Darryl Moore <[email protected]> schrieb: > I'd start by turning on sip debugging in asterisk > >sip set debug ip [your_phone_ip]
Really destroying SIP dialog '[email protected]' Method: OPTIONS Reliably Transmitting (no NAT) to 192.168.200.11:5060: OPTIONS sip:[email protected]:5060 SIP/2.0 Via: SIP/2.0/UDP 172.16.34.133:5060;branch=z9hG4bK13db26f5;rport Max-Forwards: 70 From: "asterisk" <sip:[email protected]>;tag=as1215345d To: <sip:[email protected]:5060> Contact: <sip:[email protected]> Call-ID: [email protected] CSeq: 102 OPTIONS User-Agent: Asterisk PBX 1.6.2.5-0ubuntu1.4 Date: Thu, 28 May 2015 20:39:02 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces, timer Content-Length: 0 repeated in loop... Help that? 192.168.200.11 is the IP of the phone of my wife, and 172.16.34.133 the IP of the Asterisk server. Thanks Luca Bertoncello ([email protected]) -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
