On Monday 13 October 2003 22:26, Uriel Carrasquilla wrote: > John: > are you aware of any documentation on how to configre SER to be a front-end > to Asterisk? Hi Uriel,
At TeleSIP we run a cluster of several geographically distributed SER Servers that hande all our SIP Routing. SER is a robust, fast and stable platform which has worked flawlessly for us. We use * as our company PBX and PSTN Gateway. Basically what you need to do is to device a numbering plan so that SERs routing logic can forward the call to * when it needs to. For example in ser.cfg you could put something like this: ############################################# ### PSTN ACCESS ####### ############################################# if (method=="INVITE") { if (uri=~"sip:[EMAIL PROTECTED]") { log(1, "This is a Long Distance Call\n"); route(6); break; }; }; . . . route[6] { rewritehostport("your_asterisk_box:5050"); if (!t_relay()) { sl_reply_error(); }; } Andres http://www.telesip.net > I suspect it is very inexpensive to put a SER server in a hosting facility > 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 _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users