I just ran into a problem with the spa3k and spa942's that I could not
diagnose. It "appears" as though the sipura boxes have a problem with
calls that include a CallerID with "-" in it. I can't say with 100%
certainty yet, but that's my story and I'm sticking to it (for now). ;)
Douglas Garstang wrote:
-----Original Message-----
From: Brian Capouch [mailto:[EMAIL PROTECTED]
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:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP xxx.187.142.203:5060;branch=z9hG4bK208dd2be;rport
From: "Cody XXX-527-7107" <sip:[EMAIL PROTECTED]>;tag=as3a94778b
To: <sip:[EMAIL PROTECTED]>
Contact: <sip:[EMAIL PROTECTED]>
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "Cody XXX-527-7107" <sip:[EMAIL
PROTECTED]>;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 - - - -
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