similar to: Manager Port

Hello, I have a Digium TE410P card. I get the "knight rider" lights before the module (wct4xxp) loads, but after the modules are loaded I don't get any lights. I have found the following 2 posts but still could not solve the problem http://lists.digium.com/pipermail/asterisk-users/2004-November/075277.html http://www.voip-info.org/tiki-index.php?page=Asterisk+TE410p+No+Interrupts

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

I'm pleased to announce that Flash Operator Panel 0.26 has been released! FOP is a GPL'd switchboard type application for the Asterisk PBX. It runs on a web browser with the flash plugin. It is able to display information about your Asterisk box in real time. It is included in FreePBX, Asterisk@Home, DeStar, startShop, and several other projects both free and commercial. You can grab the

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

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

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

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

Hello, I have the following situation: I have a PRI with 200 DID numbers and I have set up 200 sip extensions that matches the last 4 digit of the corresponding DID numbers so that when any of the 200 DID number is called, asterisk can pass the call to the respective sip extension. Incomming has been fine. But when making out going calls I want the called party to always see the same number

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

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, Can anybody pls tell me, how to increase the time gap between 2 digits when you transfer a call. ie, the operator answers the call, and presses hash key to transfer, and then enters the extension number, some times, it timeouts too quickly before the operator enters the whole extension number (may be bcos the operator is slow). I tried the following, but it doesn't seems to be helping

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

Nicolas, Thank you for your response. I had tried that before and it didn't work. I am trying to look up the route for a dialed number, so its a full E.164 number. Please see my query below when I try to look up the route for a USA number; mysql> SELECT * FROM routes WHERE "^13237309880" RLIKE pattern ORDER BY LENGTH(pattern) DESC;

List: This is my first manager client that I've written so please bear with me: I am trying to write a C manager interface client to interface with our CRM software. I am having an issue while reading the data from the manager interface. I am writing this in C and I have the following code: while(1) { bzero(buffer,sizeof(buffer)); readCode = read(socketHandle,buffer,sizeof(buffer));

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...

Hello all, Can I execute linux command like(ls, mkdir) through the Manager interface? I can't seem to access the manual at digium.com. I keep getting 'Forbidden' error. Looks like they are upgrading or something. CF

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 >>> between versions to find the problem but I took the lazy way out the >>>

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

Hi All, Whenever i try to start FOP using script ./op_panel_redhat.sh start given in directory /usr/local/op_panel-snapshot/init I got the following error: Starting Flash Operator Panel: execvp: No such file or directory [FAILED] Please let me know the reason for this. Thanks in Advance With Regards, newbie