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Откуда: Moscow
Сообщений: 1
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== Using SIP RTP CoS mark 5
Linux Ubuntu 10.10 2.6.35-23-generic
Asterisk 1.6.2.15
sip.conf
[TST]
type=friend
secret=welcome
context=incoming
host=dynamic
disallow=all
allow=ulaw
Пытаюсь подцепиться Ekiga, регистрируется, но при попытке совершить звонок сыплет сообщение в кли == Using SIP RTP CoS mark 5
Сделал дебаг но малый опыт не позволяет черпать из него выводы:
<--- SIP read from UDP:192.168.0.103:5060 --->
INVITE sip:xxxxxxxxxxxx@192.168.0.105 SIP/2.0
Date: Sun, 12 Dec 2010 13:16:52 GMT
CSeq: 1 INVITE
Via: SIP/2.0/UDP 192.168.0.103:5060;branch=z9hG4bKb226f8c0-5f04-e011-85b0-00216baa5106;rport
User-Agent: Ekiga/3.2.7
From: "TST" <sip:TST@192.168.0.105>;tag=0426f7c0-5f04-e011-85b0-00216baa5106
Call-ID: ce2ef7c0-5f04-e011-85b0-00216baa5106@pc
To: <sip:xxxxxxxxxxxx@192.168.0.105>
Contact: <sip:TST@192.168.0.103>
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,SUBSCRIBE,NOTIFY,REFER,MESSAGE,INFO,PING
Content-Type: application/sdp
Content-Length: 701
Max-Forwards: 70
v=0
o=- 1292159812 1 IN IP4 192.168.0.103
s=Opal SIP Session
c=IN IP4 192.168.0.103
t=0 0
m=audio 5062 RTP/AVP 110 0 8 3 9 109 119 118 101 122
a=sendrecv
a=rtpmap:110 Speex/16000/1
a=fmtp:110 sr=16000,mode=any
a=rtpmap:0 PCMU/8000/1
a=rtpmap:8 PCMA/8000/1
a=rtpmap:3 gsm/8000/1
a=rtpmap:9 G722/8000/1
a=rtpmap:109 Speex/8000/1
a=fmtp:109 sr=8000,mode=any
a=rtpmap:119 CELT/48000/1
a=rtpmap:118 CELT/32000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16,32,36
a=rtpmap:122 NSE/8000
a=fmtp:122 192-193
m=video 5064 RTP/AVP 121 31
b=AS:4096
b=TIAS:4096000
a=sendrecv
a=rtpmap:121 theora/90000
a=fmtp:121 height=576;width=704
a=rtpmap:31 h261/90000
a=fmtp:31 CIF=1;QCIF=1
<------------->
--- (13 headers 29 lines) ---
== Using SIP RTP CoS mark 5
Sending to 192.168.0.103 : 5060 (no NAT)
Using INVITE request as basis request - ce2ef7c0-5f04-e011-85b0-00216baa5106@pc
Found peer 'TST' for 'TST' from 192.168.0.103:5060
<--- Reliably Transmitting (no NAT) to 192.168.0.103:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.0.103:5060;branch=z9hG4bKb226f8c0-5f04-e011-85b0-00216baa5106;received=192.168.0.103;rport=5060
From: "TST" <sip:TST@192.168.0.105>;tag=0426f7c0-5f04-e011-85b0-00216baa5106
To: <sip:xxxxxxxxxxxx@192.168.0.105>;tag=as6f5a50e5
Call-ID: ce2ef7c0-5f04-e011-85b0-00216baa5106@pc
CSeq: 1 INVITE
Server: Asterisk PBX 1.6.2.15
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1c4cfcec"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog 'ce2ef7c0-5f04-e011-85b0-00216baa5106@pc' in 32000 ms (Method: INVITE)
<--- SIP read from UDP:192.168.0.103:5060 --->
ACK sip:xxxxxxxxxxxx@192.168.0.105 SIP/2.0
CSeq: 1 ACK
Via: SIP/2.0/UDP 192.168.0.103:5060;branch=z9hG4bKb226f8c0-5f04-e011-85b0-00216baa5106;rport
From: "TST" <sip:TST@192.168.0.105>;tag=0426f7c0-5f04-e011-85b0-00216baa5106
Call-ID: ce2ef7c0-5f04-e011-85b0-00216baa5106@pc
To: <sip:xxxxxxxxxxxx@192.168.0.105>;tag=as6f5a50e5
Content-Length: 0
Max-Forwards: 70
<------------->
--- (8 headers 0 lines) ---
<--- SIP read from UDP:192.168.0.103:5060 --->
INVITE sip:xxxxxxxxxxxx@192.168.0.105 SIP/2.0
Date: Sun, 12 Dec 2010 13:16:52 GMT
CSeq: 2 INVITE
v: SIP/2.0/UDP 192.168.0.103:5060;branch=z9hG4bK40f0fac0-5f04-e011-85b0-00216baa5106;rport
User-Agent: Ekiga/3.2.7
Authorization: Digest username="TST", realm="asterisk", nonce="1c4cfcec", uri="sip:xxxxxxxxxxxx@192.168.0.105", algorithm=MD5, response="b57008cdfdccd3656e8236cb53511932"
