[asterisk-users] Debugging a SIP / AudioCodes Problem

dubrowin.44628913 at bloglines.com dubrowin.44628913 at bloglines.com
Sun Feb 11 12:11:51 MST 2007


I have 2 identical AudioCodes MP-112s.  They have the same config except for
the SIP usernames/passwords and the device IP.  The configs in extension.conf
and sip.conf are also identical.  On one box, when I pick up the phone, I
get a fast busy and the logs/debug show an automatic hangup.  On the other
device, I can make calls without a problem.  I can even call the phone that
can't make a call.  Any ideas where I could start to figure out where the
problem is?

  Shlomo

extensions.conf
[globals]
JOE=SIP/joe
SHMOE=SIP/shmoe

SHLOMO=SIP/shlomo
DAVID=SIP/david
SIMON=SIP/simon

[internal]
exten
=> s,1,Answer()
exten => 611,1,Answer()
exten => 611,n,Playback(hello-world)

exten => 611,n,Hangup()
exten => 612,1,Answer()
exten => 9010,1,Answer()

exten => 9010,n,Dial(${JOE},10)
exten => 9010,n,Hangup()

exten => 9020,1,Answer()

exten => 9020,n,Dial(${SHMOE},10)
exten => 9020,n,Hangup()

exten => 9030,1,Answer()

exten => 9030,n,Dial(${DAVID},10)
exten => 9030,n,Hangup()

exten => 9040,1,Answer()

exten => 9040,n,Dial(${SIMON},10)
exten => 9040,n,Hangup()

exten => 9050,1,Answer()

exten => 9050,n,Dial(${SHLOMO},10)
exten => 9050,n,Hangup()

sip.conf

[joe]
type=friend
secret=joe-password
qualify=yes
nat=no
host=dynamic

canreinvite=no
context=internal

[shmoe]
type=friend
secret=shmoe-password

qualify=yes
nat=no
host=dynamic
canreinvite=no
context=internal

[david]

type=friend
secret=david-password
qualify=yes
nat=no
host=dynamic
canreinvite=no

context=internal

[simon]
type=friend
secret=simon-password
qualify=yes

nat=no
host=dynamic
canreinvite=no
context=internal

[shlomo]
type=friend

secret=shlomo-password
qualify=yes
nat=no
host=dynamic
canreinvite=no

context=internal



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