O seu asterisk está tentando fazer a troca de CODECs e nao esta conseguindo todos tem que esta Habilitados no seu TElefone IP Voiper.<br><title>Configurando Codec</title><table id="table2" width="504" border="0" height="130">
<tbody>
<tr>
<td> exemplo : e os outros <br><br><br><br></td></tr>
<tr>
<td>
<form name="sipform" action="/sip.cgi" method="post">
<table id="table3" width="100%" border="0">
<tbody>
<tr>
<td colspan="2" bgcolor="#800000"><b><font size="2" color="#ffffff" face="Arial">Codec
Preferido</font></b></td></tr>
<tr>
<td width="27%" bgcolor="#ffffdd"><font size="2" face="Arial">Codec Preferido
1:</font></td>
<td width="71%" bgcolor="#ffffdd"><font size="2" face="Arial"> <select size="1" name="PRO0"> <option value="1">G.711 lei-u</option> <option value="2">G.711
lei-a</option> <option value="3">G.723</option> <option value="4">G.729</option>
<option value="5">G.726 - 16</option> <option value="6">G.726 - 24</option> <option value="7">G.726 - 32</option> <option value="8">G.726 - 40</option> <option value="9" selected>GSM</option></select></font></td></tr>
<tr>
<td width="27%" bgcolor="#ffffcc"><font size="2" face="Arial">Codec Preferido
2:</font></td>
<td width="71%" bgcolor="#ffffcc"><font size="2" face="Arial"> <select size="1" name="PRO1"> <option value="0">Não usado</option> <option value="1" selected>G.711
lei-u</option> <option value="2">G.711 lei-a</option> <option value="3">G.723</option> <option value="4">G.729</option> <option value="5">G.726 -
16</option> <option value="6">G.726 - 24</option> <option value="7">G.726 -
32</option> <option value="8">G.726 - 40</option> <option value="9">GSM</option></select></font></td></tr></tbody></table></form></td></tr></tbody></table><br><br><div class="gmail_quote">2009/3/18 Junior Polegato - Asterisk <span dir="ltr"><<a href="mailto:asterisk@juniorpolegato.com.br">asterisk@juniorpolegato.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;">Olá,<br>
<br>
Criei um novo ambiente e consegui fazer refazer a situação do<br>
problema: quando faço uma ligação e esta não é atendida e coloco no<br>
gancho, acontece o problema, com essa troca de pacotes:<br>
<br>
<-- SIP read from <a href="http://10.1.1.101:5060" target="_blank">10.1.1.101:5060</a>:<br>
Via: SIP/2.0/UDP :5060<br>
From: "Voiper - 2222" <sip:2222@asterisk>;tag=3828945f<br>
To: <sip:3941xxxx@asterisk>;tag=as1c30ec09<br>
Call-ID: <a href="mailto:5d50123233dac07a5e92a42e6d58f883@192.168.0.12">5d50123233dac07a5e92a42e6d58f883@192.168.0.12</a><br>
CSeq: 802 ACK<br>
Content-Length: 0<br>
<br>
--- (6 headers 0 lines) ---<br>
Mar 18 15:25:41 NOTICE[27276]: chan_sip.c:3989 copy_via_headers: No<br>
<div class="im">header field 'Via' present to copy<br>
</div>Transmitting (NAT) to <a href="http://10.1.1.101:5060" target="_blank">10.1.1.101:5060</a>:<br>
SIP/2.0 481 Call leg/transaction does not exist<br>
From: "Voiper - 2222" <sip:2222@asterisk>;tag=3828945f<br>
To: <sip:3941xxxx@asterisk>;tag=as1c30ec09<br>
Call-ID: <a href="mailto:5d50123233dac07a5e92a42e6d58f883@192.168.0.12">5d50123233dac07a5e92a42e6d58f883@192.168.0.12</a><br>
