Olá pessoal,<br>Estou fazendo laboratórios de Asterisk com alta disponibilidade, por enquanto tenho um heartbeat fazendo a parte de failover, onde tenho um ip virtual, no caso 172.33.16.220, onde esse ip aponta para o host que está com o asterisk ativo. O ambiente está certinho, asterisk 1.6 + FreePBX 2.5.1, com o Zoiper eu consigo registrar no IP virtual normalmente, agora com o X-lite não funciona, segue o sip debug:<br>
<br>REGISTER sip:172.33.16.220 SIP/2.0<br>Via: SIP/2.0/UDP 172.33.33.150:38595;branch=z9hG4bK-d8754z-43603b4ff51a741d-1---d8754z-;rport<br>Max-Forwards: 70<br>Contact: <sip:2589@172.33.33.150:38595;rinstance=b73d4d015d9d419d><br>
To: "Teste Failover"<<a href="mailto:sip%3A2589@172.33.16.220">sip:2589@172.33.16.220</a>><br>From: "Teste Failover"<<a href="mailto:sip%3A2589@172.33.16.220">sip:2589@172.33.16.220</a>>;tag=2e024a16<br>
Call-ID: YTk2NmFkZjZkYTA3MWE4NTg3ZjQ4ODIwMDQzMjM1OTE.<br>CSeq: 1 REGISTER<br>Expires: 3600<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO<br>User-Agent: X-Lite release 1100l stamp 47546<br>
Content-Length: 0<br><br><br><-------------><br>--- (12 headers 0 lines) ---<br>Sending to 172.33.33.150 : 38595 (NAT)<br><br><--- Transmitting (NAT) to <a href="http://172.33.33.150:38595">172.33.33.150:38595</a> ---><br>
SIP/2.0 401 Unauthorized<br>Via: SIP/2.0/UDP 172.33.33.150:38595;branch=z9hG4bK-d8754z-43603b4ff51a741d-1---d8754z-;received=172.33.33.150;rport=38595<br>From: "Teste Failover"<<a href="mailto:sip%3A2589@172.33.16.220">sip:2589@172.33.16.220</a>>;tag=2e024a16<br>
To: "Teste Failover"<<a href="mailto:sip%3A2589@172.33.16.220">sip:2589@172.33.16.220</a>>;tag=as504997ac<br>Call-ID: YTk2NmFkZjZkYTA3MWE4NTg3ZjQ4ODIwMDQzMjM1OTE.<br>CSeq: 1 REGISTER<br>Server: Asterisk PBX 1.6.1.0-rc4<br>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY<br>Supported: replaces, timer<br>WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="07312465"<br>Content-Length: 0<br><br>
<br><------------><br>Scheduling destruction of SIP dialog 'YTk2NmFkZjZkYTA3MWE4NTg3ZjQ4ODIwMDQzMjM1OTE.' in 32000 ms (Method: REGISTER)<br>voipfailover*CLI> <br><--- SIP read from UDP://<a href="http://172.33.33.150:38595">172.33.33.150:38595</a> ---><br>
REGISTER sip:172.33.16.220 SIP/2.0<br>Via: SIP/2.0/UDP 172.33.33.150:38595;branch=z9hG4bK-d8754z-43603b4ff51a741d-1---d8754z-;rport<br>Max-Forwards: 70<br>Contact: <sip:2589@172.33.33.150:38595;rinstance=b73d4d015d9d419d><br>
To: "Teste Failover"<<a href="mailto:sip%3A2589@172.33.16.220">sip:2589@172.33.16.220</a>><br>From: "Teste Failover"<<a href="mailto:sip%3A2589@172.33.16.220">sip:2589@172.33.16.220</a>>;tag=2e024a16<br>
Call-ID: YTk2NmFkZjZkYTA3MWE4NTg3ZjQ4ODIwMDQzMjM1OTE.<br>CSeq: 1 REGISTER<br>Expires: 3600<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO<br>User-Agent: X-Lite release 1100l stamp 47546<br>
Content-Length: 0<br><br><br><-------------><br>--- (12 headers 0 lines) ---<br>Sending to 172.33.33.150 : 38595 (NAT)<br><br><--- Transmitting (NAT) to <a href="http://172.33.33.150:38595">172.33.33.150:38595</a> ---><br>
SIP/2.0 401 Unauthorized<br>Via: SIP/2.0/UDP 172.33.33.150:38595;branch=z9hG4bK-d8754z-43603b4ff51a741d-1---d8754z-;received=172.33.33.150;rport=38595<br>From: "Teste Failover"<<a href="mailto:sip%3A2589@172.33.16.220">sip:2589@172.33.16.220</a>>;tag=2e024a16<br>
To: "Teste Failover"<<a href="mailto:sip%3A2589@172.33.16.220">sip:2589@172.33.16.220</a>>;tag=as504997ac<br>Call-ID: YTk2NmFkZjZkYTA3MWE4NTg3ZjQ4ODIwMDQzMjM1OTE.<br>CSeq: 1 REGISTER<br>Server: Asterisk PBX 1.6.1.0-rc4<br>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY<br>Supported: replaces, timer<br>WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="07312465"<br>Content-Length: 0<br><br>
