<div dir="ltr">sip-server*CLI><br><-- SIP read from <a href="http://201.48.113.140:5060">201.48.113.140:5060</a>:<br>INVITE <a href="mailto:sip%3A12013198@189.50.34.1">sip:12013198@189.50.34.1</a> SIP/2.0<br>Record-Route: <sip:<a href="http://201.48.113.140">201.48.113.140</a>;lr=on><br>
Via: SIP/2.0/UDP <a href="http://201.48.113.140">201.48.113.140</a>;branch=z9hG4bK154d.332bd161.0<br>Via: SIP/2.0/UDP 201.48.113.130:5060;branch=z9hG4bK1ad62475;rport=5060<br>From: "1548394571" <<a href="mailto:sip%3A1548394571@201.48.113.130">sip:1548394571@201.48.113.130</a>>;tag=as5ba152da<br>
To: <<a href="mailto:sip%3A12013198@201.48.113.140">sip:12013198@201.48.113.140</a>><br>Contact: <<a href="mailto:sip%3A1548394571@201.48.113.130">sip:1548394571@201.48.113.130</a>><br>Call-ID: <a href="mailto:40546f06637dc18c6b6bc8080106f0a5@201.48.113.130">40546f06637dc18c6b6bc8080106f0a5@201.48.113.130</a><br>
CSeq: 102 INVITE<br>User-Agent: VoIP_CTBC_1.0<br>Max-Forwards: 69<br>Date: Wed, 30 Jul 2008 18:35:56 GMT<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY<br>Supported: replaces<br>Content-Type: application/sdp<br>
Content-Length: 291<br><br>v=0<br>o=root 21318 21318 IN IP4 <a href="http://201.48.113.130">201.48.113.130</a><br>s=session<br>c=IN IP4 <a href="http://201.48.113.130">201.48.113.130</a><br>t=0 0<br>m=audio 14084 RTP/AVP 0 8 3 101<br>
a=rtpmap:0 PCMU/8000<br>a=rtpmap:8 PCMA/8000<br>a=rtpmap:3 GSM/8000<br>a=rtpmap:101 telephone-event/8000<br>a=fmtp:101 0-16<br>a=silenceSupp:off - - - -<br>a=ptime:20<br>a=sendrecv<br><br>--- (16 headers 14 lines) ---<br>
Using INVITE request as basis request - <a href="mailto:40546f06637dc18c6b6bc8080106f0a5@201.48.113.130">40546f06637dc18c6b6bc8080106f0a5@201.48.113.130</a><br>Sending to <a href="http://201.48.113.140">201.48.113.140</a> : 5060 (non-NAT)<br>
Found peer '12015331_ctbc_trunk'<br>Found RTP audio format 0<br>Found RTP audio format 8<br>Found RTP audio format 3<br>Found RTP audio format 101<br>Peer audio RTP is at port <a href="http://201.48.113.130:14084">201.48.113.130:14084</a><br>
Found description format PCMU<br>Found description format PCMA<br>Found description format GSM<br>Found description format telephone-event<br>Capabilities: us - 0x50e (gsm|ulaw|alaw|g729|ilbc), peer - audio=0xe (gsm|ulaw|alaw)/video=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)<br>
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)<br>Looking for 12013198 in entrante (domain <a href="http://189.50.34.1">189.50.34.1</a>)<br>list_route: hop: <sip:<a href="http://201.48.113.140">201.48.113.140</a>;lr=on><br>
Transmitting (NAT) to <a href="http://201.48.113.140:5060">201.48.113.140:5060</a>:<br>SIP/2.0 100 Trying<br>Via: SIP/2.0/UDP <a href="http://201.48.113.140">201.48.113.140</a>;branch=z9hG4bK154d.332bd161.0;received=<a href="http://201.48.113.140">201.48.113.140</a><br>
Via: SIP/2.0/UDP 201.48.113.130:5060;branch=z9hG4bK1ad62475;rport=5060<br>From: "1548394571" <<a href="mailto:sip%3A1548394571@201.48.113.130">sip:1548394571@201.48.113.130</a>>;tag=as5ba152da<br>To: <<a href="mailto:sip%3A12013198@201.48.113.140">sip:12013198@201.48.113.140</a>><br>
Call-ID: <a href="mailto:40546f06637dc18c6b6bc8080106f0a5@201.48.113.130">40546f06637dc18c6b6bc8080106f0a5@201.48.113.130</a><br>CSeq: 102 INVITE<br>User-Agent: iPBX HostingVoip<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY<br>
Contact: <<a href="mailto:sip%3A12013198@189.50.34.1">sip:12013198@189.50.34.1</a>><br>Content-Length: 0<br><br><br>---<br>We're at <a href="http://189.50.34.1">189.50.34.1</a> port 19498<br>Adding codec 0x4 (ulaw) to SDP<br>
Adding codec 0x8 (alaw) to SDP<br>Adding codec 0x2 (gsm) to SDP<br>Adding non-codec 0x1 (telephone-event) to SDP<br>Reliably Transmitting (NAT) to <a href="http://201.48.113.140:5060">201.48.113.140:5060</a>:<br>SIP/2.0 200 OK<br>
Via: SIP/2.0/UDP <a href="http://201.48.113.140">201.48.113.140</a>;branch=z9hG4bK154d.332bd161.0;received=<a href="http://201.48.113.140">201.48.113.140</a><br>Via: SIP/2.0/UDP 201.48.113.130:5060;branch=z9hG4bK1ad62475;rport=5060<br>
