RE: [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: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Hall, Eric M. Sent: Wednesday, September 20, 2006 11:39 AM To: asterisk-users@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 [EMAIL PROTECTED]; 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 =-                     [EMAIL PROTECTED]             : SIP/102               State:Idle            Watchers  1                     [EMAIL PROTECTED]             : 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  [EMAIL PROTECTED]      Idle           dialog-info+xml <none>    206.173.108.25   101         499798bcfa4  [EMAIL PROTECTED]      Idle           dialog-info+xml <none>    2 active SIP subscriptions     The contents of this e-mail are intended for the named addressee only. It contains information that may be confidential. Unless you are the named addressee or an authorized designee, you may not copy or use it, or disclose it to anyone else. If you received it in error please notify us immediately and then destroy it.

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