CSeq: 802 ACK<br>
User-Agent: Asterisk PBX<br>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY<br>
Content-Length: 0<br>
<br>
Isso prossegue indefinidamente até que se reinicie o aparelho ou<br>
o Asterisk. Durante isso, nem consigo acesso via Web à configuração do<br>
aparelho. Reiniciando o Asterisk, imediatamente já aprece a tela de<br>
login no navegador.<br>
<br>
Os pacotes de quando o problema inicia:<br>
<br>
<br>
-- SIP/saida_azzu-081c88a0 is making progress passing it to<br>
SIP/22-0817f268<br>
We're at 10.1.1.254 port 11504<br>
Adding codec 0x100 (g729) to SDP<br>
Adding codec 0x2 (gsm) to SDP<br>
Adding non-codec 0x1 (telephone-event) to SDP<br>
Transmitting (NAT) to <a href="http://10.1.1.101:5060" target="_blank">10.1.1.101:5060</a>:<br>
SIP/2.0 183 Session Progress<br>
Via: SIP/2.0/UDP<br>
10.1.1.101:5060;branch=z9hG4bK36da28d784;received=10.1.1.101;rport=5060<br>
From: "Voiper - 2222" <sip:2222@asterisk>;tag=5696a690<br>
To: <sip:3941xxxx@asterisk>;tag=as5771641a<br>
Call-ID: <a href="mailto:628033914eb9a71c7e8a47a42865a0e3@10.1.1.101">628033914eb9a71c7e8a47a42865a0e3@10.1.1.101</a><br>
CSeq: 802 INVITE<br>
User-Agent: Asterisk PBX<br>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY<br>
Contact: <<a href="mailto:sip%3A3941xxxx@10.1.1.254">sip:3941xxxx@10.1.1.254</a>><br>
Content-Type: application/sdp<br>
Content-Length: 264<br>
<br>
v=0<br>
o=root 8289 8289 IN IP4 10.1.1.254<br>
s=session<br>
c=IN IP4 10.1.1.254<br>
t=0 0<br>
m=audio 11504 RTP/AVP 18 3 101<br>
a=rtpmap:18 G729/8000<br>
a=fmtp:18 annexb=no<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>
12 headers, 0 lines<br>
Reliably Transmitting (NAT) to <a href="http://10.1.1.101:5060" target="_blank">10.1.1.101:5060</a>:<br>
OPTIONS <a href="http://sip:2222@10.1.1.101:5060" target="_blank">sip:2222@10.1.1.101:5060</a> SIP/2.0<br>
Via: SIP/2.0/UDP 10.1.1.254:5060;branch=z9hG4bK172765da;rport<br>
From: "asterisk" <<a href="mailto:sip%3Aasterisk@10.1.1.254">sip:asterisk@10.1.1.254</a>>;tag=as027fc344<br>
To: <<a href="http://sip:2222@10.1.1.101:5060" target="_blank">sip:2222@10.1.1.101:5060</a>><br>
Contact: <<a href="mailto:sip%3Aasterisk@10.1.1.254">sip:asterisk@10.1.1.254</a>><br>
Call-ID: <a href="mailto:5a32290f259e06f65e35145944dd7605@10.1.1.254">5a32290f259e06f65e35145944dd7605@10.1.1.254</a><br>
CSeq: 102 OPTIONS<br>
User-Agent: Asterisk PBX<br>
Max-Forwards: 70<br>
Date: Wed, 18 Mar 2009 18:39:10 GMT<br>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY<br>
Content-Length: 0<br>
<br>
<br>
---<br>
testador*CLI><br>
<-- SIP read from <a href="http://10.1.1.101:5060" target="_blank">10.1.1.101:5060</a>:<br>
SIP/2.0 200 OK<br>
Via: SIP/2.0/UDP<br>
10.1.1.254:5060;rport=5060;received=10.1.1.254;branch=z9hG4bK172765da<br>
From: "asterisk" <<a href="mailto:sip%3Aasterisk@10.1.1.254">sip:asterisk@10.1.1.254</a>>;tag=as027fc344<br>
To: <<a href="http://sip:2222@10.1.1.101:5060" target="_blank">sip:2222@10.1.1.101:5060</a>>;tag=6ff92d4b<br>
Call-ID: <a href="mailto:5a32290f259e06f65e35145944dd7605@10.1.1.254">5a32290f259e06f65e35145944dd7605@10.1.1.254</a><br>
