<--- SIP read from 192.168.32.245:5060 ---> SIP/2.0 481 CallLeg/Transaction Does Not Exist Via: SIP/2.0/UDP 192.168.32.16:5060;branch=z9hG4bK7508a694;rport From: "asterisk"<sip:[email protected]>;tag=as2ff08179 To: <sip:[email protected]:5060;user=phone>;tag=c0a80101-2ce1bc03 Call-ID: [email protected] CSeq: 143 NOTIFY Content-Length: 0
Reliably Transmitting (no NAT) to 192.168.32.245:5060: NOTIFY sip:[email protected]:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.32.16:5060;branch=z9hG4bK7508a694;rport From: "asterisk" <sip:[email protected]>;tag=as2ff08179 To: <sip:[email protected]:5060;user=phone>;tag=c0a80101-2ce1bc03 Contact: <sip:[email protected]> Call-ID: [email protected] CSeq: 143 NOTIFY User-Agent: Asterisk PBX Max-Forwards: 70 Event: message-summary Content-Type: application/simple-message-summary Subscription-State: active Content-Length: 93 Messages-Waiting: no Message-Account: sip:[email protected] Voice-Message: 0/2 (0/0) Can anyone help me out with this? I just recently upgraded to asterisk 1.4.24.1. Use Thomson ST2030s sip phones. _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