<br><------------><br>Scheduling destruction of SIP dialog 'YTk2NmFkZjZkYTA3MWE4NTg3ZjQ4ODIwMDQzMjM1OTE.' in 32000 ms (Method: REGISTER)<br>voipfailover*CLI> <br><--- SIP read from UDP://<a href="http://172.33.33.150:38595">172.33.33.150:38595</a> ---><br>
REGISTER sip:172.33.16.220 SIP/2.0<br>Via: SIP/2.0/UDP 172.33.33.150:38595;branch=z9hG4bK-d8754z-43603b4ff51a741d-1---d8754z-;rport<br>Max-Forwards: 70<br>Contact: <sip:2589@172.33.33.150:38595;rinstance=b73d4d015d9d419d><br>
To: "Teste Failover"<<a href="mailto:sip%3A2589@172.33.16.220">sip:2589@172.33.16.220</a>><br>From: "Teste Failover"<<a href="mailto:sip%3A2589@172.33.16.220">sip:2589@172.33.16.220</a>>;tag=2e024a16<br>
Call-ID: YTk2NmFkZjZkYTA3MWE4NTg3ZjQ4ODIwMDQzMjM1OTE.<br>CSeq: 1 REGISTER<br>Expires: 3600<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO<br>User-Agent: X-Lite release 1100l stamp 47546<br>
Content-Length: 0<br><br><br><-------------><br>--- (12 headers 0 lines) ---<br>Sending to 172.33.33.150 : 38595 (NAT)<br><br><--- Transmitting (NAT) to <a href="http://172.33.33.150:38595">172.33.33.150:38595</a> ---><br>
SIP/2.0 401 Unauthorized<br>Via: SIP/2.0/UDP 172.33.33.150:38595;branch=z9hG4bK-d8754z-43603b4ff51a741d-1---d8754z-;received=172.33.33.150;rport=38595<br>From: "Teste Failover"<<a href="mailto:sip%3A2589@172.33.16.220">sip:2589@172.33.16.220</a>>;tag=2e024a16<br>
To: "Teste Failover"<<a href="mailto:sip%3A2589@172.33.16.220">sip:2589@172.33.16.220</a>>;tag=as504997ac<br>Call-ID: YTk2NmFkZjZkYTA3MWE4NTg3ZjQ4ODIwMDQzMjM1OTE.<br>CSeq: 1 REGISTER<br>Server: Asterisk PBX 1.6.1.0-rc4<br>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY<br>Supported: replaces, timer<br>WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="07312465"<br>Content-Length: 0<br><br>
<br><------------><br>Scheduling destruction of SIP dialog 'YTk2NmFkZjZkYTA3MWE4NTg3ZjQ4ODIwMDQzMjM1OTE.' in 32000 ms (Method: REGISTER)<br>voipfailover*CLI> <br><--- SIP read from UDP://<a href="http://172.33.33.150:38595">172.33.33.150:38595</a> ---><br>
REGISTER sip:172.33.16.220 SIP/2.0<br>Via: SIP/2.0/UDP 172.33.33.150:38595;branch=z9hG4bK-d8754z-43603b4ff51a741d-1---d8754z-;rport<br>Max-Forwards: 70<br>Contact: <sip:2589@172.33.33.150:38595;rinstance=b73d4d015d9d419d><br>
To: "Teste Failover"<<a href="mailto:sip%3A2589@172.33.16.220">sip:2589@172.33.16.220</a>><br>From: "Teste Failover"<<a href="mailto:sip%3A2589@172.33.16.220">sip:2589@172.33.16.220</a>>;tag=2e024a16<br>
Call-ID: YTk2NmFkZjZkYTA3MWE4NTg3ZjQ4ODIwMDQzMjM1OTE.<br>CSeq: 1 REGISTER<br>Expires: 3600<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO<br>User-Agent: X-Lite release 1100l stamp 47546<br>
Content-Length: 0<br><br><br><-------------><br>--- (12 headers 0 lines) ---<br>Sending to 172.33.33.150 : 38595 (NAT)<br><br><--- Transmitting (NAT) to <a href="http://172.33.33.150:38595">172.33.33.150:38595</a> ---><br>
SIP/2.0 401 Unauthorized<br>Via: SIP/2.0/UDP 172.33.33.150:38595;branch=z9hG4bK-d8754z-43603b4ff51a741d-1---d8754z-;received=172.33.33.150;rport=38595<br>From: "Teste Failover"<<a href="mailto:sip%3A2589@172.33.16.220">sip:2589@172.33.16.220</a>>;tag=2e024a16<br>
To: "Teste Failover"<<a href="mailto:sip%3A2589@172.33.16.220">sip:2589@172.33.16.220</a>>;tag=as504997ac<br>Call-ID: YTk2NmFkZjZkYTA3MWE4NTg3ZjQ4ODIwMDQzMjM1OTE.<br>CSeq: 1 REGISTER<br>Server: Asterisk PBX 1.6.1.0-rc4<br>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY<br>Supported: replaces, timer<br>WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="07312465"<br>Content-Length: 0<br><br>
<br><------------><br>Scheduling destruction of SIP dialog 'YTk2NmFkZjZkYTA3MWE4NTg3ZjQ4ODIwMDQzMjM1OTE.' in 32000 ms (Method: REGISTER)<br>voipfailover*CLI> <br><br>Os dados cadastrados no x-lite estão corretos, tanto que se tento registrar direto no host eu consigo, alguma dica do que pode ser?<br>
<br>Abs,<br>Weder <br>