[asterisk-users] Asterisk is not designed for University with largeuser base?

[Wolfgang Pichler]

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.li at 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,
>         
>         
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