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I have just setup up my first Asterisk server (Version 1.6.1.0) under Centos Linux.
I have got my test clients communicating OK, but in checking the SIP conversations with a packet sniffer - for interest/education as much as anything else - I have found that Asterisk is consistently rejecting all new SIP messages, such as a REGISTER or INVITE, with a 401 Unauthorized response. If the client repeats the message, the conversation proceeds as expected - i.e. a REGISTER is greeted with an OK and an INVITE results in a connected call.
This behavior is not stopping anything working but it does not seem to me to be correct. Is this just the way Asterisk behaves or have I done something wrong in the configuration. I have attached my exensions.conf and sip.conf (with IP addresses edited out for security) below as this seems to the first thing anyone asks for.
Many thanks.
Andy
extensions.conf:
[globals]
[macro-phone]
exten => s,1,Set(station=${MACRO_EXTEN:-4})
exten => s,2,Dial(SIP/${station},25)
exten => s,n,Goto(${DIALSTATUS},1)
exten => ANSWER,1,Hangup
exten => CANCEL,1,Hangup
exten => NOANSWER,1,Voicemail(${station}@default,u)
exten => BUSY,1,Voicemail(${station}@default,b)
exten => CONGESTION,1,Voicemail(${station}@default,b)
exten => CHANUNAVAIL,1,Voicemail(${station}@default,u)
exten => a,1,VoicemailMain(${station}@default)
[stations]
exten => 2220,1,Macro(phone)
exten => 2221,1,Macro(phone)
exten => 4242,1,VoicemailMain(default)
exten => 4444,1,VoicemailMain(s${CALLERID(num)}@default)
[long-distance]
exten => _NXXNXXXXXX,1,Set(DNIS=+1${EXTEN})
exten => _NXXNXXXXXX,n,Set(PATH=ld-no-cc})
exten => _NXXNXXXXXX,n,Goto(VOICECORE,1)
exten => _1NXXNXXXXXX,1,Set(DNIS=+${EXTEN})
exten => _1NXXNXXXXXX,n,Set(PATH=ld-with-cc})
exten => _1NXXNXXXXXX,n,Goto(VOICECORE,1)
exten => VOICECORE,1,Set(CALLERID(num)=+1214259${CALLERID(num)})
exten => VOICECORE,n,Set(CALLERID(name)=${CALLERID(num)})
exten => VOICECORE,n,Dial(SIP/labSBC/${DNIS})
[international]
exten => _Z.,1,Set(DNIS=+${EXTEN})
exten => _Z.,n,Set(PATH=international})
exten => _Z.,n,Goto(VOICECORE,1)
[did]
exten => _+1214259222X,1,Macro(phone)
[users]
include => stations
include => local
include => did
include => long-distance
include => international
sip.conf
[general]
videosupport=yes
context=users
port=5060
bindaddr=0.0.0.0
rtpstart=10000
rtpend=20000
[2220]
type=friend
host=dynamic
secret=smle303
context=users
subscribecontext=users
callerid=Astra Phone <2220>
disallow=all
allow=ulaw
allow=alaw
allow=g729
allow=h263
allow=h263p
allow=h264
dtmfmode=rfc2833
mailbox=2220@default
canreinvite=no
[2221]
type=friend
host=dynamic
username=asykes
secret=smle303
context=users
subscribecontext=users
callerid=Video Phone <2221>
disallow=all
allow=ulaw
allow=alaw
allow=g729
allow=h263
allow=h263p
allow=h264
dtmfmode=rfc2833
mailbox=2221@default
canreinvite=no
[2222]
type=friend
host=dynamic
username=asykes
secret=smle303
context=users
subscribecontext=users
callerid=Video Phone <2222>
disallow=all
allow=ulaw
allow=alaw
allow=g729
allow=h263
allow=h263p
allow=h264
dtmfmode=rfc2833
mailbox=2222@default
canreinvite=no
[labSBC]
type=peer
host=xx.xx.xx.xx
deny=0.0.0.0/0.0.0.0
permit=xx.xx.xx.0/255.255.255.0
disallow=all
allow=g729
dtmfmode=rfc2833
qualify=yes
transport=udp
canreinvite=no
context=users
canreinvite=no |