[Asterisk-Users] asterisk newbie and phones which don't want
tocomunicate
Mahmoud Badran
mahmoud.badran at atsint.com
Mon Jul 4 23:35:23 MST 2005
Hiii ; actually you are not allowing any codecs in the sip.conf neither
alaw nor ulaw
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=200
allow=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(${ARG1},1)
> include => dundi-e164-lookup
>
>
> [iaxtel700]
> exten => _91700XXXXXXX,1,Dial(IAX2/${IAXINFO}@iaxtel.com/${EXTEN:1}@iaxtel)
>
> [iaxprovider]
> ;switch => IAX2/user:[key]@myserver/mycontext
>
> [trunkint]
>
> exten => _9011.,1,Macro(dundi-e164,${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/guest at misery.digium.com/s at default) ; 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/${EXTEN}|20,t)
>
> [incoming]
> exten => s,1,Wait(1)
> exten => s,2,Dial(Zap/g1|20,t) ; Calls the first available channel in group 1
> exten => s,3,Voicemail,u9000 ; Directs caller to unavailable voicemailbox 9000
> exten => s,4,Hangup
> exten => s,103,Voicemail,b9000 ; Directs caller to busy voicemailbox 9000
> exten => s,104,Hangup
>
>
> [sip-incoming]
> exten => _.,1,Wait(1)
> exten => _.,2,Playback(demo-thanks)
> exten => _.,3,Hangup
>
> [from-sip]
> exten => 2010,1,Dial(SIP/2010,20)
> exten => 2010,2,Voicemail(u2000)
> exten => 2010,102,Voicemail(b2000)
> exten => 2010,103,Hangup
>
> exten => 2011,1,Dial(SIP/2011,20)
> exten => 2011,2,Voicemail(u2011)
> exten => 2011,102,Voicemail(b2011)
> exten => 2011,103,Hangup
>
> exten => 2012,1,Dial(SIP/2012,20)
> exten => 2012,2,Voicemail(u2012)
> exten => 2012,102,Voicemail(b2012)
> exten => 2012,103,Hangup
>
> [local]
> ignorepat => 9
> include => default
> include => parkedcalls
> include => trunklocal
> include => trunktollfree
> include => sip ;x included sip
>
> [sip]
> exten => 55,1,VoicemailMain
>
> exten => 2001,1,Dial(SIP/2001,20,tr)
> exten => 2001,2,VoiceMail,u2001
> exten => 2001,102,VoiceMail,b2001
>
> exten => 2002,1,Dial(SIP/2002,20,tr)
> exten => 2002,2,VoiceMail,u2002
> exten => 2002,102,VoiceMail,b2002
>
> exten => 2003,1,Dial(SIP/2003,20,tr)
> exten => 2003,2,VoiceMail,u2003
> exten => 2003,102,VoiceMail,b2003
>
> exten => 2004,1,Dial(SIP/2004,20,tr)
> exten => 2004,2,VoiceMail,u2004
> exten => 2004,102,VoiceMail,b2004
>
> exten => 2010,1,Dial(SIP/2010,20,tr)
> exten => 2010,2,VoiceMail,u2010
> exten => 2010,102,VoiceMail,b2010
>
> exten => 2011,1,Dial(SIP/2011,20,tr)
>
> exten => 2022,1,Dial(SIP/2022,20,tr)
>
> exten => _1XXX,1,Dial(IAX/asterisk2:1945 at 192.168.1.30/${EXTEN}@local)
>
> 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?
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050705/6ac90a00/attachment.htm
More information about the asterisk-users
mailing list