Kerry Garrison
2005-Oct-28 23:27 UTC
[Asterisk-Users] SPA3000 as trunk - no caller ID - solved
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@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