<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META content="text/html; charset=iso-8859-1" http-equiv=Content-Type>
<STYLE>.hmmessage P {
PADDING-BOTTOM: 0px; MARGIN: 0px; PADDING-LEFT: 0px; PADDING-RIGHT: 0px; PADDING-TOP: 0px
}
BODY.hmmessage {
FONT-FAMILY: Verdana; FONT-SIZE: 10pt
}
</STYLE>
<META name=GENERATOR content="MSHTML 8.00.6001.18812"></HEAD>
<BODY class=hmmessage bgColor=#ffffff>
<DIV><FONT face=Arial>Verifque a configuração da central pelo que me parece ela
esta mandando 2650 para o asterisk e o mesmo esta aguardando 1000, isso ocorre
quando a central esta programada para mandar o MCDU e como o asterisk não
encontrou a extensão 2650 no contexto encerra a chamada</FONT></DIV>
<DIV><FONT face=Arial></FONT> </DIV>
<DIV>From: "+2650"
<sip:+2650@10.201.201.1:5060>;tag=20934ac1daa21-208<BR>To:
<sip:+1000@10.201.201.1:5060><BR></DIV>
<BLOCKQUOTE
style="BORDER-LEFT: #000000 2px solid; PADDING-LEFT: 5px; PADDING-RIGHT: 0px; MARGIN-LEFT: 5px; MARGIN-RIGHT: 0px">
<DIV style="FONT: 10pt arial">----- Original Message ----- </DIV>
<DIV
style="FONT: 10pt arial; BACKGROUND: #e4e4e4; font-color: black"><B>From:</B>
<A title=flaviormiranda@hotmail.com
href="mailto:flaviormiranda@hotmail.com">Flavio Miranda</A> </DIV>
<DIV style="FONT: 10pt arial"><B>To:</B> <A
title=asteriskbrasil@listas.asteriskbrasil.org
href="mailto:asteriskbrasil@listas.asteriskbrasil.org">Asterisk</A> </DIV>
<DIV style="FONT: 10pt arial"><B>Sent:</B> Tuesday, September 29, 2009 10:04
AM</DIV>
<DIV style="FONT: 10pt arial"><B>Subject:</B> [AsteriskBrasil] Ligaões de
Entrada</DIV>
<DIV><BR></DIV> <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><BR><BR>__________ Informação do NOD32 IMON 4465 (20090928)
__________<BR><BR>Esta mensagem foi verificada pelo NOD32 sistema
antivírus<BR><A href="http://www.eset.com.br">http://www.eset.com.br</A><BR>
<P>
<HR>
<P></P><BR>_______________________________________________<BR>http://www.voipmania.com.br<BR>Telefone
IP sem fio Gigaset A580IP por 6 x R$59,90. <BR>Promoção por tempo
limitado!<BR>Acesse agora
http://promo.voipmania.com.br<BR><BR>_______________________________________________<BR>Lista
de discussões
AsteriskBrasil.org<BR>AsteriskBrasil@listas.asteriskbrasil.org<BR>http://listas.asteriskbrasil.org/mailman/listinfo/asteriskbrasil<BR><BR>__________
Informação do NOD32 IMON 4465 (20090928) __________<BR><BR>Esta mensagem foi
verificada pelo NOD32 sistema
antivírus<BR>http://www.eset.com.br<BR><BR></BLOCKQUOTE></BODY></HTML>