помогите решить проблему с NAT
помогите решить проблему с NAT
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Сообщений: 21
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помогите решить проблему с NAT
есть астериск, работающий в локалке(все замечательно) надо подключить возможность работать через интернет(клиент и астериск за NAT).
маршрутизатор D-Link 804HV поднимает PPPoE с статический адресом, на нем через раздел Virtual Server проброшены порты 5060 и 10000-20000 всех протоколов.
клиенты x-lite, регистрируются, а вот при звонках не слышно друг-друга и через 20-30 сек связь рвется, причем только у принимающего а у звонящего нет.
asterisk имеет такой конфиг sip.conf
[general]
vmexten=*97
context=from-sip-external
callerid=Unknown
notifyringing=yes
notifyhold=yes
limitonpeers=yes
tos_sip=cs3
tos_audio=ef
tos_video=af41
alwaysauthreject=yes
disallow=all
allow=alaw
allow=ulaw
jbenable=no
defaultexpiry=120
maxexpiry=3600
srvlookup=no
minexpiry=60
allowguest=yes
registerattempts=0
registertimeout=20
notifyhold=yes
g726nonstandard=no
t38pt_udptl=no
videosupport=no
maxcallbitrate=384
canreinvite=no
rtptimeout=30
rtpholdtimeout=300
checkmwi=10
notifyringing=yes
rtpkeepalive=0
nat=yes
externip= [my_external_ip_modem]
localnet=192.168.0.0/255.255.0.0
[600]
secret= [my_secret]
dtmfmode=rfc2833
canreinvite=no
context=default
host=dynamic
type=friend
nat=yes
port=5060
qualify=300
callgroup=
pickupgroup=
dial=SIP/600
accountcode=
sipreinvite=no
mailbox=600@device
permit=0.0.0.0/255.255.255.255
callerid=my_name<600>
call-limit=20
faxdetect=no
конфиг rtp.conf
[general]
rtpstart=10000
rtpend=20000
выходят на городские вот так(правило в extensions.conf):
exten => _0XXXXXXXXX,1,Dial(SIP/${EXTEN}@ [ip] ,60,t)
где [ip] - адрес в локальной сети циски(я так понимаю это SIP-proxy), предоставляющей выход на мир (стац. и моб. тел)
почему не слышно? |
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Сообщений: 21
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Re: помогите решить проблему с NAT
неужели никто не знает в чем проблема?
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Откуда: Саратов
Сообщений: 414
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Re: помогите решить проблему с NAT
Я знаю в чём проблема, но вам не скажу, так как не вижу вашего sip debug.
+7(925)140-7438
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Сообщений: 21
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Re: помогите решить проблему с NAT
не вижу вашего sip debug.
вот он:
-- Registered SIP '600' at 46.98.69.165 port 51828
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Executing [0631234567@default:1] Dial("SIP/600-00000010", "SIP/0631234567@10.10.0.70,60,t") in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Called 0631234567@10.10.0.70
-- SIP/10.10.0.70-00000011 is making progress passing it to SIP/6002-00000010
-- SIP/10.10.0.70-00000011 answered SIP/600-00000010
== Spawn extension (default, 0631234567, 1) exited non-zero on 'SIP/600-00000010'
asterisk*CLI>
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Откуда: Уфа
Сообщений: 5856
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Re: помогите решить проблему с NAT
это не он
sip set debug
нада набрать
потом взять в руки книжку гольдштейна по SIP и внимательно изучить, что не так
а еще лучше снять дамп и анализировать вайршарком.
