Hi Kevin,
Thanks for the reply. So purchasing TE412P with VPMOCT128 echo-cancellation
module is not going to effect the current process? It will work with
asterisk-1.2.17, zaptel-1.2.17.1. Correct?
Regards,
Kurian Thayil.
On Sat, Nov 28, 2009 at 8:14 PM, Kevin P. Fleming kpflem
a backup
card. So, I just need to know whether zaptel driver which is installed in my
server is compatible with new TE412P having VPMOCT128 echo cancellation
module. Thanks in advance.
Regards,
Kurian Thayil.
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resolved only when asterisk service is restarted which is not
a pretty good workaround. Any clue on this?
Regards,
Kurian Thayil.
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Hi,
My apologies Nicolas, a mistake from my part. And I appreciate for
correcting me. Asternic is a good piece of work.
Regards,
Kurian Thayil.
On Mon, 2009-07-06 at 09:41 -0300, Nicolás Gudiño wrote:
Hello,
Just a correction, Asternic Call Center Stats is not from
asteriskguru
asteriskguru is kind of OK. Gives you a live and
detailed report. Parses the queue_log to the MySQL DB and works. This
parse program could be used in your AGI which I mentioned in point 2.
Hope this helps.
Regards,
Kurian Thayil.
On Sun, 2009-07-05 at 22:41 +0530, Sriram wrote:
Hi
1. I want
was under a notion that I could activate an AGI only when an
extension is hit by channel. I wasn't aware that we can scan a channel
continuously using an AGI. If so, how could we do that?
Regards,
Kurian Thayil.
On Thu, 2009-06-04 at 08:35 -0500, Danny Nicholas wrote:
There was a nice post earlier
Hi,
I had similar issue which happened when record option was mentioned in
both agents.conf and queues.conf. When I commented the recordagentcalls
option in agents.conf, it started to work. Mention the monitor option
only in the queues.conf file. Do try.
Regards,
Kurian Thayil.
On Sun, 2009-06
Hi All,
I am looking for an option in Meetme or similar which will enable to
skip to next priority (a voicemail) if the person in Meetme conference
is alone and if he is there for some time (say 3 minutes)? Any hints on
this? Thanks in advance.
Regards,
--
Kurian Mathew Thayil.
(GPG KeyID:
Hi all,
I ve setup a queue with 2+ agents for managing our inbound calls from
customer. Using Asterisk 1.2.18 in a CentOS box. Agents login using
AgentCallbackLogin application and I use a BASH AGI to accomplish this
as there are some validations done with MySQL DB. Im aware that transfer
could
and the agents that are being tried
and can obtain corresponding agents that are tried for.
[inbound]
exten = 9712,1,AGI(agi_queue.sh)
Regards,
Kurian Thayil.
On Thu, 2009-05-21 at 11:09 +0200, Lenz Emilitri wrote:
What exactly are tyou trying to achieve?
l.
2009/5/20 Kurian Thayil
is not
shown above.
agi_qdial.sh
*declare -a array
while read -e ARG [ $ARG ] ; do
array=(` echo $ARG | sed -e 's/://'`)
echo ${array[0]} = ${array[1]} $LOG_FILE
export ${array[0]}=${array[1]}
done
echo EXEC Dial SIP/$agi_extension 20 tTo
*Any hint on this??
Regards,
Kurian
Hi All,
Is there a way that I can bring the callerID obtained from an incoming call
to display in a web browser. I am using eyeBeam as softphone. Please help me
with this. Thanks in advance.
Regards,
Kurian Thayil.
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On Wed, 2009-04-22 at 15:24 +1000, Lee, John (Sydney) wrote:
Daily Asterisk restart
Do you think its mandatory in production env?
Daily log rotation
Voicemail clean up for people leaving an organization.
Hi James,
I have done a similar implimentation with Asterisk. And its quite easy
to accomplish. The voice files (questions) have an entry in a MySQL DB
and then I wrote an AGI in BASH which actually picks a number for the DB
and dials the extension. The asterisk plays the voice file using STREAM
Hi James,
I guess in your case, you can use an AGI which mix the recorded voice
files alone using 'soxmix'. I am not sure, how u ll be able to
accomplish without the help of AGI.
Regards,
Kurian Thayil.
On Fri, 2009-04-10 at 12:33 -0500, James A. Shigley wrote:
Well considering php
command i have used Dial(SIP/1234,,A(beep)) so that the
agent hears a beep when they get a call.
Hope this enlightens you a bit on handling inbounds in this situation :)
Cheers
2009/3/12 Kurian Thayil kurianmtha...@gmail.com
Hi Geriant,
My apologies for the delay in reply. We won't
Kurian Thayil kurianmtha...@gmail.com
Hi Steve,
That worked beautifully. Thank you so much. But one question though.
Imagine if I keep a Hangup Button in the interface and it should terminate
the call. Will that be possible? This scenario happens when the user gets
connected to an invalid phone
Hi All,
Is there a way that I can include call dialing functionality in a
webinterface. I have EyeBeam configured with a SIP user say
8440. Will I be able to design an inteface which agent can choose a number
and the Dial without punching in the number in
Eyebeam.
I tried using the .call file. ie
-notification.com wrote:
On Tue, Mar 10, 2009 at 6:40 AM, Kurian Thayil
kurianmtha...@gmail.comwrote:
Hi All,
Is there a way that I can include call dialing functionality in a
webinterface. I have EyeBeam configured with a SIP user say
8440. Will I be able to design an inteface which
://www.speech.cs.cmu.edu/Communicator/ . Something I have not looked
into much (and I don't know if it has anything to do with Asterisk). I hope
this helps.
Regards,
Kurian Thayil.
On Sat, Jan 31, 2009 at 9:22 AM, Alfred Monticello ajmce...@yahoo.comwrote:
I wouldn't have a database to compare
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