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Проблема с возвратом голоса после приема факса через asterisk
Здравствуйте, гуру asterisk
Проблемка приема голоса после приема факса
fax1 - Станция PSTN - cisco - asterisk - cisco2811 - Станция
PSTN - fax2
fax1 звонит и отправляет факс, fax2 принимает факс
но
после прохождения факса, обратно голос не возвращается, тишина с обеих
сторон.
в Астериске после прохождения факса выходит ошибка:
WARNING[30577]: chan_sip.c:14056 handle_request_invite: RTP re-
invite after T38 session not handled yet !
SIP/2.0 488 Not Acceptable Here (unsupported)
Проблема в коде астериска или в настройках?
Просветите в чем дело может быть, пожалуйста.
За ранее Благодарен.
Детали:
настройки sip.conf
[general]
context=default
srvlookup=yes
bindport=5069
bindaddr=0.0.0.0
allowguest=no
allowoverlap=no
t38pt_udptl = yes
language=en
disallow=all
allow=alaw
allow=ulaw
allow=gsm
[VSATtestfax]
type=peer
context=from_test
host=REALIP-1
insecure=port,invite
qualify=1000
nat=no
canreinvite=no
[cisco2811]
type=peer
context=from_test
host=172.30.3.119
insecure=port,invite
qualify=1000
nat=yes
canreinvite=yes
Вот debug:
wsh*CLI>
wsh*CLI>
<--- SIP read from REALIP-1:61016 --->
INVITE sip:74112420008@REALIP-2:5069 SIP/2.0
Via: SIP/2.0/UDP REALIP-1:5060;branch=z9hG4bK2022352
From: <sip:4113999999@REALIP-1>;tag=14FA5264-65F
To: <sip:74112420008@REALIP-2>
Date: Mon, 26 Jan 2009 04:12:40 GMT
Call-ID: 687504BE-EA9611DD-9C91FD15-F90032C4@REALIP-1
Supported: 100rel,timer,replaces
Min-SE: 1800
Cisco-Guid: 1752339486-3935703517-2166554647-1493462536
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER
CSeq: 101 INVITE
Max-Forwards: 70
Remote-Party-ID: <sip:4113999999@REALIP-1>;party=calling;screen=yes;privacy=off
Timestamp: 1232943160
Contact: <sip:4113999999@REALIP-1:5060>
Expires: 300
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Length: 270
v=0
o=CiscoSystemsSIP-GW-UserAgent 499 7954 IN IP4 REALIP-1
s=SIP Call
c=IN IP4 REALIP-1
t=0 0
m=audio 19116 RTP/AVP 8 101 19
c=IN IP4 REALIP-1
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:19 CN/8000
a=ptime:20
<------------->
--- (20 headers 12 lines) ---
Sending to REALIP-1 : 5060 (no NAT)
Using INVITE request as basis request - 687504BE-EA9611DD-9C91FD15-F90032C4@REALIP-1
Found peer 'VSATtestfax'
Found RTP audio format 8
Found RTP audio format 101
Found RTP audio format 19
Peer audio RTP is at port REALIP-1:19116
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Found audio description format CN for ID 19
Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x3 (telephone-event|CN), combined - 0x1 (telephone-event)
Peer audio RTP is at port REALIP-1:19116
Looking for 74112420008 in from_test (domain REALIP-2)
list_route: hop: <sip:4113999999@REALIP-1:5060>
<--- Transmitting (no NAT) to REALIP-1:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP REALIP-1:5060;branch=z9hG4bK2022352;received=REALIP-1
From: <sip:4113999999@REALIP-1>;tag=14FA5264-65F
To: <sip:74112420008@REALIP-2>
Call-ID: 687504BE-EA9611DD-9C91FD15-F90032C4@REALIP-1
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:74112420008@REALIP-2:5069>
Content-Length: 0
<------------>
-- Executing [74112420008@from_test:1] NoCDR("SIP/REALIP-1-081e7748", "") in new stack
-- Executing [74112420008@from_test:2] Macro("SIP/REALIP-1-081e7748", "incoming-from-test1|84112420008|TEST") in new stack
-- Executing [s@macro-incoming-from-test1:1] NoCDR("SIP/REALIP-1-081e7748", "") in new stack
