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

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

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

Stefan Sayer


o Yehavi Bourvine [03/19/09 07:08]:
> - Confference is limited to 3 participants. I guess I can do more with
> external server but didn't
>   manage yet to make it working.
...
> - Same confference and "busy dial" problem.
for conference bridge you could give SEMS (http://iptel.org/sems) a try:
get http://ftp.iptel.org/pub/sems/sems-1.0.1.tar.gz; make ; make install
or on debian add deb http://ftp.iptel.org/pub/sems/debian etch free to
sources.list and apt-get install sems.

in /etc/sems/sems.conf set
sip_ip=a.b.c.d
sip_port=xxyy
media_ip=a.b.c.d
application=conference
load_plugins=wav;gsm;ilbc;adpcm;speex;l16;sipctrl;session_timer;conference

start sems (sems -f /etc/sems/sems.conf [-D 3 -E]) or /etc/init.d/sems start
and send your call to a.b.c.d:xxyy, INVITE ruri user is the room name.

if you want to have the caller enter the room number in the beginning, set
application=webconference
load_plugins=wav;gsm;ilbc;adpcm;speex;l16;sipctrl;session_timer;webconference

BR
Stefan Sayer
--
Stefan Sayer
VoIP Services

[hidden email]
www.iptego.com

IPTEGO GmbH
Wittenbergplatz 1
10789 Berlin
Germany

Amtsgericht Charlottenburg, HRB 101010
Geschaeftsfuehrer: Alexander Hoffmann

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

Bogdan-Andrei Iancu
In reply to this post by Yehavi Bourvine
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,
>
>
> ------------------------------------------------------------------------
>
> _______________________________________________
> Users mailing list
> [hidden email]
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>  


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

Yehavi Bourvine
In reply to this post by Yehavi Bourvine
 
Hello,
 
   After a long time we had a meeting with our university's management and got a green light to have a proof of concept with open source telephony. Now I have to select the right software to experiment with...
 
  Up to now I thought of going with OpenSER for the masses and Asterisk for voicemail and other media related things. However, from reading around it seems like FreeSwitch can give me the benefits of both packages. Anyone has an experience with it?
 
                                  Thanks, __Yehavi:

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

Alex Balashov
This isn't really the right mailing list for that question.

The answer, though, is, as always:  it depends.

Yehavi Bourvine wrote:

>      
>
> Hello,
>  
>    After a long time we had a meeting with our university's management
> and got a green light to have a proof of concept with open source
> telephony. Now I have to select the right software to experiment with...
>  
>   Up to now I thought of going with OpenSER for the masses and Asterisk
> for voicemail and other media related things. However, from reading
> around it seems like FreeSwitch can give me the benefits of both
> packages. Anyone has an experience with it?
>  
>                                   Thanks, __Yehavi:
>
>
> ------------------------------------------------------------------------
>
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Re: [asterisk-users] Asterisk is not designed for University with largeuser base?

Bogdan-Andrei Iancu
In reply to this post by Yehavi Bourvine
Hi Yehavi,

As Alex said, it depends of what exactly you want to implement.

You just have to evaluate your target service and to properly understand
what each piece of software is appropriate for and what it has to offer.

First of all, you have 2 complementary classes of software : you have
softswitches and and you have PBX - you have large capacity softswitches
for 100K subscribers with no media support (like OpenSER/OpenSIPS) and
you have PBX-like software with advanced and complex media capabilities.
(like Asterisk).

There is no sigle software to give the magic complete solution - so, far
the combination of the two types (opensips + asterisk) proved to be a
good solution to covers all needs and all requirements of a complex
solution.

But again, it is up to what you are looking for (as voip platform)

Regards,
Bogdan

Yehavi Bourvine wrote:

>
>      
>
> Hello,
>  
>    After a long time we had a meeting with our university's management
> and got a green light to have a proof of concept with open source
> telephony. Now I have to select the right software to experiment with...
>  
>   Up to now I thought of going with OpenSER for the masses and
> Asterisk for voicemail and other media related things. However, from
> reading around it seems like FreeSwitch can give me the benefits of
> both packages. Anyone has an experience with it?
>  
>                                   Thanks, __Yehavi:
> ------------------------------------------------------------------------
>
> _______________________________________________
> Users mailing list
> [hidden email]
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>  


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