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<BR>
Caro Felipe,<BR>
<BR>
Como solicitado em e-mail anterior, segue o Degug.<BR>
<BR>
<BR>
<BR>
Connected to Asterisk 1.4.26.2 currently running on emax (pid = 5304)<BR><BR>
####Mensagem gerada quando ligo para o ramal 1000 no Asterisk<BR>
[Sep 29 10:56:56] WARNING[5325]: chan_sip.c:14518 handle_request_invite: Received SIP INVITE with unsupported required extension: 100rel<BR><BR>
<BR>
#####DEBUG##################<BR>
emax*CLI> sip set debug ip 10.201.201.250<BR>SIP Debugging Enabled for IP: 10.201.201.250<BR>emax*CLI> <BR><--- SIP read from 10.201.201.250:5060 ---><BR>INVITE sip:+1000@10.201.201.1:5060 SIP/2.0<BR>Via: SIP/2.0/UDP 10.201.201.250:5060;branch=z9hG4bK-17d4ac1daa22-208<BR>From: "+2650" <sip:+2650@10.201.201.1:5060>;tag=20934ac1daa21-208<BR>To: <sip:+1000@10.201.201.1:5060><BR>Call-ID: <A href="mailto:2ff-4ac1daa2-0-208@10.201.201.250">2ff-4ac1daa2-0-208@10.201.201.250</A><BR>CSeq: 1 INVITE<BR>Contact: <sip:+2650@10.201.201.250:5060><BR>Allow: INVITE, ACK, CANCEL, BYE, UPDATE, PRACK<BR>user-agent: IPS<BR>Max-Forwards: 70<BR>Session-Expires: 1800<BR>Supported: timer,100rel<BR>Require: 100rel<BR>Privacy: none<BR>P-Asserted-Identity: "+2650" <sip:+2650@10.201.201.1:5060><BR>Content-Type: application/sdp<BR>Content-Length: 181<BR>
v=0<BR>o=IPS 23502 0 IN IP4 10.201.201.250<BR>s=IPS<BR>c=IN IP4 10.201.201.250<BR>t=0 0<BR>m=audio 10050 RTP/AVP 0 101<BR>a=rtpmap:0 PCMU/8000/1<BR>a=ptime:20<BR>a=rtpmap:101 telephone-event/8000<BR>
<-------------><BR>--- (17 headers 9 lines) ---<BR>
<--- Transmitting (no NAT) to 10.201.201.250:5060 ---><BR>SIP/2.0 420 Bad extension (unsupported)<BR>Via: SIP/2.0/UDP 10.201.201.250:5060;branch=z9hG4bK-17d4ac1daa22-208;received=10.201.201.250<BR>From: "+2650" <sip:+2650@10.201.201.1:5060>;tag=20934ac1daa21-208<BR>To: <sip:+1000@10.201.201.1:5060>;tag=as4153cf25<BR>Call-ID: <A href="mailto:2ff-4ac1daa2-0-208@10.201.201.250">2ff-4ac1daa2-0-208@10.201.201.250</A><BR>CSeq: 1 INVITE<BR>User-Agent: Asterisk PBX<BR>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO<BR>Supported: replaces<BR>Date: Tue, 29 Sep 2009 13:57:03 GMT<BR>Unsupported: 100rel<BR>Content-Length: 0<BR>
<BR><------------><BR>[Sep 29 10:57:03] WARNING[5325]: chan_sip.c:14518 handle_request_invite: Received SIP INVITE with unsupported required extension: 100rel<BR>Scheduling destruction of SIP dialog <A href="mailto:'2ff-4ac1daa2-0-208@10.201.201.250'">'2ff-4ac1daa2-0-208@10.201.201.250'</A> in 32000 ms (Method: INVITE)<BR>emax*CLI> <BR><--- SIP read from 10.201.201.250:5060 ---><BR>ACK sip:+1000@10.201.201.1:5060 SIP/2.0<BR>Via: SIP/2.0/UDP 10.201.201.250:5060;branch=z9hG4bK-17d4ac1daa22-208<BR>From: "+2650" <sip:+2650@10.201.201.1:5060>;tag=20934ac1daa21-208<BR>To: <sip:+1000@10.201.201.1:5060>;tag=as4153cf25<BR>Call-ID: <A href="mailto:2ff-4ac1daa2-0-208@10.201.201.250">2ff-4ac1daa2-0-208@10.201.201.250</A><BR>CSeq: 1 ACK<BR>user-agent: IPS<BR>Max-Forwards: 70<BR>Content-Length: 0<BR>
