similar to: Hints and busy lamps for phones registered on SER

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'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 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

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

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

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 regularly get asked by business people, "What's the point of VoIP?", so I put together a guide: http://integrics.com/tips/voip_for_business/ I'd be interested in hearing your feedback, and ideas for expansion. -- Alistair Cunningham, Integrics Ltd, Telephony, database, Unix consulting worldwide +44 (0)7870 699 479 http://integrics.com/

I am so far unable to get the busy lamps on a Snom 220 to work either with current cvs or asterisk 1.0. I am using the hint extension and the Snom 220 just as described in the "mini-howto" on: http://www.mail-archive.com/asterisk-users@lists.digium.com/msg49781.html There are also a couple of wiki pages referencing this: http://www.voip-info.org/wiki-Asterisk+standard+extensions

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

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

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/

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/

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 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

Hello Happy New Year to all!! I've just completed porting from Asterisk 1.2 to 1.4. I did this by doing a clean install on a new box, and moving over configuration and scripts where needed. All went surprisingly well! Anyway, one lingering issue is that the function key "lamps" on our Snom phones have all stopped working! We're using a mix of Snom 290/320/360 phones and

Hi all A quick question about busy lamps I have lamps working 'sort-of' on my GXP2000 lamps flash with ringing and go solid red when call gets answered but stay green when a call is made from the extension. Setup is Ext 200, 201, 202, each monitor the other two when 200 calls 202 - the BLF on 200 and 201 flash red - when 202 answers both 200 and 201 show BLF for 200 as red but

Hi Pan & Dhaval, In the past 8 weeks, we have delivered a load-balanced asterisks (1.4) based call center with our flexqueue application for icson.com. It has the below features, 1. 2 x asterisk 1.4 boxes, 1 x mysql db box and 1 x flexqueue box(the two are failover configured with heartbeat and custom script, and mysql master-slave replication between two svr), 2 x kamailio boxes(failover

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'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