-- Executing [s@macro-incoming-from-test1:2] NoOp("SIP/REALIP-1-081e7748", "Call from TEST to YKT 84112420008") in new stack
-- Executing [s@macro-incoming-from-test1:3] Set("SIP/REALIP-1-081e7748", "GROUP()=TEST") in new stack
-- Executing [s@macro-incoming-from-test1:4] Set("SIP/REALIP-1-081e7748", "TEST=1") in new stack
-- Executing [s@macro-incoming-from-test1:5] GotoIf("SIP/REALIP-1-081e7748", "0?s-CONGESTION|1") in new stack
-- Executing [s@macro-incoming-from-test1:6] ResetCDR("SIP/REALIP-1-081e7748", "") in new stack
-- Executing [s@macro-incoming-from-test1:7] Set("SIP/REALIP-1-081e7748", "CDR(accountcode)=from_test") in new stack
-- Executing [s@macro-incoming-from-test1:8] Dial("SIP/REALIP-1-081e7748", "SIP/84112420008@cisco2811|60") in new stack
Audio is at 172.30.3.118 port 11756
Adding codec 0x8 (alaw) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 172.30.3.119:5060:
INVITE sip:84112420008@172.30.3.119 SIP/2.0
Via: SIP/2.0/UDP 172.30.3.118:5069;branch=z9hG4bK4b733645;rport
From: "4113999999" <sip:4113999999@172.30.3.118:5069>;tag=as0993a0e0
To: <sip:84112420008@172.30.3.119>
Contact: <sip:4113999999@172.30.3.118:5069>
Call-ID: 021f0c8957a27161116586045875b092@172.30.3.118
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 26 Jan 2009 04:09:19 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 285
v=0
o=root 2932 2932 IN IP4 172.30.3.118
s=session
c=IN IP4 172.30.3.118
t=0 0
m=audio 11756 RTP/AVP 8 0 3 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
-- Called 84112420008@cisco2811
wsh*CLI>
<--- SIP read from 172.30.3.119:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.30.3.118:5069;branch=z9hG4bK4b733645;rport
From: "4113999999" <sip:4113999999@172.30.3.118:5069>;tag=as0993a0e0
To: <sip:84112420008@172.30.3.119>;tag=1E3053F0-46D
Date: Mon, 26 Jan 2009 04:14:35 GMT
Call-ID: 021f0c8957a27161116586045875b092@172.30.3.118
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 INVITE
Allow-Events: telephone-event
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
<--- SIP read from 172.30.3.119:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 172.30.3.118:5069;branch=z9hG4bK4b733645;rport
From: "4113999999" <sip:4113999999@172.30.3.118:5069>;tag=as0993a0e0
To: <sip:84112420008@172.30.3.119>;tag=1E3053F0-46D
Date: Mon, 26 Jan 2009 04:14:35 GMT
Call-ID: 021f0c8957a27161116586045875b092@172.30.3.118
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER
Allow-Events: telephone-event
Contact: <sip:84112420008@172.30.3.119:5060>
Content-Disposition: session;handling=required
Content-Type: application/sdp
Content-Length: 247
v=0
o=CiscoSystemsSIP-GW-UserAgent 1244 1895 IN IP4 172.30.3.119
s=SIP Call
c=IN IP4 172.30.3.119
t=0 0
m=audio 16656 RTP/AVP 8 101
c=IN IP4 172.30.3.119
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
<------------->
--- (14 headers 11 lines) ---
Found RTP audio format 8
Found RTP audio format 101
Peer audio RTP is at port 172.30.3.119:16656
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x8 (alaw)/video=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 172.30.3.119:16656
-- SIP/cisco2811-081fa2f0 is making progress passing it to SIP/REALIP-1-081e7748
Audio is at REALIP-2 port 24142
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
<--- Transmitting (no NAT) to REALIP-1:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP REALIP-1:5060;branch=z9hG4bK2022352;received=REALIP-1
From: <sip:4113999999@REALIP-1>;tag=14FA5264-65F