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Сообщений: 21
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Re: помогите решить проблему с NAT
=~=~=~=~=~=~=~=~=~=~=~= PuTTY log 2010.10.23 20:01:04 =~=~=~=~=~=~=~=~=~=~=~=
asterisk*CLI> <--- SIP read from UDP:46.98.69.165:19980 --->
<------------->
[Kasterisk*CLI> <--- SIP read from UDP:46.98.69.165:19980 --->
INVITE sip:0631234567@205.207.122.82 SIP/2.0Via: SIP/2.0/UDP 46.98.69.165:19980;branch=z9hG4bK-d8754z-334b0b3909033137-1---d8754z-;rportMax-Forwards:
70Contact: <sip:600@46.98.69.165:19980>To: "0631234567"<sip:0631234567@205.207.122.82>From: "Sergey"<sip:600@205.207.122.82>;tag=d5580411Call-ID:
OTVkODYwZjVlZGEzODk3ODY2MDQxYjRlZTE4YTMyNjk.CSeq: 1 INVITEAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFOContent-Type:
application/sdpUser-Agent: eyeBeam release 1102u stamp 52345Content-Length: 502v=0o=- 4 2 IN IP4 46.98.69.165s=CounterPath eyeBeam 1.5c=IN IP4
46.98.69.165t=0 0m=audio 8580 RTP/AVP 107 119 100 106 6 9 0 97 105 98 8 18 3 5 101a=alt:1 1 : O0PZntOQ WQ7Zmor6 46.98.69.165 8580a=fmtp:18
annexb=yesa=fmtp:101 0-15a=rtpmap:107 BV32/16000a=rtpmap:119 BV32-FEC/16000a=rtpmap:100 SPEEX/16000a=rtpmap:106 SPEEX-FEC/16000a=rtpmap:97
SPEEX/8000a=rtpmap:105 SPEEX-FEC/8000a=rtpmap:98 iLBC/8000a=rtpmap:18 G729/8000a=rtpmap:101 telephone-event/8000a=sendrecv
<------------->
--- (12 headers 19 lines) ---
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
Sending to 46.98.69.165 : 19980 (NAT)
Using INVITE request as basis request - OTVkODYwZjVlZGEzODk3ODY2MDQxYjRlZTE4YTMyNjk.
Found peer '600' for '600' from 46.98.69.165:19980
<--- Reliably Transmitting (NAT) to 46.98.69.165:19980 --->
SIP/2.0 401 UnauthorizedVia: SIP/2.0/UDP 46.98.69.165:19980;branch=z9hG4bK-d8754z-334b0b3909033137-1---d8754z-;received=46.98.69.165;rport=19980From:
"Sergey"<sip:600@205.207.122.82>;tag=d5580411To: "0631234567"<sip:0631234567@205.207.122.82>;tag=as4ab2e5ecCall-ID:
OTVkODYwZjVlZGEzODk3ODY2MDQxYjRlZTE4YTMyNjk.CSeq: 1 INVITEServer: Asterisk PBX 1.6.2.7Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFOSupported: replaces, timerWWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="08aaa117"Content-Length: 0
<------------>
S
[Kasterisk*CLI>
cheduling destruction of SIP dialog 'OTVkODYwZjVlZGEzODk3ODY2MDQxYjRlZTE4YTMyNjk.' in 12480 ms (Method: INVITE)
[Kasterisk*CLI> <--- SIP read from UDP:46.98.69.165:19980 --->
ACK sip:0631234567@205.207.122.82 SIP/2.0Via: SIP/2.0/UDP 46.98.69.165:19980;branch=z9hG4bK-d8754z-334b0b3909033137-1---d8754z-;rportTo:
"0631234567"<sip:0631234567@205.207.122.82>;tag=as4ab2e5ecFrom: "Sergey"<sip:600@205.207.122.82>;tag=d5580411Call-ID:
OTVkODYwZjVlZGEzODk3ODY2MDQxYjRlZTE4YTMyNjk.CSeq: 1 ACKContent-Length: 0
<------------->
--- (7 headers 0 lines) ---
[Kasterisk*CLI> <--- SIP read from UDP:46.98.69.165:19980 --->
INVITE sip:0631234567@205.207.122.82 SIP/2.0Via: SIP/2.0/UDP 46.98.69.165:19980;branch=z9hG4bK-d8754z-9c242c029747836b-1---d8754z-;rportMax-Forwards:
