no seu sip.conf tah <br><br>nat=yes<br>canreinvite=no<br><br>???<br><br><div class="gmail_quote">2010/5/28 Saulo Quinteiro <span dir="ltr"><<a href="mailto:sauloquinteiro@gmail.com">sauloquinteiro@gmail.com</a>></span><br>
<blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;">Dai galera, to com problemas no nat com sip, tem nat nas duas pontas.<br>
O rtp não passa.<br>
<br>
Segue debug da ligação em uma fila o audio da fila não passa. (tirei o<br>
final dos ip para evitar engraçadinhos. querendo fazer brincadeiras)<br>
<br>
<br>
<--- SIP read from 200.160.xxx.xxx:51151 ---><br>
INVITE sip:7000@189.38.xxx.xxx SIP/2.0<br>
Via: SIP/2.0/UDP<br>
172.30.1.64:51151;branch=z9hG4bK-d8754z-8d2c602a3a15574b-1---d8754z-;rport<br>
Max-Forwards: 70<br>
Contact: <sip:702@200.160.xxx.xxx:51151><br>
To: "7000"<sip:7000@189.38.xxx.xxx><br>
From: "702"<sip:702@189.38.xxx.xxx>;tag=027a0f27<br>
Call-ID: ZWZkZGU2N2E2NzhlMTJjZjMwNDgzYjU0ZDliNjg4NDI.<br>
CSeq: 1 INVITE<br>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE,<br>
SUBSCRIBE, INFO<br>
Content-Type: application/sdp<br>
User-Agent: X-Lite release 1104o stamp 56125<br>
Content-Length: 261<br>
<br>
v=0<br>
o=- 2 2 IN IP4 172.30.1.64<br>
s=CounterPath X-Lite 3.0<br>
c=IN IP4 172.30.1.64<br>
t=0 0<br>
m=audio 13626 RTP/AVP 107 0 8 101<br>
a=alt:1 1 : QdrDUv3A fbjZtXfs 172.30.1.64 13626<br>
a=fmtp:101 0-15<br>
a=rtpmap:107 BV32/16000<br>
a=rtpmap:101 telephone-event/8000<br>
a=sendrecv<br>
<br>
<-------------><br>
--- (12 headers 11 lines) ---<br>
Sending to 200.160.xxx.xxx : 51151 (NAT)<br>
Using INVITE request as basis request -<br>
ZWZkZGU2N2E2NzhlMTJjZjMwNDgzYjU0ZDliNjg4NDI.<br>
<br>
<--- Reliably Transmitting (NAT) to 200.160.xxx.xxx:51151 ---><br>
SIP/2.0 407 Proxy Authentication Required<br>
Via: SIP/2.0/UDP<br>
172.30.1.64:51151;branch=z9hG4bK-d8754z-8d2c602a3a15574b-1---d8754z-;received=200.160.xxx.xxx;rport=51151<br>
From: "702"<sip:702@189.38.xxx.xxx>;tag=027a0f27<br>
To: "7000"<sip:7000@189.38.xxx.xxx>;tag=as7f5973d2<br>
Call-ID: ZWZkZGU2N2E2NzhlMTJjZjMwNDgzYjU0ZDliNjg4NDI.<br>
CSeq: 1 INVITE<br>
User-Agent: Asterisk PBX<br>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO<br>
Supported: replaces<br>
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="17f78f20"<br>
Content-Length: 0<br>
<br>
<br>
<------------><br>
Scheduling destruction of SIP dialog<br>
'ZWZkZGU2N2E2NzhlMTJjZjMwNDgzYjU0ZDliNjg4NDI.' in 32000 ms (Method: INVITE)<br>
Found user '702'<br>
vispbx01*CLI><br>
<--- SIP read from 200.160.xxx.xxx:51151 ---><br>
ACK sip:7000@189.38.xxx.xxx SIP/2.0<br>
Via: SIP/2.0/UDP<br>
172.30.1.64:51151;branch=z9hG4bK-d8754z-8d2c602a3a15574b-1---d8754z-;rport<br>
To: "7000"<sip:7000@189.38.xxx.xxx>;tag=as7f5973d2<br>
From: "702"<sip:702@189.38.xxx.xxx>;tag=027a0f27<br>