<BR><-------------><BR>--- (9 headers 0 lines) ---<BR>Really destroying SIP dialog <A href="mailto:'2ff-4ac1daa2-0-208@10.201.201.250'">'2ff-4ac1daa2-0-208@10.201.201.250'</A> Method: ACK<BR>Reliably Transmitting (no NAT) to 10.201.201.250:5060:<BR>OPTIONS sip:10.201.201.250 SIP/2.0<BR>Via: SIP/2.0/UDP 10.201.201.1:5060;branch=z9hG4bK4f8f17f8;rport<BR>From: "asterisk" <sip:asterisk@10.201.201.1>;tag=as7754e9ac<BR>To: <sip:10.201.201.250><BR>Contact: <sip:asterisk@10.201.201.1><BR>Call-ID: <A href="mailto:3353e7306217b72d0a0f992d6e530f39@10.201.201.1">3353e7306217b72d0a0f992d6e530f39@10.201.201.1</A><BR>CSeq: 102 OPTIONS<BR>User-Agent: Asterisk PBX<BR>Max-Forwards: 70<BR>Date: Tue, 29 Sep 2009 13:57:04 GMT<BR>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO<BR>Supported: replaces<BR>Content-Length: 0<BR>
<BR>---<BR>emax*CLI> <BR><--- SIP read from 10.201.201.250:5060 ---><BR>SIP/2.0 200 OK<BR>Via: SIP/2.0/UDP 10.201.201.1:5060;branch=z9hG4bK4f8f17f8;rport<BR>From: "asterisk" <sip:asterisk@10.201.201.1>;tag=as7754e9ac<BR>To: <sip:10.201.201.250><BR>Call-ID: <A href="mailto:3353e7306217b72d0a0f992d6e530f39@10.201.201.1">3353e7306217b72d0a0f992d6e530f39@10.201.201.1</A><BR>CSeq: 102 OPTIONS<BR>Allow: INVITE, ACK, CANCEL, BYE, UPDATE, PRACK<BR>user-agent: IPS<BR>Supported: timer,100rel<BR>Content-Length: 0<BR>
<BR><-------------><BR>--- (10 headers 0 lines) ---<BR>Really destroying SIP dialog <A href="mailto:'3353e7306217b72d0a0f992d6e530f39@10.201.201.1'">'3353e7306217b72d0a0f992d6e530f39@10.201.201.1'</A> Method: OPTIONS<BR>emax*CLI> <BR><--- SIP read from 10.201.201.250:5060 ---><BR>INVITE sip:+1000@10.201.201.1:5060 SIP/2.0<BR>Via: SIP/2.0/UDP 10.201.201.250:5060;branch=z9hG4bK-7a884ac1daac2-210<BR>From: "+2650" <sip:+2650@10.201.201.1:5060>;tag=4c474ac1daac1-210<BR>To: <sip:+1000@10.201.201.1:5060><BR>Call-ID: <A href="mailto:7d3e-4ac1daac-0-210@10.201.201.250">7d3e-4ac1daac-0-210@10.201.201.250</A><BR>CSeq: 1 INVITE<BR>Contact: <sip:+2650@10.201.201.250:5060><BR>Allow: INVITE, ACK, CANCEL, BYE, UPDATE, PRACK<BR>user-agent: IPS<BR>Max-Forwards: 70<BR>Session-Expires: 1800<BR>Supported: timer,100rel<BR>Require: 100rel<BR>Privacy: none<BR>P-Asserted-Identity: "+2650" <sip:+2650@10.201.201.1:5060><BR>Content-Type: application/sdp<BR>Content-Length: 181<BR>
