similar to: WebCall application

Hello all, I need to conect an Asterisk with an Alcatel OmniPBX 4400 using an E1 port. It is the first time I make this kind of connection and I do not know exactly how to get it working. Someone has experience with this kind of connection? Could you paste a zapata.con and zaptel.conf files with QSIG configuration? Any clue will be wellcomed. Thanks Voipcrazy -------------- next part

hello is any web based sip phone? for example: a user after logining in, view a configured sip phone, and ...... best MAni ____________________________________________________________________________________ Finding fabulous fares is fun. Let Yahoo! FareChase search your favorite travel sites to find flight and hotel bargains. http://farechase.yahoo.com/promo-generic-14795097

Anybody uses Asterisk Java to register an extension to an Asterisk server ??? Is there any solution for this ? Phan Anh Vu DT12.K49.HUT RDLab ( C9.410 ) HUT --------------------------------- Be a better Globetrotter. Get better travel answers from someone who knows. Yahoo! Answers - Check it out. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Does anyone know if you can use softphones on thin clients? I have a new customer that wants to use Eyebeam (about 10 users) on a thin client platform. Each user has a little box on their desk that has a USB port, mic and headphone jacks and monitor. I am worried about conflicts when running 10 softphones on the same server since they will all try to use por 5060. -- Telecomunicaciones

Hi I''d like to create a webcal link to share a calendar with iCal like it''s done in Basecamp for example. I know how to create the file using icalendar ruby library and iCal can open it, but if I send this file using an url like webcal:// localhost:3000/ical/1151495837 iCal says that data are not valid. Any ideas? Thanks. PS here is my rb code def ical_data cal

Hi, I am looking for a very low cost way of receiving and sending T38 fax reliably. Is there any possible solution using Asterisk as the PSTN SIP gateay and Digium E1/T1 card? Is there other open source package that can help to accomplish this purpose? Regards, Mark -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi guys! We have a call centre that has been moved across from an old Ericsson MD110 PABX to an Asterisk server with those in the call centre using X-Lite as their softphone. I'm trying to get Agent presence configured so that X-Lite gives the operators a visual indicator of their status - logged on, off and on "pause". I'm using chan_agent for the agents, so agents are

I have a DID from IPKall.com which is forwarded to my asterisk box. Then this extension should call my ip phone using Dial application. Everything works fine, except when I pickup the phone, I can talk, the other party can hear me, but I cannot hear anything the person says on the ip phone. Then after a couple of seconds, the call hangs up. I don't know why. Here is the message I get:

Yell at me if you will, but I hate func_realtime - it's not very usable nor is it change-friendly (update your database and your dialplan completely breaks). I'm getting a new 1.6 box built out and working, and wanted to emulate the functionality of APP_realtime somehow, so I started digging around in the func_realtime source - here's what I came up with: For 1.6.0, look at line 86

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

This is a bit off topic, because I 'think' it isn't an Asterisk problem. However I'm not sure and anyhow I'm hoping someone may recognize the symptom. We moved offices a month ago. Our trusty SNOM190s (all between 3 and 5 years old) were packed up for the move, then unpacked a couple of weeks later. On unpacking them and connecting them to the new network, several of

hi, there is T.38 fax gateway for asterisk http://bugs.digium.com/view.php?id=12931 please test it and report bugs for people from http://www.voip-info.org/wiki-Asterisk+T.38+Bounty if you still want donate t.38 development please contact me at cervajs at fpf.slu.cz --------------------------------------- Marek Cervenka =======================================

Hi guys! I'm having one or two issues with queues hosted by an Asterisk machine where the clients are on a legacy PABX - at least for the interim. I fully expect most of these issues to be non-resolvable, but thought I'd at least ask to find out if there is some way of working around the issues. The legacy PABX is an NEC 7400 ICS connected to Asterisk via an E1 ISDN link. Calls are

Hello users, Can you share with me your experiences with Asterisk 1.6? Is it stable enough for commercial service? Thanks. > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080901/d849d412/attachment.htm

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi guys, I've been working on a little dialplan fragment for roaming extensions, however the customer wants us to set the MWI indicator for the roaming extension that has just logged in. We're using MySQL realtime, so I've figured out that RealTimeUpdate will happily update the realtime database with the correct mailbox. My problem

I've hit an odd issue in a test 1.8 deployment, playback() stalls mid file. The call stays up, but asterisk stops sending packets. It doesn't always happen - but on demo-congrats it happens about half the time. It only happens in IAX calls. Anyone else experienced it ? (I filed an issue just in case it isn't just me) T. Tim Panton - Web/VoIP consultant and implementor

Good day Im look at http://www.voip-info.org/wiki-Asterisk+GUI And I see there are a few GUI for asterisk What do you guys prefer? What is the best and simplest? Id like something that give me access to backend for a little bit of customization Thanks for you help and time -------------- next part -------------- An HTML attachment was scrubbed... URL:

I'm looking to connect a BMC 450 to an asterisk with a Digium Quad E1 card. Am I right in thinking that I'll need a special 'crossover-E1' RJ45 cable? If so, any clues where I might buy one in the UK? The Digium card sellers don't seem to stock such a thing. Thanks. Tim. Tim Panton - Web/VoIP consultant and implementor www.westhawk.co.uk

Does anyone know of an IAX softphone in Java, whether applet or application? Even the most minimum featureset, just voice and dialing, or even embedded in some other app/let. Preferably GPL. Thanks. -- (C) Matthew Rubenstein

Hi, I've noticed that sometimes instead of getting a regular ring tone when calling out on a Zap channel, I get this obnoxious loud noise which forces me to hang up. Is this a problem in the Zaptel driver? I seem to recall that ringback tones are generated by zaptel when dialing out from a SIP phone over a Zap trunk. Thanks.