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suddenly stopped working

Added by tim forman almost 8 years ago

I'm pretty sure I haven't changed anything.. but, all of a sudden, every single call I make fails.
NOTICE7359: chan_sip.c:23534 handle_request_invite: Call from '4030' (198.57.82.173:9061) to extension '4020' rejected because extension not found in context 'default'.
I've deleted and recreated the carrier. Here's the carrier:

[twilio]
type=peer
host=detail.pstn.twilio.com
dtmfmode=rfc2833
canreinvite=no
disallow=all
allow=ulaw
insecure=port,invite
fromuser= 1423xxxxxxx
fromdomain=xxxxx.pstn.twilio.com

DIAL9TRUNK = SIP/twilio

exten => _91NXXNXXXXXX,1,AGI
exten => _91NXXNXXXXXX,n,Dial(${DIAL9TRUNK}/1${EXTEN:2},,To)
exten => _91NXXNXXXXXX,n,Hangup

exten => _9NXXNXXXXXX,1,AGI
exten => _9NXXNXXXXXX,n,Dial(${DIAL9TRUNK}/1${EXTEN:1},,tTo)
exten => _9NXXNXXXXXX,n,Hangup

That was working 12 hours ago, now it's not
I've got about 5 extensions, and every single one gives same error (extension not found in context default)

Here is what the phone shows:
type=friend
host=dynamic
secret=xxxxxx
canreinvite=no
context=default
qualify=yes
disallow=all
allow=ulaw
allow=g729
allow=gsm

I have no problem registering the phone with the server..
The only thing I can think of is my sip.conf
[general]
context=default ; Default context for incoming calls
allowguest=yes ; Allow or reject guest calls (default is yes)
bindport=9061 ; UDP Port to bind to (SIP standard port is 5060)
bindaddr=10.40.10.5 ; IP address to bind to (0.0.0.0 binds to all)
srvlookup=yes ; Enable DNS SRV lookups on outbound calls
externhost=detailxperts.asuscomm.com:9061
externrefresh=300
localnet=10.40.0.0/255.255.0.0
externip=104.128.172.226
nat=yes
disallow=all ; First disallow all codecs
allow=ulaw ; Allow codecs in order of preference
mohinterpret=default
mohsuggest=default
language=en ; Default language setting for all users/peers
relaxdtmf=yes ; Relax dtmf handling
trustrpid = no ; If Remote-Party-ID should be trusted
sendrpid = yes ; If Remote-Party-ID should be sent
progressinband=no ; If we should generate in-band ringing always
dtmfmode = rfc2833 ; Set default dtmfmode for sending DTMF. Default: rfc2833
videosupport=no ; Turn on support for SIP video. You need to turn this on
callevents=yes ; generate manager events when sip ua
rtptimeout=60 ; Terminate call if 60 seconds of no RTP or RTCP activity
notifyringing = yes ; Notify subscriptions on RINGING state (default: no)
notifyhold = yes ; Notify subscriptions on HOLD state (default: no)
limitonpeers = yes ; Apply call limits on peers only. This will improve
canreinvite=no ; Asterisk by default tries to redirect the
jbenable = yes ; Enables the use jitterbuffer on the receiving side of a
jbforce = no ; Forces the use a jitterbuffer on the receive side of a SIP
jbmaxsize = 100 ; Max length of the jitterbuffer in milliseconds.
jbresyncthreshold = 1000 ; Jump in the frame timestamps o jitterbuffer is
jbimpl = fixed ; Jitterbuffer implementation, used he receiving side of a SIP
jblog = no ; Enables jitterbuffer frame logging. Defaults to "no".
qualify=yes ; By default, qualify all peers at 2000ms
limitonpeer = yes ; enable call limit on a per different from limitonpeers
include sip-vicidial.conf

And extensons.conf (just the relevant parts)

[general]
static=yes
writeprotect=no

[globals]
CONSOLE=Console/dsp ; Console interface for demo
TRUNK=DAHDI/r1 ; Trunk interface
TRUNKX=DAHDI/r2 ; 2nd trunk interface
TRUNKIAX=IAX2/ASTtest1::4569 ; IAX trunk interface
TRUNKIAX1=IAX2/ASTtest1::4569 ; IAX trunk interface
TRUNKBINFONE=IAX2/1112223333: ; IAX trunk interface
SIPtrunk=SIP/1234: ; SIP trunk

include extensions-vicidial.conf

[default]
exten => _+X.,n,Hangup()
exten => h,1,AGI

The extensions-vicidial.conf are auto generated, but appear ok
; Phones direct dial extensions:
exten => 4030,1,Dial(SIP/4030,60,)
exten => 4030,2,Goto(default,850266666666664030,1)
exten => 4030,3,Hangup()

I'm stuck, no idea why it suddenly has rejected because extension not found in context 'default' all the time


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