v=0<BR>o=IPS 13436 0 IN IP4 10.201.201.250<BR>s=IPS<BR>c=IN IP4 10.201.201.250<BR>t=0 0<BR>m=audio 10060 RTP/AVP 0 101<BR>a=rtpmap:0 PCMU/8000/1<BR>a=ptime:20<BR>a=rtpmap:101 telephone-event/8000<BR>
<-------------><BR>--- (17 headers 9 lines) ---<BR>
<--- Transmitting (no NAT) to 10.201.201.250:5060 ---><BR>SIP/2.0 420 Bad extension (unsupported)<BR>Via: SIP/2.0/UDP 10.201.201.250:5060;branch=z9hG4bK-7a884ac1daac2-210;received=10.201.201.250<BR>From: "+2650" <sip:+2650@10.201.201.1:5060>;tag=4c474ac1daac1-210<BR>To: <sip:+1000@10.201.201.1:5060>;tag=as3b199582<BR>Call-ID: <A href="mailto:7d3e-4ac1daac-0-210@10.201.201.250">7d3e-4ac1daac-0-210@10.201.201.250</A><BR>CSeq: 1 INVITE<BR>User-Agent: Asterisk PBX<BR>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO<BR>Supported: replaces<BR>Date: Tue, 29 Sep 2009 13:57:12 GMT<BR>Unsupported: 100rel<BR>Content-Length: 0<BR>
<BR><------------><BR>[Sep 29 10:57:12] WARNING[5325]: chan_sip.c:14518 handle_request_invite: Received SIP INVITE with unsupported required extension: 100rel<BR>Scheduling destruction of SIP dialog <A href="mailto:'7d3e-4ac1daac-0-210@10.201.201.250'">'7d3e-4ac1daac-0-210@10.201.201.250'</A> in 32000 ms (Method: INVITE)<BR>emax*CLI> <BR><--- SIP read from 10.201.201.250:5060 ---><BR>ACK sip:+1000@10.201.201.1:5060 SIP/2.0<BR>Via: SIP/2.0/UDP 10.201.201.250:5060;branch=z9hG4bK-7a884ac1daac2-210<BR>From: "+2650" <sip:+2650@10.201.201.1:5060>;tag=4c474ac1daac1-210<BR>To: <sip:+1000@10.201.201.1:5060>;tag=as3b199582<BR>Call-ID: <A href="mailto:7d3e-4ac1daac-0-210@10.201.201.250">7d3e-4ac1daac-0-210@10.201.201.250</A><BR>CSeq: 1 ACK<BR>user-agent: IPS<BR>Max-Forwards: 70<BR>Content-Length: 0<BR>
<BR><-------------><BR>--- (9 headers 0 lines) ---<BR>Really destroying SIP dialog <A href="mailto:'7d3e-4ac1daac-0-210@10.201.201.250'">'7d3e-4ac1daac-0-210@10.201.201.250'</A> Method: ACK<BR>Reliably Transmitting (no NAT) to 10.201.201.250:5060:<BR>OPTIONS sip:10.201.201.250 SIP/2.0<BR>Via: SIP/2.0/UDP 10.201.201.1:5060;branch=z9hG4bK6b8cee75;rport<BR>From: "asterisk" <sip:asterisk@10.201.201.1>;tag=as7c556238<BR>To: <sip:10.201.201.250><BR>Contact: <sip:asterisk@10.201.201.1><BR>Call-ID: <A href="mailto:04fb1766622cc1855bbf67905f6c3c88@10.201.201.1">04fb1766622cc1855bbf67905f6c3c88@10.201.201.1</A><BR>CSeq: 102 OPTIONS<BR>User-Agent: Asterisk PBX<BR>Max-Forwards: 70<BR>Date: Tue, 29 Sep 2009 13:57:13 GMT<BR>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO<BR>Supported: replaces<BR>Content-Length: 0<BR>
