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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
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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.

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