Anyone know what's causing this:
<-- SIP read from x.x.x.x:56800: ACK sip:[EMAIL PROTECTED]:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.200.16;branch=z9hG4bKba5c00f818387D67 From: "Xxx xxx" <sip:[EMAIL PROTECTED]>;tag=69375B3E-ACF6C78B To: <sip:[EMAIL PROTECTED];user=phone>;tag=as57402fc2 CSeq: 1 ACK Call-ID: [EMAIL PROTECTED] Contact: <sip:[EMAIL PROTECTED]> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_301-UA/1.5.2.0054 Max-Forwards: 70 Content-Length: 0 --- (11 headers 0 lines)--- Destroying call '[EMAIL PROTECTED]' lou01*CLI> <-- SIP read from x.x.x.x:56800: INVITE sip:[EMAIL PROTECTED]:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.200.16;branch=z9hG4bKc56387b5FF26D6A0 From: "Xxx xxx" <sip:[EMAIL PROTECTED]>;tag=69375B3E-ACF6C78B To: <sip:[EMAIL PROTECTED];user=phone> CSeq: 2 INVITE Call-ID: [EMAIL PROTECTED] Contact: <sip:[EMAIL PROTECTED]> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_301-UA/1.5.2.0054 Supported: 100rel,replace Allow-Events: talk,hold,conference Proxy-Authorization: Digest username="user1", realm="asterisk", nonce="07b9f9a3", uri="sip:[EMAIL PROTECTED]:5060;user=phone", response="a8f005540682f07a88e023d50135cce0", algorithm=MD5 Max-Forwards: 70 Content-Type: application/sdp Content-Length: 253 v=0 o=- 1130439113 1130439113 IN IP4 192.168.200.16 s=Polycom IP Phone c=IN IP4 192.168.200.16 t=0 0 a=sendrecv m=audio 2228 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 --- (15 headers 11 lines)--- Using INVITE request as basis request - [EMAIL PROTECTED] Reliably Transmitting (NAT) to x.x.x.x:56800: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.200.16;branch=z9hG4bKc56387b5FF26D6A0;received=x.x.x.x;rport=568 00 From: "Xxx xxx" <sip:[EMAIL PROTECTED]>;tag=69375B3E-ACF6C78B To: <sip:[EMAIL PROTECTED];user=phone>;tag=as71adaedb Call-ID: [EMAIL PROTECTED] CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:[EMAIL PROTECTED]> Proxy-Authenticate: Digest realm="asterisk", nonce="56bff437" Content-Length: 0 --- Scheduling destruction of call '[EMAIL PROTECTED]' in 15000 ms <-- SIP read from x.x.x.x:56800: INVITE sip:[EMAIL PROTECTED]:5060;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.200.16;branch=z9hG4bKc56387b5FF26D6A0 From: "xxx xxx" <sip:[EMAIL PROTECTED]>;tag=69375B3E-ACF6C78B To: <sip:[EMAIL PROTECTED];user=phone> CSeq: 2 INVITE Call-ID: [EMAIL PROTECTED] Contact: <sip:[EMAIL PROTECTED]> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER User-Agent: PolycomSoundPointIP-SPIP_301-UA/1.5.2.0054 Supported: 100rel,replace Allow-Events: talk,hold,conference Proxy-Authorization: Digest username="user1", realm="asterisk", nonce="07b9f9a3", uri="sip:[EMAIL PROTECTED]:5060;user=phone", response="a8f005540682f07a88e023d50135cce0", algorithm=MD5 Max-Forwards: 70 Content-Type: application/sdp Content-Length: 253 v=0 o=- 1130439113 1130439113 IN IP4 192.168.200.16 s=Polycom IP Phone c=IN IP4 192.168.200.16 t=0 0 a=sendrecv m=audio 2228 RTP/AVP 0 8 18 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 --- (15 headers 11 lines)--- Ignoring this INVITE request Transmitting (NAT) to x.x.x.x:56800: SIP/2.0 488 Not Acceptable Here (codec error) Via: SIP/2.0/UDP 192.168.200.16;branch=z9hG4bKc56387b5FF26D6A0;received=x.x.x.x;rport=568 00 From: "xxx xxx" <sip:[EMAIL PROTECTED]>;tag=69375B3E-ACF6C78B To: <sip:[EMAIL PROTECTED];user=phone>;tag=as71adaedb Call-ID: [EMAIL PROTECTED] CSeq: 2 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY Contact: <sip:[EMAIL PROTECTED]> Content-Length: 0 --- _______________________________________________ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
