similar to: asterisk 1.2.14 with GUI

Hello, I want to safely delegate ACD edition to a system administrator who has no knowledge of Linux nor Asterisk. More precisely, I want him to be able to edit and change menus such as : "Type 1 for management; 2 for support; 3 for sales department". I could teach this administrator what Asterisk config files are but I'm wondering if any GUI exists for such task (editing a vocal

I'm a network admin that maintains 3 commercial Asterisk servers for my employer. I am wanting to move away from the "pre-packaged" commercial PBXs to a more "pure" asterisk setup. The systems I have utilize a nice web GUI to make changes, but it really limits what I can do beyond what they have programmed into their GUI. Would I be better off starting with: a) Plain

I was searching thru the internet and I found a wide variety of different web interfaces for asterisks I was curious which one is best suited for asterisks. Thanks

Hello, In case there are other users of the AsteriskWin32 port... I haven't really used the AGI feature of Asterisk to run an application from extensions.conf. *Win32 supports Perl, which I don't know. Apparently, it's also possible to write AGI applications as EXE's (there's a eagi-test.exe file installed by default). => When a call comes in, I'd like an AGI

How to execute some system command from AMI? -- Tomislav Parcina firstname.lastname@email.t-com.hr

Hello all, Here is the requirement from my side to use Asterisk.NET API to generate an automated call (outgoing) from asterisk and then link to one of the extensions which plays a sound file for the callee. For this i have worked out in the follwing way 1)modified manager.conf to facilitate this API to talk to asterisk 2)used the command Originate to call a Registered user under

I am generating an outbound call through the Manager API and bridging it to an internal Extension, my problem is I am not able to find the logs for the call generated by the Manger API, Since on the same Asterisk server there are many users connected and I am receiving lot of Events back, not able to recognize which was the call generated by me as same time multiple users are dialing out.

Hello all, I think Lee has given me a head start, but I'm still running in a circle (at least i'm in the lead). The problem is with my queues. The phones go to their own voicemail after 5 rings. That's about the same time I allow the phone to ring before trying another phone in the queue. Is there a way to tell asterisk....? If this call is coming from a queue, do not follow a

I have all my SIP users in a realtime database. I would like to use MySQL command to query the database and use the results from the query to page all the phones found in the query. The results from the MySQL query will be multiple rows of extension: Something like: mysql> Select extension from sip where extension like '6%' 6001 6002 6003 ex.... I need to put all the results into a

Hi, I am having some problems mentioned below, the box is in production live environment with traffic around 30 - 100 calls. I am running T/E410P in a Dual P4 xeon with HT disabled. I am using zaptel 0.9.0 and asterisk stable 1 release. There is no gui, just mysql, perl (small script) and asterisk. System runs very smoothly if the calls are around 40-50 and comes one by one , however sometimes

Hi all! We've released VoiceOne 0.4.0, a web-based and open source solution which allows to fully manage an Asterisk service hosted on a LAMP server. We focused on an charming and overall user-friendly interface. Thanks to the authentication based on roles, once configured by a super user, the PBX may be easily maintained even by an Asterisk unskilled users. From a technical point of

Hi, Im new, but trying real hard! I just need general direction, not details yet..i'll try to figure those...just looking to avoid brick walls...bottlenecks...inefficiencies etc upfront. Hardware: motorola vt2442 - trixbox Apps: Dot Net application that operates the Manager API and the FASTAGI interfaces. I have the 2442 set as a PLAR so as soon as the ext is off-hook, it dials into the *61

Are there any nice GUIs out there for Asterisk Realtime? Google doesn't yield much. I spent a day trying to get VoiceOne to work without much success. Thanks, Mike Clark

Hi! anyone from here, who uses voiceone <http://www.voiceone.it> as their web gui for asterisk pbx? I know it's still under development but i wish someone would be joining on the development 'cause i think it's a great project to finish. I started some things on the validation forms on the zapata/zaptel part which is not included on the demo site. I hope I can get more help

Hi there, I want to use asterisk for sip comminication with max 1000 users Which gui shuld i use for adding users and managing asterisk? I tried AMPortal, it added extensions to mysql but asterisk did not find users i added ? installed asterisk 1.2.2 on FC4 Toygun

Is there a version of the pv_ops kernel that will do HVM and work with fc11? I have been failing miserably trying to get it to work. I have followed Boris''s guide at: http://bderzhavets.wordpress.com/2009/06/10/setup-fedora-11-pv-domu-at-xen-3-4-1-dom0-kernel-2-6-30-rc6-tip-on-top-of-fedora-11/ But have had no success so far. I am wanting to know what is the version of the pv_ops

Hey all, After figuring out a problem with AGI and freepascal, I have finished writing a small Cepstral (http://www.cepstral.com) AGI app. I wrote a small readme for it at http://www.datatrakpos.com/misc/dial/readme.txt. I'd like to give it to the community (source/binary) and was wondering where to post it? The wiki? Also, anyone have suggestion on licensing? LGPL? FreeBSD? Thanks

People, I have a Debian box with Asterisk and I've installed the Destar package in order to get web managing of my voip system. After I installed Destar, it runs on "localhost:8080", but my server does not have X-Window to access to it so I can engter the web interface.. So how can I change localhost:8080 to IP_ASTERISK:8080 in order to access Destar via web from another PC ???

Hi. Has anyone had any success in implementing a callback or ringback function in Asterisk? I've had a look at the callback-voicemail example on voip-info.org http://www.voip-info.org/wiki/view/Asterisk+tips+callback However it won't quite work for me. I need it for local SIP users which most of them don't have voicemail. If one SIP user calls another SIP user and the second user is

Hello, I'm glad to announce that DeStar 0.2.2 version has been released. This release contains a large number of bugfixes and new features, see CHANGELOG.txt for the full list. You can find it in the usual place: http://developer.berlios.de/project/showfiles.php?group_id=2112 Thanks for using DeStar, Santiago Ruano Rinc?n http://destar.berlios.de -------------- next part -------------- A