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нет звука до истечения rtpkeepalive

Новичком считается только что прочитавший «Астериск - будущее телефонии»
http://asterisk.ru/knowledgebase/books
и пытающийся сделать большее

Модераторы: april22, Zavr2008

d771
Сообщения: 28
Зарегистрирован: 29 дек 2012, 21:18

нет звука до истечения rtpkeepalive

Сообщение d771 »

Здравствуйте!
Столкнулся с непоняткой, которая моей логике пока не поддалась:
звонки проходят, но в трубке тишина, пока не пойдет столько секунд, сколько написано в rtpkeepalive.
Естественно, firewall отключал, nat нет как класса.
От типа клиентского девайса не зависит - пробовал и пару fxs, и пару софтфонов.

вот лог (это два разных звонка, иначе мешанина из sip и rtp получается)
PRIME_BBCODE_SPOILER_SHOW PRIME_BBCODE_SPOILER:

Код: Выделить всё

------------------- SIP --------------------------
cs17*CLI> 
[Dec 28 13:54:27] Really destroying SIP dialog '877d1552-aa99cdc0@192.168.184.20' Method: REGISTER
[Dec 28 13:54:27] Really destroying SIP dialog '570f60c8-c6cc08c2@192.168.184.20' Method: REGISTER
[Dec 28 13:54:27] Really destroying SIP dialog '877d1552-aa99cdc0@192.168.184.20' Method: REGISTER
[Dec 28 13:54:27] Really destroying SIP dialog '570f60c8-c6cc08c2@192.168.184.20' Method: REGISTER
[Dec 28 13:54:38] 
<--- SIP read from UDP:192.168.184.20:5060 --->
INVITE sip:530@192.168.185.196 SIP/2.0
Via: SIP/2.0/UDP 192.168.184.20:5060;branch=z9hG4bK-7b542470
From: 431 <sip:431@192.168.185.196>;tag=445df880e3cba6c4o0
To: <sip:530@192.168.185.196>
Remote-Party-ID: 431 <sip:431@192.168.185.196>;screen=yes;party=calling
Call-ID: 8b22f830-1e2b2ad4@192.168.184.20
CSeq: 101 INVITE
Max-Forwards: 70
Contact: 431 <sip:431@192.168.184.20:5060>
Expires: 240
User-Agent: Linksys/SPA3102-5.2.13(GW002)
Content-Length: 442
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura, replaces
Content-Type: application/sdp

v=0
o=- 4805 4805 IN IP4 192.168.184.20
s=-
c=IN IP4 192.168.184.20
t=0 0
m=audio 16468 RTP/AVP 8 0 2 4 18 96 97 98 100 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:4 G723/8000
a=rtpmap:18 G729a/8000
a=rtpmap:96 G726-40/8000
a=rtpmap:97 G726-24/8000
a=rtpmap:98 G726-16/8000
a=rtpmap:100 NSE/8000
a=fmtp:100 192-193
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
<------------->
[Dec 28 13:54:38] --- (15 headers 20 lines) ---
[Dec 28 13:54:38] Sending to 192.168.184.20:5060 (NAT)
[Dec 28 13:54:38] Using INVITE request as basis request - 8b22f830-1e2b2ad4@192.168.184.20
[Dec 28 13:54:38] Found peer '431' for '431' from 192.168.184.20:5060
[Dec 28 13:54:38] 
<--- Reliably Transmitting (NAT) to 192.168.184.20:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.184.20:5060;branch=z9hG4bK-7b542470;received=192.168.184.20;rport=5060
From: 431 <sip:431@192.168.185.196>;tag=445df880e3cba6c4o0
To: <sip:530@192.168.185.196>;tag=as6c0bf908
Call-ID: 8b22f830-1e2b2ad4@192.168.184.20
CSeq: 101 INVITE
Server: Asterisk PBX 1.8.19.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="880.ru", nonce="48ee131f"
Content-Length: 0


<------------>
[Dec 28 13:54:38] Scheduling destruction of SIP dialog '8b22f830-1e2b2ad4@192.168.184.20' in 6400 ms (Method: INVITE)
[Dec 28 13:54:38] 
<--- SIP read from UDP:192.168.184.20:5060 --->
ACK sip:530@192.168.185.196 SIP/2.0
Via: SIP/2.0/UDP 192.168.184.20:5060;branch=z9hG4bK-7b542470
From: 431 <sip:431@192.168.185.196>;tag=445df880e3cba6c4o0
To: <sip:530@192.168.185.196>;tag=as6c0bf908
Call-ID: 8b22f830-1e2b2ad4@192.168.184.20
CSeq: 101 ACK
Max-Forwards: 70
Contact: 431 <sip:431@192.168.184.20:5060>
User-Agent: Linksys/SPA3102-5.2.13(GW002)
Content-Length: 0

<------------->
[Dec 28 13:54:38] --- (10 headers 0 lines) ---
[Dec 28 13:54:38] 
<--- SIP read from UDP:192.168.184.20:5060 --->
INVITE sip:530@192.168.185.196 SIP/2.0
Via: SIP/2.0/UDP 192.168.184.20:5060;branch=z9hG4bK-cde3c6b4
From: 431 <sip:431@192.168.185.196>;tag=445df880e3cba6c4o0
To: <sip:530@192.168.185.196>
Remote-Party-ID: 431 <sip:431@192.168.185.196>;screen=yes;party=calling
Call-ID: 8b22f830-1e2b2ad4@192.168.184.20
CSeq: 102 INVITE
Max-Forwards: 70
Authorization: Digest username="431",realm="880.ru",nonce="48ee131f",uri="sip:530@192.168.185.196",algorithm=MD5,response="bfa8879fb2ac6efe1bc0b628be54a0a6"
Contact: 431 <sip:431@192.168.184.20:5060>
Expires: 240
User-Agent: Linksys/SPA3102-5.2.13(GW002)
Content-Length: 442
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura, replaces
Content-Type: application/sdp

