[AsteriskBrasil] Problema com Nat SIP

Sexta Maio 28 16:32:33 BRT 2010

no seu sip.conf tah

nat=yes
canreinvite=no

???

2010/5/28 Saulo Quinteiro <sauloquinteiro em gmail.com>

> Dai galera, to com problemas no nat com sip, tem nat nas duas pontas.
> O rtp não passa.
>
> Segue debug da ligação em uma fila o audio da fila não passa. (tirei o
> final dos ip para evitar engraçadinhos. querendo fazer brincadeiras)
>
>
> <--- SIP read from 200.160.xxx.xxx:51151 --->
> INVITE sip:7000 em 189.38.xxx.xxx SIP/2.0
> Via: SIP/2.0/UDP
> 172.30.1.64:51151;branch=z9hG4bK-d8754z-8d2c602a3a15574b-1---d8754z-;rport
> Max-Forwards: 70
> Contact: <sip:702 em 200.160.xxx.xxx:51151>
> To: "7000"<sip:7000 em 189.38.xxx.xxx>
> From: "702"<sip:702 em 189.38.xxx.xxx>;tag=027a0f27
> Call-ID: ZWZkZGU2N2E2NzhlMTJjZjMwNDgzYjU0ZDliNjg4NDI.
> CSeq: 1 INVITE
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE,
> SUBSCRIBE, INFO
> Content-Type: application/sdp
> User-Agent: X-Lite release 1104o stamp 56125
> Content-Length: 261
>
> v=0
> o=- 2 2 IN IP4 172.30.1.64
> s=CounterPath X-Lite 3.0
> c=IN IP4 172.30.1.64
> t=0 0
> m=audio 13626 RTP/AVP 107 0 8 101
> a=alt:1 1 : QdrDUv3A fbjZtXfs 172.30.1.64 13626
> a=fmtp:101 0-15
> a=rtpmap:107 BV32/16000
> a=rtpmap:101 telephone-event/8000
> a=sendrecv
>
> <------------->
> --- (12 headers 11 lines) ---
> Sending to 200.160.xxx.xxx : 51151 (NAT)
> Using INVITE request as basis request -
> ZWZkZGU2N2E2NzhlMTJjZjMwNDgzYjU0ZDliNjg4NDI.
>
> <--- Reliably Transmitting (NAT) to 200.160.xxx.xxx:51151 --->
> SIP/2.0 407 Proxy Authentication Required
> Via: SIP/2.0/UDP
> 172.30.1.64:51151
> ;branch=z9hG4bK-d8754z-8d2c602a3a15574b-1---d8754z-;received=200.160.xxx.xxx;rport=51151
> From: "702"<sip:702 em 189.38.xxx.xxx>;tag=027a0f27
> To: "7000"<sip:7000 em 189.38.xxx.xxx>;tag=as7f5973d2
> Call-ID: ZWZkZGU2N2E2NzhlMTJjZjMwNDgzYjU0ZDliNjg4NDI.
> CSeq: 1 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
> Supported: replaces
> Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk",
> nonce="17f78f20"
> Content-Length: 0
>
>
> <------------>
> Scheduling destruction of SIP dialog
> 'ZWZkZGU2N2E2NzhlMTJjZjMwNDgzYjU0ZDliNjg4NDI.' in 32000 ms (Method: INVITE)
> Found user '702'
> vispbx01*CLI>
> <--- SIP read from 200.160.xxx.xxx:51151 --->
> ACK sip:7000 em 189.38.xxx.xxx SIP/2.0
> Via: SIP/2.0/UDP
> 172.30.1.64:51151;branch=z9hG4bK-d8754z-8d2c602a3a15574b-1---d8754z-;rport
> To: "7000"<sip:7000 em 189.38.xxx.xxx>;tag=as7f5973d2
> From: "702"<sip:702 em 189.38.xxx.xxx>;tag=027a0f27
> Call-ID: ZWZkZGU2N2E2NzhlMTJjZjMwNDgzYjU0ZDliNjg4NDI.
> CSeq: 1 ACK
> Content-Length: 0
>
>
> <------------->
> --- (7 headers 0 lines) ---
> vispbx01*CLI>
> <--- SIP read from 200.160.xxx.xxx:51151 --->
> INVITE sip:7000 em 189.38.xxx.xxx SIP/2.0
> Via: SIP/2.0/UDP
> 172.30.1.64:51151;branch=z9hG4bK-d8754z-0d75b23fb408b74f-1---d8754z-;rport
> Max-Forwards: 70
> Contact: <sip:702 em 200.160.xxx.xxx:51151>
> To: "7000"<sip:7000 em 189.38.xxx.xxx>
> From: "702"<sip:702 em 189.38.xxx.xxx>;tag=027a0f27
> Call-ID: ZWZkZGU2N2E2NzhlMTJjZjMwNDgzYjU0ZDliNjg4NDI.
> CSeq: 2 INVITE
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE,
> SUBSCRIBE, INFO
> Content-Type: application/sdp
> Proxy-Authorization: Digest
>
> username="702",realm="asterisk",nonce="17f78f20",uri="sip:7000 em 189.38.xxx.xxx
> ",response="f680a98eeae53cdb9e632d820f942017",algorithm=MD5
> User-Agent: X-Lite release 1104o stamp 56125
> Content-Length: 261
>
> v=0
> o=- 2 2 IN IP4 172.30.1.64
> s=CounterPath X-Lite 3.0
> c=IN IP4 172.30.1.64
> t=0 0
> m=audio 13626 RTP/AVP 107 0 8 101
> a=alt:1 1 : QdrDUv3A fbjZtXfs 172.30.1.64 13626
> a=fmtp:101 0-15
> a=rtpmap:107 BV32/16000
> a=rtpmap:101 telephone-event/8000
> a=sendrecv
>
> <------------->
> --- (13 headers 11 lines) ---
> Sending to 200.160.xxx.xxx : 51151 (NAT)
> Using INVITE request as basis request -
> ZWZkZGU2N2E2NzhlMTJjZjMwNDgzYjU0ZDliNjg4NDI.
> Found user '702'
> Found RTP audio format 107
> Found RTP audio format 0
> Found RTP audio format 8
> Found RTP audio format 101
> Found unknown media description format BV32 for ID 107
> Found audio description format telephone-event for ID 101
> Capabilities: us - 0xc (ulaw|alaw), peer - audio=0xc
> (ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
> Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1
> (telephone-event), combined - 0x1 (telephone-event)
> Peer audio RTP is at port 172.30.1.64:13626
> Looking for 7000 in from-internal (domain 189.38.xxx.xxx)
> list_route: hop: <sip:702 em 200.160.xxx.xxx:51151>
>
> <--- Transmitting (NAT) to 200.160.xxx.xxx:51151 --->
> SIP/2.0 100 Trying
> Via: SIP/2.0/UDP
> 172.30.1.64:51151
> ;branch=z9hG4bK-d8754z-0d75b23fb408b74f-1---d8754z-;received=200.160.xxx.xxx;rport=51151
> From: "702"<sip:702 em 189.38.xxx.xxx>;tag=027a0f27
> To: "7000"<sip:7000 em 189.38.xxx.xxx>
> Call-ID: ZWZkZGU2N2E2NzhlMTJjZjMwNDgzYjU0ZDliNjg4NDI.
> CSeq: 2 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
> Supported: replaces
> Contact: <sip:7000 em 192.168.0.102 <sip%3A7000 em 192.168.0.102>>
> Content-Length: 0
>
>
> <------------>
> Audio is at 192.168.0.102 port 12906
> Adding codec 0x4 (ulaw) to SDP
> Adding codec 0x8 (alaw) to SDP
> Adding non-codec 0x1 (telephone-event) to SDP
>
> <--- Reliably Transmitting (NAT) to 200.160.xxx.xxx:51151 --->
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP
> 172.30.1.64:51151
> ;branch=z9hG4bK-d8754z-0d75b23fb408b74f-1---d8754z-;received=200.160.xxx.xxx;rport=51151
> From: "702"<sip:702 em 189.38.xxx.xxx>;tag=027a0f27
> To: "7000"<sip:7000 em 189.38.xxx.xxx>;tag=as5616376f
> Call-ID: ZWZkZGU2N2E2NzhlMTJjZjMwNDgzYjU0ZDliNjg4NDI.
> CSeq: 2 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
> Supported: replaces
> Contact: <sip:7000 em 192.168.0.102 <sip%3A7000 em 192.168.0.102>>
> Content-Type: application/sdp
> Content-Length: 264
>
> v=0
> o=root 2752 2752 IN IP4 192.168.0.102
> s=session
> c=IN IP4 192.168.0.102
> t=0 0
> m=audio 12906 RTP/AVP 0 8 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
> a=ptime:20
> a=sendrecv
>
> <------------>
> Really destroying SIP dialog
> '3b9e133d3529d1585aa5e1bd67f4dbb0 em 127.0.1.1' Method: INVITE
> [May 28 13:56:30] WARNING[3058]: app_dial.c:1296 dial_exec_full: Unable
> to create channel of type 'SIP' (cause 20 - Unknown)
> Really destroying SIP dialog
> '56bcb2c0785cc1e025834c291c9c6d04 em 127.0.1.1' Method: INVITE
> [May 28 13:56:30] WARNING[3062]: app_dial.c:1296 dial_exec_full: Unable
> to create channel of type 'SIP' (cause 20 - Unknown)
