so try this to all phones in sip.conf or put it in the general context (allow=all)
[2011]
type=friend username=2011 secret=1945 nat=yes host=dynamic dtmfmode=rfc2833 canreinvite=no qualify=200allow=all
On Mon, 2005-07-04 at 18:00 +0200, Sistemista WebSolvingJaa wrote:
with some trials configuration,and a couple of hours now i can make a
call from a phone to another phone. typing the code of phone A from
phone B, the ring-tone of phone A rings but neither phone A and phone
B can comunicate as voice (i hope my explaination can be understood by
all of you). so my extension.conf is now like this:
[general]
static=yes
writeprotect=yes
autofallthrough=yes
[globals]
CONSOLE=Console/dsp ; Console interface for demo
CONSOLE=Zap/1
CONSOLE=Phone/phone0
IAXINFO=guest ; IAXtel username/password
TRUNK=Zap/g2 ; Trunk interface
TRUNKMSD=1 ; MSD digits to strip
(usually 1 or 0)
[dundi-e164-local]
include => dundi-e164-canonical
include => dundi-e164-customers
include => dundi-e164-via-pstn
[dundi-e164-switch]
switch => DUNDi/e164
[dundi-e164-lookup]
include => dundi-e164-local
include => dundi-e164-switch
[macro-dundi-e164]
exten => s,1,Goto(${ARG1},1)
include => dundi-e164-lookup
[iaxtel700]
exten => _91700XXXXXXX,1,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN:[EMAIL PROTECTED])
[iaxprovider]
;switch => IAX2/user:[EMAIL PROTECTED]/mycontext
[trunkint]
exten => _9011.,1,Macro(dundi-e164,${EXTEN:4})
exten => _9011.,n,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
[trunkld]
exten => _91NXXNXXXXXX,1,Macro(dundi-e164,${EXTEN:1})
exten => _91NXXNXXXXXX,n,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
[trunklocal]
exten => _9NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
[trunktollfree]
exten => _91800NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten => _91888NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten => _91877NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten => _91866NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
[international]
ignorepat => 9
include => longdistance
include => trunkint
[longdistance]
ignorepat => 9
include => local
include => trunkld
[local]
ignorepat => 9
include => default
include => parkedcalls
include => trunklocal
include => iaxtel700
include => trunktollfree
include => iaxprovider
[macro-stdexten];
exten => s,1,Dial(${ARG2},20) ; Ring
the interface, 20 seconds maximum
exten => s,2,Goto(s-${DIALSTATUS},1) ; Jump
based on status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
exten => s-NOANSWER,1,Voicemail(u${ARG1}) ; If
unavailable, send to voicemail w/ unavail announce
exten => s-NOANSWER,2,Goto(default,s,1) ; If they
press #, return to start
exten => s-BUSY,1,Voicemail(b${ARG1}) ; If busy,
send to voicemail w/ busy announce
exten => s-BUSY,2,Goto(default,s,1) ; If
they press #, return to start
exten => _s-.,1,Goto(s-NOANSWER,1) ;
Treat anything else as no answer
exten => a,1,VoicemailMain(${ARG1}) ; If
they press *, send the user into VoicemailMain
[demo]
exten => s,1,Wait,1 ; Wait a second, just for fun
exten => s,n,Answer ; Answer the line
exten => s,n,SetVar(TIMEOUT(digit)=5) ; Set Digit Timeout to 5 seconds
exten => s,n,SetVar(TIMEOUT(response)=10) ; Set Response Timeout
to 10 seconds
exten => s,n(restart),BackGround(demo-congrats) ; Play a congratulatory message
exten => s,n(instruct),BackGround(demo-instruct) ; Play some instructions
exten => s,n,WaitExten ; Wait for an extension to be dialed.
exten => 2,1,BackGround(demo-moreinfo) ; Give some more information.
exten => 2,n,Goto(s,instruct)
exten => 3,1,SetVar(LANGUAGE()=fr) ; Set language to french
exten => 3,n,Goto(s,restart) ; Start with the congratulations
exten => 1000,1,Goto(default,s,1)
exten => 1234,1,Playback(transfer,skip) ; "Please hold while..."
; (but skip if channel is not up)
exten => 1234,n,Macro(stdexten,1234,${CONSOLE})
exten => 1235,1,Voicemail(u1234) ; Right to voicemail
exten => 1236,1,Dial(Console/dsp) ; Ring forever
exten => 1236,n,Voicemail(u1234) ; Unless busy
exten => #,1,Playback(demo-thanks) ; "Thanks for trying the demo"
exten => #,n,Hangup ; Hang them up.
exten => t,1,Goto(#,1) ; If they take too long, give up
exten => i,1,Playback(invalid) ; "That's not valid, try again"
exten => 500,1,Playback(demo-abouttotry); Let them know what's going on
exten => 500,n,Dial(IAX2/[EMAIL PROTECTED]/[EMAIL PROTECTED]) ; Call
the Asterisk demo
exten => 500,n,Playback(demo-nogo) ; Couldn't connect to the demo site
exten => 500,n,Goto(s,6) ; Return to the start over message.