v=0
o=- 4805 4805 IN IP4 192.168.184.20
s=-
c=IN IP4 192.168.184.20
t=0 0
m=audio 16468 RTP/AVP 8 0 2 4 18 96 97 98 100 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:4 G723/8000
a=rtpmap:18 G729a/8000
a=rtpmap:96 G726-40/8000
a=rtpmap:97 G726-24/8000
a=rtpmap:98 G726-16/8000
a=rtpmap:100 NSE/8000
a=fmtp:100 192-193
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
<------------->
[Dec 28 13:54:38] --- (16 headers 20 lines) ---
[Dec 28 13:54:38] Sending to 192.168.184.20:5060 (NAT)
[Dec 28 13:54:38] Using INVITE request as basis request - 8b22f830-1e2b2ad4@192.168.184.20
[Dec 28 13:54:38] Found peer '431' for '431' from 192.168.184.20:5060
[Dec 28 13:54:38]   == Using SIP RTP CoS mark 5
[Dec 28 13:54:38] Found RTP audio format 8
[Dec 28 13:54:38] Found RTP audio format 0
[Dec 28 13:54:38] Found RTP audio format 2
[Dec 28 13:54:38] Found RTP audio format 4
[Dec 28 13:54:38] Found RTP audio format 18
[Dec 28 13:54:38] Found RTP audio format 96
[Dec 28 13:54:38] Found RTP audio format 97
[Dec 28 13:54:38] Found RTP audio format 98
[Dec 28 13:54:38] Found RTP audio format 100
[Dec 28 13:54:38] Found RTP audio format 101
[Dec 28 13:54:38] Found audio description format PCMA for ID 8
[Dec 28 13:54:38] Found audio description format PCMU for ID 0
[Dec 28 13:54:38] Found audio description format G726-32 for ID 2
[Dec 28 13:54:38] Found audio description format G723 for ID 4
[Dec 28 13:54:38] Found audio description format G729a for ID 18
[Dec 28 13:54:38] Found unknown media description format G726-40 for ID 96
[Dec 28 13:54:38] Found unknown media description format G726-24 for ID 97
[Dec 28 13:54:38] Found unknown media description format G726-16 for ID 98
[Dec 28 13:54:38] Found unknown media description format NSE for ID 100
[Dec 28 13:54:38] Found audio description format telephone-event for ID 101
[Dec 28 13:54:38] Capabilities: us - 0x8 (alaw), peer - audio=0x90d (g723|ulaw|alaw|g726|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw)
[Dec 28 13:54:38] Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
[Dec 28 13:54:38] Peer audio RTP is at port 192.168.184.20:16468
[Dec 28 13:54:38] Looking for 530 in simplegate (domain 192.168.185.196)
[Dec 28 13:54:38] list_route: hop: <sip:431@192.168.184.20:5060>
[Dec 28 13:54:38] 
<--- Transmitting (NAT) to 192.168.184.20:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.184.20:5060;branch=z9hG4bK-cde3c6b4;received=192.168.184.20;rport=5060
From: 431 <sip:431@192.168.185.196>;tag=445df880e3cba6c4o0
To: <sip:530@192.168.185.196>
Call-ID: 8b22f830-1e2b2ad4@192.168.184.20
CSeq: 102 INVITE
Server: Asterisk PBX 1.8.19.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:530@172.16.254.4:5060>
Content-Length: 0


<------------>
[Dec 28 13:54:38]     -- Executing [530@simplegate:1] Dial("SIP/431-00000004", "SIP/530,,tg") in new stack
[Dec 28 13:54:38]   == Using SIP RTP CoS mark 5
[Dec 28 13:54:38] Audio is at 15236
[Dec 28 13:54:38] Adding codec 0x8 (alaw) to SDP
[Dec 28 13:54:38] Adding non-codec 0x1 (telephone-event) to SDP
[Dec 28 13:54:38] Reliably Transmitting (NAT) to 192.168.184.195:5060:
INVITE sip:530@192.168.184.195:5060;rinstance=a8c81eb06df965a2;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 172.16.254.4:5060;branch=z9hG4bK0ef87780;rport
Max-Forwards: 70
From: "431" <sip:431@172.16.254.4>;tag=as113a3166
To: <sip:530@192.168.184.195:5060;rinstance=a8c81eb06df965a2;transport=UDP>
Contact: <sip:431@172.16.254.4:5060>
Call-ID: 0cb4b56653b1ff7f4fa7e8ea53d86031@172.16.254.4:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.19.0
Date: Fri, 28 Dec 2012 06:54:38 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 237

v=0
o=root 1372612708 1372612708 IN IP4 172.16.254.4
s=Asterisk PBX 1.8.19.0
c=IN IP4 172.16.254.4
t=0 0
m=audio 15236 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
[Dec 28 13:54:38]     -- Called SIP/530
[Dec 28 13:54:38] Retransmitting #1 (NAT) to 192.168.184.195:5060:
INVITE sip:530@192.168.184.195:5060;rinstance=a8c81eb06df965a2;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 172.16.254.4:5060;branch=z9hG4bK0ef87780;rport
Max-Forwards: 70
From: "431" <sip:431@172.16.254.4>;tag=as113a3166
To: <sip:530@192.168.184.195:5060;rinstance=a8c81eb06df965a2;transport=UDP>
Contact: <sip:431@172.16.254.4:5060>
Call-ID: 0cb4b56653b1ff7f4fa7e8ea53d86031@172.16.254.4:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.19.0
Date: Fri, 28 Dec 2012 06:54:38 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 237

v=0
o=root 1372612708 1372612708 IN IP4 172.16.254.4
s=Asterisk PBX 1.8.19.0
c=IN IP4 172.16.254.4
t=0 0
m=audio 15236 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
[Dec 28 13:54:38] 
<--- SIP read from UDP:192.168.184.195:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 172.16.254.4:5060;branch=z9hG4bK0ef87780;rport=5060
Contact: <sip:530@192.168.184.195:5060;rinstance=a8c81eb06df965a2;transport=UDP>
To: <sip:530@192.168.184.195:5060;rinstance=a8c81eb06df965a2;transport=UDP>;tag=23676357
From: "431"<sip:431@172.16.254.4>;tag=as113a3166
Call-ID: 0cb4b56653b1ff7f4fa7e8ea53d86031@172.16.254.4:5060
CSeq: 102 INVITE
User-Agent: Zoiper for Windows 2.38 rev.16635
Content-Length: 0

