[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 11:20 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Sipura ATA's Forwarding PSTN Calls to
> Asterisk
> 
> 
> Douglas Garstang wrote:
> > I'm working with a Sipura 3000 ATA here. I'm trying to get 
> incoming PSTN calls on the FXO port to go automatically to 
> Asterisk. I have it working, but I had to configure the ATA 
> to register with Asterisk, which means that all calls are 
> being sent to Asterisk with a caller id of the username used 
> to register with Asterisk.
> > 
> > I want the real caller ID to be sent to Asterisk, which 
> means I don't want the ATA to register. The badly written 
> Sipura docs aren't clear about how to do this. Anyone set this up?
> > 
> 
> That's not correct.
> 
> My SPA-3000 FXO port registers with my Asterisk server, and when the 
> PSTN calls come in, it uses the incoming caller's CallerID 
> for the call.
> 
> Sounds like you have something misconfigured.

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>
Call-ID: 6946cb0d3fc1b6d6763e1dea7e5c1d8c at xxx.187.142.203
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "Cody XXX-527-7107" <sip:atacody1 at xxx.187.142.203>;privacy=off;screen=no
Date: Fri, 21 Jul 2006 17:44:20 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 269

v=0
o=root 28771 28771 IN IP4 xxx.187.142.203
s=session
c=IN IP4 xxx.187.142.203
t=0 0
m=audio 21652 RTP/AVP 0 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -