I am using asterisk 1.4.26.2 and cisco call manager 6.1 I am getting a "503 service unavailable" message CCM.
Incoming calls work, outgoing calls get 503 message. What do I do about this??? Jerry ------------------ Asterisk is sending the OPTION: 09/08/2009 18:55:03.150 CCM|//SIP/SIPUdp/wait_UdpDataInd: Incoming SIP UDP message size 498 from ast_server:[5060]: OPTIONS sip:cisco <sip:10.101.66.14> SIP/2.0 Via: SIP/2.0/UDP ast_server:5060;branch=z9hG4bK5a6cdc74;rport From: "asterisk" <sip:aster...@ast_server>;tag=as481c0f21 To: <sip:cisco> Contact: <sip:aster...@ast_server> Call-ID: 047d027a06a762e93a7c9c7f23d46...@ast_server CSeq: 102 OPTIONS User-Agent: Asterisk PBX Max-Forwards: 70 Date: Tue, 08 Sep 2009 23:55:03 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Length: 0 Call Manager reply to this option by 09/08/2009 18:55:03.152 CCM|//SIP/SIPUdp/wait_SdlSPISignal: Outgoing SIP UDP message to ast_server:[5060]: SIP/2.0 503 Service Unavailable Date: Tue, 08 Sep 2009 23:55:03 GMT Warning: 399 "Routing failed: ccbid=6361 socket=ast_server:5060" From: "asterisk" <sip:aster...@ast_server>;tag=as481c0f21 Content-Length: 0 To: <sip:cisco>;tag=1924075766 Call-ID: 047d027a06a762e93a7c9c7f23d46...@ast_server Via: SIP/2.0/UDP ast_server:5060;branch=z9hG4bK5a6cdc74;rport CSeq: 102 OPTIONS _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