<------------->
[Dec 28 13:54:38] --- (9 headers 0 lines) ---
[Dec 28 13:54:38] list_route: hop: <sip:530@192.168.184.195:5060;rinstance=a8c81eb06df965a2;transport=UDP>
[Dec 28 13:54:38]     -- SIP/530-00000005 is ringing
[Dec 28 13:54:38] 
<--- Transmitting (NAT) to 192.168.184.20:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.184.20:5060;branch=z9hG4bK-cde3c6b4;received=192.168.184.20;rport=5060
From: 431 <sip:431@192.168.185.196>;tag=445df880e3cba6c4o0
To: <sip:530@192.168.185.196>;tag=as729c8fed
Call-ID: 8b22f830-1e2b2ad4@192.168.184.20
CSeq: 102 INVITE
Server: Asterisk PBX 1.8.19.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:530@172.16.254.4:5060>
Content-Length: 0


<------------>
[Dec 28 13:54:38] 
<--- SIP read from UDP:192.168.184.195:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 172.16.254.4:5060;branch=z9hG4bK0ef87780;rport=5060
Contact: <sip:530@192.168.184.195:5060;rinstance=a8c81eb06df965a2;transport=UDP>
To: <sip:530@192.168.184.195:5060;rinstance=a8c81eb06df965a2;transport=UDP>;tag=23676357
From: "431"<sip:431@172.16.254.4>;tag=as113a3166
Call-ID: 0cb4b56653b1ff7f4fa7e8ea53d86031@172.16.254.4:5060
CSeq: 102 INVITE
User-Agent: Zoiper for Windows 2.38 rev.16635
Content-Length: 0

<------------->
[Dec 28 13:54:38] --- (9 headers 0 lines) ---
[Dec 28 13:54:38] list_route: hop: <sip:530@192.168.184.195:5060;rinstance=a8c81eb06df965a2;transport=UDP>
[Dec 28 13:54:38]     -- SIP/530-00000005 is ringing
cs17*CLI> 
cs17*CLI> 
cs17*CLI> 
cs17*CLI> 
[Dec 28 13:54:46] 
<--- SIP read from UDP:192.168.184.195:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.254.4:5060;branch=z9hG4bK0ef87780;rport=5060
Contact: <sip:530@192.168.184.195:5060;rinstance=a8c81eb06df965a2;transport=UDP>
To: <sip:530@192.168.184.195:5060;rinstance=a8c81eb06df965a2;transport=UDP>;tag=23676357
From: "431"<sip:431@172.16.254.4>;tag=as113a3166
Call-ID: 0cb4b56653b1ff7f4fa7e8ea53d86031@172.16.254.4:5060
CSeq: 102 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/sdp
Supported: replaces, norefersub, extended-refer, X-cisco-serviceuri
User-Agent: Zoiper for Windows 2.38 rev.16635
Allow-Events: presence, kpml
Content-Length: 333

v=0
o=Zoiper_user 0 2 IN IP4 192.168.184.195
s=Zoiper_session
c=IN IP4 192.168.184.195
t=0 0
m=audio 8000 RTP/AVP 8 3 0 110 98 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:110 speex/8000
a=rtpmap:98 iLBC/8000
a=fmtp:98 mode=30
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->
[Dec 28 13:54:46] --- (13 headers 15 lines) ---
[Dec 28 13:54:46] Found RTP audio format 8
[Dec 28 13:54:46] Found RTP audio format 3
[Dec 28 13:54:46] Found RTP audio format 0
[Dec 28 13:54:46] Found RTP audio format 110
[Dec 28 13:54:46] Found RTP audio format 98
[Dec 28 13:54:46] Found RTP audio format 101
[Dec 28 13:54:46] Found audio description format PCMA for ID 8
[Dec 28 13:54:46] Found audio description format GSM for ID 3
[Dec 28 13:54:46] Found audio description format PCMU for ID 0
[Dec 28 13:54:46] Found audio description format speex for ID 110
[Dec 28 13:54:46] Found audio description format iLBC for ID 98
[Dec 28 13:54:46] Found audio description format telephone-event for ID 101
[Dec 28 13:54:46] Capabilities: us - 0x8 (alaw), peer - audio=0x60e (gsm|ulaw|alaw|speex|ilbc)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw)
[Dec 28 13:54:46] Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
[Dec 28 13:54:46] Peer audio RTP is at port 192.168.184.195:8000
[Dec 28 13:54:46] list_route: hop: <sip:530@192.168.184.195:5060;rinstance=a8c81eb06df965a2;transport=UDP>
[Dec 28 13:54:46] set_destination: Parsing <sip:530@192.168.184.195:5060;rinstance=a8c81eb06df965a2;transport=UDP> for address/port to send to
[Dec 28 13:54:46] set_destination: set destination to 192.168.184.195:5060
[Dec 28 13:54:46] Transmitting (NAT) to 192.168.184.195:5060:
ACK sip:530@192.168.184.195:5060;rinstance=a8c81eb06df965a2;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 172.16.254.4:5060;branch=z9hG4bK5eb6c5b1;rport
Max-Forwards: 70
From: "431" <sip:431@172.16.254.4>;tag=as113a3166
To: <sip:530@192.168.184.195:5060;rinstance=a8c81eb06df965a2;transport=UDP>;tag=23676357
Contact: <sip:431@172.16.254.4:5060>
Call-ID: 0cb4b56653b1ff7f4fa7e8ea53d86031@172.16.254.4:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.8.19.0
Content-Length: 0


---
[Dec 28 13:54:46]     -- SIP/530-00000005 answered SIP/431-00000004
[Dec 28 13:54:46] Audio is at 7246
[Dec 28 13:54:46] Adding codec 0x8 (alaw) to SDP
[Dec 28 13:54:46] Adding non-codec 0x1 (telephone-event) to SDP
[Dec 28 13:54:46] 
<--- Reliably Transmitting (NAT) to 192.168.184.20:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.184.20:5060;branch=z9hG4bK-cde3c6b4;received=192.168.184.20;rport=5060
From: 431 <sip:431@192.168.185.196>;tag=445df880e3cba6c4o0
To: <sip:530@192.168.185.196>;tag=as729c8fed
Call-ID: 8b22f830-1e2b2ad4@192.168.184.20
CSeq: 102 INVITE
Server: Asterisk PBX 1.8.19.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:530@172.16.254.4:5060>
Content-Type: application/sdp
Content-Length: 236

v=0
o=root 1027083962 1027083962 IN IP4 172.16.254.4
s=Asterisk PBX 1.8.19.0
c=IN IP4 172.16.254.4
t=0 0
m=audio 7246 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>
[Dec 28 13:54:46] 
<--- SIP read from UDP:192.168.184.20:5060 --->
ACK sip:530@172.16.254.4:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.184.20:5060;branch=z9hG4bK-b32b113e
From: 431 <sip:431@192.168.185.196>;tag=445df880e3cba6c4o0
To: <sip:530@192.168.185.196>;tag=as729c8fed
Call-ID: 8b22f830-1e2b2ad4@192.168.184.20
CSeq: 102 ACK
Max-Forwards: 70
Authorization: Digest username="431",realm="880.ru",nonce="48ee131f",uri="sip:530@192.168.185.196",algorithm=MD5,response="bfa8879fb2ac6efe1bc0b628be54a0a6"
Contact: 431 <sip:431@192.168.184.20:5060>
User-Agent: Linksys/SPA3102-5.2.13(GW002)
Content-Length: 0

