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 <[EMAIL PROTECTED]> 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:[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 - - - - > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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