Contact: <<a href="http://sip:2222@10.1.1.101:5060" target="_blank">sip:2222@10.1.1.101:5060</a>><br>
CSeq: 102 OPTIONS<br>
Allow: INVITE,CANCEL,ACK,BYE,NOTIFY,REFER,OPTIONS<br>
Content-Length: 0<br>
<br>
--- (9 headers 0 lines) ---<br>
Destroying call '<a href="mailto:5a32290f259e06f65e35145944dd7605@10.1.1.254">5a32290f259e06f65e35145944dd7605@10.1.1.254</a>'<br>
testador*CLI><br>
<-- SIP read from <a href="http://10.1.1.101:5060" target="_blank">10.1.1.101:5060</a>:<br>
<br>
--- (0 headers 0 lines) Nat keepalive ---<br>
testador*CLI><br>
<-- SIP read from <a href="http://10.1.1.101:5060" target="_blank">10.1.1.101:5060</a>:<br>
<br>
--- (0 headers 0 lines) Nat keepalive ---<br>
testador*CLI><br>
<-- SIP read from <a href="http://10.1.1.101:5060" target="_blank">10.1.1.101:5060</a>:<br>
CANCEL sip:3941xxxx@asterisk SIP/2.0<br>
Via: SIP/2.0/UDP 10.1.1.101:5060;rport;branch=z9hG4bK36da28d784<br>
From: "Voiper - 2222" <sip:2222@asterisk>;tag=5696a690<br>
To: <sip:3941xxxx@asterisk>;tag=as5771641a<br>
Call-ID: <a href="mailto:628033914eb9a71c7e8a47a42865a0e3@10.1.1.101">628033914eb9a71c7e8a47a42865a0e3@10.1.1.101</a><br>
Contact: <<a href="http://sip:2222@10.1.1.101:5060" target="_blank">sip:2222@10.1.1.101:5060</a>><br>
CSeq: 802 CANCEL<br>
Proxy-Authorization: Digest<br>
username="2222",realm="yyyyyyyyyyyy",nonce="12345678",response="xxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxx",uri="sip:3941xxxx@asterisk",algorithm=MD5<br>
Content-Length: 0<br>
<br>
--- (9 headers 0 lines) ---<br>
Sending to 10.1.1.101 : 5060 (NAT)<br>
Reliably Transmitting (NAT) to <a href="http://10.1.1.101:5060" target="_blank">10.1.1.101:5060</a>:<br>
SIP/2.0 487 Request Terminated<br>
Via: SIP/2.0/UDP<br>
10.1.1.101:5060;branch=z9hG4bK36da28d784;received=10.1.1.101;rport=5060<br>
From: "Voiper - 2222" <sip:2222@asterisk>;tag=5696a690<br>
To: <sip:3941xxxx@asterisk>;tag=as5771641a<br>
Call-ID: <a href="mailto:628033914eb9a71c7e8a47a42865a0e3@10.1.1.101">628033914eb9a71c7e8a47a42865a0e3@10.1.1.101</a><br>
CSeq: 802 INVITE<br>
User-Agent: Asterisk PBX<br>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY<br>
Content-Length: 0<br>
<br>
<br>
---<br>
Transmitting (NAT) to <a href="http://10.1.1.101:5060" target="_blank">10.1.1.101:5060</a>:<br>
SIP/2.0 200 OK<br>
Via: SIP/2.0/UDP<br>
10.1.1.101:5060;branch=z9hG4bK36da28d784;received=10.1.1.101;rport=5060<br>
From: "Voiper - 2222" <sip:2222@asterisk>;tag=5696a690<br>
To: <sip:3941xxxx@asterisk>;tag=as5771641a<br>
Call-ID: <a href="mailto:628033914eb9a71c7e8a47a42865a0e3@10.1.1.101">628033914eb9a71c7e8a47a42865a0e3@10.1.1.101</a><br>
CSeq: 802 CANCEL<br>
User-Agent: Asterisk PBX<br>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY<br>
Contact: <<a href="mailto:sip%3A3941xxxx@10.1.1.254">sip:3941xxxx@10.1.1.254</a>><br>
Content-Length: 0<br>
<br>
<br>
---<br>
<div class="im"> == Spawn extension (macro-saida_azzu, s, 15) exited non-zero on<br>
</div>'SIP/22-0817f268' in macro 'saida_azzu'<br>
<div class="im"> == Spawn extension (macro-saida_azzu, s, 15) exited non-zero on<br>