<------------->
[Dec 28 13:54:46] --- (11 headers 0 lines) ---
[Dec 28 13:54:46] Reliably Transmitting (NAT) to 192.168.184.195:5060:
OPTIONS sip:530@192.168.184.195:5060;rinstance=a8c81eb06df965a2;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 172.16.254.4:5060;branch=z9hG4bK57b6c22f;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.16.254.4>;tag=as172a1975
To: <sip:530@192.168.184.195:5060;rinstance=a8c81eb06df965a2;transport=UDP>
Contact: <sip:asterisk@172.16.254.4:5060>
Call-ID: 5dce5a7647cb1bb0782b4e3d4ce91740@172.16.254.4:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.19.0
Date: Fri, 28 Dec 2012 06:54:46 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[Dec 28 13:54:46] 
<--- SIP read from UDP:192.168.184.195:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.254.4:5060;branch=z9hG4bK57b6c22f;rport=5060
Contact: <sip:192.168.184.195:5060>
To: <sip:530@192.168.184.195:5060;rinstance=a8c81eb06df965a2;transport=UDP>;tag=6d41012c
From: "asterisk"<sip:asterisk@172.16.254.4>;tag=as172a1975
Call-ID: 5dce5a7647cb1bb0782b4e3d4ce91740@172.16.254.4:5060
CSeq: 102 OPTIONS
Accept: application/sdp, application/sdp
Accept-Language: en
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, X-cisco-serviceuri
User-Agent: Zoiper for Windows 2.38 rev.16635
Allow-Events: presence, kpml
Content-Length: 0

<------------->
[Dec 28 13:54:46] --- (14 headers 0 lines) ---
[Dec 28 13:54:46] Really destroying SIP dialog '5dce5a7647cb1bb0782b4e3d4ce91740@172.16.254.4:5060' Method: OPTIONS
[Dec 28 13:54:46] 
<--- SIP read from UDP:192.168.184.195:5060 --->


<------------->
[Dec 28 13:54:55] Reliably Transmitting (NAT) to 192.168.184.20:5060:
OPTIONS sip:431@192.168.184.20:5060 SIP/2.0
Via: SIP/2.0/UDP 172.16.254.4:5060;branch=z9hG4bK5af4ddf7;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.16.254.4>;tag=as56d1103e
To: <sip:431@192.168.184.20:5060>
Contact: <sip:asterisk@172.16.254.4:5060>
Call-ID: 4030efc73fa38818710effdb1635b2ef@172.16.254.4:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.19.0
Date: Fri, 28 Dec 2012 06:54:55 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[Dec 28 13:54:55] 
<--- SIP read from UDP:192.168.184.20:5060 --->
SIP/2.0 200 OK
To: <sip:431@192.168.184.20:5060>;tag=709518e227af20eci0
From: "asterisk" <sip:asterisk@172.16.254.4>;tag=as56d1103e
Call-ID: 4030efc73fa38818710effdb1635b2ef@172.16.254.4:5060
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP 172.16.254.4:5060;branch=z9hG4bK5af4ddf7
Server: Linksys/SPA3102-5.2.13(GW002)
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura, replaces

<------------->
[Dec 28 13:54:55] --- (10 headers 0 lines) ---
[Dec 28 13:54:55] Really destroying SIP dialog '4030efc73fa38818710effdb1635b2ef@172.16.254.4:5060' Method: OPTIONS
[Dec 28 13:54:55] Reliably Transmitting (NAT) to 192.168.184.20:5061:
OPTIONS sip:430@192.168.184.20:5061 SIP/2.0
Via: SIP/2.0/UDP 172.16.254.4:5060;branch=z9hG4bK3de133ca;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@172.16.254.4>;tag=as1c314699
To: <sip:430@192.168.184.20:5061>
Contact: <sip:asterisk@172.16.254.4:5060>
Call-ID: 24feee8d283b566b4b398fbe6dfe43a4@172.16.254.4:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.19.0
Date: Fri, 28 Dec 2012 06:54:55 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


---
[Dec 28 13:54:55] 
<--- SIP read from UDP:192.168.184.20:5061 --->
SIP/2.0 200 OK
To: <sip:430@192.168.184.20:5061>;tag=3e6ee9321cbf0cdci1
From: "asterisk" <sip:asterisk@172.16.254.4>;tag=as1c314699
Call-ID: 24feee8d283b566b4b398fbe6dfe43a4@172.16.254.4:5060
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP 172.16.254.4:5060;branch=z9hG4bK3de133ca
Server: Linksys/SPA3102-5.2.13(GW002)
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura, replaces

<------------->
[Dec 28 13:54:55] --- (10 headers 0 lines) ---
[Dec 28 13:54:55] Really destroying SIP dialog '24feee8d283b566b4b398fbe6dfe43a4@172.16.254.4:5060' Method: OPTIONS
cs17*CLI> 
cs17*CLI> 
cs17*CLI> 
cs17*CLI> 
[Dec 28 13:55:04] 
<--- SIP read from UDP:192.168.184.20:5060 --->
BYE sip:530@172.16.254.4:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.184.20:5060;branch=z9hG4bK-cafc3f6f
From: 431 <sip:431@192.168.185.196>;tag=445df880e3cba6c4o0
To: <sip:530@192.168.185.196>;tag=as729c8fed
Call-ID: 8b22f830-1e2b2ad4@192.168.184.20
CSeq: 103 BYE
Max-Forwards: 70
Authorization: Digest username="431",realm="880.ru",nonce="48ee131f",uri="sip:530@172.16.254.4:5060",algorithm=MD5,response="ac5430eed9c56ed4eba1e379d0abdcf6"
User-Agent: Linksys/SPA3102-5.2.13(GW002)
Content-Length: 0

