[Asterisk-Users] New to IP-PBX

[Jay Milk]

If you don't do any transcoding, and turn canreinvite=on for your
sip-clients, there shouldn't be a reason why you couldn't run hundreds
or thousands of extensions on a Celeron 500.  Once you get into
transcoding (or you turn canreinvite=off in order to allow for recording
of conversations), processor speed matters.  AFAIK, the #1 reason for
recommending POTS over SIP is that in an all-IP system, you'll need a
timing source, and that can be tricky on some systems.

> -----Original Message-----
> From: James Richards [mailto:jimmy at kissyfish.org] 
> Sent: Friday, July 30, 2004 4:18 PM
> To: asterisk-users at lists.digium.com
> Subject: Re: [Asterisk-Users] New to IP-PBX
> 
> 
> I have been seeing reccomendations for using asterisk as a 
> soft-pbx with the reccomendation being to use regular analog 
> phones via FXS rather than SIP.
> 
> Is this still a big issue? Or is this a left-over from 
> previous bad experiences?  I have been doing demos with SIP 
> phones, and some IAXYs to whet their apetites, and people are 
> really biting at the feature set I can provide, and I have 
> run into no problems yet,  but I would love to know at what 
> threshold of SIP phones does the system start to have problems.
> 
>   The assumption in my scenario is a quality ASUS 
> motherboard, running RedHat/Debian, 512 MB RAM 10/100 
> Ethernet, P4 2.4 Ghz processor.
> 
>   I am trying to hit the small office market, with up to 20 
> SIP phones, and up to 8 POTS lines. (These have been my 
> current limits until I see the system inaction a bit more)
> 
>   Is the problem in using dissimilar SIP phones with 
> different codecs? Thus burdening the processor with 
> conversion on top of all of the other work it is doing?
> 
> PS, I am having a whale of a time with this software,  and I 
> appreciate the helpfullness of members of the community...
> 
> Jim Richards
> Kissyfish
> 
> On Fri, 2004-07-30 at 15:31, Nicolas Gudino wrote:
> > Hello,
> > 
> > On Fri, 2004-07-30 at 15:39, Duraid Abbas wrote:
> > > I have a D/41JCT-LS Dialogic board and I want to use it as an 
> > > IP-PBX. I'm new to IP Telephony and telephony and general and I 
> > > researched a lot but still confused about what I really need.
> > > 
> > > I know that I can setup an IP-Telephony for my LAN using a SIP 
> > > server and SIP compatible software phones. But the 
> challenge is how 
> > > can I connect to the PSTN so that I can send and receive calls?
> > 
> > Asterisk will do a wonderfull job as a soft PBX, but my 
> advice is to 
> > use hardware from Digium to connet to the PSTN (FXO or 
> T1/E1) and to 
> > connect regular analog phones (FXS or T1/E1+ChannelBank):
> > 
> > http://www.digium.com/index.php?menu=hardware_products
> > 
> > Before purchasing hardware, you can try to set up Asterisk 
> just with 
> > SIP softphones and get it to know the platform. Once you are 
> > comfortable you can jump on buying some hardware.
> > 
> > If you do not have time to investigate yourself search for 
> "Asterisk 
> > consultants" on http://www.voip-info.org
> > 
> > Best regards,
> 
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