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
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