similar to: Asterisk in India?

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

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'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

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.

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

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

Hi all, Can someone point me to some information on the type of hardware that might / should be used for a high load on an asterisk machine ? I know that this is dependant on what services you plan to have running, and it's relevant to what you plan to do. We are likely to be running 4 E1's, Voicemail, IVR menus, Music on Hold, Pay-Over-The-Phone, lots of interdepartmental

Hi All. I am beginning a project of Call center and predictive diales, my call center have 50 operators, I have 50 analog phone line with the company PTT in my country. I have the following questions: 1- Can I to work this project with Asterisk? 2- What caracteristic of hardware need for my servers? 3- For 50 analog phone line what tipe of card digium I need? Thanks in advanced, Regards.

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

I'm planning a system of 120 SIP or PRI channels using speech recognition (fixed grammar of 500 words) menus. I could use a Cisco router and VoiceXML, but would prefer not to on cost grounds. Has anyone tried Asterisk and Sphinx (bonus points if in a production environment)? If so, what's your opinion on quality of recognition, stability, resource usage, etc? Anyone have any

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

Hi all! We use a PHP-portal for management of our projects & contacts. Now I would like to make it possible to dial contacts directly from the portal. Since users have to log in, I can use that to determine which office phone the call should originate from. And the number-to-be-dialed is of course also listed. How do I commence here? I'm pretty sure others have done this already, so

Hello list readers, I've been trying out * for a week, setting up three SIP softphones with SJPhone, calling from each other, conferencing, and it seems to be what I was looking for, but I have yet one more question that I haven't been able to find out by myself, is it possible to create conferences in Asterisk with a (some) listener(s), that is, people who can only listen to the

I have some simple questions and i need your help guys. I have ser server which working fine, between users. I am trying to add some more features to the ser. Most important is the IVR. I installed Asterisk and i am trying to register user in asterisk with no success. Part of ser.cfg file where i am trying to redirect the call to the asterisk.

Hi all, I have Asterisk talking to my call manager 4.0 with SIP trunk as mentioned in the wiki. I also have SER talking to Asterisk. I need the SER talking to my Call manager. The reason why CCM cannot talk to SER is because SER is a on a public ip address, and CCM is on a private ip address. The asterisk how ever has 2 nics, which talks to both and external. Is it possible to allow

Hey guys, I'm looking for a programming or Development Team/Company to do some custom coding for Asterisk. What we need is not exactly simple. In fact, I'm not sure the extent of the coding as far as technical terms go at all. Currently we have a "call center" with 4 phones. There will be a total of 8 people using the phones. Obviously, no more than 4 people will use