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

[Bogdan-Andrei Iancu]

Hi Yehavi,

Please see my inline comments:

Yehavi Bourvine wrote:
> 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).
Actual this is the natural way of doing. You have two pieces of 
software, with different purposes, but complementary in the same time.
   Asterisk is an IPPBX handling media and implementing a lot of nice 
class5 features - and an PBX is not more large numbers of lines
   OpenSIPS is an softswitch, no media, limited class 5 features, but 
nice routing  and able to handle hundreds of thousands of line and 
subscribers

See: http://www.opensips.org/index.php?n=Resources.DocsTutLoadbalancing
> 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
yes, there is such a module (thanks to the combination of dialog and 
presence features)
> - 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 :-)
Well, a commercial solution does not exclude open source - there are lot 
of companies offering commercial products based on OSS.  So, you can get 
a good price (no licenses) and you can still sleep well :).

James Body from Truphone was asked (during a VON when he had an OpenSER 
talk) if using OSS is cheaper - the answer was no, it is not, but the 
difference comes in what you get in the end - instead of paying for 70% 
licence code and 30% for customization, with OSS you can pay 100% for 
customization/tunings. So, at the end you get exactly what you want and 
not a compromise solution (cost versus requirements).
>  
> 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,
Bogdan
>  
>                         Regards, __Yehavi:
>
> 2009/3/17 Vincent Li <vincent.mc.li <http://vincent.mc.li/>@gmail.com 
> <http://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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