Freepbx 3.2rc1 Disconected my DID?

Im not actually accusing anyone, its more of a question: Are there any changes in 2.3rc1 that would cause a DID to disconnecte after 5-10 seconds?

Befor I upgraded my sipgate.co.uk DID’s where working fine. Now they disconect after 5-10 seconds. Here is the log file output:

[code:1]
Aug 12 11:52:55 VERBOSE[8554] logger.c: – Executing Answer(“SIP/2721246-b7a0e9e0”, “”) in new stack
Aug 12 11:52:55 VERBOSE[8554] logger.c: – Executing Wait(“SIP/2721246-b7a0e9e0”, “1”) in new stack
Aug 12 11:52:55 DEBUG[6870] chan_sip.c: Stopping retransmission on ‘02bcdf357c7ac68341d4f53[email protected]’ of Response 102: Match Found
Aug 12 11:52:56 VERBOSE[8549] logger.c: – IAX2/Rapidvox-6 is making progress passing it to IAX2/ToTest-3
Aug 12 11:52:56 DEBUG[6878] chan_iax2.c: Ooh, voice format changed to 4
Aug 12 11:52:56 DEBUG[6870] acl.c: ##### Testing 192.168.0.105 with 192.168.0.0
Aug 12 11:52:56 DEBUG[6878] chan_iax2.c: Ooh, voice format changed to 4
Aug 12 11:52:56 VERBOSE[8549] logger.c: – IAX2/Rapidvox-6 answered IAX2/ToTest-3
Aug 12 11:52:56 VERBOSE[8554] logger.c: – Executing Set(“SIP/2721246-b7a0e9e0”, “LANGUAGE()=en”) in new stack
Aug 12 11:52:56 VERBOSE[8554] logger.c: – Executing SetMusicOnHold(“SIP/2721246-b7a0e9e0”, “Classical”) in new stack
Aug 12 11:52:56 VERBOSE[8554] logger.c: – Executing System(“SIP/2721246-b7a0e9e0”, “rm /var/lib/asterisk/sounds/priv-callerintros/.”) in new stack
Aug 12 11:52:56 VERBOSE[8554] logger.c: – Executing Dial(“SIP/2721246-b7a0e9e0”, “Zap/g3/0987654#|40|pmA(beep)”) in new stack
Aug 12 11:52:56 VERBOSE[8554] logger.c: – Privacy Screening, clid is '0123456’
Aug 12 11:52:56 DEBUG[8554] app.c: play_and_record: priv-recordintro, priv-callerintros/0123456, 'gsm’
Aug 12 11:52:56 DEBUG[8554] channel.c: Scheduling timer at 160 sample intervals
Aug 12 11:52:56 VERBOSE[8554] logger.c: – Playing ‘priv-recordintro’ (language ‘en’)
Aug 12 11:52:58 DEBUG[8554] channel.c: Scheduling timer at 105 sample intervals
Aug 12 11:52:58 DEBUG[8554] channel.c: Scheduling timer at 0 sample intervals
Aug 12 11:52:58 DEBUG[8554] channel.c: Scheduling timer at 0 sample intervals
Aug 12 11:52:58 DEBUG[8554] channel.c: Scheduling timer at 160 sample intervals
Aug 12 11:52:58 VERBOSE[8554] logger.c: – Playing ‘beep’ (language ‘en’)
Aug 12 11:52:58 DEBUG[6870] acl.c: ##### Testing 204.147.183.18 with 192.168.0.0
Aug 12 11:52:58 DEBUG[6870] chan_sip.c: Target address 204.147.183.18 is not local, substituting externip
Aug 12 11:52:58 DEBUG[6870] chan_sip.c: Scheduled a registration timeout for sip.stanaphone.com id #37925
Aug 12 11:52:58 DEBUG[8554] channel.c: Scheduling timer at 0 sample intervals
Aug 12 11:52:58 DEBUG[8554] channel.c: Scheduling timer at 0 sample intervals
Aug 12 11:52:58 DEBUG[8554] app.c: Recording Formats: sfmts=gsm
Aug 12 11:52:58 VERBOSE[8554] logger.c: – x=0, open writing: priv-callerintros/0123456 format: gsm, 0xa0653c0
Aug 12 11:52:58 DEBUG[6870] chan_sip.c: Stopping retransmission on ‘[email protected]’ of Request 105: Match Found
Aug 12 11:52:58 DEBUG[6870] chan_sip.c: Registration successful
Aug 12 11:52:58 DEBUG[6870] chan_sip.c: Cancelling timeout 37925
Aug 12 11:53:00 DEBUG[8554] app.c: One waitfor failed, trying another
Aug 12 11:53:02 WARNING[8554] app.c: No audio available on SIP/2721246-b7a0e9e0??
Aug 12 11:53:02 VERBOSE[8554] logger.c: – User hung up
Aug 12 11:53:02 VERBOSE[8554] logger.c: – Successfully deleted priv-callerintros/0123456 intro file
Aug 12 11:53:02 DEBUG[8554] app_dial.c: Exiting with DIALSTATUS=.
Aug 12 11:53:02 VERBOSE[8554] logger.c: == Spawn extension (custom-moshe-cell, s, 6) exited non-zero on ‘SIP/2721246-b7a0e9e0’
[/code:1]