To: <sip:74112420008@REALIP-2>;tag=as0a6ffa18
Call-ID: 687504BE-EA9611DD-9C91FD15-F90032C4@REALIP-1
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:74112420008@REALIP-2:5069>
Content-Type: application/sdp
Content-Length: 238
v=0
o=root 2932 2932 IN IP4 REALIP-2
s=session
c=IN IP4 REALIP-2
t=0 0
m=audio 24142 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
<------------>
wsh*CLI>
<--- SIP read from 172.30.3.119:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 172.30.3.118:5069;branch=z9hG4bK4b733645;rport
From: "4113999999" <sip:4113999999@172.30.3.118:5069>;tag=as0993a0e0
To: <sip:84112420008@172.30.3.119>;tag=1E3053F0-46D
Date: Mon, 26 Jan 2009 04:14:35 GMT
Call-ID: 021f0c8957a27161116586045875b092@172.30.3.118
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER
Allow-Events: telephone-event
Contact: <sip:84112420008@172.30.3.119:5060>
Content-Disposition: session;handling=required
Content-Type: application/sdp
Content-Length: 247
v=0
o=CiscoSystemsSIP-GW-UserAgent 1244 1895 IN IP4 172.30.3.119
s=SIP Call
c=IN IP4 172.30.3.119
t=0 0
m=audio 16656 RTP/AVP 8 101
c=IN IP4 172.30.3.119
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
<------------->
--- (14 headers 11 lines) ---
Found RTP audio format 8
Found RTP audio format 101
Peer audio RTP is at port 172.30.3.119:16656
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x8 (alaw)/video=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 172.30.3.119:16656
-- SIP/cisco2811-081fa2f0 is making progress passing it to SIP/REALIP-1-081e7748
wsh*CLI>
<--- SIP read from 172.30.3.119:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.3.118:5069;branch=z9hG4bK4b733645;rport
From: "4113999999" <sip:4113999999@172.30.3.118:5069>;tag=as0993a0e0
To: <sip:84112420008@172.30.3.119>;tag=1E3053F0-46D
Date: Mon, 26 Jan 2009 04:14:35 GMT
Call-ID: 021f0c8957a27161116586045875b092@172.30.3.118
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER
Supported: replaces
Allow-Events: telephone-event
Contact: <sip:84112420008@172.30.3.119:5060>
Content-Type: application/sdp
Content-Length: 247
v=0
o=CiscoSystemsSIP-GW-UserAgent 1244 1895 IN IP4 172.30.3.119
s=SIP Call
c=IN IP4 172.30.3.119
t=0 0
m=audio 16656 RTP/AVP 8 101
c=IN IP4 172.30.3.119
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
<------------->
--- (14 headers 11 lines) ---
Found RTP audio format 8
Found RTP audio format 101
Peer audio RTP is at port 172.30.3.119:16656
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x8 (alaw)/video=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 172.30.3.119:16656
list_route: hop: <sip:84112420008@172.30.3.119:5060>
set_destination: Parsing <sip:84112420008@172.30.3.119:5060> for address/port to send to
set_destination: set destination to 172.30.3.119, port 5060
Transmitting (NAT) to 172.30.3.119:5060:
ACK sip:84112420008@172.30.3.119:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.3.118:5069;branch=z9hG4bK074966c2;rport
From: "4113999999" <sip:4113999999@172.30.3.118:5069>;tag=as0993a0e0
To: <sip:84112420008@172.30.3.119>;tag=1E3053F0-46D
Contact: <sip:4113999999@172.30.3.118:5069>
Call-ID: 021f0c8957a27161116586045875b092@172.30.3.118
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
-- SIP/cisco2811-081fa2f0 answered SIP/REALIP-1-081e7748
Audio is at REALIP-2 port 24142
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
<--- Reliably Transmitting (no NAT) to REALIP-1:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP REALIP-1:5060;branch=z9hG4bK2022352;received=REALIP-1
