[Asterisk-Users] SPA3000 as trunk - no caller ID
Kerry Garrison
support at techdatapros.com
Thu Oct 27 09:45:31 MST 2005
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 at lists.digium.com
[mailto:asterisk-users-bounces at 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 at 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 at 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 at 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 at 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
More information about the asterisk-users
mailing list