Hi, i do have a request for an installation with about 1800 sip extensions - as addon to a exisiting system - connected to it using qsig. The requirement here is also that the system should have SIP over TCP with TLS and SRTP (snom phones should get supported) I know there are patches out there to get this working - but have someone already used these patches with so large installations ?
About the database polling - i think for such a installation you could create something like a database to config files script - so not to use realtime. This should solve this problem. What do you people think about using freeswitch to handle the sip extensions - do BLF/SLA/SRTP/SIP over TCP TLS - and use asterisk for interconnection to the other PBX - doing the rest. Have anyone here already tried freeswitch in such a combination ? regards, Wolfgang Am Donnerstag, den 19.03.2009, 08:08 +0200 schrieb Yehavi Bourvine: > Hello, > > Sorry for the delay - was out of office. I also cross-posting it to > OpenSIPS list. > > I have a small pilot (20-30 phones) which also does some sort of SIP > to PRI transcode for our old PBX. The pilot is base on Asterisk and > mostly Polycom-501 phones. It works quite well, but I have a few > minor/missing issues: > - I have the RPID patch, and unattended transfers fails with it. > - No SLA, only BLF. I know there is SLA, but it is cumbersome to > deploy. > - Confference is limited to 3 participants. I guess I can do more with > external server but didn't > manage yet to make it working. > - No "busy dial again" which is required by our users. > > Now, to the original issue: I tried adding 1000 extensions to the SIP > database, and then use SIPP to send one REGISTER for each extension. > After doing so Asterisk still worked, but it was continously accessing > the database for all these extensions, just polling them. This raised > a red flag to me, and I decided to check the following config: > OpenSIPS/Kamailo/etc. as registrar and "SIP switch" for the phones, > while using Asterisk only for media related issues (which is the > common suggestion here). Now, I have new problems: > > - SLA works, but very "fragile". > - Not BLF, although I think it will be solve with the dialog handling > on OpenSIPS 1.5 > - Same confference and "busy dial" problem. > > Next week our management is going to decide (I hope...) how to > proceed: Do nothing (stay with the Nortel as we are tight on budget), > go to open source or to a commercial solution. > > Although a commercial solution allows me so sleep well at night, I am > going to recommend the open source direction. If accepted, then I will > continue the development and you'll hear me quite a lot here asking > hard questions :-) > > BTW, If I didn't say so far: we have around 8,000 extensions on 4 > Notel PBX'es, using around 10 PRI's to the world. > > Regards, __Yehavi: > > > 2009/3/17 Vincent Li <vincent.mc...@gmail.com> > > > On Tue, 17 Mar 2009, Yehavi Bourvine wrote: > > Hello' > > I am at the same situation as you. I also work at a > university and we have > over 8.000 extensions on a Nortel PBX. I also run a > small Asterisk pilot. > > I am using a realtime users database and the main > problem is that Aaterisk > does too mcuh database access to inquire for the > currently registered users. > (I am using direct RTP path between the phones so this > is not a limiting > issue here). > > I am checking now a combination of OpenSIPS and > Asterisk, where OpenSIPS > will serve the phones and Asterisk the more complicate > things (voicemail, > transcoding, etc.). OpenSIPS still lacks some of > Asterisk features, but they > are being worked on. > > Regards, __Yehavi: > > > > Hi Yehavi, > > Could you please keep us informed with your research, That > would be very interesting case that all other Universities > could study. There seems no known large Asterisk deployment in > University enviroment at this time. > > Regards, > > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users