Call disconnect problems

I am using Trixbox 2.2.12 with Asterisk 1.6, FreePBX 2.5.1.1 and Centos 4.7 Final.

I have noticed that when I place a call to my mobile or my analogue line if I do not aswer and hang up the PBX handset the phone keeps ringing until I answer, it then goes dead like a ghost call. Does anybody have any ideas about this one? I have recently taken over from an experienced asterisk engineer and am quite new to this system. I would be happy to supply whatever information is needed to diagnose this problem.

Here is the end of the output from "asterisk cli" I don't see what is wrong with this as I can see the hangup macro being used. But I do not really understand asterisk yet.

*TEXT*
-- SIP/voip-unlimited-0997c410 is making progress passing it to SIP/200-b7a08778
-- SIP/voip-unlimited-0997c410 is making progress passing it to SIP/200-b7a08778
== Spawn extension (macro-dialout-trunk, s, 19) exited non-zero on 'SIP/200-b7a08778' in macro 'dialout-trunk'
== Spawn extension (macro-dialout-trunk, s, 19) exited non-zero on 'SIP/200-b7a08778'
-- Executing Macro("SIP/200-b7a08778", "hangupcall|") in new stack
-- Executing ResetCDR("SIP/200-b7a08778", "w") in new stack
-- Executing NoCDR("SIP/200-b7a08778", "") in new stack
-- Executing GotoIf("SIP/200-b7a08778", "1?skiprg") in new stack
-- Goto (macro-hangupcall,s,6)
-- Executing GotoIf("SIP/200-b7a08778", "1?skipblkvm") in new stack
-- Goto (macro-hangupcall,s,9)
-- Executing GotoIf("SIP/200-b7a08778", "1?theend") in new stack
-- Goto (macro-hangupcall,s,11)
-- Executing Hangup("SIP/200-b7a08778", "") in new stack
== Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'SIP/200-b7a08778' in macro 'hangupcall'
== Spawn extension (macro-hangupcall, s, 11) exited non-zero on 'SIP/200-b7a08778'
-- Saved useragent "Grandstream GXP2010 1.1.5.15" for peer 201
*TEXT*

Thank you in advance.
Phil Garner