This may sound stupid but could you not off load the 4 pri max limit off to a voice router. I know it's not the cheapest way to go since dsp for a hdv-2mft (cisco card) can run you a few K but would make it so your limited by the Voice router limit so you get higher number of pri to a * box? If so could you not stack a bunch of these Voice routers together in * so that if X number of call goto Voice router 1 then spill over to Voice router 2 and so on. I know 1750 cisco router with 2 mft and dsp are a lot cheaper then buying a big box router with hundreds of DSP's.
I am still learning * but I am planning on replacing a CCM solution with * as CCM is starting to get to buggy and costly for simple addons like monitoring and multi phone hunt groups that ring all of the phone at once like a 2or 4 wire PBX system. We can no longer spend 50k -100k for such a simple option. I just hope I can learn * well enough to replace the CCM's sccp with sip w/o losing to much fuctionality. My 2c Doug -----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Steve Underwood Sent: Thursday, May 27, 2004 8:09 PM To: [EMAIL PROTECTED] Subject: Re: dialogic was RE: [Asterisk-Users] "Glare" condition - How well does asteriskhandle? Scott Stingel wrote: >Hi Steve- > >Just briefly: > >I was mentioning the old days to illustrate what an even low clock rate >DSP can do. More recently (2000-2001), using D/600's we were able to >drive a large number of channels (8-12 E1's) for IVR. > > Ah, the D/600 - damned big heat sink, lots of heat, and lots of cooling troubles. I remember them well :-) Actually, we are still using quite a few. >All I'm trying to do is to illustrate both the beauty and the >limitations of taking the processor horsepower off the line interface >card and doing everything in the central processor. The Digium boards >are MUCH less expensive than the Dialogic boards (about 15% the cost, >if that, per channel), but are not a plug-in replacement. I've hit a >real-world limit, in my less-common environment, of about 4 E1's per >chassis. I believe this limitation is not so much in the bit-rate i/o, >but the PRI call setup overhead. I have communicated with a number of >other asterisk developers who have experienced this limitation, again in a high-volume IVR >environment. I have demonstrated it to Mark as well... > >To get back to the original subject a bit, Dialogic developed an >elegant API called Global Call, which maybe we can use, or at least >learn something from. > >I'll let you have the last word if you like, Steve.... <s> > >Cheers >Scott > > If the PRI setup is loading things that much there must be something wrong. That should be a very lightweight activity. It sounds like that bears investigation. Actually, with host based DSP short PRI calls should be a lighter load than long ones. The trunks spend rather more time not sending any audio, than with fewer longer calls. Global Call is a mix of elegance and botchups. I actually have private code that implements something I call UniCall for the Digium cards, and a chan_unicall channel driver that works with it. This tries to be a lot like Globall Call, but without the botchups. I generally agree that an abstraction layer like that is a good thing. I'm not trying to get the last word. I am trying to illuminate and improve things. If there are serious limitations they need kicking hard, not accepting. :-) There is no sound reason why things should be limited in the way you find. If you are doing call switching the Dialogic cards will win hands down on throughput. The audio never hits the CPU, and the limit is based on the number of call setup/cleardowns rather than active channels. For simple IVR it is certainly not the heavy DSP load on the host CPU that is the limiting factor, and the Digium approach should do much better than you are finding it actually does. Heck, the Dialogic cards still don't bus master. The bus mastering on a TE405P/TE410P should give that a substantial win. Regards, Steve _______________________________________________ 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 _______________________________________________ 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
