similar to: Voice Prompts with no sound

Guys, Im looking for a good text file editor for asterisk config files that can be embedded on a web page for online editing (on an interface), any recommendations? ________________________________________________________________ Anton Krall Direccion General Intruder Consulting A Division of IntruderEnterprises S.A. de C.V. www.Intruder.com.mx www.IntruderStore.com.mx Tel. 3872-2200 ext. 201

The method we use for web popups on incoming calls in the astGUIclient client app that we are working on for release next week is to use AJAX(Javascript + XMLHTTPRequest) It works in Firefox and IE5+ and doesn't require any META refreshes. We've been using this internally for the last month and it works great. MATT--- -----Original Message----- From: Anton Krall

Guys.. which free softphone is the best,grandest,most recommended one out there? based on your own experiences..

Guys... Ive been having problems with my callerid and I have no more clues as to what I could be.. dates and times stamped on voicemail and info received on the phones display are off by +6 hours and also the date for example today is Jan 02 :) What can I do to modify this? __________________________________________________________________ Anton Krall

Guys. How do those callcenter apps work with Asterisk where a call comes in and * send a URL and some screen popup up based on callerid or something or username or id and shows all the customers info? Anybody done that? What do you need to do that? If you are using ATAs or IP Phones, how do those integrate with the PC so the screen would popup?

Guys.. I just noticed that my grandstream handytone 286 ata are having problems with voice cutoffs... We can listen to the person on the zap channel (x100p cards) without problems but they sometimes listen to us with cutoffs.. like "He ...lo..... ow...r.. you" and it comes and goes.. this doesnt happen all the time but often enough. Any ideas what might be happening or what do I need

Hi, I was just wondering if anyone knows how predictive dialers detect voicemail and answering machines, and if they could explain to me how that works. Thanks! Brian.

I am also very interested in CRM integration. Anything I can do to help? One thing I don't understand is how is the browser being launched on the person's PC. Or is it not launched automatically? Anyone know of a simple app running on the desktop to do this? I looked into IPSwithcBoard and it appears like it should be able to do the job, but: a) It is pretty heavy - it does a lot of

Guys.. Ive noticed that I have 2 mpg123 processes running, is that ok? also... can you make MOH random? Also, I dont know if there is a problem with my config but when listening to MOH, every 3 or so second I get a click sound which notices because music gets a hickup every 3 or so seconds... is this ok? __________________________________________________________________ Anton Krall

Guys, anybody has a 1.2.x compatible app_voicemail patched for Spanish prompts that can handle for example, instead of saying "trabajo mensjes" would say "mensajes de trabajo o mensajes trabajo" (inverse)? Also can handle singular and plural (mensaje vs. mensajes)? Anton

Guys, Im new to asterisk and voip but Im have a couple of questions regarding the initial setup. 1. Im going to install an asterisk server at home, where I have 2 phone lines, what kind of card do I need to get? I was thinking about 2 X100P Cards, so 1 can have 2 FXO ports and regarding phones, what else do I need? Ive seen the Grandstream HandyTone HT-286, I guess that servers as and FXS devide

Uys, Steve Underwood I just got a Sangoma A101 card and Im using unicall 0.0.3.pre9 for R2MFC, I get the far and local end unblocked but as soon as I try to make a call I get dialing and then protocol failure.. Do you guys know if there are any issues with sangoma and unicall? Anybody has an a101 card working with unicall and r2mfc? Are you out there Steve? :)

Hello All, Wondering what sort of real world mileage people are getting out of different internet connecions - i.e. different DSL connection speeds, cable modems, etc... Is it reasonable to hope to carry 10 - 15 concurrent calls on a 768K DSL? I'm not talking about theoretical BW or looking for any difinitive absolute guarantee... With DSL and Cable - there is no guarantee, so I'm

Guys I just read on the wiki: "2005-07-19 - long awaited extension lights (hint priority) and call pickup on various phones work with newly released asterisk patch digium bugtracker - feel free to test and report findings to the bugtracker to have this commited to cvs." How does this work? And will it work only on certain phones or can it work with the gxp2000?

I just downloaded the zip file from grandstreams website to upgrade my gxp2000 firmware from 1.0.0.3 to the latest but seems there are some files missing on the zip file... Anybody been able to upgrade their firmware? My website shows this files as missing: 201.133.125.152 - - [11/May/2005:16:47:16 -0500] "GET /firmware/ring1.bin HTTP/1.0" 200 12737 "-" "Grandstream

Guys, anybody has seen or is using some kind of softphone on any square screen device with WM5? Ive tried sjlabs one and xten for pocket pc and they do work on Wm5 but they are designed for standard screens, anybody using anything on square ones?

Guys. Anybody doing ChanisAvail on IAX2 channels? Im trying to do this: exten => s,7,ChanIsAvail(IAX2/anton:intrudercom@armando-gw) But I get that the chan is unavailable eventhough I can make calls to that channel. Is there any chatch? The channels is defined as peer and Ialso tried doing a register on iax.conf for that channel. Everything is registering ok and I CAN make the call. Any

Guys, what does hint do in a dialplan and how do you use it?

I just installed the latest cvs head and seems a lot of commands haven been depricated. Where can I see the changes on all cvs head versions in order to keep up with the changes needed on my side. I checked the wiki and it still shows all the old commands and no mentions about the changes.

Guys, is anybody using PHP sockets to connect to the Manager and send command like "show voicemail users" for example or any other? My question is, how to parse the return info in a way that can be shown back to the user via web (discard all the manager responses not needed)? Thx in advance for any help you can provide.