Call-ID: ZWZkZGU2N2E2NzhlMTJjZjMwNDgzYjU0ZDliNjg4NDI.<br>
CSeq: 1 ACK<br>
Content-Length: 0<br>
<br>
<br>
<-------------><br>
--- (7 headers 0 lines) ---<br>
vispbx01*CLI><br>
<--- SIP read from 200.160.xxx.xxx:51151 ---><br>
INVITE sip:7000@189.38.xxx.xxx SIP/2.0<br>
Via: SIP/2.0/UDP<br>
172.30.1.64:51151;branch=z9hG4bK-d8754z-0d75b23fb408b74f-1---d8754z-;rport<br>
Max-Forwards: 70<br>
Contact: <sip:702@200.160.xxx.xxx:51151><br>
To: "7000"<sip:7000@189.38.xxx.xxx><br>
From: "702"<sip:702@189.38.xxx.xxx>;tag=027a0f27<br>
Call-ID: ZWZkZGU2N2E2NzhlMTJjZjMwNDgzYjU0ZDliNjg4NDI.<br>
CSeq: 2 INVITE<br>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE,<br>
SUBSCRIBE, INFO<br>
Content-Type: application/sdp<br>
Proxy-Authorization: Digest<br>
username="702",realm="asterisk",nonce="17f78f20",uri="sip:7000@189.38.xxx.xxx",response="f680a98eeae53cdb9e632d820f942017",algorithm=MD5<br>
User-Agent: X-Lite release 1104o stamp 56125<br>
Content-Length: 261<br>
<br>
v=0<br>
o=- 2 2 IN IP4 172.30.1.64<br>
s=CounterPath X-Lite 3.0<br>
c=IN IP4 172.30.1.64<br>
t=0 0<br>
m=audio 13626 RTP/AVP 107 0 8 101<br>
a=alt:1 1 : QdrDUv3A fbjZtXfs 172.30.1.64 13626<br>
a=fmtp:101 0-15<br>
a=rtpmap:107 BV32/16000<br>
a=rtpmap:101 telephone-event/8000<br>
a=sendrecv<br>
<br>
<-------------><br>
--- (13 headers 11 lines) ---<br>
Sending to 200.160.xxx.xxx : 51151 (NAT)<br>
Using INVITE request as basis request -<br>
ZWZkZGU2N2E2NzhlMTJjZjMwNDgzYjU0ZDliNjg4NDI.<br>
Found user '702'<br>
Found RTP audio format 107<br>
Found RTP audio format 0<br>
Found RTP audio format 8<br>
Found RTP audio format 101<br>
Found unknown media description format BV32 for ID 107<br>
Found audio description format telephone-event for ID 101<br>
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0xc<br>
(ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)<br>
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1<br>
(telephone-event), combined - 0x1 (telephone-event)<br>
Peer audio RTP is at port <a href="http://172.30.1.64:13626" target="_blank">172.30.1.64:13626</a><br>
Looking for 7000 in from-internal (domain 189.38.xxx.xxx)<br>
list_route: hop: <sip:702@200.160.xxx.xxx:51151><br>
<br>
<--- Transmitting (NAT) to 200.160.xxx.xxx:51151 ---><br>
SIP/2.0 100 Trying<br>
Via: SIP/2.0/UDP<br>
172.30.1.64:51151;branch=z9hG4bK-d8754z-0d75b23fb408b74f-1---d8754z-;received=200.160.xxx.xxx;rport=51151<br>
From: "702"<sip:702@189.38.xxx.xxx>;tag=027a0f27<br>
To: "7000"<sip:7000@189.38.xxx.xxx><br>
Call-ID: ZWZkZGU2N2E2NzhlMTJjZjMwNDgzYjU0ZDliNjg4NDI.<br>
CSeq: 2 INVITE<br>
User-Agent: Asterisk PBX<br>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO<br>
Supported: replaces<br>
Contact: <<a href="mailto:sip%3A7000@192.168.0.102">sip:7000@192.168.0.102</a>><br>
