://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 agents so one agent is
able to handle calls of different languages and different queues. For
the agent to identify
)
[sub-QueueConnected]
; this runs on the agent/member's channel
exten =s,1,NoOp()
; whatever you 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
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),
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
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
:
On Friday 04 Feb 2011, Timothy Smith wrote:
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.
Some distros used to use mpg321 instead of mpg123 (early versions of which
used to suffer from
On Fri, Feb 4, 2011 at 7:32 PM, A J Stiles
asterisk_l...@earthshod.co.uk 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
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
(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, Timothy Smith a écrit :
Hi,
I am implimenting a solution
, 2009 at 3:44 PM, David Backeberg dbackeb...@gmail.com wrote:
On Sat, May 16, 2009 at 10:22 AM, Timothy Smith timotsm...@gmail.com 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 are atleast
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
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
:
On 16 May 2009, at 12:46, Timothy Smith wrote:
blah
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
the past for similar stuff. I think it was 'insecure=very
...@gmail.com wrote:
On Sat, May 16, 2009 at 7:46 AM, Timothy Smith timotsm...@gmail.com 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. It works properly but the problem is there is NO AUDIO! I
have
:
On Thu, Apr 2, 2009 at 12:07 PM, Timothy Smith timotsm...@gmail.com 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 Asterisk doing what Asterisk does
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
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
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
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
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
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
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