70Contact: <sip:600@46.98.69.165:19980>To: "0631234567"<sip:0631234567@205.207.122.82>From: "Sergey"<sip:600@205.207.122.82>;tag=d5580411Call-ID:
OTVkODYwZjVlZGEzODk3ODY2MDQxYjRlZTE4YTMyNjk.CSeq: 2 INVITEAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFOContent-Type:
application/sdpUser-Agent: eyeBeam release 1102u stamp 52345Authorization: Digest
username="600",realm="asterisk",nonce="08aaa117",uri="sip:0631234567@205.207.122.82",response="8478e0c134a2c6c545305f0729c2c3ce",algorithm=MD5Content-Length:
502v=0o=- 4 2 IN IP4 46.98.69.165s=CounterPath eyeBeam 1.5c=IN IP4 46.98.69.165t=0 0m=audio 8580 RTP/AVP 107 119 100 106 6 9 0 97 105 98 8 18 3 5 101a=alt:1
1 : O0PZntOQ WQ7Zmor6 46.98.69.165 8580a=fmtp:18 annexb=yesa=fmtp:101 0-15a=rtpmap:107 BV32/16000a=rtpmap:119 BV32-FEC/16000a=rtpmap:100
SPEEX/16000a=rtpmap:106 SPEEX-FEC/16000a=rtpmap:97 SPEEX/8000a=rtpmap:105 SPEEX-FEC/8000a=rtpmap:98 iLBC/8000a=rtpmap:18 G729/8000a=rtpmap:101
telephone-event/8000a=sendrecv
<------------->
--- (13 headers 19 lines) ---
Sending to 46.98.69.165 : 19980 (NAT)
Using INVITE request as basis request - OTVkODYwZjVlZGEzODk3ODY2MDQxYjRlZTE4YTMyNjk.
Found peer '600' for '600' from 46.98.69.165:19980
Found RTP audio format 107
Found RTP audio format 119
Found RTP audio format 100
Found RTP audio format 106
Found RTP audio format 6
Found RTP audio format 9
Found RTP audio format 0
Found RTP audio format 97
Found RTP audio format 105
Found RTP audio format 98
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 3
Found RTP audio format 5
Found RTP audio format 101
Found audio description format BV32 for ID 107
Found audio description format BV32-FEC for ID 119
Found audio description format SPEEX for ID 100
Found audio description format SPEEX-FEC for ID 106
Found audio description format SPEEX for ID 97
Found audio description format SPEEX-FEC for ID 105
Found audio description format iLBC for ID 98
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 101
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0xc00172e (gsm|ulaw|alaw|adpcm|g729|speex|ilbc|g722|red|t140)/video=0x0 (nothing)/text=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 46.98.69.165:8580
Looking for 0631234567 in default (domain 205.207.122.82)
list_route: hop: <sip:600@46.98.69.165:19980>
<--- Transmitting (NAT) to 46.98.69.165:19980 --->
SIP/2.0 100 TryingVia: SIP/2.0/UDP 46.98.69.165:19980;branch=z9hG4bK-d8754z-9c242c029747836b-1---d8754z-;received=46.98.69.165;rport=19980From:
"Sergey"<sip:600@205.207.122.82>;tag=d5580411T
[Kasterisk*CLI>
o: "0631234567"<sip:0631234567@205.207.122.82>Call-ID:
OTVkODYwZjVlZGEzODk3ODY2MDQxYjRlZTE4YTMyNjk.CSeq: 2 INVITEServer: Asterisk PBX 1.6.2.7Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFOSupported: replaces, timerContact: <sip:0631234567@205.207.122.82>Content-Length: 0
<------------>
[Kasterisk*CLI>
-- Executing [0631234567@default:1] [1;36mDial[0m("[1;35mSIP/600-00000006[0m", "[1;35mSIP/0631234567@10.10.12.70,60,t[0m") in
new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
