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

[Watkins, Bradley]

You will need to change the type=friend to type=peer and also define
call-limit to some value (it can be large if you don't care about the
actual limit).  That should fix hints for you.
 
Regards,
- Brad


________________________________

	From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Hall, Eric
M.
	Sent: Wednesday, September 20, 2006 11:39 AM
	To: asterisk-users at lists.digium.com
	Subject: [asterisk-users] HINT problems with SVN-trunk-r43322
	
	

	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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