[asterisk-users] Sipura ATA's Forwarding PSTN Calls to Asterisk

[Douglas Garstang]

> -----Original Message-----
> From: Brian Capouch [mailto:brianc at palaver.net]
> Sent: Friday, July 21, 2006 12:15 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Sipura ATA's Forwarding PSTN Calls to
> Asterisk
> 
> 
> Douglas Garstang wrote:
> 
> > 
> > Here's my invite Brian. The From: is always going to 
> contain the auth id the ATA used to register with Asterisk.
> > 
> > INVITE sip:2944009 at xxx.187.130.42 SIP/2.0
> > Via: SIP/2.0/UDP xxx.187.142.203:5060;branch=z9hG4bK208dd2be;rport
> > From: "Cody XXX-527-7107" 
> <sip:atacody1 at xxx.187.142.203>;tag=as3a94778b
> > To: <sip:2944009 at xxx.187.130.42>
> > Contact: <sip:atacody1 at xxx.187.142.203>
> 
> And here's one from a call I just placed.  Note the dissimilarities 
> between the From: and Contact: fields on mine and the snippet 
> of yours 
> shown above.
> 
> I suspect there is an option somewhere on one of the "PSTN" 
> tabs on the 
> SPA-3000 that has to be set correctly to enable the pass-through.  I 
> don't have time right now to play around with it--my system 
> is working 
> just fine :-)  192.168.1.1 is my Asterisk server, and the ATA is at 
> 192.168.1.113.
> 
> "AstIn" is the display name I chose for the registration, btw.
> 
> B.
> 
> INVITE sip:s at 192.168.1.1 SIP/2.0
> Via: SIP/2.0/UDP 192.168.1.113:5061;branch=z9hG4bK-3c04a2ec
> From: "Capouch B" <sip:12192538181 at 192.168.1.1>;tag=5e2ab9e072a1a2cco1
> To: <sip:s at 192.168.1.1>
> Call-ID: 1db84670-5816ff7c at 192.168.1.113
> CSeq: 101 INVITE
> Max-Forwards: 70
> Contact: AstIn <sip:12192538181 at 192.168.1.113:5061>

Well in that case, what's the point in having the ATA register with Asterisk? You just direct all PSTN->VOIP calls to Asterisk with their PSTN CID and destination, and VOIP->PSTN calls you just route with Dial(${EXTEN}@ata).

I tried removing the registration info, thinking maybe that would make the ATA not register, and send all PSTN->VOIP calls to Asterisk, but then the ATA didn't even answer the incoming PSTN call.

Doug.