Content-Length: 0<br>
<br>
<br>
<------------><br>
Audio is at 192.168.0.102 port 12906<br>
Adding codec 0x4 (ulaw) to SDP<br>
Adding codec 0x8 (alaw) to SDP<br>
Adding non-codec 0x1 (telephone-event) to SDP<br>
<br>
<--- Reliably Transmitting (NAT) to 200.160.xxx.xxx:51151 ---><br>
SIP/2.0 200 OK<br>
Via: SIP/2.0/UDP<br>
172.30.1.64:51151;branch=z9hG4bK-d8754z-0d75b23fb408b74f-1---d8754z-;received=200.160.xxx.xxx;rport=51151<br>
From: "702"<sip:702@189.38.xxx.xxx>;tag=027a0f27<br>
To: "7000"<sip:7000@189.38.xxx.xxx>;tag=as5616376f<br>
Call-ID: ZWZkZGU2N2E2NzhlMTJjZjMwNDgzYjU0ZDliNjg4NDI.<br>
CSeq: 2 INVITE<br>
User-Agent: Asterisk PBX<br>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO<br>
Supported: replaces<br>
Contact: <<a href="mailto:sip%3A7000@192.168.0.102">sip:7000@192.168.0.102</a>><br>
Content-Type: application/sdp<br>
Content-Length: 264<br>
<br>
v=0<br>
o=root 2752 2752 IN IP4 192.168.0.102<br>
s=session<br>
c=IN IP4 192.168.0.102<br>
t=0 0<br>
m=audio 12906 RTP/AVP 0 8 101<br>
a=rtpmap:0 PCMU/8000<br>
a=rtpmap:8 PCMA/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>
<------------><br>
Really destroying SIP dialog<br>
'<a href="mailto:3b9e133d3529d1585aa5e1bd67f4dbb0@127.0.1.1">3b9e133d3529d1585aa5e1bd67f4dbb0@127.0.1.1</a>' Method: INVITE<br>
[May 28 13:56:30] WARNING[3058]: app_dial.c:1296 dial_exec_full: Unable<br>
to create channel of type 'SIP' (cause 20 - Unknown)<br>
Really destroying SIP dialog<br>
'<a href="mailto:56bcb2c0785cc1e025834c291c9c6d04@127.0.1.1">56bcb2c0785cc1e025834c291c9c6d04@127.0.1.1</a>' Method: INVITE<br>
[May 28 13:56:30] WARNING[3062]: app_dial.c:1296 dial_exec_full: Unable<br>
to create channel of type 'SIP' (cause 20 - Unknown)<br>
Retransmitting #1 (NAT) to 200.160.xxx.xxx:51151:<br>
SIP/2.0 200 OK<br>
Via: SIP/2.0/UDP<br>
172.30.1.64:51151;branch=z9hG4bK-d8754z-0d75b23fb408b74f-1---d8754z-;received=200.160.xxx.xxx;rport=51151<br>
From: "702"<sip:702@189.38.xxx.xxx>;tag=027a0f27<br>
To: "7000"<sip:7000@189.38.xxx.xxx>;tag=as5616376f<br>
Call-ID: ZWZkZGU2N2E2NzhlMTJjZjMwNDgzYjU0ZDliNjg4NDI.<br>
CSeq: 2 INVITE<br>
User-Agent: Asterisk PBX<br>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO<br>
Supported: replaces<br>
Contact: <<a href="mailto:sip%3A7000@192.168.0.102">sip:7000@192.168.0.102</a>><br>
Content-Type: application/sdp<br>
Content-Length: 264<br>
<br>
v=0<br>
o=root 2752 2752 IN IP4 192.168.0.102<br>
s=session<br>
c=IN IP4 192.168.0.102<br>
t=0 0<br>
m=audio 12906 RTP/AVP 0 8 101<br>
a=rtpmap:0 PCMU/8000<br>
a=rtpmap:8 PCMA/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>
---<br>
Retransmitting #2 (NAT) to 200.160.xxx.xxx:51151:<br>
SIP/2.0 200 OK<br>
Via: SIP/2.0/UDP<br>