> Retransmitting #1 (NAT) to 200.160.xxx.xxx:51151:
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP
> 172.30.1.64:51151
> ;branch=z9hG4bK-d8754z-0d75b23fb408b74f-1---d8754z-;received=200.160.xxx.xxx;rport=51151
> From: "702"<sip:702 em 189.38.xxx.xxx>;tag=027a0f27
> To: "7000"<sip:7000 em 189.38.xxx.xxx>;tag=as5616376f
> Call-ID: ZWZkZGU2N2E2NzhlMTJjZjMwNDgzYjU0ZDliNjg4NDI.
> CSeq: 2 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
> Supported: replaces
> Contact: <sip:7000 em 192.168.0.102 <sip%3A7000 em 192.168.0.102>>
> Content-Type: application/sdp
> Content-Length: 264
>
> v=0
> o=root 2752 2752 IN IP4 192.168.0.102
> s=session
> c=IN IP4 192.168.0.102
> t=0 0
> m=audio 12906 RTP/AVP 0 8 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
> a=ptime:20
> a=sendrecv
>
> ---
> Retransmitting #2 (NAT) to 200.160.xxx.xxx:51151:
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP
> 172.30.1.64:51151
> ;branch=z9hG4bK-d8754z-0d75b23fb408b74f-1---d8754z-;received=200.160.xxx.xxx;rport=51151
> From: "702"<sip:702 em 189.38.xxx.xxx>;tag=027a0f27
> To: "7000"<sip:7000 em 189.38.xxx.xxx>;tag=as5616376f
> Call-ID: ZWZkZGU2N2E2NzhlMTJjZjMwNDgzYjU0ZDliNjg4NDI.
> CSeq: 2 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
> Supported: replaces
> Contact: <sip:7000 em 192.168.0.102 <sip%3A7000 em 192.168.0.102>>
> Content-Type: application/sdp
> Content-Length: 264
>
> v=0
> o=root 2752 2752 IN IP4 192.168.0.102
> s=session
> c=IN IP4 192.168.0.102
> t=0 0
> m=audio 12906 RTP/AVP 0 8 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
> a=ptime:20
> a=sendrecv
>
> ---
> Retransmitting #3 (NAT) to 200.160.xxx.xxx:51151:
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP
> 172.30.1.64:51151
> ;branch=z9hG4bK-d8754z-0d75b23fb408b74f-1---d8754z-;received=200.160.xxx.xxx;rport=51151
> From: "702"<sip:702 em 189.38.xxx.xxx>;tag=027a0f27
> To: "7000"<sip:7000 em 189.38.xxx.xxx>;tag=as5616376f
> Call-ID: ZWZkZGU2N2E2NzhlMTJjZjMwNDgzYjU0ZDliNjg4NDI.
> CSeq: 2 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
> Supported: replaces
> Contact: <sip:7000 em 192.168.0.102 <sip%3A7000 em 192.168.0.102>>
> Content-Type: application/sdp
> Content-Length: 264
>
> v=0
> o=root 2752 2752 IN IP4 192.168.0.102
> s=session
> c=IN IP4 192.168.0.102
> t=0 0
> m=audio 12906 RTP/AVP 0 8 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
> a=ptime:20
> a=sendrecv
>
> ---
> Really destroying SIP dialog
> '002ef8cd391fb9515bb446eb1f3bd90c em 127.0.1.1' Method: INVITE
> [May 28 13:56:35] WARNING[3074]: app_dial.c:1296 dial_exec_full: Unable
> to create channel of type 'SIP' (cause 20 - Unknown)
> Really destroying SIP dialog
> '7fd5945017d1ecc7701cbcd30dc2e9f8 em 127.0.1.1' Method: INVITE
> [May 28 13:56:35] WARNING[3070]: app_dial.c:1296 dial_exec_full: Unable
> to create channel of type 'SIP' (cause 20 - Unknown)
> vispbx01*CLI>
> <--- SIP read from 200.160.xxx.xxx:51151 --->
>
>
>
> <------------->
>
>
> Ja configurei externip=189.38.xxx.xxx ,
> localhost=192.168.0.0/255.255.255.0, nat=yes, canreinvite=no
> Agradeço qualquer ajuda.
>
> Att,
>
> --
> Saulo Quinteiro dos Santos
> Bacharel em Ciências da Computação UFPR
> Cel:    (041) 9927-5236
> Com:    (041) 2141-9567
>
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