<BR>---<BR>emax*CLI> <BR><--- SIP read from 10.201.201.250:5060 ---><BR>SIP/2.0 200 OK<BR>Via: SIP/2.0/UDP 10.201.201.1:5060;branch=z9hG4bK6b8cee75;rport<BR>From: "asterisk" <sip:asterisk@10.201.201.1>;tag=as7c556238<BR>To: <sip:10.201.201.250><BR>Call-ID: <A href="mailto:04fb1766622cc1855bbf67905f6c3c88@10.201.201.1">04fb1766622cc1855bbf67905f6c3c88@10.201.201.1</A><BR>CSeq: 102 OPTIONS<BR>Allow: INVITE, ACK, CANCEL, BYE, UPDATE, PRACK<BR>user-agent: IPS<BR>Supported: timer,100rel<BR>Content-Length: 0<BR>
<BR><-------------><BR>--- (10 headers 0 lines) ---<BR>Really destroying SIP dialog <A href="mailto:'04fb1766622cc1855bbf67905f6c3c88@10.201.201.1'">'04fb1766622cc1855bbf67905f6c3c88@10.201.201.1'</A> Method: OPTIONS<BR>Reliably Transmitting (no NAT) to 10.201.201.250:5060:<BR>OPTIONS sip:10.201.201.250 SIP/2.0<BR>Via: SIP/2.0/UDP 10.201.201.1:5060;branch=z9hG4bK5e2e8d2f;rport<BR>From: "asterisk" <sip:asterisk@10.201.201.1>;tag=as3429ba79<BR>To: <sip:10.201.201.250><BR>Contact: <sip:asterisk@10.201.201.1><BR>Call-ID: <A href="mailto:22c4ca806944474364ca70753048a417@10.201.201.1">22c4ca806944474364ca70753048a417@10.201.201.1</A><BR>CSeq: 102 OPTIONS<BR>User-Agent: Asterisk PBX<BR>Max-Forwards: 70<BR>Date: Tue, 29 Sep 2009 13:57:13 GMT<BR>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO<BR>Supported: replaces<BR>Content-Length: 0<BR>
<BR>---<BR>Reliably Transmitting (no NAT) to 10.201.201.250:5060:<BR>OPTIONS sip:10.201.201.250 SIP/2.0<BR>Via: SIP/2.0/UDP 10.201.201.1:5060;branch=z9hG4bK3a337600;rport<BR>From: "asterisk" <sip:asterisk@10.201.201.1>;tag=as18f7ee3d<BR>To: <sip:10.201.201.250><BR>Contact: <sip:asterisk@10.201.201.1><BR>Call-ID: <A href="mailto:626ceab217965f8b39c3ae212a74ca25@10.201.201.1">626ceab217965f8b39c3ae212a74ca25@10.201.201.1</A><BR>CSeq: 102 OPTIONS<BR>User-Agent: Asterisk PBX<BR>Max-Forwards: 70<BR>Date: Tue, 29 Sep 2009 13:57:14 GMT<BR>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO<BR>Supported: replaces<BR>Content-Length: 0<BR>
<BR>---<BR>emax*CLI> <BR><--- SIP read from 10.201.201.250:5060 ---><BR>SIP/2.0 200 OK<BR>Via: SIP/2.0/UDP 10.201.201.1:5060;branch=z9hG4bK5e2e8d2f;rport<BR>From: "asterisk" <sip:asterisk@10.201.201.1>;tag=as3429ba79<BR>To: <sip:10.201.201.250><BR>Call-ID: <A href="mailto:22c4ca806944474364ca70753048a417@10.201.201.1">22c4ca806944474364ca70753048a417@10.201.201.1</A><BR>CSeq: 102 OPTIONS<BR>Allow: INVITE, ACK, CANCEL, BYE, UPDATE, PRACK<BR>user-agent: IPS<BR>Supported: timer,100rel<BR>Content-Length: 0<BR>
