[asterisk-users] HINT problems with SVN-trunk-r43322

[Hall, Eric M.]

I'm unable to get HINTS working with the new SVN-Trunk

State never changed when ringing or on the phone.

 

 

Below is my configs (Maybe I missed something)

 

Thanks for any help you could give!!

 

 

##sip.conf##

 

[general]

callerid=unavailable

context=default                 ; Default context for incoming calls

bindport=5060                   ; UDP Port to bind to (SIP standard port
is 5060)

bindaddr=0.0.0.0                ; IP address to bind to (0.0.0.0 binds
to all)

;allow=all

allow=ulaw

allow=g729

;allow=gsm

;maxexpirey=3600                ; Max length of incoming registration we
allow

;defaultexpirey=120             ; Default length of incoming/outoing
registration

;notifymimetype=text/plain      ; Allow overriding of mime type in MWI
NOTIFY

videosupport=yes

allow=h263 ; H.263 is our video codec

allow=h263p ; H.263p is the enhanced video codec

qualify=yes

notifyringing=yes

 

[101]

type=friend               ; "friend" means this device takes and makes
calls

username=101             ; Username on device

callerid=Eric <102>

secret=101             ; Password for device

host=dynamic              ; This host is not on the same IP addr every
time

context=default ; Inbound calls from this host go here

mailbox=101 at default; Activate the message waiting light if this

canreinvite=no            ; Leave this alone for now; see archives for
details

nat=1

qualify=yes

Subscribecontext=default

notifyringing=yes

 

##extensions.conf##

 

[general]

static=yes

writeprotect=no

autofallthrough=yes

priorityjumping=yes

[globals]

CONSOLE=Console/dsp                             ; Console interface for
demo

;CONSOLE=Zap/1

;CONSOLE=Phone/phone0

IAXINFO=guest                                   ; IAXtel
username/password

;IAXINFO=myuser:mypass

TRUNK=Zap/g2                

 

[default]

 

 

exten => 101,hint,SIP/101

exten => 102,hint,SIP/102

 

 

exten => 101,1,dial(sip/101,20,tw)

exten => 101,n,voicemail(101)

exten => 101,n,hanup()

 

exten => 102,1,dial(sip/102,20,tw)

exten => 102,n,voicemail(102)

exten => 102,n,hanup()

 

 

 

 

 

Commands from the CLI

 

 

 

CLI> sip show peers

Name/username              Host            Dyn Nat ACL Port     Status


102/102                    206.173.108.30   D   N      5060     OK (5
ms)            

101/101                    206.173.108.25   D   N      5060     OK (5
ms)            

2 sip peers [Monitored: 2 online, 0 offline Unmonitored: 0 online, 0
offline]

 

CLI> show hints

    -= Registered Asterisk Dial Plan Hints =-

                    102 at default             : SIP/102
State:Idle            Watchers  1

                    101 at default             : SIP/101
State:Idle            Watchers  1

----------------

- 2 hints registered

 

CLI> sip show subscriptions 

Peer             User        Call ID      Extension        Last state
Type            Mailbox   

206.173.108.30   102         fb84429adb2  101 at default      Idle
dialog-info+xml <none>    

206.173.108.25   101         499798bcfa4  102 at default      Idle
dialog-info+xml <none>    

2 active SIP subscriptions

 

 

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