I am trying to get Asterisk to respond correctly to a SIP OPTIONS request sent to it from an external PSTN switch.
According to SIP RFC 3261 a SIP OPTIONS request should be responded with a SIP 200 OK response or the UA would be determined to be off-line by the other party.
Our carrier’s PSTN switch is sending our Asterisk server SIP OPTIONS as a heartbeat signal so they know when our equipment is off-line so they can route calls accordingly. Without a SIP 200 OK response to their SIP OPTIONS request we can not make or receive any calls.
I am using Asterisk 1.2.14 (Asterisk SVN-branch-1.2-r51197 ). I have extensions and SIP accounts setup that correspond to the usernames presented within the SIP OPTIONS request.
Asterisk is responding with: SIP/2.0 404 Not Found to the switches SIP OPTIONS request
Asterisk Verbose Debug response to the SIP OPTIONS shows:-
Am I missing something?
Is this fixed in Asterisk 1.4?
Any help would be appreciated.
OPTIONS sip:GDSSUDTARCQLD0 SIP/2.0. From: <sip:PSDFFS>;tag=c57d2-13c4-b17a7-17c17df6-b17a7. To: <sip:GDSSUDTARCQLD0>. Call-ID: 9175470-202c57d2-13c4-b17a7-6fa155f7-b17a7@PSDFFS. CSeq: 632505012 OPTIONS. Via: SIP/2.0/UDP PSDFFS:5060;maddr=188.8.131.52;branch=z9hG4bK-b17a7-2b546642-35cdfc47. User-agent: CS2000_NGSS/9.0. Max-Forwards: 70. Accept: application/isup, application/sdp, application/dtmf-relay, audio/telephone-event, application/simple-message-summary. Supported: 100rel. Allow: ACK,BYE,CANCEL,INVITE,OPTIONS,INFO,SUBSCRIBE,REFER,NOTIFY,PRACK. Content-Length: 0. . SIP/2.0 404 Not Found. Via: SIP/2.0/UDP PSDFFS:5060;maddr=184.108.40.206;branch=z9hG4bK-b17a7-2b546642-35cdfc47;received=220.127.116.11. From: <sip:PSDFFS>;tag=c57d2-13c4-b17a7-17c17df6-b17a7. To: <sip:GDSSUDTARCQLD0>;tag=as29688414. Call-ID: 9175470-202c57d2-13c4-b17a7-6fa155f7-b17a7@PSDFFS. CSeq: 632505012 OPTIONS. User-Agent: Asterisk PBX. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. Accept: application/sdp. Content-Length: 0.