<BR><-------------><BR>--- (10 headers 0 lines) ---<BR>Really destroying SIP dialog <A href="mailto:'22c4ca806944474364ca70753048a417@10.201.201.1'">'22c4ca806944474364ca70753048a417@10.201.201.1'</A> Method: OPTIONS<BR>emax*CLI> <BR><--- SIP read from 10.201.201.250:5060 ---><BR>SIP/2.0 200 OK<BR>Via: SIP/2.0/UDP 10.201.201.1:5060;branch=z9hG4bK3a337600;rport<BR>From: "asterisk" <sip:asterisk@10.201.201.1>;tag=as18f7ee3d<BR>To: <sip:10.201.201.250><BR>Call-ID: <A href="mailto:626ceab217965f8b39c3ae212a74ca25@10.201.201.1">626ceab217965f8b39c3ae212a74ca25@10.201.201.1</A><BR>CSeq: 102 OPTIONS<BR>Allow: INVITE, ACK, CANCEL, BYE, UPDATE, PRACK<BR>user-agent: IPS<BR>Supported: timer,100rel<BR>Content-Length: 0<BR>
<BR><-------------><BR>--- (10 headers 0 lines) ---<BR>Really destroying SIP dialog <A href="mailto:'626ceab217965f8b39c3ae212a74ca25@10.201.201.1'">'626ceab217965f8b39c3ae212a74ca25@10.201.201.1'</A> Method: OPTIONS<BR>Reliably Transmitting (no NAT) to 10.201.201.250:5060:<BR>OPTIONS sip:10.201.201.250 SIP/2.0<BR>Via: SIP/2.0/UDP 10.201.201.1:5060;branch=z9hG4bK66ad830a;rport<BR>From: "asterisk" <sip:asterisk@10.201.201.1>;tag=as7eecb8c9<BR>To: <sip:10.201.201.250><BR>Contact: <sip:asterisk@10.201.201.1><BR>Call-ID: <A href="mailto:2dedd7755f6bd5eb19473f6602150428@10.201.201.1">2dedd7755f6bd5eb19473f6602150428@10.201.201.1</A><BR>CSeq: 102 OPTIONS<BR>User-Agent: Asterisk PBX<BR>Max-Forwards: 70<BR>Date: Tue, 29 Sep 2009 13:57:14 GMT<BR>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO<BR>Supported: replaces<BR>Content-Length: 0<BR>
<BR>---<BR>emax*CLI> <BR><--- SIP read from 10.201.201.250:5060 ---><BR>SIP/2.0 200 OK<BR>Via: SIP/2.0/UDP 10.201.201.1:5060;branch=z9hG4bK66ad830a;rport<BR>From: "asterisk" <sip:asterisk@10.201.201.1>;tag=as7eecb8c9<BR>To: <sip:10.201.201.250><BR>Call-ID: <A href="mailto:2dedd7755f6bd5eb19473f6602150428@10.201.201.1">2dedd7755f6bd5eb19473f6602150428@10.201.201.1</A><BR>CSeq: 102 OPTIONS<BR>Allow: INVITE, ACK, CANCEL, BYE, UPDATE, PRACK<BR>user-agent: IPS<BR>Supported: timer,100rel<BR>Content-Length: 0<BR>
<BR><-------------><BR>--- (10 headers 0 lines) ---<BR>Really destroying SIP dialog <A href="mailto:'2dedd7755f6bd5eb19473f6602150428@10.201.201.1'">'2dedd7755f6bd5eb19473f6602150428@10.201.201.1'</A> Method: OPTIONS<BR>Reliably Transmitting (no NAT) to 10.201.201.250:5060:<BR>OPTIONS sip:1000@10.201.201.6:56702;rinstance=09d7015650b931f8 SIP/2.0<BR>Via: SIP/2.0/UDP 10.201.201.1:5060;branch=z9hG4bK7ac0c8f5;rport<BR>From: "asterisk" <sip:asterisk@10.201.201.1>;tag=as0d2c4b3a<BR>To: <sip:1000@10.201.201.6:56702;rinstance=09d7015650b931f8><BR>Contact: <sip:asterisk@10.201.201.1><BR>Call-ID: <A href="mailto:59d07ee263e0180f5233274833d14d1d@10.201.201.1">59d07ee263e0180f5233274833d14d1d@10.201.201.1</A><BR>CSeq: 102 OPTIONS<BR>User-Agent: Asterisk PBX<BR>Max-Forwards: 70<BR>Date: Tue, 29 Sep 2009 13:57:14 GMT<BR>llow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO<BR>Supported: replaces<BR>Content-Length: 0<BR>
