Asterisk cached an old dial plan? rejected because extension not found in context

asterisk

I'm trying to fix my asterisk server which has been quite stable up until recently. Here's my problem:

a) if i run a sip reload and/or dialplan reload from asterisk terminal, it sometimes breaks the calling service, so that when i dial into the server from a real phone, it says can't complete the call. But then if i wait a few minutes, it might automatically start working again sometimes. If it still doesn't work, i repeat these steps until it works. When the error does happen, I get rejected because extension not found in context [exts]. I can't consistently reproduce the problem. I don't edit my sip.conf or extensions.conf

b) When the server does work, and after calling to it from a real phone I dial the extension 99, it calls my cell phone at 555 555 5555. But no where in app.extensions.conf is there that rule! Last year I had a rule where extension 99 dials 555 555 5555, but that has long since been replaced! I greped my server and don't find any instance of 555 555 5555. Is there something i should be doing beside sip reload and dialplan reload?

For debugging purposes, I have attached my sip.conf and extensions.conf

sip.conf

[general]
context=sipdefault
allowoverlap=no
allowtransfer=no
maxexpiry=3600
minexpiry=60
defaultexpiry=3600
checkmwi=10
buggymwi=no
vmexten=voicemail
disallow=all
allow=ulaw
language=en
relaxdtmf=yes
useragent=Asterisk PBX
dtmfmode = rfc2833
rfc2833compensate=yes
alwaysauthreject = yes
rtptimeout=60
rtpholdtimeout=300

register => 8888888888:PASS@sip06.unlimitel.ca/8888888888


canreinvite=yes
jbenable = yes
jbforce = no
jbmaxsize = 200
jbresyncthreshold = 1000
jbimpl = fixed
jblog = no
; END OF GENERAL

[8888888888]
context=exts
type=peer
auth=md5
username=8888888888
fromuser=8888888888
fromdomain=unlimitel.ca
secret=PASS
host=sip06.unlimitel.ca
port=5060
nat=yes
canreinvite=no
qualify=no
disallow=all
allow=ulaw
dtmfmode=rfc2833
insecure=port,invite

extensions.conf

[general]
static=yes
writeprotect=no
clearglobalvars=no

[globals]

[default]
exten => 2005,1,Answer
exten => 2005,2,SetMusicOnHold(default)
exten => 2005,3,WaitMusicOnHold(9000)
exten => 2005,4,Hangup

exten => _X.,1,Hangup(3)


[sipdefault]

exten => _X.,1,Hangup(3)

exten => 2005,1,Answer
exten => 2005,2,SetMusicOnHold(default)
exten => 2005,3,WaitMusicOnHold(9000)
exten => 2005,4,Hangup

[exts]

exten => s,1,Answer
exten => s,n,Wait(1)
exten => s,n,WaitExten(7)


exten => 0011,1,Goto(outbound,s,1)


[outbound]
exten => s,1,Answer
exten => s,n,Wait(1)
exten => s,n,Playback(vm-extension)
exten => s,n,WaitExten(10)

exten => _NXXNXXXXXX,1,Dial(SIP/8888888888/${EXTEN})
exten => _NXXNXXXXXX,n,Hangup

Best Answer

Use "sip set debug on" and see on consoel how EXACTLY come that packet.

Very likly it go not to s extension,but other one.