f: "TST" <sip:TST@192.168.0.105>;tag=0426f7c0-5f04-e011-85b0-00216baa5106
i: ce2ef7c0-5f04-e011-85b0-00216baa5106@pc
t: <sip:xxxxxxxxxxxx@192.168.0.105>
m: <sip:TST@192.168.0.103>
Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,SUBSCRIBE,NOTIFY,REFER,MESSAGE,INFO,PING
c: application/sdp
l: 701
Max-Forwards: 70
v=0
o=- 1292159812 1 IN IP4 192.168.0.103
s=Opal SIP Session
c=IN IP4 192.168.0.103
t=0 0
m=audio 5062 RTP/AVP 110 0 8 3 9 109 119 118 101 122
a=sendrecv
a=rtpmap:110 Speex/16000/1
a=fmtp:110 sr=16000,mode=any
a=rtpmap:0 PCMU/8000/1
a=rtpmap:8 PCMA/8000/1
a=rtpmap:3 gsm/8000/1
a=rtpmap:9 G722/8000/1
a=rtpmap:109 Speex/8000/1
a=fmtp:109 sr=8000,mode=any
a=rtpmap:119 CELT/48000/1
a=rtpmap:118 CELT/32000/1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16,32,36
a=rtpmap:122 NSE/8000
a=fmtp:122 192-193
m=video 5064 RTP/AVP 121 31
b=AS:4096
b=TIAS:4096000
a=sendrecv
a=rtpmap:121 theora/90000
a=fmtp:121 height=576;width=704
a=rtpmap:31 h261/90000
a=fmtp:31 CIF=1;QCIF=1
<------------->
--- (14 headers 29 lines) ---
Sending to 192.168.0.103 : 5060 (no NAT)
Using INVITE request as basis request - ce2ef7c0-5f04-e011-85b0-00216baa5106@pc
Found peer 'TST' for 'TST' from 192.168.0.103:5060
Found RTP audio format 110
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 9
Found RTP audio format 109
Found RTP audio format 119
Found RTP audio format 118
Found RTP audio format 101
Found RTP audio format 122
Found audio description format Speex for ID 110
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format gsm for ID 3
Found audio description format G722 for ID 9
Found audio description format Speex for ID 109
Found audio description format CELT for ID 119
Found audio description format CELT for ID 118
Found audio description format telephone-event for ID 101
Found audio description format NSE for ID 122
Found RTP video format 121
Found RTP video format 31
Found video description format h261 for ID 31
Capabilities: us - 0x4 (ulaw), peer - audio=0x120e (gsm|ulaw|alaw|speex|g722)/video=0x40000 (h261)/text=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 192.168.0.103:5062
Looking for xxxxxxxxxxxx in incoming (domain 192.168.0.105)
<--- Reliably Transmitting (no NAT) to 192.168.0.103:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.0.103:5060;branch=z9hG4bK40f0fac0-5f04-e011-85b0-00216baa5106;received=192.168.0.103;rport=5060
From: "TST" <sip:TST@192.168.0.105>;tag=0426f7c0-5f04-e011-85b0-00216baa5106
To: <sip:xxxxxxxxxxxx@192.168.0.105>;tag=as6f5a50e5
Call-ID: ce2ef7c0-5f04-e011-85b0-00216baa5106@pc
CSeq: 2 INVITE
Server: Asterisk PBX 1.6.2.15
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog 'ce2ef7c0-5f04-e011-85b0-00216baa5106@pc' in 32000 ms (Method: INVITE)
<--- SIP read from UDP:192.168.0.103:5060 --->
ACK sip:xxxxxxxxxxxx@192.168.0.105 SIP/2.0
CSeq: 2 ACK
Via: SIP/2.0/UDP 192.168.0.103:5060;branch=z9hG4bK40f0fac0-5f04-e011-85b0-00216baa5106;rport
Authorization: Digest username="TST", realm="asterisk", nonce="1c4cfcec", uri="sip:xxxxxxxxxxxx@192.168.0.105", algorithm=MD5, response="33c2b75bf04ba2639abd67eac25065ee"
From: "TST" <sip:TST@192.168.0.105>;tag=0426f7c0-5f04-e011-85b0-00216baa5106
Call-ID: ce2ef7c0-5f04-e011-85b0-00216baa5106@pc
To: <sip:xxxxxxxxxxxx@192.168.0.105>;tag=as6f5a50e5
Content-Length: 0
Max-Forwards: 70
Спасибо за помощь!
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