Record-Route: <sip:<a href="http://201.48.113.140">201.48.113.140</a>;lr=on><br>From: "1548394571" <<a href="mailto:sip%3A1548394571@201.48.113.130">sip:1548394571@201.48.113.130</a>>;tag=as5ba152da<br>
To: <<a href="mailto:sip%3A12013198@201.48.113.140">sip:12013198@201.48.113.140</a>>;tag=as7b63f52d<br>Call-ID: <a href="mailto:40546f06637dc18c6b6bc8080106f0a5@201.48.113.130">40546f06637dc18c6b6bc8080106f0a5@201.48.113.130</a><br>
CSeq: 102 INVITE<br>User-Agent: iPBX HostingVoip<br>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY<br>Contact: <<a href="mailto:sip%3A12013198@189.50.34.1">sip:12013198@189.50.34.1</a>><br>Content-Type: application/sdp<br>
Content-Length: 261<br><br>v=0<br>o=root 3739 3739 IN IP4 <a href="http://189.50.34.1">189.50.34.1</a><br>s=session<br>c=IN IP4 <a href="http://189.50.34.1">189.50.34.1</a><br>t=0 0<br>m=audio 19498 RTP/AVP 0 8 3 101<br>a=rtpmap:0 PCMU/8000<br>
a=rtpmap:8 PCMA/8000<br>a=rtpmap:3 GSM/8000<br>a=rtpmap:101 telephone-event/8000<br>a=fmtp:101 0-16<br>a=silenceSupp:off - - - -<br><br>---<br>sip-server*CLI><br><-- SIP read from <a href="http://201.48.113.140:5060">201.48.113.140:5060</a>:<br>
ACK <a href="mailto:sip%3A12013198@189.50.34.1">sip:12013198@189.50.34.1</a> SIP/2.0<br>Record-Route: <sip:<a href="http://201.48.113.140">201.48.113.140</a>;lr=on><br>Via: SIP/2.0/UDP <a href="http://201.48.113.140">201.48.113.140</a>;branch=z9hG4bK154d.332bd161.2<br>
Via: SIP/2.0/UDP 201.48.113.130:5060;branch=z9hG4bK673787b8;rport=5060<br>From: "1548394571" <<a href="mailto:sip%3A1548394571@201.48.113.130">sip:1548394571@201.48.113.130</a>>;tag=as5ba152da<br>To: <<a href="mailto:sip%3A12013198@201.48.113.140">sip:12013198@201.48.113.140</a>>;tag=as7b63f52d<br>
Contact: <<a href="mailto:sip%3A1548394571@201.48.113.130">sip:1548394571@201.48.113.130</a>><br>Call-ID: <a href="mailto:40546f06637dc18c6b6bc8080106f0a5@201.48.113.130">40546f06637dc18c6b6bc8080106f0a5@201.48.113.130</a><br>
CSeq: 102 ACK<br>User-Agent: VoIP_CTBC_1.0<br>Max-Forwards: 69<br>Content-Length: 0<br><br><br>--- (12 headers 0 lines) ---<br><br><div class="gmail_quote">2008/7/30 Ricardo Landim <span dir="ltr"><<a href="mailto:pangole@bol.com.br">pangole@bol.com.br</a>></span><br>
<blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;"><div dir="ltr">Manda a sinalização SIP...<br><br><div class="gmail_quote">2008/7/30 hamasterisk <span dir="ltr"><<a href="mailto:listas.publicas@gmail.com" target="_blank">listas.publicas@gmail.com</a>></span><br>
<blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;"><div><div></div><div class="Wj3C7c">
<div dir="ltr"> Caros colegas, nossa empresa tem uma central de atendimento via SIP, usando números entrante da CTBC.<br> Temos cerca de 25 números entrantes em cidades que temos clientes, de forma a facilitar o suporte.<br>
Me incumbiram de levantar os dados a respeito de quantas chamadas cada número vem recebendo. Contudo, pelo que estive monitorando no Debug, a CTBC envia a chamada para o nosso login e não para o número que foi discado. Sendo assim, não tenho a menor idéia de como levantar esses dados pra gerência.<br>
Algúem tem alguma vaga idéia? No estado em que estou, qualquer coisa ajuda.<br></div>
<br></div></div><div class="Ih2E3d">_______________________________________________<br>
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Lista de discussões AsteriskBrasil.org<br>
<a href="mailto:AsteriskBrasil@listas.asteriskbrasil.org" target="_blank">AsteriskBrasil@listas.asteriskbrasil.org</a><br>
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<br>_______________________________________________<br>
Compre uma camiseta da AsteriskBrasil.org!<br>
<a href="http://www.voipmania.com.br" target="_blank">http://www.voipmania.com.br</a><br>
<br>
Acesse o canal IRC de discussão sobre Asterisk em Português Brasileiro na rede Freenode.net: #asterisk-br<br>
_______________________________________________<br>
Lista de discussões AsteriskBrasil.org<br>
<a href="mailto:AsteriskBrasil@listas.asteriskbrasil.org">AsteriskBrasil@listas.asteriskbrasil.org</a><br>
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