similar to: Opinions of Sphinx?

This is a request to the website manager for asterisk.org. The build scripts for our ITSP product include the URLs to download the Asterisk files, such as: wget "http://ftp.digium.com/pub/asterisk/asterisk-1.2.5.tar.gz" However, if a new version is released, asterisk-1.2.5.tar.gz is moved to the "old" directory. This breaks our scripts until we can update them and send

Integrics is pleased to announce the release of ITSP version 1.6. This version has the following new features: - Comes in 2 editions: * Carrier edition, for 250 to tens of thousands of users on hosted systems. Integrics sells this edition directly and through partners. * Office edition, for 10 to 250 users. This edition is sold only through our partners, for them to sell as PBX systems at

I've been considering doing a web based database system, where you can post your termination offerings or wanted, then search by location, price, minimum volumes, etc. I'd probably make it free, supported by advertising my consulting company, or Google Adwords, or something like that. I've got the design written down, all ready to start coding. I could probably have a prototype

I'm planning an Asterisk Voicemail system of around 3000 users spread across several sites, each site connected by a fast network to a central site. We're considering 2 models: - Central Voicemail with VoIP calls from remote sites (easier to administer the system(s)). - Voicemail server at each site with shared database and NFS server at the central site (easier to connect to the

We've been asked to quote for a large cluster running Asterisk and our ITSP in a box product. The system will be SIP throughout, with mixed codecs. We're considering using Dell blade servers, 1855 or similar, on the grounds that we normally use Dell machines and they work well, but we need higher rack density. Has anyone used these? Any feedback on whether they're

A feature we're often asked for in our ITSP product is to allow customers to upload their own music on hold, or to have it recorded for them by a recording studio with the latest news, weather, etc, punctuated by "Welcome to <customer>, please hold". Since there may be thousands or tens of thousands of customers, and perhaps 10% of customers may want this feature with a

Hello Anyone here uses Vitelity as voip provider ? Their pplans looks good but i need some feedback from existing customers if any here . -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070324/8bb0be73/attachment.htm

Hello! I need to: 1) call special number (or run special application) on mobile phone 2) establish connection between mobile phone and server 3) allow server to recognize spoken numbers (Polish language) and some other control words 4) let the server to say some short answers (prerecorded in mp3) according to some algorithm and recognized words 5) let the server to save little text file on its

Integrics is pleased to announce version 2.0 of Enswitch, the most integrated platform available for offering commercial telephony services such as ITSP, hosted PBX, calling cards, call shops, number translation services, and much more. Enswitch was formerly known as ITSP in a box, and Enswitch 2.0 is effectively the same product as ITSP 1.7. The product has been rebranded as, although it

I'm sure this is a stupid question, but I'm not finding an answer anywhere. Do I need a dedicated box to run asterisk, or can I put in my server (running Fedora) and leverage some of the free cpu cycles and disk space? Thanks, Dunc

Hi, Alistair Cunningham of Integrics was our guest yesterday. We talked about Integrics new product Geons, a suite of software for building large-scale distributed enterprise applications. The recorded session is now available here: http://www.voipusersconference.org/2010/geons/ The extremely rare John Todd was sighted (and heard) at this event. If you are developing a product or service

Hi all, I have seen the term E1 and PRI used interchangably when referring to a voice service with 30B channels and 1 D channel. Are they just different terms for the same thing or is there some technical difference. Even Newton's telco dictonary seemed a bit fuzzy on this topic. I have seen it said the PRi is a protocol that runs on top of E1. Is this true?

Hi, I'd like to terminate IAX call on PRI interface. What conditions should be met to be able to send arbitrary caller numbers to called party, so he can call back to supplied ISDN number (that is different for every IAX user) - not through PRI interface, but plain ISDN call !! Thanks in advance, regards, Rob.

Hi. I'm looking at ordering a 30-channel ISDN connection from telia (a swedish operator) and then using a Wildcard TE110P card with that and asterisk to do IVR. Can I be certain that the TE110P card will work with that ISDN connection? A 30 channel ISDN certainly sounds like an E1 connection, but I couldn't get any clear answers from the operator if it is. Has anyone used the TE110P

I have a customer who wants to do large volumes of H.323 to H.323 hairpinning. We haven't tested this scenario for large volumes before; maybe someone on asterisk-users has. If they buy a top of the line PC, how many concurrent calls are we likely to get? Routing logic will be simple, the machine won't be doing anything else, and let's assume no transcoding for now. We're not

I think I have 4 options. 1, Modify chan_sip.c to update a new field in sipusers realtime table with the status of the sip peer/user. Then use agi to dial sip calls. Check the status field if OK then dial the fullcontact from the sip table. If not goto voicemail or where ever else I want the call to go.. The UA would only register to one server, so only one server *should* be writing to the

I am working on a project on 3rd party call control for a call center, for which I think Asterisk may be useful. What I would like to do is: - Have a call come in to Asterisk. - Asterisk asks another machine, over a slow IP link, such as a modem, how it should route the call. Asterisk passes the called and calling numbers. - This other machine looks up the destination, based on called and

Hello, I just started with asterisk and I start to get it, but there is one thing that I don't seem to get: If I put an FXS-card into my asterisk server, then I can phone to the server with a normal phone, but can that phone also be reached by de server, so someone can caal back that phone? Or do I have to provide another connection? And if I connect two asterisk servers, can this be thone

Hi all, I'm currently looking for a VoIP platform to support the following features: Caller ID Call Waiting with caller ID Call Hold/Retrieve Three-way conference Calling Line Identity Presentation Call back last missed call Last called number redial User line locking/Call Barring (all current levels) Itemised bill Call Forward Call Forward on No Reply Call Forward on Busy Call Forward

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello everyone, As you know, we released Asterisk 1.0.4 earlier this week. However, there was a callerid bug in chan_zap that has caused us to go ahead and make another release. Asterisk 1.0.5 is available at all of the usual locations. I'm sorry for any inconvenience this may cause. Russell Bryant -----BEGIN PGP SIGNATURE----- Version: GnuPG