SER + Astrersik can handle UNLIMITED simultaneous callas, but for that you have to have unlimited servers working as a cluster :) I would never recommend to have asterisk and ser on one machnine, asterisk that will work on separate machine with P4 3.X/Xeon terminationg calls to PSTN can handle up to 100 calls easily. SER with rtpproxy on one server can handle REGISTRATIONS from 10000 clients. Even if they will do it every 30 sec (depend of course how u gonna authenticate them) RTPPROXY can handle much more then 100 calls (what codec btw ?) plus you can distribute SER to different machines to work in a cluster OR you can use mediaproxy or nathelper in same cluster mode. Btw, I believe you not gonna let all calls trough asterisk ? asterisk is only for termination to PSTN ? otherwise forget what I wrote :) On 2/25/2005, "alexey(a)home.kovyrin.net" <alexey(a)home.kovyrin.net> wrote: _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

In this case Asterisk will be bottle neck of whole system.. 50-100 calls because of g729 But why do you need SER in this case? that gonna be extremely unscallable system (not because of SER but because of asterisk). You can use simple asterisk to get same functionality. In my system, SER is registrar and NAT traversal (you need it to do SIP-SIP calls). Then I have b2bua to do CDR and digit manipulation and then it goes to termination carrier Asterisk as b2bua is kind of expensive in my opinion. Not so stable and take a lot of resources. Whenever I send call thru Asterisk I notice that overall quality go down. May be it just because I do not like asterisk :) If you in your setup will do same as in mine but instead of B2BUA use Asterisk, then it will be scalable quite nice by just increasing amount of asterisk servers -----Original Message----- From: Mik Cheez [mailto:michael_bulk@wildgate.com] Sent: Friday, February 25, 2005 11:55 AM To: Vitaly Nikolaev; serusers(a)lists.iptel.org Subject: Re: [Serusers] SER+Asterisk on high loads How about the following scenario: One box acting as a proxy to a PSTN. Asterisk is handed the initial request (port 5060) in order to do some modifying of the packet (callID), and to keep a stateful connection for CDR's. Asterisk passes it to SER (port 5061 on the same box) which uses its logic to route the call to one of several PSTN's. All of this is done in g729. Why do it this way? SER cannot modify the callID without having issues. I don't trust SER to do the CDR, and logging on the endpoint is very difficult compared to Asterisk. Any idea what the load maximum would be with that same P4 3.X/Xeon? Best regards Vitaly Nikolaev wrote: clients. can

Nothing to comment regarding digit manipulation and so on.. btw good solution is to use something like nexton softswithc :)) Regarding tracking: Ngrep for quick stuff Ethereal for.. I never used it in production, too slow Etherealdump + SIP Scenario - very nice but kind of slow... it very good for showing off (presentation, communication with UAs developers, etc) Hammer Call Analyzer - extremely nice software but extremely expensive like 15k per license but it can show you calls with weird path, show you media jitter/delay, play it, show faxes everything you might dreamed of :).. They have 10 days demo May be somebody knows something better? -----Original Message----- From: Iqbal [mailto:iqbal@gigo.co.uk] Sent: Friday, February 25, 2005 12:39 PM To: vitaly(a)voipsonic.com; michael_bulk(a)wildgate.com; serusers(a)lists.iptel.org Subject: RE: [Serusers] SER+Asterisk on high loads I would agree, at one point I was taking calls on asterisk etc, but its the wrong way of doing it, get ser to do all the donkey work, and then use asterisk or whatever b2bua to take care of billing, if like me you need voicemail etc, then just use asterisk to do both function.. Using ser you can redirect to various backend asterisk boxes to help with scaling, and in the middle also take care of the mediastream, and natting issues. As for the pstn , just either handoff from ser, or you could habd off from asterisk (waste), if u want toi manipulate callerid, let me know how also, cause I am having the same problem, ser really cant do it, since it does not maintain state, hence I am thiking of having a special route for calls who need to have CLID changed, route them to asterisk get them changed and then forward them to a SIP device...this sounds a bit crazy but I cant see how else to do it Now if someone can just tell me whats the easiest way to debug calls, I have logs logs and more logs, but it never makes sense...well not easily because of the multiple invites etc, tracing one call path...there must be a way Iqbal On 2/25/2005, "Vitaly Nikolaev" <vitaly(a)voipsonic.com> wrote: take overall