<------------->
[Dec 28 13:55:04] --- (10 headers 0 lines) ---
[Dec 28 13:55:04] Sending to 192.168.184.20:5060 (NAT)
[Dec 28 13:55:04] Scheduling destruction of SIP dialog '8b22f830-1e2b2ad4@192.168.184.20' in 6400 ms (Method: BYE)
[Dec 28 13:55:04] 
<--- Transmitting (NAT) to 192.168.184.20:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.184.20:5060;branch=z9hG4bK-cafc3f6f;received=192.168.184.20;rport=5060
From: 431 <sip:431@192.168.185.196>;tag=445df880e3cba6c4o0
To: <sip:530@192.168.185.196>;tag=as729c8fed
Call-ID: 8b22f830-1e2b2ad4@192.168.184.20
CSeq: 103 BYE
Server: Asterisk PBX 1.8.19.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0


<------------>
[Dec 28 13:55:04] Scheduling destruction of SIP dialog '0cb4b56653b1ff7f4fa7e8ea53d86031@172.16.254.4:5060' in 6400 ms (Method: INVITE)
[Dec 28 13:55:04] set_destination: Parsing <sip:530@192.168.184.195:5060;rinstance=a8c81eb06df965a2;transport=UDP> for address/port to send to
[Dec 28 13:55:04] set_destination: set destination to 192.168.184.195:5060
[Dec 28 13:55:04] Reliably Transmitting (NAT) to 192.168.184.195:5060:
BYE sip:530@192.168.184.195:5060;rinstance=a8c81eb06df965a2;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 172.16.254.4:5060;branch=z9hG4bK59a09d08;rport
Max-Forwards: 70
From: "431" <sip:431@172.16.254.4>;tag=as113a3166
To: <sip:530@192.168.184.195:5060;rinstance=a8c81eb06df965a2;transport=UDP>;tag=23676357
Call-ID: 0cb4b56653b1ff7f4fa7e8ea53d86031@172.16.254.4:5060
CSeq: 103 BYE
User-Agent: Asterisk PBX 1.8.19.0
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


---
[Dec 28 13:55:04]   == Spawn extension (simplegate, 530, 1) exited non-zero on 'SIP/431-00000004'
[Dec 28 13:55:04] Retransmitting #1 (NAT) to 192.168.184.195:5060:
BYE sip:530@192.168.184.195:5060;rinstance=a8c81eb06df965a2;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 172.16.254.4:5060;branch=z9hG4bK59a09d08;rport
Max-Forwards: 70
From: "431" <sip:431@172.16.254.4>;tag=as113a3166
To: <sip:530@192.168.184.195:5060;rinstance=a8c81eb06df965a2;transport=UDP>;tag=23676357
Call-ID: 0cb4b56653b1ff7f4fa7e8ea53d86031@172.16.254.4:5060
CSeq: 103 BYE
User-Agent: Asterisk PBX 1.8.19.0
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


---
[Dec 28 13:55:04] 
<--- SIP read from UDP:192.168.184.195:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.254.4:5060;branch=z9hG4bK59a09d08;rport=5060
Contact: <sip:530@192.168.184.195:5060;rinstance=a8c81eb06df965a2;transport=UDP>
To: <sip:530@192.168.184.195:5060;rinstance=a8c81eb06df965a2;transport=UDP>;tag=23676357
From: "431"<sip:431@172.16.254.4>;tag=as113a3166
Call-ID: 0cb4b56653b1ff7f4fa7e8ea53d86031@172.16.254.4:5060
CSeq: 103 BYE
User-Agent: Zoiper for Windows 2.38 rev.16635
Content-Length: 0

<------------->
[Dec 28 13:55:04] --- (9 headers 0 lines) ---
[Dec 28 13:55:04] Really destroying SIP dialog '0cb4b56653b1ff7f4fa7e8ea53d86031@172.16.254.4:5060' Method: INVITE
[Dec 28 13:55:04] 
<--- SIP read from UDP:192.168.184.195:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.16.254.4:5060;branch=z9hG4bK59a09d08;rport=5060
Contact: <sip:530@192.168.184.195:5060;rinstance=a8c81eb06df965a2;transport=UDP>
To: <sip:530@192.168.184.195:5060;rinstance=a8c81eb06df965a2;transport=UDP>;tag=23676357
From: "431"<sip:431@172.16.254.4>;tag=as113a3166
Call-ID: 0cb4b56653b1ff7f4fa7e8ea53d86031@172.16.254.4:5060
CSeq: 103 BYE
User-Agent: Zoiper for Windows 2.38 rev.16635
Content-Length: 0

<------------->
[Dec 28 13:55:04] --- (9 headers 0 lines) ---
cs17*CLI> 
cs17*CLI> 
cs17*CLI> 
cs17*CLI> 