Audio is at 205.207.122.82 port 10066
Adding codec 0x8 (alaw) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 10.10.12.70:5060:
INVITE sip:0631234567@10.10.12.70 SIP/2.0Via: SIP/2.0/UDP 205.207.122.82:5060;branch=z9hG4bK6d88f373;rportMax-Forwards: 70From: "Serg"
<sip:600@205.207.122.82>;tag=as08054e83To: <sip:0631234567@10.10.12.70>Contact: <sip:600@205.207.122.82>Call-ID:
468ae18e1c13003662e45bcb42b05723@205.207.122.82CSeq: 102 INVITEUser-Agent: Asterisk PBX 1.6.2.7Date: Sat, 23 Oct 2010 17:00:26 GMTAllow: INVITE, ACK, CANCEL,
OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFOSupported: replaces, timerContent-Type: application/sdpContent-Length: 291v=0o=root 1206405265 1206405265 IN IP4
205.207.122.82s=Asterisk PBX 1.6.2.7c=IN IP4 205.207.122.82t=0 0m=audio 10066 RTP/AVP 8 0 101a=rtpmap:8 PCMA/8000a=rtpmap:0 PCMU/8000a=rtpmap:101
telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=ptime:20a=sendrecv
---
-- Called 0631234567@10.10.12.70
[Kasterisk*CLI> <--- SIP read from UDP:10.10.12.70:5060 --->
SIP/2.0 100 TryingVia: SIP/2.0/UDP 205.207.122.82:5060;branch=z9hG4bK6d88f373;rport;received=192.168.178.240From: "Serg"
<sip:600@205.207.122.82>;tag=as08054e83To: <sip:0631234567@10.10.12.70>;tag=85C34758-4EDDate: Sat, 23 Oct 2010 17:54:52 GMTCall-ID:
468ae18e1c13003662e45bcb42b05723@205.207.122.82Server: Cisco-SIPGateway/IOS-12.xCSeq: 102 INVITEAllow-Events: telephone-eventContent-Length: 0
<------------->
--- (14 headers 11 lines) ---
Found RTP audio format 8
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 172.17.17.205:17006
[Kasterisk*CLI>
-- SIP/10.10.12.70-00000007 is making progress passing it to SIP/600-00000006
Audio is at 205.207.122.82 port 10060
Adding codec 0x8 (alaw) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
<--- Transmitting (NAT) to 46.98.69.165:19980 --->
SIP/2.0 183 Session ProgressVia: SIP/2.0/UDP 46.98.69.165:19980;branch=z9hG4bK-d8754z-9c242c029747836b-1---d8754z-;received=46.98.69.165;rport=19980From:
"Sergey"<sip:600@205.207.122.82>;tag=d5580411To: "0631234567"<sip:0631234567@205.207.122.82>;tag=as54032f82Call-ID:
OTVkODYwZjVlZGEzODk3ODY2MDQxYjRlZTE4YTMyNjk.CSeq: 2 INVITEServer: Asterisk PBX 1.6.2.7Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFOSupported: replaces, timerContact: <sip:0631234567@205.207.122.82>Content-Type: application/sdpContent-Length: 289v=0o=root 969271786 969271786 IN IP4
205.207.122.82s=Asterisk PBX 1.6.2.7c=IN IP4 205.207.122.82t=0 0m=audio 10060 RTP/AVP 8 0 101a=rtpmap:8 PCMA/8000a=rtpmap:0 PCMU/8000a=rtpmap:101
telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=ptime:20a=sendrecv
<------------>
[Kasterisk*CLI> <--- SIP read from UDP:10.10.12.70:5060 --->
SIP/2.0 200 OKVia: SIP/2.0/UDP 205.207.122.82:5060;branch=z9hG4bK6d88f373;rport;received=192.168.178.240From: "Serg"
<sip:600@205.207.122.82>;tag=as08054e83To: <sip:0631234567@10.10.12.70>;tag=85C34758-4EDDate: Sat, 23 Oct 2010 17:54:52 GMTCall-ID:
468ae18e1c13003662e45bcb42b05723@205.207.122.82Server: Cisco-SIPGateway/IOS-12.xCSeq: 102 INVITEAllow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET,
REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTERSupported: replacesAllow-Events: telephone-eventContact: <sip:0631234567@10.10.12.70:5060>Content-Type:
application/sdpContent-Length: 249v=0o=CiscoSystemsSIP-GW-UserAgent 4771 1832 IN IP4 10.10.12.70s=SIP Callc=IN IP4 172.17.17.205t=0 0m=audio 17006 RTP/AVP 8
101c=IN IP4 172.17.17.205a=rtpmap:8 PCMA/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=ptime:20
<------------->
--- (14 headers 11 lines) ---
[Kasterisk*CLI>
list_route: hop: <sip:0631234567@10.10.12.70:5060>
set_destination: Parsing <sip:0631234567@10.10.12.70:5060> for address/port to send to
set_destination: set destination to 10.10.12.70, port 5060
Transmitting (NAT) to 10.10.12.70:5060:
ACK sip:0631234567@10.10.12.70:5060 SIP/2.0Via: SIP/2.0/UDP 205.207.122.82:5060;branch=z9hG4bK3df43e23;rportMax-Forwards: 70From: "Serg"
<sip:600@205.207.122.82>;tag=as08054e83To: <sip:0631234567@10.10.12.70>;tag=85C34758-4EDContact: <sip:600@205.207.122.82>Call-ID:
468ae18e1c13003662e45bcb42b05723@205.207.122.82CSeq: 102 ACKUser-Agent: Asterisk PBX 1.6.2.7Content-Length: 0
---
[Kasterisk*CLI>
-- SIP/10.10.12.70-00000007 answered SIP/600-00000006
[Kasterisk*CLI>
Audio is at 205.207.122.82 port 10060
Adding codec 0x8 (alaw) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
<--- Reliably Transmitting (NAT) to 46.98.69.165:19980 --->
SIP/2.0 200 OKVia: SIP/2.0/UDP 46.98.69.165:19980;branch=z9hG4bK-d8754z-9c242c029747836b-1---d8754z-;received=46.98.69.165;rport=19980From:
"Sergey"<sip:600@205.207.122.82>;tag=d5580411To: "0631234567"<sip:0631234567@205.207.122.82>;tag=as54032f82Call-ID:
OTVkODYwZjVlZGEzODk3ODY2MDQxYjRlZTE4YTMyNjk.CSeq: 2 INVITEServer: Asterisk PBX 1.6.2.7Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFOSupported: replaces, timerContact: <sip:0631234567@205.207.122.82>Content-Type: application/sdpContent-Length: 289v=0o=root 969271786 969271787 IN IP4
205.207.122.82s=Asterisk PBX 1.6.2.7c=IN IP4 205.207.122.82t=0 0m=audio 10060 RTP/AVP 8 0 101a=rtpmap:8 PCMA/8000a=rtpmap:0 PCMU/8000a=rtpmap:101
telephone-event/8000a=fmtp:101 0-16a=silenceSupp:off - - - -a=ptime:20a=sendrecv
<------------>
[Kasterisk*CLI> <--- SIP read from UDP:46.98.69.165:19980 --->
ACK sip:0631234567@205.207.122.82 SIP/2.0Via: SIP/2.0/UDP 46.98.69.165:19980;branch=z9hG4bK-d8754z-db186f602a45d257-1---d8754z-;rportMax-Forwards: 70Contact:
<sip:600@46.98.69.165:19980>To: "0631234567"<sip:0631234567@205.207.122.82>;tag=as54032f82From: "Sergey"<sip:600@205.207.122.82>;tag=d5580411Call-ID:
OTVkODYwZjVlZGEzODk3ODY2MDQxYjRlZTE4YTMyNjk.CSeq: 2 ACKUser-Agent: eyeBeam release 1102u stamp 52345Authorization: Digest
username="600",realm="asterisk",nonce="08aaa117",uri="sip:0631234567@205.207.122.82",response="8478e0c134a2c6c545305f0729c2c3ce",algorithm=MD5Content-Length:
0
<------------->
--- (11 headers 0 lines) ---
[Kasterisk*CLI> <--- SIP read from UDP:46.98.69.165:19980 --->
<------------->