<BR>---<BR>emax*CLI> <BR><--- SIP read from 10.201.201.250:5060 ---><BR>SIP/2.0 200 OK<BR>Via: SIP/2.0/UDP 10.201.201.1:5060;branch=z9hG4bK7ac0c8f5;rport<BR>From: "asterisk" <sip:asterisk@10.201.201.1>;tag=as0d2c4b3a<BR>To: <sip:1000@10.201.201.6:56702;rinstance=09d7015650b931f8><BR>Call-ID: <A href="mailto:59d07ee263e0180f5233274833d14d1d@10.201.201.1">59d07ee263e0180f5233274833d14d1d@10.201.201.1</A><BR>CSeq: 102 OPTIONS<BR>Allow: INVITE, ACK, CANCEL, BYE, UPDATE, PRACK<BR>user-agent: IPS<BR>Supported: timer,100rel<BR>Content-Length: 0<BR>
<BR><-------------><BR>--- (10 headers 0 lines) ---<BR>Really destroying SIP dialog <A href="mailto:'59d07ee263e0180f5233274833d14d1d@10.201.201.1'">'59d07ee263e0180f5233274833d14d1d@10.201.201.1'</A> Method: OPTIONS<BR>Reliably Transmitting (no NAT) to 10.201.201.250:5060:<BR>OPTIONS sip:10.201.201.250 SIP/2.0<BR>Via: SIP/2.0/UDP 10.201.201.1:5060;branch=z9hG4bK4b9d60e2;rport<BR>From: "asterisk" <sip:asterisk@10.201.201.1>;tag=as7becd38c<BR>To: <sip:10.201.201.250><BR>Contact: <sip:asterisk@10.201.201.1><BR>Call-ID: <A href="mailto:31bc7fe044f8ea6b6312f10a5f685342@10.201.201.1">31bc7fe044f8ea6b6312f10a5f685342@10.201.201.1</A><BR>CSeq: 102 OPTIONS<BR>User-Agent: Asterisk PBX<BR>Max-Forwards: 70<BR>Date: Tue, 29 Sep 2009 13:57:16 GMT<BR>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO<BR>Supported: replaces<BR>Content-Length: 0<BR>
<BR>---<BR>emax*CLI> sip set debug ip 10.201.201.250<BR><--- SIP read from 10.201.201.250:5060 ---><BR>SIP/2.0 200 OK<BR>Via: SIP/2.0/UDP 10.201.201.1:5060;branch=z9hG4bK4b9d60e2;rport<BR>From: "asterisk" <sip:asterisk@10.201.201.1>;tag=as7becd38c<BR>To: <sip:10.201.201.250><BR>Call-ID: <A href="mailto:31bc7fe044f8ea6b6312f10a5f685342@10.201.201.1">31bc7fe044f8ea6b6312f10a5f685342@10.201.201.1</A><BR>CSeq: 102 OPTIONS<BR>Allow: INVITE, ACK, CANCEL, BYE, UPDATE, PRACK<BR>user-agent: IPS<BR>Supported: timer,100rel<BR>Content-Length: 0<BR>
<BR><-------------><BR>--- (10 headers 0 lines) ---<BR>Really destroying SIP dialog <A href="mailto:'31bc7fe044f8ea6b6312f10a5f685342@10.201.201.1'">'31bc7fe044f8ea6b6312f10a5f685342@10.201.201.1'</A> Method: OPTIONS<BR>emax*CLI> sip set debug off<BR>SIP Debugging Disabled<BR>emax*CLI> <BR><BR><BR>Att,<BR> <BR>Flavio Roberto Miranda<BR>MSN:flaviormiranda@hotmail.com<BR>Skype: flaviormiranda<BR><BR><BR> <br /><hr />Conheça os novos produtos Windows Live. <a href='http://www.windowslive.com.br' target='_new'>Clique aqui!</a></body>
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