similar to: VoIP guide for business people

All, I've posted a new tip on the Integrics website. It's on how ISPs can offer VoIP service to their customers, and why it makes good business sense to do so. http://integrics.com/tips/voip_for_isps/ Older tips can be found at: http://integrics.com/tips/ -- Alistair Cunningham, Integrics Ltd, Telephony, Database, Unix consulting worldwide +44 (0)7870 699 479 http://integrics.com/

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

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

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

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

All, I've put a new Integrics Tip. This one is on how to go about recording voice prompts for your IVR. It's available at: http://integrics.com/tips/recording_prompts/ -- Alistair Cunningham, Integrics Ltd, Telephony, Database, Unix consulting worldwide +44 (0)7870 699 479 http://integrics.com/

We use SER to front several Asterisk systems. Phones register on SER, which also acts as a load balancing and failover proxy for the Asterisks. Phone account details are held in MySQL, which Asterisk could access but does not currently do so. At present, call routing is done on the Asterisks using a FastAGI program which does the database access. We've been asked to implement busy lamps.

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

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

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

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

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

I have an AGI program with an array containing a set of ${UNIQUEID} variables for channels that may be active on the system. I need a way for the program to tell if they are or not. It's certainly possible using the manager interface, or appropriate "asterisk -rx" commands, but I'd prefer to do it directly from AGI for performance, security, and ease of configuration. Does

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

We've been asked to add H.264 and and AAC codecs pass through support to Asterisk. Looking at the latest 1.2 SVN branch, I see H.264 has already been added. Does anyone have experience of using it? Any problems encountered? Would anyone have a how-to guide (or just hints) on adding a new codec such as AAC for pass through only? -- Alistair Cunningham, Integrics Ltd, +44 20 799 39 799

Wrong list. asterisk-dev is for changing the C source code of Asterisk. I don't think AGI's "count" or are considered for inclusion into the subversion repository as stated by one of your conditions for payment. On Wed, 29 Apr 2009, Alistair Cunningham wrote: > I'd like to offer a bounty for a feature for Asterisk where an AGI > program can park and retrieve calls

Hello, Voisonics is hiring a VoIP support engineer to assist our customers running Asterisk based hosted PBX platforms. This is a part-time contract work-from-home position. For communication reasons we're looking for someone in a timezone encompassing Far East Asia, Australia, New Zealand, Canada, the USA, and Mexico. If you are not physically located in that area please do not apply -

Hello, Voisonics is hiring a VoIP support engineer to assist our customers running Asterisk based hosted PBX platforms. This is a part-time contract work-from-home position. For communication reasons we're looking for someone in a timezone encompassing New Zealand, Canada, the USA, and Mexico. If you are not physically located in that area please do not apply - being "flexible"

Does Asterisk have the capability to re-route calls that have already been connected? By this, I mean: 1. A call comes in to Asterisk. 2. It is routed to an extension as normal. 3. This extension answers, and the conversation starts. 4. After a few minutes, a plugin that I am writing decides that it wants to transfer the call to somewhere else. 5. It signals this to the core of Asterisk (this