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:[EMAIL PROTECTED] > Sent: Friday, July 30, 2004 4:18 PM > To: [EMAIL PROTECTED] > 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, > > _______________________________________________ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/aster> isk-users > To > UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
