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

[Jonathan Attwood]

For the OP, do you have an entry against "Display Name" on the PSTN
tab, whilst logged in as admin/advanced? If I have an entry in this,
what you describe happens for me. If the field is empty, CLID is sent
correctly to my Asterisk box.



On 21/07/06, Rich Adamson <radamson at routers.com> wrote:
> 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: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 - - - -
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