I still cannot get this to work on 1.0.9. I am trying to test with two extensions:
Here is the config I am using: exten => 451,hint,sip/451 exten => 451,1,Dial(SIP/451,20,tr) exten => 451,2,Voicemail([EMAIL PROTECTED]) exten => 451,102,Voicemail([EMAIL PROTECTED]) exten => 453,hint,sip/453 exten => 453,1,Dial(SIP/453,20,tr) exten => 453,2,Voicemail([EMAIL PROTECTED]) exten => 453,102,Voicemail([EMAIL PROTECTED]) On the SNOM, the SIP trace shows the initial subscription: NOTIFY sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK13ea176f From: <sip:[EMAIL PROTECTED];user=phone>;tag=as77402d3b To: <sip:[EMAIL PROTECTED]>;tag=c0av8f2x4v Contact: <sip:[EMAIL PROTECTED]> Call-ID: [EMAIL PROTECTED] CSeq: 102 NOTIFY User-Agent: Asterisk PBX Event: dialog Content-Type: application/dialog-info+xml Content-Length: 203 <?xml version="1.0"?> <dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="0" state="full" entity="sip:[EMAIL PROTECTED]"> <dialog id="453"> <state>terminated</state> </dialog> </dialog-info> The SNOM shows the light off for this extension. This is a hardphone, and is always registered. NOTIFY sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP 195.27.242.8:5060;branch=z9hG4bK258fb569 From: <sip:[EMAIL PROTECTED];user=phone>;tag=as26ba79ca To: <sip:[EMAIL PROTECTED]>;tag=8ioo4i3sp7 Contact: <sip:[EMAIL PROTECTED]> Call-ID: [EMAIL PROTECTED] CSeq: 102 NOTIFY User-Agent: Asterisk PBX Event: dialog Content-Type: application/dialog-info+xml Content-Length: 202 <?xml version="1.0"?> <dialog-info xmlns="urn:ietf:params:xml:ns:dialog-info" version="0" state="full" entity="sip:[EMAIL PROTECTED]"> <dialog id="451"> <state>confirmed</state> </dialog> </dialog-info> This is a softphone that is not registered, and the light on the keyboard is on. Light is one unavailable, light is off available. When I make a call from extension 453, and am on the phone, nothing is sent to the SNOM. I see no SIP packets leaving Asterisk either. This is what Asterisk shows: asterisk_test*CLI> sip show subscriptions Peer User Call ID URI 195.27.242.113 320 3c26700c30d4-libo7sf1 195.27.242.113 320 3c26700c2bf2-wfqpeg34 0 active SIP subscriptions(s) asterisk_test*CLI> If anyone has any additional ideas, or a snippet of config that works, please post it. I will try to upgrade to 1.2 and see how this works. Thanks, Joe _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users