---------------------- RTP -----------------------------
[Dec 30 00:00:24]   == Using SIP RTP CoS mark 5
[Dec 30 00:00:24]     -- Executing [530@simplegate:1] Dial("SIP/431-00000016", "SIP/530,,tg") in new stack
[Dec 30 00:00:24]   == Using SIP RTP CoS mark 5
[Dec 30 00:00:24]     -- Called SIP/530
[Dec 30 00:00:25]     -- SIP/530-00000017 is ringing
[Dec 30 00:00:25]     -- SIP/530-00000017 is ringing
[Dec 30 00:00:25]     -- SIP/530-00000017 is ringing
[Dec 30 00:00:27]     -- SIP/530-00000017 answered SIP/431-00000016
[Dec 30 00:00:38] Sent Comfort Noise RTP packet to 192.168.184.20:16482 (type 02, seq 041015, ts 000000, len 000001)
[Dec 30 00:00:38] Sent Comfort Noise RTP packet to 192.168.184.195:8000 (type 02, seq 021875, ts 000000, len 000001)
[Dec 30 00:00:38] Got  RTP packet from    192.168.184.20:16482 (type 08, seq 003217, ts 521354364, len 000160)
[Dec 30 00:00:38] Sent RTP packet to      192.168.184.195:8000 (type 08, seq 021875, ts 521354360, len 000160)
[Dec 30 00:00:38] Got  RTP packet from    192.168.184.195:8000 (type 08, seq 060944, ts 1815478832, len 000160)
[Dec 30 00:00:38] Sent RTP packet to      192.168.184.20:16482 (type 08, seq 041015, ts 1815478832, len 000160)
[Dec 30 00:00:38] Got  RTP packet from    192.168.184.20:16482 (type 08, seq 003218, ts 521354524, len 000160)
[Dec 30 00:00:38] Sent RTP packet to      192.168.184.195:8000 (type 08, seq 021876, ts 521354520, len 000160)
[Dec 30 00:00:38] Got  RTP packet from    192.168.184.195:8000 (type 08, seq 060945, ts 1815478992, len 000160)
[Dec 30 00:00:38] Sent RTP packet to      192.168.184.20:16482 (type 08, seq 041016, ts 1815478992, len 000160)
[Dec 30 00:00:38] Got  RTP packet from    192.168.184.20:16482 (type 08, seq 003219, ts 521354684, len 000160)
[Dec 30 00:00:38] Sent RTP packet to      192.168.184.195:8000 (type 08, seq 021877, ts 521354680, len 000160)
[Dec 30 00:00:38] Got  RTP packet from    192.168.184.195:8000 (type 08, seq 060946, ts 1815479152, len 000160)
[Dec 30 00:00:38] Sent RTP packet to      192.168.184.20:16482 (type 08, seq 041017, ts 1815479152, len 000160)
[Dec 30 00:00:38] Got  RTP packet from    192.168.184.20:16482 (type 08, seq 003220, ts 521354844, len 000160)
[Dec 30 00:00:38] Sent RTP packet to      192.168.184.195:8000 (type 08, seq 021878, ts 521354840, len 000160)
[Dec 30 00:00:38] Got  RTP packet from    192.168.184.195:8000 (type 08, seq 060947, ts 1815479312, len 000160)
[Dec 30 00:00:38] Sent RTP packet to      192.168.184.20:16482 (type 08, seq 041018, ts 1815479312, len 000160)
[Dec 30 00:00:38] Got  RTP packet from    192.168.184.20:16482 (type 08, seq 003221, ts 521355004, len 000160)
[Dec 30 00:00:38] Sent RTP packet to      192.168.184.195:8000 (type 08, seq 021879, ts 521355000, len 000160)
[Dec 30 00:00:38] Got  RTP packet from    192.168.184.195:8000 (type 08, seq 060948, ts 1815479472, len 000160)
[Dec 30 00:00:38] Sent RTP packet to      192.168.184.20:16482 (type 08, seq 041019, ts 1815479472, len 000160)
[Dec 30 00:00:38] Got  RTP packet from    192.168.184.20:16482 (type 08, seq 003222, ts 521355164, len 000160)
[Dec 30 00:00:38] Sent RTP packet to      192.168.184.195:8000 (type 08, seq 021880, ts 521355160, len 000160)
[Dec 30 00:00:38] Got  RTP packet from    192.168.184.195:8000 (type 08, seq 060949, ts 1815479632, len 000160)
[Dec 30 00:00:38] Sent RTP packet to      192.168.184.20:16482 (type 08, seq 041020, ts 1815479632, len 000160)
[Dec 30 00:00:38] Got  RTP packet from    192.168.184.20:16482 (type 08, seq 003223, ts 521355324, len 000160)
[Dec 30 00:00:38] Sent RTP packet to      192.168.184.195:8000 (type 08, seq 021881, ts 521355320, len 000160)
[Dec 30 00:00:38] Got  RTP packet from    192.168.184.195:8000 (type 08, seq 060950, ts 1815479792, len 000160)
[Dec 30 00:00:38] Sent RTP packet to      192.168.184.20:16482 (type 08, seq 041021, ts 1815479792, len 000160)
[Dec 30 00:00:38] Got  RTP packet from    192.168.184.20:16482 (type 08, seq 003224, ts 521355484, len 000160)
[Dec 30 00:00:38] Sent RTP packet to      192.168.184.195:8000 (type 08, seq 021882, ts 521355480, len 000160)
[Dec 30 00:00:38] Got  RTP packet from    192.168.184.195:8000 (type 08, seq 060951, ts 1815479952, len 000160)
[Dec 30 00:00:38] Sent RTP packet to      192.168.184.20:16482 (type 08, seq 041022, ts 1815479952, len 000160)
[Dec 30 00:00:38] Got  RTP packet from    192.168.184.195:8000 (type 08, seq 060952, ts 1815480112, len 000160)
[Dec 30 00:00:38] Sent RTP packet to      192.168.184.20:16482 (type 08, seq 041023, ts 1815480112, len 000160)
[Dec 30 00:00:38] Got  RTP packet from    192.168.184.20:16482 (type 08, seq 003225, ts 521355644, len 000160)
[Dec 30 00:00:38] Sent RTP packet to      192.168.184.195:8000 (type 08, seq 021883, ts 521355640, len 000160)
[Dec 30 00:00:38] Got  RTP packet from    192.168.184.195:8000 (type 08, seq 060953, ts 1815480272, len 000160)
[Dec 30 00:00:38] Sent RTP packet to      192.168.184.20:16482 (type 08, seq 041024, ts 1815480272, len 000160)
[Dec 30 00:00:38] Got  RTP packet from    192.168.184.20:16482 (type 08, seq 003226, ts 521355804, len 000160)
[Dec 30 00:00:38] Sent RTP packet to      192.168.184.195:8000 (type 08, seq 021884, ts 521355800, len 000160)
[Dec 30 00:00:38] Got  RTP packet from    192.168.184.195:8000 (type 08, seq 060954, ts 1815480432, len 000160)
[Dec 30 00:00:38] Sent RTP packet to      192.168.184.20:16482 (type 08, seq 041025, ts 1815480432, len 000160)
[Dec 30 00:00:38] Got  RTP packet from    192.168.184.20:16482 (type 08, seq 003227, ts 521355964, len 000160)
[Dec 30 00:00:38] Sent RTP packet to      192.168.184.195:8000 (type 08, seq 021885, ts 521355960, len 000160)
[Dec 30 00:00:38] Got  RTP packet from    192.168.184.195:8000 (type 08, seq 060955, ts 1815480592, len 000160)
[Dec 30 00:00:38] Sent RTP packet to      192.168.184.20:16482 (type 08, seq 041026, ts 1815480592, len 000160)
[Dec 30 00:00:39] Got  RTP packet from    192.168.184.20:16482 (type 08, seq 003228, ts 521356124, len 000160)
[Dec 30 00:00:39] Sent RTP packet to      192.168.184.195:8000 (type 08, seq 021886, ts 521356120, len 000160)
[Dec 30 00:00:39] Got  RTP packet from    192.168.184.195:8000 (type 08, seq 060956, ts 1815480752, len 000160)
[Dec 30 00:00:39] Sent RTP packet to      192.168.184.20:16482 (type 08, seq 041027, ts 1815480752, len 000160)
[Dec 30 00:00:39] Got  RTP packet from    192.168.184.20:16482 (type 08, seq 003229, ts 521356284, len 000160)
[Dec 30 00:00:39] Sent RTP packet to      192.168.184.195:8000 (type 08, seq 021887, ts 521356280, len 000160)
[Dec 30 00:00:39] Got  RTP packet from    192.168.184.195:8000 (type 08, seq 060957, ts 1815480912, len 000160)
[Dec 30 00:00:39] Sent RTP packet to      192.168.184.20:16482 (type 08, seq 041028, ts 1815480912, len 000160)
[Dec 30 00:00:39] Got  RTP packet from    192.168.184.20:16482 (type 08, seq 003230, ts 521356444, len 000160)
[Dec 30 00:00:39] Sent RTP packet to      192.168.184.195:8000 (type 08, seq 021888, ts 521356440, len 000160)
[Dec 30 00:00:39] Got  RTP packet from    192.168.184.195:8000 (type 08, seq 060958, ts 1815481072, len 000160)
[Dec 30 00:00:39] Sent RTP packet to      192.168.184.20:16482 (type 08, seq 041029, ts 1815481072, len 000160)
[Dec 30 00:00:39] Got  RTP packet from    192.168.184.20:16482 (type 08, seq 003231, ts 521356604, len 000160)
[Dec 30 00:00:39] Sent RTP packet to      192.168.184.195:8000 (type 08, seq 021889, ts 521356600, len 000160)
[Dec 30 00:00:39] Got  RTP packet from    192.168.184.195:8000 (type 08, seq 060959, ts 1815481232, len 000160)
[Dec 30 00:00:39] Sent RTP packet to      192.168.184.20:16482 (type 08, seq 041030, ts 1815481232, len 000160)
[Dec 30 00:00:39] Got  RTP packet from    192.168.184.20:16482 (type 08, seq 003232, ts 521356764, len 000160)
[Dec 30 00:00:39] Sent RTP packet to      192.168.184.195:8000 (type 08, seq 021890, ts 521356760, len 000160)
[Dec 30 00:00:39] Got  RTP packet from    192.168.184.195:8000 (type 08, seq 060960, ts 1815481392, len 000160)
[Dec 30 00:00:39] Sent RTP packet to      192.168.184.20:16482 (type 08, seq 041031, ts 1815481392, len 000160)
[Dec 30 00:00:39] Got  RTP packet from    192.168.184.20:16482 (type 08, seq 003233, ts 521356924, len 000160)
[Dec 30 00:00:39] Sent RTP packet to      192.168.184.195:8000 (type 08, seq 021891, ts 521356920, len 000160)
[Dec 30 00:00:39] Got  RTP packet from    192.168.184.195:8000 (type 08, seq 060961, ts 1815481552, len 000160)
[Dec 30 00:00:39] Sent RTP packet to      192.168.184.20:16482 (type 08, seq 041032, ts 1815481552, len 000160)
[Dec 30 00:00:39] Got  RTP packet from    192.168.184.20:16482 (type 08, seq 003234, ts 521357084, len 000160)
[Dec 30 00:00:39] Sent RTP packet to      192.168.184.195:8000 (type 08, seq 021892, ts 521357080, len 000160)
[Dec 30 00:00:39] Got  RTP packet from    192.168.184.195:8000 (type 08, seq 060962, ts 1815481712, len 000160)
[Dec 30 00:00:39] Sent RTP packet to      192.168.184.20:16482 (type 08, seq 041033, ts 1815481712, len 000160)
[Dec 30 00:00:39] Got  RTP packet from    192.168.184.20:16482 (type 08, seq 003235, ts 521357244, len 000160)
[Dec 30 00:00:39] Sent RTP packet to      192.168.184.195:8000 (type 08, seq 021893, ts 521357240, len 000160)
[Dec 30 00:00:39] Got  RTP packet from    192.168.184.195:8000 (type 08, seq 060963, ts 1815481872, len 000160)
[Dec 30 00:00:39] Sent RTP packet to      192.168.184.20:16482 (type 08, seq 041034, ts 1815481872, len 000160)
cs17*CLI> rtp set debug off
RTP Debugging Disabled
[Dec 30 00:00:42]   == Spawn extension (simplegate, 530, 1) exited non-zero on 'SIP/431-00000016'
cs17*CLI>
awsswa
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Re: нет звука до истечения rtpkeepalive

