstyn wrote:
I'm having problems getting results from a PHP file. This is what the
CLI is showing.
-- Executing [...@internal:1] AGI(Console/dsp, GoTalk.php) in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/GoTalk.php [Dec 14
11:57:25]
On Sun, Dec 13, 2009 at 8:45 PM, meetmecall i...@meetmecall.nl wrote:
Siax is working great for me and as far as I know/remember well, you
can get it from the app store for a reasonable price. It supports SIP
and IAX2 and works easy with Asterisk.
It looks like it requires a jailbroken iPhone,
Hi Jerry,
I use the built-in function queue_member
http://www.asterisk.org/docs/asterisk/trunk/functions/queue_member?type=functionsvalue=QUEUE_MEMBER
and check with a GotoIf statement to check if the number is equal to zero. If
it is not I send the call to the queue, if it is I pass the
Fring, it's free and works perfectly with an Asterisk server..
On 13 Dec 2009, at 10:15, Alex Samad wrote:
Hi
Got a new iphone, want to know about peoples experience with any apps
that work well with asterisk and run on a iphone
Alex
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