From: <sip:4113999999@REALIP-1>;tag=14FA5264-65F
To: <sip:74112420008@REALIP-2>;tag=as0a6ffa18
Call-ID: 687504BE-EA9611DD-9C91FD15-F90032C4@REALIP-1
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:74112420008@REALIP-2:5069>
Content-Type: application/sdp
Content-Length: 238
v=0
o=root 2932 2933 IN IP4 REALIP-2
s=session
c=IN IP4 REALIP-2
t=0 0
m=audio 24142 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
<------------>
-- Packet2Packet bridging SIP/REALIP-1-081e7748 and SIP/cisco2811-081fa2f0
wsh*CLI>
<--- SIP read from REALIP-1:61016 --->
ACK sip:74112420008@REALIP-2:5069 SIP/2.0
Via: SIP/2.0/UDP REALIP-1:5060;branch=z9hG4bK2031494
From: <sip:4113999999@REALIP-1>;tag=14FA5264-65F
To: <sip:74112420008@REALIP-2>;tag=as0a6ffa18
Date: Mon, 26 Jan 2009 04:12:40 GMT
Call-ID: 687504BE-EA9611DD-9C91FD15-F90032C4@REALIP-1
Max-Forwards: 70
CSeq: 101 ACK
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
<--- SIP read from REALIP-1:61016 --->
INVITE sip:74112420008@REALIP-2:5069 SIP/2.0
Via: SIP/2.0/UDP REALIP-1:5060;branch=z9hG4bK20477F
From: <sip:4113999999@REALIP-1>;tag=14FA5264-65F
To: <sip:74112420008@REALIP-2>;tag=as0a6ffa18
Date: Mon, 26 Jan 2009 04:12:52 GMT
Call-ID: 687504BE-EA9611DD-9C91FD15-F90032C4@REALIP-1
Supported: 100rel,timer,replaces
Min-SE: 1800
Cisco-Guid: 1752339486-3935703517-2166554647-1493462536
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER
CSeq: 102 INVITE
Max-Forwards: 70
Remote-Party-ID: <sip:4113999999@REALIP-1>;party=calling;screen=yes;privacy=off
Timestamp: 1232943172
Contact: <sip:4113999999@REALIP-1:5060>
Expires: 300
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Length: 395
v=0
o=CiscoSystemsSIP-GW-UserAgent 499 7955 IN IP4 REALIP-1
s=SIP Call
c=IN IP4 REALIP-1
t=0 0
m=image 19116 udptl t38
c=IN IP4 REALIP-1
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxFillBitRemoval:0
a=T38FaxTranscodingMMR:0
a=T38FaxTranscodingJBIG:0
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:200
a=T38FaxMaxDatagram:72
a=T38FaxUdpEC:t38UDPRedundancy
<------------->
--- (20 headers 16 lines) ---
Sending to REALIP-1 : 5060 (no NAT)
Got T.38 offer in SDP in dialog 687504BE-EA9611DD-9C91FD15-F90032C4@REALIP-1
Got T.38 Re-invite without audio. Keeping RTP active during T.38 session. Callid 687504BE-EA9611DD-9C91FD15-F90032C4@REALIP-1
Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x0 (nothing)/video=0x0 (nothing), combined - 0x0 (nothing)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
<--- Transmitting (no NAT) to REALIP-1:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP REALIP-1:5060;branch=z9hG4bK20477F;received=REALIP-1
From: <sip:4113999999@REALIP-1>;tag=14FA5264-65F
To: <sip:74112420008@REALIP-2>;tag=as0a6ffa18
Call-ID: 687504BE-EA9611DD-9C91FD15-F90032C4@REALIP-1
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:74112420008@REALIP-2:5069>
Content-Length: 0
<------------>
set_destination: Parsing <sip:84112420008@172.30.3.119:5060> for address/port to send to
set_destination: set destination to 172.30.3.119, port 5060
Reliably Transmitting (NAT) to 172.30.3.119:5060:
INVITE sip:84112420008@172.30.3.119:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.3.118:5069;branch=z9hG4bK1c85804d;rport
From: "4113999999" <sip:4113999999@172.30.3.118:5069>;tag=as0993a0e0
To: <sip:84112420008@172.30.3.119>;tag=1E3053F0-46D
Contact: <sip:4113999999@172.30.3.118:5069>
Call-ID: 021f0c8957a27161116586045875b092@172.30.3.118
