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

[Brian Capouch]

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>

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