172.30.1.64:51151;branch=z9hG4bK-d8754z-0d75b23fb408b74f-1---d8754z-;received=200.160.xxx.xxx;rport=51151<br>
From: "702"<sip:702@189.38.xxx.xxx>;tag=027a0f27<br>
To: "7000"<sip:7000@189.38.xxx.xxx>;tag=as5616376f<br>
Call-ID: ZWZkZGU2N2E2NzhlMTJjZjMwNDgzYjU0ZDliNjg4NDI.<br>
CSeq: 2 INVITE<br>
User-Agent: Asterisk PBX<br>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO<br>
Supported: replaces<br>
Contact: <<a href="mailto:sip%3A7000@192.168.0.102">sip:7000@192.168.0.102</a>><br>
Content-Type: application/sdp<br>
Content-Length: 264<br>
<br>
v=0<br>
o=root 2752 2752 IN IP4 192.168.0.102<br>
s=session<br>
c=IN IP4 192.168.0.102<br>
t=0 0<br>
m=audio 12906 RTP/AVP 0 8 101<br>
a=rtpmap:0 PCMU/8000<br>
a=rtpmap:8 PCMA/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>
---<br>
Retransmitting #3 (NAT) to 200.160.xxx.xxx:51151:<br>
SIP/2.0 200 OK<br>
Via: SIP/2.0/UDP<br>
172.30.1.64:51151;branch=z9hG4bK-d8754z-0d75b23fb408b74f-1---d8754z-;received=200.160.xxx.xxx;rport=51151<br>
From: "702"<sip:702@189.38.xxx.xxx>;tag=027a0f27<br>
To: "7000"<sip:7000@189.38.xxx.xxx>;tag=as5616376f<br>
Call-ID: ZWZkZGU2N2E2NzhlMTJjZjMwNDgzYjU0ZDliNjg4NDI.<br>
CSeq: 2 INVITE<br>
User-Agent: Asterisk PBX<br>
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO<br>
Supported: replaces<br>
Contact: <<a href="mailto:sip%3A7000@192.168.0.102">sip:7000@192.168.0.102</a>><br>
Content-Type: application/sdp<br>
Content-Length: 264<br>
<br>
v=0<br>
o=root 2752 2752 IN IP4 192.168.0.102<br>
s=session<br>
c=IN IP4 192.168.0.102<br>
t=0 0<br>
m=audio 12906 RTP/AVP 0 8 101<br>
a=rtpmap:0 PCMU/8000<br>
a=rtpmap:8 PCMA/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>
---<br>
Really destroying SIP dialog<br>
'<a href="mailto:002ef8cd391fb9515bb446eb1f3bd90c@127.0.1.1">002ef8cd391fb9515bb446eb1f3bd90c@127.0.1.1</a>' Method: INVITE<br>
[May 28 13:56:35] WARNING[3074]: app_dial.c:1296 dial_exec_full: Unable<br>
to create channel of type 'SIP' (cause 20 - Unknown)<br>
Really destroying SIP dialog<br>
'<a href="mailto:7fd5945017d1ecc7701cbcd30dc2e9f8@127.0.1.1">7fd5945017d1ecc7701cbcd30dc2e9f8@127.0.1.1</a>' Method: INVITE<br>
[May 28 13:56:35] WARNING[3070]: app_dial.c:1296 dial_exec_full: Unable<br>
to create channel of type 'SIP' (cause 20 - Unknown)<br>
vispbx01*CLI><br>
<--- SIP read from 200.160.xxx.xxx:51151 ---><br>
<br>
<br>
<br>
<-------------><br>
<br>
<br>
Ja configurei externip=189.38.xxx.xxx ,<br>
localhost=<a href="http://192.168.0.0/255.255.255.0" target="_blank">192.168.0.0/255.255.255.0</a>, nat=yes, canreinvite=no<br>
Agradeço qualquer ajuda.<br>
<br>
Att,<br>
<br>
--<br>
Saulo Quinteiro dos Santos<br>
Bacharel em Ciências da Computação UFPR<br>
Cel: (041) 9927-5236<br>
Com: (041) 2141-9567<br>
<br>
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</blockquote></div><br>