A phone plugged into it will grab the CID on about the second ring and I have adjusted the SPA3000 out to 5 rings with no difference. What gets passed to asterisk is whatever is set in the 3000's Display Name field. If the Display Name field is blank, then nothing comes across and the phones display 'Unknown'. I have been wondering if there is a variable you can put into the display field. There are some fields that use variables like $PROXY and $NOTIFY, of course simply trying $CID uses '$CID' as the caller ID. On trick I saw before was to add something like A to the caller id number prefix and then strip it out in Asterisk. However, possibly with the latest SPA3000 firmware, if you put an A into the caller id number prefix, the SPA3000 will not answer. -Kerry -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Paul Sent: Wednesday, October 26, 2005 7:58 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] SPA3000 as trunk - no caller ID Kerry Garrison wrote:>Following the instructions at http://geekgazette.com, I have the >SPA-3000 setup as a SIP trunk. This is working flawlessly with one >exception, it isn't passing caller ID. Regardless of what settings I >have tried, I can't seem to figure this out. Has anyone else got it towork?>-Kerry > > >Is it answering too soon? Plug a phone with caller ID feature or a caller ID display in parallel so you can be sure that the caller ID info is received before the SPA-3000 goes off-hook. _______________________________________________ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Kerry Garrison wrote:> A phone plugged into it will grab the CID on about the second ring and I > have adjusted the SPA3000 out to 5 rings with no difference. What gets > passed to asterisk is whatever is set in the 3000's Display Name field. If > the Display Name field is blank, then nothing comes across and the phones > display 'Unknown'. I have been wondering if there is a variable you can put > into the display field. There are some fields that use variables like $PROXY > and $NOTIFY, of course simply trying $CID uses '$CID' as the caller ID.You don't need any clever manipulation tricks with the current firmware. Have you got PSTN CID for VOIP CID set to yes ? jd -- John Daragon john@argv.co.uk argv[0] limited Lambs Lawn Cottage, Staple Fitzpaine, Taunton, TA3 5SL, UK v +44 (0) 1460 234068 f +44 (0) 1460 234069 m +44 (0) 7836 576127
Yes I do have that set to Yes. -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of John Daragon Sent: Wednesday, October 26, 2005 11:05 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] SPA3000 as trunk - no caller ID Kerry Garrison wrote:> A phone plugged into it will grab the CID on about the second ring and > I have adjusted the SPA3000 out to 5 rings with no difference. What > gets passed to asterisk is whatever is set in the 3000's Display Name > field. If the Display Name field is blank, then nothing comes across > and the phones display 'Unknown'. I have been wondering if there is a > variable you can put into the display field. There are some fields > that use variables like $PROXY and $NOTIFY, of course simply trying $CIDuses '$CID' as the caller ID. You don't need any clever manipulation tricks with the current firmware. Have you got PSTN CID for VOIP CID set to yes ? jd -- John Daragon john@argv.co.uk argv[0] limited Lambs Lawn Cottage, Staple Fitzpaine, Taunton, TA3 5SL, UK v +44 (0) 1460 234068 f +44 (0) 1460 234069 m +44 (0) 7836 576127 _______________________________________________ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
I am connected to Cox Cable telephone service which nicely passes CID when using an X100P card but am not getting CID when using the SPA3000. I am using 3.1.5(GWb) -Kerry -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of asterisk@txpe.net Sent: Wednesday, October 26, 2005 9:57 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] SPA3000 as trunk - no caller ID What are you plugging into the FXO port? I have two SPA-2000's. One is connected to an SBC POTS line and the other to a Vonage Cisco ATA186. The SPA3k connected to the SBC POTS passes CID info, the one connected to the Cisco ATA will not. I have tried many things mentioned at AAH SF.net site and Voxilla's forums (and probably a few others) and was never able to get it to pass the CID info. According to the Samurize program and the SPA3k web page, the SPA3k is receiving the CID info from the Cisco ATA. I gave up and now I have my Vonage line forward all calls to my Teliax account. I'm running 3.1.3(GWa) on both units. At 09:37 AM 10/26/2005, you wrote:>Following the instructions at http://geekgazette.com, I have the >SPA-3000 setup as a SIP trunk. This is working flawlessly with one >exception, it isn't passing caller ID. Regardless of what settings I >have tried, I can't seem to figure this out. Has anyone else got it towork?>-Kerry_______________________________________________ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Kerry Garrison wrote:> Yes I do have that set to Yes. >Does the SPA-3000 show the caller ID in the "last call" field in the summary page ? It's capable of interpreting a bewildering array of callerid schemes - is it set to what your local telco is generating ? jd -- John Daragon john@argv.co.uk argv[0] limited Lambs Lawn Cottage, Staple Fitzpaine, Taunton, TA3 5SL, UK v +44 (0) 1460 234068 f +44 (0) 1460 234069 m +44 (0) 7836 576127
REAAAALLLY??? Hell, I can do that. Anything is worth a try at this point. I have it fully documented so restoring the settings shouldn't take but a few minutes. I am just not going to be in the office for about 5 hours now and not going to ask my wife to do it. I will certainly try it, its had half a dozen firmware updates and a bajillion setting changes, it certainly wont hurt to try it. -Kerry -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of InetUID Sent: Thursday, October 27, 2005 9:14 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: john@argv.co.uk Subject: Re: [Asterisk-Users] SPA3000 as trunk - no caller ID I've had a very similar thing on my SPA-3000 and they only way to fix it was a full default reset on the SPA and reconfigure it from scratch 8-( Matt. On 27/10/05, Kerry Garrison <support@techdatapros.com> wrote:> Upgraded to 3.1.7 > > Excerpts from Asterisk Log: > > Oct 27 07:43:50 DEBUG[1531]: cdr_mysql: SQL command as follows: INSERT > INTO cdr > (calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,du > ration > ,billsec,disposition,amaflags,accountcode) VALUES ('2005-10-27 > 07:43:50','\"Garrison Kerry\" > <9496799285>','9496799285','s','from-sip-external', > 'SIP/192.168.5.200-083279d0','','Congestion','',0,0,'NO ANSWER',3,'') > Oct 27 07:43:56 VERBOSE[1531]: -- Executing GotoIf("SIP/spa3000-8d99", > "0?from-pstn-reghours|s|1:") in new stack Oct 27 07:43:56 DEBUG[1531]: > Check for res for spa3000 Oct 27 07:43:56 DEBUG[1531]: Call from user > 'spa3000' is 1 out of 0 Oct 27 07:43:56 DEBUG[1531]: build_route: > Contact hop: > Oct 27 07:43:56 DEBUG[1531]: Expression is '0' > Oct 27 07:43:56 VERBOSE[1531]: -- Executing GotoIf("SIP/spa3000-8d99", > "0?from-pstn-reghours|s|1:") in new stack > > The log is interesting in that it actually is pushing the CID across > but then something strange is happening, if I look at my CDR it shows > the > following: > > The call comes in to SIP/192.168.5.200 Source is the correct source > phone number, Clid is correct CID, Dst is s, Disposition is NO ANSWER > 6-7 seconds later it there is another entry The call comes in to > SIP/spa3000 Source is now empty, Clid is spa3000, Dst is 201, > Disposition is ANSWERED > > Here is a link to a screenshot of the SPA3000 settings: > http://techdatapros.com/temp/spa3000.gif > > -Kerry > > > _______________________________________________ > --Bandwidth and Colocation sponsored by Easynews.com -- > > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >_______________________________________________ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users