[Asterisk-Users] SPA3000 as trunk - no caller ID - solved
Kerry Garrison
support at techdatapros.com
Fri Oct 28 23:27:02 MST 2005
Well, we figured it out. It wasn't a factory reset that fixed it either.
Here is the info:
Corrected article:
http://voipspeak.net/index.php?option=com_content&task=view&id=24
The change that got it working was in the Peer Details. We said to put the
IP address of the asterisk server in the host field, but changing it to the
IP address of the SPA-3000 fixed the problem.
-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