search for: saarshalom

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2010 Jun 10
1
Am I having problems with codecs? or am I not receiving an invite at all from my DID provider?
...can make calls out: *<--- Transmitting (no NAT) to 82.80.252.29:5090 --->* *SIP/2.0 489 Bad event* *Via: SIP/2.0/UDP 82.80.252.234:5090 ;branch=z9hG4bK463b703d;received=82.80.252.29;rport=5090* *From: "asterisk" <sip:asterisk at 82.80.252.234:5090>;tag=as4af8cf81* *To: <sip:saarshalom at 173.203.29.102 <sip%3Asaarshalom at 173.203.29.102> >;tag=as64c0ba34* *Call-ID: 497197a679122f5d448d324f571f3c3e at 82.80.252.234* *CSeq: 102 NOTIFY* *Server: Asterisk PBX 1.6.2.7* *Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO* *Supported: replaces, timer* *C...