Сообщение awsswa »

NAT нету - а везде только надпись про нат и сверкает

Sending to 192.168.184.20:5060 (NAT)

<--- SIP read from UDP:192.168.184.20:5060 --->
ACK sip:530@192.168.185.196 SIP/2.0
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d771
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Re: нет звука до истечения rtpkeepalive

Сообщение d771 »

Это включено в профиле (nat=yes) - мало ли откуда софтфон подключится?
В итоге попробовал отключить - ничего не изменилось...

Есть какие-нибудь еще идеи?
ded
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Re: нет звука до истечения rtpkeepalive

Сообщение ded »

Изображение
Изображение
awsswa
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Re: нет звука до истечения rtpkeepalive

Сообщение awsswa »

Не отключать надо а настраивать
192.168.--184--.20 ---> 192.168.--185--.196

http://asterisk-support.ru/question/153 ... oi-pochte/
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d771
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Re: нет звука до истечения rtpkeepalive

Сообщение d771 »

Картинки эти видел еще при регистрации, если бы по ним удалось решить задачу - не писал бы :)

184.* и 185.* это корректные адреса.
Действительно, астериск находится в одной подсети (185), клиенты - в другой (184)
Но трафик бегает, пинги, все остальное, без исключений, фильтрации на L3 форвардинге тоже никакой.

