I have just setup a small queue implementation for one
of my branch offices, replacing a 16 year old key system
that had a hacked together pseudo call queuing feature.
The 'agents' are not dedicated to the queues and want to
be able to logon and get one call only from the queue.
I know this is
show application RemoveQueueMember
-= Info about application 'RemoveQueueMember' =-
[Synopsis]
Dynamically removes queue members
[Description]
RemoveQueueMember(queuename[|interface[|options]]):
Dynamically removes interface to an existing queue
If the interface is NOT in the queue and
Hi. I have a SPA3102 updated with with Software Version: 5.1.7(GW).
I have this settings on Voice/Regional:
Interdigit Long Timer: 10
Interdigit Short Timer: 3
Anyway, when hooking up (without dialing anything), the timeout starts
after 3 seconds. It's like the Long Timer is unused. After
Julian wrote:
show application RemoveQueueMember
-= Info about application 'RemoveQueueMember' =-
[Synopsis]
Dynamically removes queue members
[Description]
RemoveQueueMember(queuename[|interface[|options]]):
Dynamically removes interface to an existing queue
If the interface is NOT
hi;
i bought 2 numbers:
971-228-0707
and
971-228-0708
and then got the random number i wanted. I am asking for $3.95 for
these numbers a month, unlimited incoming, no out going. Can if
someone wants these let me know? They can be sent to whom ever you
wanted in the us or canada. It gives
I have a DID from IPKall.com which is forwarded to my asterisk box.
Then this extension should call my ip phone using Dial application.
Everything works fine, except when I pickup the phone, I can talk, the other
party can hear me, but I cannot hear anything the person says on the ip
phone.
Then
Emmanuel Pascal Bruno wrote:
I have a DID from IPKall.com which is forwarded to my asterisk box.
Then this extension should call my ip phone using Dial application.
Everything works fine, except when I pickup the phone, I can talk, the
other party can hear me, but I cannot hear anything the
Hi Rob,
Also try without the r option to the dial command:
http://www.voip-info.org/wiki-Asterisk+cmd+dial
Rob
I should have read the wiki instead of a number of other dokumentations.
I used to add the r option to make sure that the caller always hears
the ringing tone. I my case this
Try moving the card to another slot in the computer. Also disable everything
in the BIOS that you aren't using like the sound card, serial ports,
parallel ports, etc.
On Fri, Oct 31, 2008 at 12:30 PM, Edwin Quijada
[EMAIL PROTECTED]wrote:
Date: Fri, 31 Oct 2008 11:39:43 +0200
From: [EMAIL
Oh ok, I knew it was something like that. I have tried many different
settings on my router. I'll dig into it some more.
Thanks
On Sat, Nov 1, 2008 at 2:04 PM, Rob Hillis [EMAIL PROTECTED] wrote:
Emmanuel Pascal Bruno wrote:
I have a DID from IPKall.com which is forwarded to my asterisk
Hi fellows..
I have 2 asterisk servers in which the following line
exten = _09049.,111,SetVar(LIMIT_PLAYAUDIO_CALLER=YES)
exten = _09049.,112,SetVar(LIMIT_WARNING_FILE=beep)
exten = _09049.,113,Dial(${TYPE}${DESTINO}|30|L(3:1))
works OK on my Asterisk 1.2.9, it plays the beep 10
Dan Austin wrote:
Julian wrote:
show application RemoveQueueMember
-= Info about application 'RemoveQueueMember' =-
[Synopsis]
Dynamically removes queue members
[Description]
RemoveQueueMember(queuename[|interface[|options]]):
Dynamically removes interface to
Hi.
I registered a Gizmo trunk in trixbox, last version.
registers fine.
trixbox - Gizmo5/windows/laptop: work fine.
Gizmo/Windows/laptop - trixbox: phone rings, but no audio when answer.
What could be the problem? Tried changing codecs, nothing.
Trunk name: 192
type=peer
insecure=very
No need to compile ! out of asterisk source
Just put SHELL=/bin/false in your login script
The ! command will not work...
Alex
Kindly consider the environment before printing this e-mail.
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL
On Saturday 01 November 2008 18:52:41 Alexander Lopez wrote:
No need to compile ! out of asterisk source
Just put SHELL=/bin/false in your login script
The ! command will not work...
That's not completely true. The only thing that will prevent is the ability
to get a shell prompt
Hi,
I have installed Asterisk 1.4.20 on Debian Etch. The server has no telephony
card
installed, but I have anyhow installed Zaptel (Zaptel-1.4.9) in order to be
able to use MeetMe.
The Zaptel modules load normally. I obtain the following prompts:
kerplunk:/# /etc/init.d/zaptel start
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