-----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
At 11:26 PM -0400 10/13/03, Uriel Carrasquilla wrote:
From: "Uriel Carrasquilla" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Subject: RE: [Asterisk-Users] NAT, SIP (was: No sound with SIP Phones on the Internet)
Reply-To: [EMAIL PROTECTED]
Date: Mon, 13 Oct 2003 23:26:59 -0400
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 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
Uriel - 1) Please stop top-posting.
2) I'm afraid I don't have any data on specifics of creating a front-end. I know how to do it, but my time these days is spent writing lots of other projects that I have been doing. :-) I would suggest you get SER and set it up - it's quite easy, and the documentation on SER itself is very well written, and if you have a good idea of how SIP works you should be able to patch together an appropriate system. However, if you aren't 100% familiar with how SIP works, I would stick to just an Asterisk system; SER doesn't allow for any of the "shortcuts" that Asterisk has.
3) Use Google and do some searching. I found some quick links with a few of the keywords that would seem obvious, but I don't have enough time to review them...
JT _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
