[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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