--- Uriel Carrasquilla <[EMAIL PROTECTED]> wrote: > John: > are you aware of any documentation on how to configre SER to be a > front-end > to Asterisk? > I suspect it is very inexpensive to put a SER server in a hosting > facility
I think the cost is about the same as for putting a web server at a hosting facility. But I don't think you need high bandwidth. SER simply sets up the call. I don't think the audio data actually goes through SER. It goes directly between the two end points. This is the big problem with using Asterisk for SIP. With Asterisk the audio data between two SIP extensions has to actualy go into then out of the Asterisk box. This does not scale well to thousands of users like in a university campus or a comercial SIP service. > to forward traffic to multiple Asterisks based on Least Cost Routing. > My problem is that my experience is with Asterisk and not with SER. > Uriel > > -----Original Message----- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] Behalf Of John Todd > Sent: Monday, October 13, 2003 8:11 PM > To: [EMAIL PROTECTED] > Subject: Re: [Asterisk-Users] NAT, SIP (was: No sound with SIP Phones > on > the Internet) > > > >I'm curently looking into using SER to front end SIP calls for > >Asterisk. > >Basicaly all SIP users would register with SER not Asterisk and then > >Asterisk and SER exchange registrations. > > > >SER is a very capable SIP router, much more sophisticated than > Asterisk > >as it can look inside packets and route based on what it finds or > even > >re-write packets based on user specified logic. > > > >SER is GPL'd and has very good user documentation. Don't know how > well > >the above will work. The claim by the authors or SER that it can > >handle thousands of calls per second is quite impressive > > > >One other nice feature is that SER users can set up their own SIP > >accounts using a web interface and not needing to edit *.conf > files. > > > >See here for details http://www.iptel.org/ser/ > > > > > >===== > >Chris Albertson > > Home: 310-376-1029 [EMAIL PROTECTED] > > Cell: 310-990-7550 > > Office: 310-336-5189 [EMAIL PROTECTED] > > KG6OMK > > SER is an excellent option as a front end to Asterisk. It is a > "true" SIP proxy, whereas Asterisk is a hybrid, and SIP has not been > the primary focus of Asterisk development. In fact, Asterisk's SIP > implementation is very limited (though it is extremely pragmatic.) > > However, moving to SER does not solve any of the issues about the > proxy being behind a NAT, and I believe that SER will have the same > problems (though I could be wrong on this; I haven't experimented > with SER's ability to work from behind a NAT.) SIP clients work > well enough behind NAT (most of them, anyway) but the servers are a > different story. > > I really like SER's third-party addons for account administration; > Asterisk is significantly more complex, and probably would not be as > easily converted to such a front end. In fact, SER has a very > complex routing/scripting language that is not easily administered > with a web front end, so I think that SER and Asterisk suffer from > the same problems. If someone were to come up with a simple way to > administer voicemail.conf and sip.conf from a web tool, that would go > far to making Asterisk a bit more user-accessible... > > JT > _______________________________________________ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > > > _______________________________________________ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users ===== Chris Albertson Home: 310-376-1029 [EMAIL PROTECTED] Cell: 310-990-7550 Office: 310-336-5189 [EMAIL PROTECTED] KG6OMK __________________________________ Do you Yahoo!? The New Yahoo! Shopping - with improved product search http://shopping.yahoo.com _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