exten => 600,1,Playback(demo-echotest) ; Let them know what's going on
exten => 600,n,Echo ; Do the echo test
exten => 600,n,Playback(demo-echodone) ; Let them know it's over
exten => 600,n,Goto(s,6) ; Start over
exten => 8500,1,VoicemailMain
exten => 8500,n,Goto(s,6)
[default]
include => from-sip
exten => 1000,1,Dial,Zap/1|20 ; Exten 1000 dials Zap channel 1 with a
ring time option of 20 secs, which is the analog telephone plugged
into the first port of the TDM400P.
exten => 1000,2,Voicemail,u1000
exten => 1000,3,Hangup
exten => 1000,102,Voicemail,b1000
exten => 1000,103,Hangup
exten => 2000,1,Dial,Zap/2|20
exten => 2000,2,Voicemail,u2000
exten => 2000,3,Hangup
exten => 2000,102,Voicemail,b2000
exten => 2000,103,Hangup
exten => 3000,1,Dial,Zap/3|20
exten => 3000,2,Voicemail,u3000
exten => 3000,3,Hangup
exten => 3000,102,Voicemail,b3000
exten => 3000,103,Hangup
exten => _NXXXXXX,1,Dial(Zap/4/${EXTEN}|20,t)
[incoming]
exten => s,1,Wait(1)
exten => s,2,Dial(Zap/g1|20,t) ; Calls the first available channel in group 1
exten => s,3,Voicemail,u9000 ; Directs caller to unavailable voicemailbox 9000
exten => s,4,Hangup
exten => s,103,Voicemail,b9000 ; Directs caller to busy voicemailbox 9000
exten => s,104,Hangup
[sip-incoming]
exten => _.,1,Wait(1)
exten => _.,2,Playback(demo-thanks)
exten => _.,3,Hangup
[from-sip]
exten => 2010,1,Dial(SIP/2010,20)
exten => 2010,2,Voicemail(u2000)
exten => 2010,102,Voicemail(b2000)
exten => 2010,103,Hangup
exten => 2011,1,Dial(SIP/2011,20)
exten => 2011,2,Voicemail(u2011)
exten => 2011,102,Voicemail(b2011)
exten => 2011,103,Hangup
exten => 2012,1,Dial(SIP/2012,20)
exten => 2012,2,Voicemail(u2012)
exten => 2012,102,Voicemail(b2012)
exten => 2012,103,Hangup
[local]
ignorepat => 9
include => default
include => parkedcalls
include => trunklocal
include => trunktollfree
include => sip ;x included sip
[sip]
exten => 55,1,VoicemailMain
exten => 2001,1,Dial(SIP/2001,20,tr)
exten => 2001,2,VoiceMail,u2001
exten => 2001,102,VoiceMail,b2001
exten => 2002,1,Dial(SIP/2002,20,tr)
exten => 2002,2,VoiceMail,u2002
exten => 2002,102,VoiceMail,b2002
exten => 2003,1,Dial(SIP/2003,20,tr)
exten => 2003,2,VoiceMail,u2003
exten => 2003,102,VoiceMail,b2003
exten => 2004,1,Dial(SIP/2004,20,tr)
exten => 2004,2,VoiceMail,u2004
exten => 2004,102,VoiceMail,b2004
exten => 2010,1,Dial(SIP/2010,20,tr)
exten => 2010,2,VoiceMail,u2010
exten => 2010,102,VoiceMail,b2010
exten => 2011,1,Dial(SIP/2011,20,tr)
exten => 2022,1,Dial(SIP/2022,20,tr)
exten => _1XXX,1,Dial(IAX/asterisk2:[EMAIL PROTECTED]/[EMAIL PROTECTED])
and the sip.conf file is like this:
[general]
port = 5060
bindaddr = 0.0.0.0
context = from-sip ;x changed from default to sip
[2001]
type=friend
username=2001
secret=1945
canreinvite=no
host=dynamic
dtmfmode=rfc2833
qualify=200
mailbox=2001
nat=1
[2002]
type=friend
username=2002
secret=1945
canreinvite=no
host=dynamic
dtmfmode=rfc2833
qualify=200
mailbox=2002
nat=1
[2010]
type=friend
username=2010
secret=1945
nat=yes
host=dynamic
dtmfmode=rfc2833
canreinvite=no
qualify=200
[2011]
type=friend
username=2011
secret=1945
nat=yes
host=dynamic
dtmfmode=rfc2833
canreinvite=no
qualify=200
[2012]
type=friend
username=2012
secret=1945
nat=yes
host=dynamic
dtmfmode=rfc2833
canreinvite=no
qualify=200
can somebody tell me where are the mistakes?
_______________________________________________ Asterisk-Users mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
