l plan when the agent answers the call.
>>
>> same =>n,Queue(sales,tc,,sub-QueueConnected)
>>
>> [sub-QueueConnected]
>> ; this runs on the agent/member's channel
>> exten =>s,1,NoOp()
>> ; whatever you need to do here
>> same =>n,Return(
ou need to do here
> same =>n,Return()
>
> See https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Application_Queue
>
>
> Mitch
>
>
> On 08/03/2013 12:45 PM, Timothy Smith wrote:
>>
>> Hello Folks,
>>
>> I am setting up a call center but we have few age
Hello Folks,
I am setting up a call center but we have few agents so one agent is
able to handle calls of different languages and different queues. For
the agent to identify the caller, I want a popup to appear as the
phone starts to ring with the caller's number, language (selected in
the IVR), Q
Hi,
I am running a service where I play full songs but MP3 files kept on
crashing my server. I resorted to wav but the quality is really poor
after converting..or even sometimes not audible at all! Do you guys
know of a better way I can convert mp3 to wav and restore quality?
Below is the script I
On Fri, Feb 4, 2011 at 7:32 PM, A J Stiles
wrote:
> (Putting everything back into the right order, and stripping out unnecessary
> bits, for the sake of anybody searching the archives in future.)
>
Thanks!
> On Friday 04 Feb 2011, Timothy Smith wrote:
>> On Fri, Feb 4, 20
think I installed it using yum, however, i can still install a
version from sources, just to be sure. Could you please give me the
exact URLwhere I can download a version that works well with asterisk?
Thank alot!
Tim
On Fri, Feb 4, 2011 at 5:37 PM, A J Stiles
wrote:
> On Friday 04 Feb 2011, Timo
Hi Users,
I have a problem with some of my mp3 files. they crash the system
(Asterisk 1.6.2.14 on a x86_64 running Fedora 13 ) when it tries to
play them. Unfortunately the logs do not give me a clear fault or
cause of crash but i can clearly see that ts because of the MP3 files.
Its the way some
o have a visual solution (a good
> receptionist console), but we don't have time to create one for our own
> usage, and many solutions over the web are not compatible with our Asterisk
> version (1.6.2.x).
>
> Hope this helps.
>
> Hoggins!
>
> Le 04/09/2010 14:42, Timo
Hi,
I am implimenting a solution for a radio station where by calls are
first received by an attendant, who interviews the caller and then
places the call in a queue along with some information about the
caller. The radio presenter can then choose which call to pick up
depending on those in the qu
Thank David and Neeraj for your input.
Neeraj, I posted the configs in my first post, but i've also attached
some extracts here. they haven't changed much.
David, You're absolutely right and i think the problem could be the
reverse dial-peer or DTMF configuration. I think I have the
corresponding
Tim
On Sun, May 17, 2009 at 3:44 PM, David Backeberg wrote:
> On Sat, May 16, 2009 at 10:22 AM, Timothy Smith wrote:
>> I have finally managed to get voice working. I both parties can hear
>> each other. The problem was nating. Our network is fairly big and
>> these machines
ance.
On Sat, May 16, 2009 at 4:02 PM, David Backeberg wrote:
> On Sat, May 16, 2009 at 7:46 AM, Timothy Smith wrote:
>> I have added the 3845 router as my SIP gateway (on asterisk 1.6.0.9),
>> and also a dialpeer to forward on the router to forward calls to my
>> asterisk
y 2009, at 12:46, Timothy Smith wrote:
>>
>>
>> Has anyone had the above set up working successfully? Attached are
>> some confs.
>>
>> Thanks a lot for your assistance.
>
> Check about the sip.conf 'insecure' option. I have had to use it in
> t
Hi,
In our office, we're slowly migrating from a cisco call manager set up
to asterisk. Problem is management doesn't want to buy any other
hardware as they had already invested a lot in cisco. The main cause
of this is asterisk's added features like unique FAX number for
everyone in the company
ckeberg wrote:
> On Thu, Apr 2, 2009 at 12:07 PM, Timothy Smith wrote:
>> In our office, we're migrating from a Cisco set up to Asterisk.
>
> What is the goal of doing this migration?
> Plenty of people do a blended environment with Cisco doing what Cisco
> does well and
Hi,
In our office, we're migrating from a Cisco set up to Asterisk. We'd
like to do it gradually, so I've added an asterisk server as an H.323
gateway to the call manager so out going calls are going through
asterisk. So far so good.
Am now faced with the challenge relaying incoming calls from as
Hi,
In our office, we're migrating from a Cisco set up to Asterisk. We'd
like to do it gradually, so I've added an asterisk server as an H.323
gateway to the call manager so out going calls are going through
asterisk. So far so good.
Am now faced with the challenge relaying incoming calls from as
Hi,
I would like to get musiconhold from a sound card. This is because I
want to kind of be a DJ and easily change the music playing, etc.
However, I followed the instructions at
http://www.voip-info.org/wiki-Asterisk+config+musiconhold.conf and
other tutorials on the net but no success. I have
[
Hi,
I would like to get musiconhold from a sound card. This is because I want to
kind of be a DJ and easily change the music playing, etc. However, I
followed the instructions at
http://www.voip-info.org/wiki-Asterisk+config+musiconhold.conf but no
success. i have
[mycustom]
mode=custom
directory
Hi,
Could someone please help me with this?
I have a new asterisk box running asterisk 1.2.24 on open suse 10.3 on
an acer aspire motherboard. It has a TDM card with 3 fxos and 1 FXS,
where an incoming line is plugged and also analog phone plugged to the
FXS port. Am faced with the problems below
Hi,
I have a new asterisk box running asterisk 1.2.24 on open suse 10.3 on
an acer aspire motherboard. It has a TDM card with 3 fxos and 1 FXS,
where an incoming line is plugged and also analog phone plugged to the
FXS port. Am faced with the problems below.
- For conversations between analog pho
Hi,
I'm having problems using a TDM400P Card. I put my SIM card in a Nokia
Premicell and connected it to a TDM400P card but when I make calls to
the number, it hangs up automatically. The card also can't call out.
Any ideas? I've searched the archives without much success. I
appreciate all your he
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