[Asterisk-Users] NAT, SIP (was: No sound with SIP Phones on the Internet)

[Andres]

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:01[1-9][0-9]+ at .*") {
                      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: asterisk-users-admin at lists.digium.com
> [mailto:asterisk-users-admin at lists.digium.com]On Behalf Of John Todd
> Sent: Monday, October 13, 2003 8:11 PM
> To: asterisk-users at lists.digium.com
> 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  chrisalbertson90278 at yahoo.com
> >   Cell:   310-990-7550
> >   Office: 310-336-5189  Christopher.J.Albertson at aero.org
> >   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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>
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