similar to: ASTCC and Pattern question

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

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 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

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;

Hi all. Just to update list and increase the "souls-save" database. The patch solved the problem. Now I have an asterisk-1.2.0-beta1 with asterisk-perl-0.08 and mysql-server-3.23.58-16.FC2.1 machine working fine with ASTCC and "inuse" flag. The link of the patch is: http://bugs.digium.com/view.php?id=5400 Best regards to all you in the list. Ricardo Poppi.

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

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?

I'm looking for a good web based voicemail client that can use mysql or realtime drivers. I can't seem to get vmail.cgi to work with realtime. Thanks for any help you can give.

Not able to get the auto dialer part of asterisk to work with the zap channel. It works great with the sip channel. Here is the call file and the CLI output Call File Channel: ZAP/G1/6144994925 MaxRetries: 3 RetryTime: 40 WaitTime: 2 Context: amaxx Extension: 36652 Priority: 1 CLI Output Connected to Asterisk SVN-branch-1.4-r57207 currently running on VoIP-PBX (pid

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

I search Google to find how to get the message light to flash on my Cisco 7960 running (Application Load ID POS3-06-3-00) (Boot Load ID PC03M030) (DSP Load ID PS03AT38) All I see is about the sip.conf file witch mine has the mailbox=XXXX but still no light. Also the messages button does not work. Any ideas?

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, 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

I have created the following stuff: BRANDS: Brand Name Language Published Number DID Inc Markup (in 1/100 of 1%) Gamma en 5000 6 0 TRUNKS: Trunk Name Technology Peer/Trunk Gamma SIP 200.68.XXX.XXX ROUTES: Pattern Comment Trunks Connect Fee Inc. Seconds Cost per additional minute 63 Parana Gamma 0 60 500 CARDS: I have created some custom numbered cards. So, every call started with

I have astcc up and running with one problem no matter what I call I get the number you have dial is unavailable. I think my trunks and routes are correct but I'm not 100% sure. I was wondering if anyone else has gotten this error? and if anyone had some basic examples of how a config should look like using either sip or the zap interfaces for the Trunks. Any Help is Appreciated Eric

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

I have a question for the group. To get this running do I need any Digium Cards? I understand I will need them to connect to the public phone system. I'm looking at just using IP Phones or IP Softphones just to test this app. Thanks for any help you could give.

Hi, alhtough i googled for details concerning ASTCC i did not found an aswer to the following: Should i put in my extensions.conf the configuration of the astcc? I ask this because as i see it, in the end of the extensions.conf there is an include statement : #include /var/lib/astcc/astcc-exten.conf Should the config been done in the astcc-exten.conf file or the initial extensions.conf

Hi: Astcc is working fine, except for one thing. It doesn't give the called party enough time to answer the phone. If nobody picks up in two rings, astcc reports back no answer and hangs-up. The only instant NOANSWER "value" was mentioned in astcc.agi script is: elsif ($res eq "NOANSWER") { $res = &mystreamfile("astcc-noanswer");

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