Попробовал transmit_silence=yes - результата нет.
ded
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Re: нет звука до истечения rtpkeepalive

Сообщение ded »

Пинги - не показатель, и не даёт ответа оа вопрос: есть НАТ между 192.168.184.0/25 и 192.168.185.0/24 или там просто маршрутизация.
И совсем уж непонятно Via: SIP/2.0/UDP 172.16.254.4:5060;branch=z9hG4bK5eb6c5b1;rport
Откуда в этой схеме 172.16.254.4?

Простой тест - начать со звонка на голосовой сервис, например - голосовую почту, или Music on hold. Должно всё слышать.
В общем - если дампить тишину - раговор, при которм нет голоса, оба плеча, то будет видно куда и откуда бегут rtp пакеты.
d771
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Re: нет звука до истечения rtpkeepalive

Сообщение d771 »

172.16.254.4 это второй интерфейс на астериске, он не подключен к этой сети.
в sip.conf при этом udpbindaddr=192.168.185.196

Зачем астериск его берет - для меня тоже загадка. Он даже в другом vlan, хоть и на том же физическом интерфейсе. В любом случае INPUT на диапазон портов rtp открыт с любого интерфейса и ип.

НАТ'а 100% нет. на машине с астериском в таблице NAT 0 правил, по дороге есть L3 свич (но он NAT не умеет).
про голосовую почту - спасибо, попробую.
ded
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Re: нет звука до истечения rtpkeepalive

Сообщение ded »

Что-то намудрили там с интерфейсами. ЧТобы на одном физическом интерфейсе наклеить несколько ИП адресов и VLANs надо сильно разбираться в загадках как на уровне L3 так и на L7.
Я бы сильно упростил сетевую конфигурацию, и всё бы у вас пошло. Смотрите sip show settings, там не должно быть 172.16.254.4, добавляйте localnet=

Схему соединения Астериск и клиентов я так и не понял.
d771
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Re: нет звука до истечения rtpkeepalive

Сообщение d771 »

Хм, ну я сильно разбираюсь в загадках L3 и L7, просто в астериске слабо :)

Сервер астериска смотрит в два vlan. Одни из них (на котором у астериска адрес 192.168.185.196), терминируется на L3 свиче.
Свичу разрешено форвардить эту подсеть куда угодной на любой из его L3 ipif.

астериск => серверный vlan => L3 свич => клиентский vlan => клиенты

второй vlan для управления, я уже пробовал оставить всего один vlan, его адрес действительно исчезает, но это ничего не меняет.
PRIME_BBCODE_SPOILER_SHOW PRIME_BBCODE_SPOILER:

Код: Выделить всё

cs17*CLI> sip show settings


Global Settings:
----------------
  UDP Bindaddress:        192.168.185.196:5060
  TCP SIP Bindaddress:    192.168.185.196:5060
  TLS SIP Bindaddress:    Disabled
  Videosupport:           No
  Textsupport:            No
  Ignore SDP sess. ver.:  No
  AutoCreate Peer:        No
  Match Auth Username:    No
  Allow unknown access:   No
  Allow subscriptions:    Yes
  Allow overlap dialing:  No
  Allow promisc. redir:   No
  Enable call counters:   No
  SIP domain support:     No
  Realm. auth:            No
  Our auth realm          test.ru
  Use domains as realms:  No
  Call to non-local dom.: Yes
  URI user is phone no:   No
  Always auth rejects:    Yes
  Direct RTP setup:       No
  User Agent:             Asterisk PBX 1.8.19.0
  SDP Session Name:       Asterisk PBX 1.8.19.0
  SDP Owner Name:         root
  Reg. context:           (not set)
  Regexten on Qualify:    No
  Legacy userfield parse: No
  Caller ID:              asterisk
  From: Domain:           
  Record SIP history:     Off
  Call Events:            Off
  Auth. Failure Events:   Off
  T.38 support:           No
  T.38 EC mode:           Unknown
  T.38 MaxDtgrm:          -1
  SIP realtime:           Disabled
  Qualify Freq :          60000 ms
  Q.850 Reason header:    No
  Store SIP_CAUSE:        No

Network QoS Settings:
---------------------------
  IP ToS SIP:             CS0
  IP ToS RTP audio:       CS0
  IP ToS RTP video:       CS0
  IP ToS RTP text:        CS0
  802.1p CoS SIP:         4
  802.1p CoS RTP audio:   5
  802.1p CoS RTP video:   6
  802.1p CoS RTP text:    5
  Jitterbuffer enabled:   No

Network Settings:
---------------------------
  SIP address remapping:  Disabled, no localnet list
  Externhost:             <none>
  Externaddr:             (null)
  Externrefresh:          10

Global Signalling Settings:
---------------------------
  Codecs:                 0x80000008000e (gsm|ulaw|alaw|h263|testlaw)
  Codec Order:            none
  Relax DTMF:             Yes
  RFC2833 Compensation:   No
  Symmetric RTP:          No
  Compact SIP headers:    No
  RTP Keepalive:          1 
  RTP Timeout:            60 
  RTP Hold Timeout:       0 (Disabled)
  MWI NOTIFY mime type:   application/simple-message-summary
  DNS SRV lookup:         No
  Pedantic SIP support:   Yes
  Reg. min duration       60 secs
  Reg. max duration:      3600 secs
  Reg. default duration:  120 secs
  Outbound reg. timeout:  20 secs
  Outbound reg. attempts: 0
  Notify ringing state:   Yes
    Include CID:          No
  Notify hold state:      No
  SIP Transfer mode:      open
  Max Call Bitrate:       384 kbps
  Auto-Framing:           No
  Outb. proxy:            <not set> 
  Session Timers:         Accept
  Session Refresher:      uas
  Session Expires:        1800 secs
  Session Min-SE:         90 secs
  Timer T1:               500
  Timer T1 minimum:       100
  Timer B:                32000
  No premature media:     Yes
  Max forwards:           70

Default Settings:
-----------------
  Allowed transports:     UDP
  Outbound transport:     UDP
  Context:                incoming
  Force rport:            Yes
  DTMF:                   rfc2833
  Qualify:                0
  Use ClientCode:         No
  Progress inband:        Never
  Language:               en
  MOH Interpret:          default
  MOH Suggest:            
  Voice Mail Extension:   asterisk

----
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