Hi All, Our experience with * to date has been a bit limited. It's a 4xCisco 7960 network, linking our head office with a faraday caged datacenter. As a way of putting voicecomms into a sealed room, it was quick and easy to deploy, and works very well. As typically happens, we've now thought about extending the use of asterisk - and a new opportunity has cropped up. In three months time, we have a couple of new small offices coming on stream. Prices quoted for POTS comms have been expensive, so we've decided to look at providing an Asterisk/VOIP solution for the 30 or so users, and just having a couple of local analogue lines for fax and emergency use. The aim of the asterisk solution is to provide both external POTS connectivity and the ability to make internal calls to our existing head office network. (350 extension Mitel SX2000). In theory we should be able to assign a head-office DDI number for each desk in the new offices, and route calls transparently via the asterisk server to the VOIP extension. Current plan is that we buy a digium TE110P Card, and crossover-connect it to the Mitel PABX as a secondary exchange. Asterisk will have no direct PSTN connection, but will route all non VOIP to VOIP calls via the Mitel. Unfortunately I can't find any info on the * Wiki or in the list archives about how to go about configuring this combination. Mitel have confirmed that the E1/EuroISDN option should connect to the PABX, and that 'QSIG can be enabled on the exchange' but have stopped short of saying that the two will talk to each other. 1. Does anybody have any experience of trying to get * talking to a Mitel SX2000 Lite? 2. Is there a definitive list of what asterisks implementation of QSIG supports, and what it doesn't? Is the current support level likely to be sufficient for the above, or do we need to look at some alternative method/protocol? 3. What pitfalls do we need to look out for when implementing the above, over and above the usual datacomms latency/capacity issues? 4. Coming from a datacomms/systems background, I find all this talk of channel banks, spans, signalling protocols, TDM and so on a bit of a foreign language. Does anyone know of a good primer on the web somewhere that would help in getting up to speed with voicecomms terminology? Any help gratefully received, Chris Morgan UFI Limited -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20041210/a80c1abd/attachment.htm
Hi Chris I've done quite a bit of work with Mitel SX2000s and got to admit I like them a lot (give me those forms based admin screens over a Nortel command line interface any day!). I've not interfaced an asterisk box with an SX2000 before but it should be pretty easy (hehe, note my use of the word "should") ;-) I'd be tempted to set up a dialplan on the Mitel so that all VoIP extensions start with the same digit - it'll make it less of a routing headache in the long run. You can then subdivide on the asterisk box as needed.. For example, on the Mitel, set 4xxx to go to the asterisk PRI. Then on the Asterisk box you could have 40xx be assigned to users at site A, 41xx be users at site B etc.. The sites can be SIP devices registered to the asterisk box, or even an asterisk server of their own if it warranted it. There's a bit of work to do with the asterisk dialplan and also the Mitel, handling COS and barring/restrictions etc, but it should be fairly simple to set up. Give me a shout if you want to discuss further. Paul
John Mylchreest
2005-Sep-20 05:51 UTC
[Asterisk-Users] Integrating * with Mitel SX2000 Lite
Hello there. I have noticed you were trying to place an asterisk box infront of an SX2000. I am trying to do this also, but no matter what I try the connection to the SX2000 causes a major alarm. Did you get this working? If so could I please see what signalling you used to achieve this? Thanks, John