</div>'SIP/22-0817f268'<br>
Retransmitting #1 (NAT) to <a href="http://10.1.1.101:5060" target="_blank">10.1.1.101:5060</a>:<br>
SIP/2.0 487 Request Terminated<br>
Via: SIP/2.0/UDP<br>
10.1.1.101:5060;branch=z9hG4bK36da28d784;received=10.1.1.101;rport=5060<br>
From: "Voiper - 2222" <sip:2222@asterisk>;tag=5696a690<br>
To: <sip:3941xxxx@asterisk>;tag=as5771641a<br>
Call-ID: <a href="mailto:628033914eb9a71c7e8a47a42865a0e3@10.1.1.101">628033914eb9a71c7e8a47a42865a0e3@10.1.1.101</a><br>
CSeq: 802 INVITE<br>
User-Agent: Asterisk PBX<br>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY<br>
Content-Length: 0<br>
<br>
<br>
---<br>
Retransmitting #2 (NAT) to <a href="http://10.1.1.101:5060" target="_blank">10.1.1.101:5060</a>:<br>
SIP/2.0 487 Request Terminated<br>
Via: SIP/2.0/UDP<br>
10.1.1.101:5060;branch=z9hG4bK36da28d784;received=10.1.1.101;rport=5060<br>
From: "Voiper - 2222" <sip:2222@asterisk>;tag=5696a690<br>
To: <sip:3941xxxx@asterisk>;tag=as5771641a<br>
Call-ID: <a href="mailto:628033914eb9a71c7e8a47a42865a0e3@10.1.1.101">628033914eb9a71c7e8a47a42865a0e3@10.1.1.101</a><br>
CSeq: 802 INVITE<br>
User-Agent: Asterisk PBX<br>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY<br>
Content-Length: 0<br>
<br>
<br>
---<br>
testador*CLI><br>
<-- SIP read from <a href="http://10.1.1.101:5060" target="_blank">10.1.1.101:5060</a>:<br>
ACK <a href="mailto:sip%3A3941xxxx@10.1.1.254">sip:3941xxxx@10.1.1.254</a> SIP/2.0<br>
Via: SIP/2.0/UDP 10.1.1.101:5060;branch=z9hG4bK36da28d784<br>
From: "Voiper - 2222" <sip:2222@asterisk>;tag=5696a690<br>
To: <sip:3941xxxx@asterisk>;tag=as5771641a<br>
Call-ID: <a href="mailto:628033914eb9a71c7e8a47a42865a0e3@10.1.1.101">628033914eb9a71c7e8a47a42865a0e3@10.1.1.101</a><br>
Contact: <<a href="http://sip:2222@10.1.1.101:5060" target="_blank">sip:2222@10.1.1.101:5060</a>><br>
CSeq: 802 ACK<br>
Content-Length: 0<br>
<br>
--- (8 headers 0 lines) ---<br>
Destroying call '<a href="mailto:628033914eb9a71c7e8a47a42865a0e3@10.1.1.101">628033914eb9a71c7e8a47a42865a0e3@10.1.1.101</a>'<br>
testador*CLI><br>
<-- SIP read from <a href="http://10.1.1.101:5060" target="_blank">10.1.1.101:5060</a>:<br>
ACK sip:3941xxxx@asterisk SIP/2.0<br>
Via: SIP/2.0/UDP 10.1.1.101:5060;branch=z9hG4bK36da28d784<br>
From: "Voiper - 2222" <sip:2222@asterisk>;tag=5696a690<br>
To: <sip:3941xxxx@asterisk>;tag=as5771641a<br>
Call-ID: <a href="mailto:628033914eb9a71c7e8a47a42865a0e3@10.1.1.101">628033914eb9a71c7e8a47a42865a0e3@10.1.1.101</a><br>
CSeq: 802 ACK<br>
Content-Length: 0<br>
<br>
--- (7 headers 0 lines) ---<br>
Transmitting (NAT) to <a href="http://10.1.1.101:5060" target="_blank">10.1.1.101:5060</a>:<br>
SIP/2.0 481 Call leg/transaction does not exist<br>
Via: SIP/2.0/UDP<br>
10.1.1.101:5060;branch=z9hG4bK36da28d784;received=10.1.1.101<br>
From: "Voiper - 2222" <sip:2222@asterisk>;tag=5696a690<br>
To: <sip:3941xxxx@asterisk>;tag=as5771641a<br>
Call-ID: <a href="mailto:628033914eb9a71c7e8a47a42865a0e3@10.1.1.101">628033914eb9a71c7e8a47a42865a0e3@10.1.1.101</a><br>
CSeq: 802 ACK<br>