I would like to add that I noticed that these disconects only happen if im using the “p” option to bridge the call. This was working perfectly fine befor i upgraded…

Here is my dial plan:

[code:1]
exten => s,1,Answer
exten => s,n,Wait(1)
exten => s,n,Set(LANGUAGE()=en)
exten => s,n,SetMusicOnHold(Classical)
exten => s,n,System(rm /var/lib/asterisk/sounds/priv-callerintros/.) ;delete all previously saved “name” files
exten => s,n,Dial(Zap/g3/0987654#,40,mA(beep))
;exten => s,n,Dial(Zap/g3/0987654#,40,pmA(beep))
exten => s,n,Goto(s-${DIALSTATUS},1) ;catch errors or other events

exten => s-ANSWER,1,Goto(s-NOANSWER,1)

exten => s-NOANSWER,1,Noop(Call rejected on [custom-moshe-cell] by callee)
exten => s-NOANSWER,n,Background(vm-nobodyavail&to-send-num-msg&press-1);1-to send an sms
exten => s-NOANSWER,n,Background(T-to-leave-msg&press-2) ;2 to send a voicemail
exten => s-NOANSWER,n,DigitTimeout(1)
exten => s-NOANSWER,n,WaitExten(6)
exten => s-NOANSWER,n,Playback(vm-goodbye)
exten => s-NOANSWER,n,Hangup
exten => 1,1,Set(__sms-from=moshe-cell)
exten => 1,n,Goto(moshe-sms,s,1)
exten => 2,1,Playback(pls-wait-connect-call)
exten => 2,2,Dial(Zap/g3/1515987654#,m)

exten => s-TORTURE,1,Noop(The caller didnt record their voice…) ; in case people perposly dont want to record their voice or are otherwise annoying…
exten => s-TORTURE,n,Set(LOOPCOUNT=$[${LOOPCOUNT} + 1])
exten => s-TORTURE,n,GotoIf($[${LOOPCOUNT} > 1]?s-TORTURE-hang,1)
exten => s-TORTURE,n,Playback(vm-sorry&please-try-again)
exten => s-TORTURE,n,Waitmusiconhold(1)
exten => s-TORTURE,n,Goto(s,1)

exten => s-TORTURE-hang,1,System(/usr/sbin/smsq --motx-channel “Zap/g1/14974800” “+9725987654” “The caller trying to reach you is: ${CALLERID}. However they didnt record their voice at the promt.”)
exten => s-TORTURE-hang,n,Playback(Goodbye)
exten => s-TORTURE-hang,n,Hangup

exten => s-CONGESTION,1,Noop(Apperently Zap/6 is currently in use)
exten => s-CONGESTION,n,Playback(im-sorry&all-circuits-busy-now&sorry-youre-having-problems&please-try-again-later&auth-thankyou&goodbye)
exten => s-CONGESTION,n,System(/usr/sbin/smsq --motx-channel “Zap/g1/14974800” “+972987654” “You have a missed call from: ${CALLERID}. \n (DIALSTATUS=CONGESTION)”)
exten => s-CONGESTION,n,Hangup

[/code:1]