Running FreePBX 12.0.45 Asterisk 13.02 and fully updated, we have a few PJSIP extensions created and a single VVX310 physically connected. Our logs are full of errors that keep rotating about every 30 seconds when the phone is logged in. Not sure what to do but don’t want to add any more phones yet or the console will become unusable for us.
– end of packet.
[2015-03-23 10:32:38] ERROR[2076] pjsip: sip_transport. Error processing 529 bytes packet from UDP 65.34.111.113:5060 : PJSIP syntax error exception when parsing ‘’ header on line 1 col 12:
SUBSCRIBE SIP/2.0
Via: SIP/2.0/UDP 192.168.101.91;branch=z9hG4bK873dc5c5A3DCFC2F
From: “1060” sip:[email protected];tag=44F91737-460170E5
To:
CSeq: 1 SUBSCRIBE
Call-ID: [email protected]
Contact: sip:[email protected]
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: dialog
User-Agent: PolycomVVX-VVX_310-UA/5.2.0.8330
Accept-Language: en
Accept: application/dialog-info+xml
Max-Forwards: 70
Expires: 3600
Content-Length: 0
– end of packet.
[2015-03-23 10:32:38] NOTICE[5454] res_pjsip_exten_state.c: Extension 611 does not exist or has no associated hint
[2015-03-23 10:32:40] NOTICE[5454] res_pjsip_exten_state.c: Extension *97 does not exist or has no associated hint
[2015-03-23 10:32:42] ERROR[2076] pjsip: sip_transport. Error processing 529 bytes packet from UDP 65.34.111.113:5060 : PJSIP syntax error exception when parsing ‘’ header on line 1 col 12:
SUBSCRIBE SIP/2.0
Via: SIP/2.0/UDP 192.168.101.91;branch=z9hG4bK873dc5c5A3DCFC2F
From: “1060” sip:[email protected];tag=44F91737-460170E5
To:
CSeq: 1 SUBSCRIBE
Call-ID: [email protected]
Contact: sip:[email protected]
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
Event: dialog
User-Agent: PolycomVVX-VVX_310-UA/5.2.0.8330
Accept-Language: en
Accept: application/dialog-info+xml
Max-Forwards: 70
Expires: 3600
Content-Length: 0