Hi all, I regret that I don't know much about telephony as I'm a networking bod, but here goes... We are thinking about implementing a VoIP service so that staff and students can make VoIP calls from home or using our wireless LAN on campus. Clearly, we would like it to integrate with our PBX so VoIP users can talk to the PSTN as well. We don't actually control the University's PBX, so it is highly desirable that any changes to the PBX are very minimal! We were thinking of setting up a single extension which is associated with a huntgroup on the PBX that somehow connects (via an E1?) to the Asterisk box. If a user leaves his office he sets his extension to re-direct calls to that extension number (or to re-direct on no-reply). The PBX would route the call to the asterisk huntgroup. Asterisk would "see" the original extension number called (assuming this is possible!!), and (knowing which extension maps to which user to which IP address) route the call to the user over the IP network. I can handle the IP stuff without any problems. My question is: is this a possible/sensible approach to implementing this type of service? If not, what's a better solution? TIA for any suggestions/comments, josh. -- ----------------------------------------------------------- Josh Howlett, Networking & Digital Communications, Information Systems & Computing, University of Bristol, U.K. 'phone: 0117 928 7850 email: josh.howlett@bris.ac.uk ------------------------------------------------------------
> I regret that I don't know much about telephony as I'm a networking bod, > but here goes... > > We are thinking about implementing a VoIP service so that staff and > students can make VoIP calls from home or using our wireless LAN on > campus. > > Clearly, we would like it to integrate with our PBX so VoIP users can > talk to the PSTN as well. > > We don't actually control the University's PBX, so it is highly > desirable that any changes to the PBX are very minimal!Hi, we're currently trying to do something like that here at Saarland University. We will give each VoIP-User his own extension that will be routed to Astersik over an E1 (We will use extensions like 69XXX for that). I think this routing can be done easily in the PBX. We will also set up a SIP Express Router and give an account to every student here. Students will eventually also get their own extension that can be reached from everywhere (something like 70XXXXX). The students will be able to call every phone here on campus over the Asterisk-System (including other students of course :) ). This is mainly to prove that a simple cheap VoIP system can handle that traffic. In a second step we will try to replace the existing PBX with Asterisk and SIP phones or channel banks. This will of course take some time. :) Another goal of the project is to route calls to other universities over IP if they support it. You can find some information about our project on our Website (see below). -- Maik Schmitt http://graphics.cs.uni-sb.de/VoIP -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 232 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20030709/285a2ecc/attachment.pgp