User-Agent: Asterisk PBX<br>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY<br>
Content-Length: 0<br>
<br>
<br>
---<br>
testador*CLI><br>
<-- SIP read from <a href="http://10.1.1.101:5060" target="_blank">10.1.1.101:5060</a>:<br>
ACK sip:3941xxxx@asterisk SIP/2.0<br>
Via: SIP/2.0/UDP 10.1.1.101:5060;branch=z9hG4bK36da28d784<br>
From: "Voiper - 2222" <sip:2222@asterisk>;tag=5696a690<br>
To: <sip:3941xxxx@asterisk>;tag=as5771641a<br>
Call-ID: <a href="mailto:628033914eb9a71c7e8a47a42865a0e3@10.1.1.101">628033914eb9a71c7e8a47a42865a0e3@10.1.1.101</a><br>
CSeq: 802 ACK<br>
Content-Length: 0<br>
<br>
--- (7 headers 0 lines) ---<br>
Transmitting (NAT) to <a href="http://10.1.1.101:5060" target="_blank">10.1.1.101:5060</a>:<br>
SIP/2.0 481 Call leg/transaction does not exist<br>
Via: SIP/2.0/UDP<br>
10.1.1.101:5060;branch=z9hG4bK36da28d784;received=10.1.1.101<br>
From: "Voiper - 2222" <sip:2222@asterisk>;tag=5696a690<br>
To: <sip:3941xxxx@asterisk>;tag=as5771641a<br>
Call-ID: <a href="mailto:628033914eb9a71c7e8a47a42865a0e3@10.1.1.101">628033914eb9a71c7e8a47a42865a0e3@10.1.1.101</a><br>
CSeq: 802 ACK<br>
User-Agent: Asterisk PBX<br>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY<br>
Content-Length: 0<br>
<br>
<br>
---<br>
testador*CLI><br>
<-- SIP read from <a href="http://10.1.1.101:5060" target="_blank">10.1.1.101:5060</a>:<br>
Via: SIP/2.0/UDP 10.1.1.101:5060;branch=z9hG4bK36da28d784<br>
From: "Voiper - 2222" <sip:2222@asterisk>;tag=5696a690<br>
To: <sip:3941xxxx@asterisk>;tag=as5771641a<br>
Call-ID: <a href="mailto:628033914eb9a71c7e8a47a42865a0e3@10.1.1.101">628033914eb9a71c7e8a47a42865a0e3@10.1.1.101</a><br>
CSeq: 802 ACK<br>
Content-Length: 0<br>
<br>
--- (6 headers 0 lines) ---<br>
Mar 18 15:39:31 NOTICE[8300]: chan_sip.c:3989 copy_via_headers: No<br>
<div class="im">header field 'Via' present to copy<br>
</div>Transmitting (NAT) to <a href="http://10.1.1.101:5060" target="_blank">10.1.1.101:5060</a>:<br>
SIP/2.0 481 Call leg/transaction does not exist<br>
From: "Voiper - 2222" <sip:2222@asterisk>;tag=5696a690<br>
To: <sip:3941xxxx@asterisk>;tag=as5771641a<br>
Call-ID: <a href="mailto:628033914eb9a71c7e8a47a42865a0e3@10.1.1.101">628033914eb9a71c7e8a47a42865a0e3@10.1.1.101</a><br>
CSeq: 802 ACK<br>
User-Agent: Asterisk PBX<br>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY<br>
Content-Length: 0<br>
<br>
<br>
---<br>
testador*CLI><br>
<-- SIP read from <a href="http://10.1.1.101:5060" target="_blank">10.1.1.101:5060</a>:<br>
Via: SIP/2.0/UDP 10.1.1.101:5060;branch=z9hG4bK36da28d784<br>
From: "Voiper - 2222" <sip:2222@asterisk>;tag=5696a690<br>
To: <sip:3941xxxx@asterisk>;tag=as5771641a<br>
Call-ID: <a href="mailto:628033914eb9a71c7e8a47a42865a0e3@10.1.1.101">628033914eb9a71c7e8a47a42865a0e3@10.1.1.101</a><br>
CSeq: 802 ACK<br>
Content-Length: 0<br>
<br>
--- (6 headers 0 lines) ---<br>
Mar 18 15:39:31 NOTICE[8300]: chan_sip.c:3989 copy_via_headers: No<br>
<div class="im">header field 'Via' present to copy<br>
</div>Transmitting (NAT) to <a href="http://10.1.1.101:5060" target="_blank">10.1.1.101:5060</a>:<br>
SIP/2.0 481 Call leg/transaction does not exist<br>
