similar to: Manager proxy

Hi folks -- For those of us trying to develop interactive web-based Asterisk applications, it can be a challenge getting Asterisk integrated in a cgi/mod_perl/php environment. The load associated with making multiple connections to asterisk via the manager port, having to teach our applications to parse "manager-ese", and a variety of other factors have all pointed to the need for

Hello all, Anyone expecialized with call center monitoring and reporting solution based on asterisk. A client of us, want to install a call center reporting solution for an asterisk server but I do not know which could be the best tool for that. I need a tool for reporting queue calls, agent calls, and disconnect cause. Any clue will be appreciated. Thanks in advance. VoipCrazy

Hi, With more recent version of v1.2 and with v1.4 are things like the AstManProxy still recommended if you want to have a bunch of applications talking directly to Asterisk? Cheers! -- Andrew Ruthven, Wellington, New Zealand At work: andrew.ruthven at catalyst.net.nz At home: andrew at etc.gen.nz GPG fpr: 34CA 12A3 C6F8 B156 72C2 D0D7 D286 CE0C 0C62 B791 -------------- next part

Hello Voipcrazy, It's funny you should mention that - we've just released (as in today) a FREE version of our OrderlyStats service for call centre and queue monitoring and management. OrderlyStats features realtime (synchronous/message-based) display of all queue, agent and caller events so you can see what's happening in your call centre as it happens. The control panel feature

Greetings everyone, I'm pleased to announce the release of Astmanproxy 1.20, the fast, flexible proxy server for Asterisk's Manager Interface. Astmanproxy allows you to communicate with multiple Asterisk boxes from a single point of contact using a variety of I/O formats, now including support for XML, HTTP, HTTPS, SSL, CSV, and the Asterisk-native standard format. Astmanproxy is

Hi All, I want to track a call that is originated using originate AMI command through AstManProxy server. I m using AstManProxy server and I developed an AstManProxy client. By using my AstManClient program I can able to login AstManProxy server. Now I can able to issue/send originate command to generate a call but I m very confuse that I cannot able to track my call. The AMI events were

Hello, I cannot find documnetation explaining how to access AstManProxy. I am working with Asterisk Java and accessing Asterisk Manager. I wonder if AStManProxy is using the same API as Manager? Can I access it with Asterisk java too ? Regards, Daniel

Hey there folks -- I have been continuing development on the multi-threaded, c-based Asterisk Manager Proxy program, AstManProxy. I've incorporated several ideas I received at the recent Astricon Europe, including: - Supports proxying of multiple Asterisk servers at once - Abstracted, modular I/O handlers (implemented as shared objects) - Existing handlers: XML, Standard, CSV, HTTP

Greetings -- Many of you have downloaded and tried out Astmanproxy, a multi-threaded C-based proxy for Asterisk's Manager Interface. It has been under development since April 2005 and was presented at the Madrid Astricon in June, and will also be presented at Astricon in Anaheim in October. There has been interest in setting up an Astmanproxy mailing list specifically devoted to

Dear list, We have an weird problem with our FXO card (TDM01B). When I made a call using this channel, all goes well, the called phone rings but when the called phone answers the call. In me handset I can hear an weird sound like a "Clack". I tryed diferents TDM cards and modules, and my zapata.conf is like, language=en context=from-zaptel switchtype=national usecallerid=yes

For a long time now we've used the astmanproxy process to handle manager connections (75+ clients) so that these clients can tap into the dialplan / send commands etc. We use astmanproxy because at that time the manager connection routines of asterisk did not cope well with numerous connections, either permanent or as-needed. Does anyone have any experience of later versions of asterisk

After having misunderstood some key elements of AstManProxy, I started to write my own proxy server for Asterisk AMI. I was under the impression that it required a mysql database to cache its data for some reason. (Is there another AMI proxy that uses a mysql database?) At any rate, I had written about 70% of the core functionality so I decided to continue on. I'm not a C programmer so

Hello there, I'm new to Asterisk and I'm trying to figure out a way to make the Asterisk Manager Interface (AMI) accessible to multiple users at the same time. Would anyone recommend an AMI proxy that could be accessed from PHP code? Thanks in advance, Leo

Hi all, I am looking for phones witch support POE, with a good relation between quality and price to work with asterisk. I just see the Thompson st2030 and the Linksys SPA 922 an SPA 942. Witch of this phones or another ones gave you the best results in a productivity enviroment? Thanks in advance. VoipCrazy. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, Looking at /etc/init.d/astmanproxy file, it seems astmanproxy should create a /var/run/astmanproxy.pid file when run. I can't find any such astmanproxy.pid (not in /var/run directory, nor anywhere else). Is it normal ? (I'm using Proxy/1.22pre081119) Regards -------------- next part -------------- An HTML attachment was scrubbed... URL:

Dear list, I need to buy a phone which could monitor the state of the maximun number of sip extensions about 200. It is for an attendant. I just saw Snom 370 with keypad and Linksys 962 but they do not let me to monitor 200 extensions states adding keypads. Do you know any kind of phone that let me do that? Which is the maximun number of extensions your phones can monitor and which models phones

Hello all, I am using the Snom 370 phone with firmware Snom370-SIP 7.0.17* *connected to an asterisk 1.2.14 and I can't record any calls using the "Recording" button on this phone. The extension I configured on this phone has the values "Recording on demand", an the voicemail enabled. I am using FreePBX to manage my PBX. How should I configure the "Function

I'd like to be able to generate a Manager Event from the dialplan but can't seem to find a way to do it. Alternatively, trigger and Event when a record in AstDB gets changed. Can anyone point me in the right direction? Thanks. By way of explanation, I've a app that connects to astmanproxy and I'd like it to know when a call group gets put into Nightservice. Putting the call

When the asterisk a queue reset their counters? I 'm talking about this kind of info in asterisk console. >show queue 600 600 has 0 calls (max unlimited) in 'ringall' strategy (4s holdtime), W:0, C:14, A:8, SL:0.0% within 0s I just say that because I have a queue with strategy "Fewest Calls" working for a couple of mouths, and a new agent has been added this

Hello list, hope someone can help me - I'm going crazy using the FXO port a SPA3012. I would like the SPA 3012 to act as a simple FXO port to an Asterisk, that is, once it detects a call, it should simply send it over to the local Asterisk server. No intelligent routing, PIN, anything else.... I configured it like this: PSTN-To-VoIP Gateway Setup PSTN-To-VoIP Gateway Enable: yes PSTN