search for: gudi

Is there anyway in asterisk to figure out how much time an agent has spent on the phone? I know I can see total time for a call (inbound or outbound) but where/how do I view queue stats?

Hi all, How are things going ? Is there a way for me to individually identify each BRI channel on the Sirrix quad BRI board. The reason I ask is because our client uses the "Asterisk Flash Operator Panel" to monitor its external lines and transfer calls from the lines to the various SIP phones. The "Flash Operator Panel" requires that we set a static value for each line or

If a user calls and hangs up before the destination party rings, than the in-use flag remains set! This is one case, but maybe there are many other cases. I have created a number the user can dial to reset this flag. However, that is written in the manual!!! Who reads a manual anyway!!!! I want to make to reset all in use flag with a program. Has anybody done it, or has a better idea? My idea

...m not a MYSQL expert, so I am not sure how to make the query return the result with the first digit of the dialed number as the first row, which would be the USA route. Thanks, Voipcarib ----- Original Message ----- > Message: 9 > Date: Tue, 14 Dec 2004 10:04:08 -0300 > From: Nicol?s Gudi?o <asternic@gmail.com> > Subject: Re: [Asterisk-Users] ASTCC > To: Asterisk Users Mailing List - Non-Commercial Discussion > <asterisk-users@lists.digium.com> > Message-ID: <e7d413aa04121405042f00bcf4@mail.gmail.com> > Content-Type: text/plain; charset=ISO-8859-1 &g...

Hello, Just confirmed this on my end, because of the massive changes that have been made to callerID handling in asterisk 1.0.5 many of the features of the astGUIclient suite will not work on this new version. The latest stable version recommended is Asterisk 1.0.3. We will work on trying to find ways around the new callerID rules that the asterisk developers have put in place and hope to have

Hello Guys. I recently contacted polycoms tech support asking if their phones supported XML pushed information to which they replied that only model 600 had a microbrwoser capable of reading dhtml files and such. My question to the community is: is somebody doing any XML info push to any brand of phones except Cisco? How are you doing it? One of the wonders of VoIP should be the means to send

Been using mpg123 for moh for the last two years or so. However, when I have * config errors, often times get a endless stream of console messages and need to kill the two mpg123 processes. Is there an alternative to mpg123 that eliminates that issue? I see references in musiconhold.conf relative to madplay, native file format, asterisk-addons, etc. Not sure why the asterisk-addon approach

...fg. * Several bug fixes, internal refactoring, profiling and optimizations. The upgrade instructions are on the tarball UPGRADE file. Remember to upgrade the .swf file and to flush your browser cache! Many thanks to everyone who provided feedback, patches, ideas and suggestions. -- Nicol?s Gudi?o Buenos Aires - Argentina

...t, I'd hate to have to install asterisk@home from scratch just to fix this FOP (conference rooms themselves work fine). Any suggestions? Thanks, Dean -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Nicol?s Gudi?o Sent: Thursday, February 10, 2005 8:19 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] really easy FOP asterisk@home question Hello, > > I deleted the config examples in the op_buttons.conf folder for how to set > up the meetme representati...

I have setup flash pannel, ... looks nice, but so far I could not configure it to get more than 4x7 buttons. I tried to make the buttons smaller, but than just the entire picture is smaller. The description says you can have a hundred buttons, .... Can I have multiple flash pannels? E.g. for each department? bye Ronald

Nicol?s Gudi?o <asternic@gmail.com> wrote: >>> Paul, 1.0.5 stable suffers from caller id issues as well, at least for >>> SIP channels. What fixed things for me was swapping in app_dial.c from >>> 1.0.2 stable (didn't try others). You could also just diff app_dial.c >&gt...

Ok, we're trying to use Asterisk as a PBX in our office. Our original plan was to use a Cisco 7960 with a 7914 attached. Short story is, no one updated chan_sccp in a long time and the 7914 is questionable at best anyway from what I've heard. We couldn't ever get chan_sccp to compile, I went to an older version of Asterisk and that broke some of our SIP devices. We tried using a couple

Dear All, I have a requirement to 'originate' a number of calls to various external users from within a conference room, so that the end users does not pay for the call. I know that within Astman I can define an extension and then originate the call from that extension. Can I define a conference room (how would I configure that on astman? What channel would it use?) and then generate a

Hi, Does anyone know of a Print-to-Fax client that works with asterisk & spandsp? Astfax is a partial solution but that only lets us email the fax in, we'ld like to set it up so the user can hit the print button and send the fax (even if all it does is email - transparently to the user - the fax to astfax). -------------- next part -------------- An HTML attachment was scrubbed...

I download and installed ASTCC over the weekend and I am having an issue where the INUSE flag will not get set back to 0 if the user drops a call while the balance is being played. All other times it seems to reset the flag correctly. I have tried both AGI and DeadAGI with the same results. Those of you using it for a while, how did you get around this? Just for fun this is all I am doing in

I'm trying to set up xinetd to run an asterisk console on a tcp port. So far I've added a file in /etc/xinetd.d/ like: service actl { disable = no socket_type = stream protocol = tcp port = 1234 wait = no user = root server = /usr/sbin/asterisk

Hello! I am new to both the list and to "*". Can someone direct me to some documentation concerning the reporting tools available for use with "*" as a call-center system? Specifically, things like ACD offer/taken, wrap-time, and such? Thanks very much. This looks like an exciting project. I'm looking forward to playing with it! -- Michel R Vaillancourt Avaya

I just installed ASTCC and it wroks ok. But there is one problem whenever I want to generate any new cards it seems to work for a long time and at the end it fils w/o any error message. Any suggestions ? Thanks. Ehsanul Karim

Some days ago there was a subject regarding BLF (SIP Phone-receptionist Setup). We are the developers of a Price Verify Terminal for a French company. We have developed the hardware (small board based on a PPC 823e), working with Linux embedded (based on Wolfgang Denk's work). I think that it can be a good BLF. Probably it is possible to integrate the Nicolas's FOP or a new application.

Is it possible to execute a script upon successful registration and authentication of a SIP device in Asterisk? For instance, have a script log all successful registrations in a database or authenticate users instead of using the secret=password in the sip.conf file? Thanks - -- Brian Wilkins Software Engineer brian@hcc.net Heritage Communications Corporation Melbourne, FL USA