From: "Voiper - 2222" <sip:2222@asterisk>;tag=5696a690<br>
To: <sip:3941xxxx@asterisk>;tag=as5771641a<br>
Call-ID: <a href="mailto:628033914eb9a71c7e8a47a42865a0e3@10.1.1.101">628033914eb9a71c7e8a47a42865a0e3@10.1.1.101</a><br>
CSeq: 802 ACK<br>
User-Agent: Asterisk PBX<br>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY<br>
Content-Length: 0<br>
<br>
<br>
---<br>
testador*CLI><br>
<-- SIP read from <a href="http://10.1.1.101:5060" target="_blank">10.1.1.101:5060</a>:<br>
Via: SIP/2.0/UDP :5060<br>
From: "Voiper - 2222" <sip:2222@asterisk>;tag=5696a690<br>
To: <sip:3941xxxx@asterisk>;tag=as5771641a<br>
Call-ID: <a href="mailto:628033914eb9a71c7e8a47a42865a0e3@10.1.1.101">628033914eb9a71c7e8a47a42865a0e3@10.1.1.101</a><br>
CSeq: 802 ACK<br>
Content-Length: 0<br>
<br>
--- (6 headers 0 lines) ---<br>
Mar 18 15:39:31 NOTICE[8300]: chan_sip.c:3989 copy_via_headers: No<br>
<div class="im">header field 'Via' present to copy<br>
</div>Transmitting (NAT) to <a href="http://10.1.1.101:5060" target="_blank">10.1.1.101:5060</a>:<br>
SIP/2.0 481 Call leg/transaction does not exist<br>
From: "Voiper - 2222" <sip:2222@asterisk>;tag=5696a690<br>
To: <sip:3941xxxx@asterisk>;tag=as5771641a<br>
Call-ID: <a href="mailto:628033914eb9a71c7e8a47a42865a0e3@10.1.1.101">628033914eb9a71c7e8a47a42865a0e3@10.1.1.101</a><br>
CSeq: 802 ACK<br>
User-Agent: Asterisk PBX<br>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY<br>
Content-Length: 0<br>
<br>
<br>
---<br>
testador*CLI><br>
<-- SIP read from <a href="http://10.1.1.101:5060" target="_blank">10.1.1.101:5060</a>:<br>
Via: SIP/2.0/UDP :5060<br>
From: "Voiper - 2222" <sip:2222@asterisk>;tag=5696a690<br>
To: <sip:3941xxxx@asterisk>;tag=as5771641a<br>
Call-ID: <a href="mailto:628033914eb9a71c7e8a47a42865a0e3@10.1.1.101">628033914eb9a71c7e8a47a42865a0e3@10.1.1.101</a><br>
CSeq: 802 ACK<br>
Content-Length: 0<br>
<br>
--- (6 headers 0 lines) ---<br>
Mar 18 15:39:32 NOTICE[8300]: chan_sip.c:3989 copy_via_headers: No<br>
<div class="im">header field 'Via' present to copy<br>
</div>Transmitting (NAT) to <a href="http://10.1.1.101:5060" target="_blank">10.1.1.101:5060</a>:<br>
SIP/2.0 481 Call leg/transaction does not exist<br>
From: "Voiper - 2222" <sip:2222@asterisk>;tag=5696a690<br>
To: <sip:3941xxxx@asterisk>;tag=as5771641a<br>
Call-ID: <a href="mailto:628033914eb9a71c7e8a47a42865a0e3@10.1.1.101">628033914eb9a71c7e8a47a42865a0e3@10.1.1.101</a><br>
CSeq: 802 ACK<br>
User-Agent: Asterisk PBX<br>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY<br>
Content-Length: 0<br>
<div><div></div><div class="h5"><br>
_______________________________________________<br>
Openmoko Freerunner, primeiro telefone open source, disponível no Brasil rodando o Android da Google.<br>
<a href="http://www.neodroid.com" target="_blank">http://www.neodroid.com</a><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>
<a href="http://listas.asteriskbrasil.org/mailman/listinfo/asteriskbrasil" target="_blank">http://listas.asteriskbrasil.org/mailman/listinfo/asteriskbrasil</a><br>
</div></div></blockquote></div><br>