[Kasterisk*CLI> <--- SIP read from UDP:46.98.69.165:19980 --->
SIP/2.0 200 OKVia: SIP/2.0/UDP 205.207.122.82:5060;branch=z9hG4bK230d070a;rport=5060Contact: <sip:46.98.69.165:19980>To:
<sip:600@46.98.69.165:19980;rinstance=65b5ef9080309be0>;tag=257b4462From: "Unknown"<sip:Unknown@205.207.122.82>;tag=as301dad0eCall-ID:
09637a723608840d7d1f541856a923d7@205.207.122.82CSeq: 102 OPTIONSAccept: application/sdpAccept-Language: enAllow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER,
NOTIFY, MESSAGE, SUBSCRIBE, INFOUser-Agent: eyeBeam release 1102u stamp 52345Content-Length: 0
<------------->
--- (12 headers 0 lines) ---
Really destroying SIP dialog '09637a723608840d7d1f541856a923d7@205.207.122.82' Method: OPTIONS
[Kasterisk*CLI>
Scheduling destruction of SIP dialog '468ae18e1c13003662e45bcb42b05723@205.207.122.82' in 32000 ms (Method: INVITE)
[Kasterisk*CLI>
set_destination: Parsing <sip:0631234567@10.10.12.70:5060> for address/port to send to
set_destination: set destination to 10.10.12.70, port 5060
Reliably Transmitting (NAT) to 10.10.12.70:5060:
BYE sip:0631234567@10.10.12.70:5060 SIP/2.0Via: SIP/2.0/UDP 205.207.122.82:5060;branch=z9hG4bK6e6ea25e;rportMax-Forwards: 70From: "Serg"
<sip:600@205.207.122.82>;tag=as08054e83To: <sip:0631234567@10.10.12.70>;tag=85C34758-4EDCall-ID: 468ae18e1c13003662e45bcb42b05723@205.207.122.82CSeq: 103
BYE
[Kasterisk*CLI>
User-Agent: Asterisk PBX 1.6.2.7X-Asterisk-HangupCause: Normal ClearingX-Asterisk-HangupCauseCode: 16Content-Length: 0
---
== Spawn extension (default, 0631234567, 1) exited non-zero on 'SIP/600-00000006'
Scheduling destruction of SIP dialog 'OTVkODYwZjVlZGEzODk3ODY2MDQxYjRlZTE4YTMyNjk.' in 12480 ms (Method: ACK)
set_destination: Parsing <sip:600@46.98.69.165:19980> for address/port to send to
set_destination: set destination to 46.98.69.165, port 19980
Reliably Transmitting (NAT) to 46.98.69.165:19980:
BYE sip:600@46.98.69.165:19980 SIP/2.0Via: SIP/2.0/UDP 205.207.122.82:5060;branch=z9hG4bK52a9bc15;rportMax-Forwards: 70From:
"0631234567"<sip:0631234567@205.207.122.82>;tag=as54032f82To: "Sergey"<sip:600@205.207.122.82>;tag=d5580411Call-ID:
OTVkODYwZjVlZGEzODk3ODY2MDQxYjRlZTE4YTMyNjk.CSeq: 102 BYEUser-Agent: Asterisk PBX 1.6.2.7X-Asterisk-HangupCause: Normal ClearingX-Asterisk-HangupCauseCode:
16Content-Length: 0
---
[Kasterisk*CLI> <--- SIP read from UDP:10.10.12.70:5060 --->
SIP/2.0 200 OKVia: SIP/2.0/UDP 205.207.122.82:5060;branch=z9hG4bK6e6ea25e;rport;received=192.168.178.240From: "Serg"
<sip:600@205.207.122.82>;tag=as08054e83To: <sip:0631234567@10.10.12.70>;tag=85C34758-4EDDate: Sat, 23 Oct 2010 17:55:30 GMTCall-ID:
468ae18e1c13003662e45bcb42b05723@205.207.122.82Server: Cisco-SIPGateway/IOS-12.xContent-Length: 0CSeq: 103 BYE
<------------->
--- (9 headers 0 lines) ---
Really destroying SIP dialog '468ae18e1c13003662e45bcb42b05723@205.207.122.82' Method: INVITE
[Kasterisk*CLI> <--- SIP read from UDP:46.98.69.165:19980 --->
SIP/2.0 200 OKVia: SIP/2.0/UDP 205.207.122.82:5060;branch=z9hG4bK52a9bc15;rport=5060Contact: <sip:600@46.98.69.165:19980>To:
"Sergey"<sip:600@205.207.122.82>;tag=d5580411From: "0631234567"<sip:0631234567@205.207.122.82>;tag=as54032f82Call-ID:
OTVkODYwZjVlZGEzODk3ODY2MDQxYjRlZTE4YTMyNjk.CSeq: 102 BYEUser-Agent: eyeBeam release 1102u stamp 52345Content-Length: 0
<------------->
Really destroying SIP dialog 'NTVmZTU3MDE0ZTU1NTZiMzdkN2M4ZDlhYmFkYWNkMGQ.' Method: SUBSCRIBE
[Kasterisk*CLI> sip set debug on[Kffasterisk*CLI>
SIP Debugging Disabled
[Kasterisk*CLI>
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Откуда: Саратов
Сообщений: 414
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Re: помогите решить проблему с NAT
Это был лог неудачного звонка?
И если ещё будете постить логи, то сделайте нормальное форматирование. Такое очень трудно читать.
+7(925)140-7438
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Сообщений: 21
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Re: помогите решить проблему с NAT
вызов идет, но при снятии трубки ничего не слышно, и вызов сам завершается через 20-30 секунд
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Откуда: Саратов
Сообщений: 414
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Re: помогите решить проблему с NAT
set_destination: Parsing <sip:0631234567@10.10.12.70:5060> for address/port to send to
set_destination: set destination to 10.10.12.70, port 5060
Reliably Transmitting (NAT) to 10.10.12.70:5060:
BYE sip:0631234567@10.10.12.70:5060 SIP/2.0
Via: SIP/2.0/UDP 205.207.122.82:5060;branch=z9hG4bK6e6ea25e;rport
Max-Forwards: 70
From: "Serg" <sip:600@205.207.122.82>;tag=as08054e83
To: <sip:0631234567@10.10.12.70>;tag=85C34758-4ED
Call-ID: 468ae18e1c13003662e45bcb42b05723@205.207.122.82
CSeq: 103 BYE
User-Agent: Asterisk PBX 1.6.2.7
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0
== Spawn extension (default, 0631234567, 1) exited non-zero on 'SIP/600-00000006'
Scheduling destruction of SIP dialog 'OTVkODYwZjVlZGEzODk3ODY2MDQxYjRlZTE4YTMyNjk.' in 12480 ms (Method: ACK)
set_destination: Parsing <sip:600@46.98.69.165:19980> for address/port to send to
set_destination: set destination to 46.98.69.165, port 19980
Reliably Transmitting (NAT) to 46.98.69.165:19980:
BYE sip:600@46.98.69.165:19980 SIP/2.0
Via: SIP/2.0/UDP 205.207.122.82:5060;branch=z9hG4bK52a9bc15;rport
Max-Forwards: 70
From: "0631234567"<sip:0631234567@205.207.122.82>;tag=as54032f82
To: "Sergey"<sip:600@205.207.122.82>;tag=d5580411
Call-ID: OTVkODYwZjVlZGEzODk3ODY2MDQxYjRlZTE4YTMyNjk.
CSeq: 102 BYE
User-Agent: Asterisk PBX 1.6.2.7
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0
У вас астериск в оба конца звонка по непонятной причине посылает сигнал разъединения.То есть он является инициатором разъединения. Включайте core set debug 5 и копайте, почему он так делает.
+7(925)140-7438
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Сообщений: 21
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Re: помогите решить проблему с NAT
при снятии трубки ничего не слышно
может в этом причина?
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