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
X-asterisk-info: SIP re-invite (T38 switchover)
Content-Type: application/sdp
Content-Length: 345
v=0
o=root 2932 2933 IN IP4 172.30.3.118
s=session
c=IN IP4 172.30.3.118
t=0 0
m=image 4443 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:9600
a=T38FaxFillBitRemoval:0
a=T38FaxTranscodingMMR:0
a=T38FaxTranscodingJBIG:0
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:72
a=T38FaxMaxDatagram:72
a=T38FaxUdpEC:t38UDPRedundancy
---
wsh*CLI>
<--- SIP read from 172.30.3.119:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.30.3.118:5069;branch=z9hG4bK1c85804d;rport
From: "4113999999" <sip:4113999999@172.30.3.118:5069>;tag=as0993a0e0
To: <sip:84112420008@172.30.3.119>;tag=1E3053F0-46D
Date: Mon, 26 Jan 2009 04:14:47 GMT
Call-ID: 021f0c8957a27161116586045875b092@172.30.3.118
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 103 INVITE
Allow-Events: telephone-event
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
<--- SIP read from 172.30.3.119:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.3.118:5069;branch=z9hG4bK1c85804d;rport
From: "4113999999" <sip:4113999999@172.30.3.118:5069>;tag=as0993a0e0
To: <sip:84112420008@172.30.3.119>;tag=1E3053F0-46D
Date: Mon, 26 Jan 2009 04:14:47 GMT
Call-ID: 021f0c8957a27161116586045875b092@172.30.3.118
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 103 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER
Supported: replaces
Allow-Events: telephone-event
Contact: <sip:84112420008@172.30.3.119:5060>
Content-Type: application/sdp
Content-Length: 395
v=0
o=CiscoSystemsSIP-GW-UserAgent 1244 1896 IN IP4 172.30.3.119
s=SIP Call
c=IN IP4 172.30.3.119
t=0 0
m=image 16656 udptl t38
c=IN IP4 172.30.3.119
a=T38FaxVersion:0
a=T38MaxBitRate:9600
a=T38FaxFillBitRemoval:0
a=T38FaxTranscodingMMR:0
a=T38FaxTranscodingJBIG:0
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:200
a=T38FaxMaxDatagram:72
a=T38FaxUdpEC:t38UDPRedundancy
<------------->
--- (14 headers 16 lines) ---
Got T.38 offer in SDP in dialog 021f0c8957a27161116586045875b092@172.30.3.118
Got T.38 Re-invite without audio. Keeping RTP active during T.38 session. Callid 021f0c8957a27161116586045875b092@172.30.3.118
Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x0 (nothing)/video=0x0 (nothing), combined - 0x0 (nothing)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
<--- Reliably Transmitting (no NAT) to REALIP-1:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP REALIP-1:5060;branch=z9hG4bK20477F;received=REALIP-1
From: <sip:4113999999@REALIP-1>;tag=14FA5264-65F
To: <sip:74112420008@REALIP-2>;tag=as0a6ffa18
Call-ID: 687504BE-EA9611DD-9C91FD15-F90032C4@REALIP-1
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:74112420008@REALIP-2:5069>
Content-Type: application/sdp
Content-Length: 345
v=0
o=root 2932 2934 IN IP4 REALIP-2
s=session
c=IN IP4 REALIP-2
t=0 0
m=image 4441 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:9600
a=T38FaxFillBitRemoval:0
a=T38FaxTranscodingMMR:0
a=T38FaxTranscodingJBIG:0
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:72
a=T38FaxMaxDatagram:72
a=T38FaxUdpEC:t38UDPRedundancy
<------------>
set_destination: Parsing <sip:84112420008@172.30.3.119:5060> for address/port to send to
set_destination: set destination to 172.30.3.119, port 5060
Transmitting (NAT) to 172.30.3.119:5060:
ACK sip:84112420008@172.30.3.119:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.3.118:5069;branch=z9hG4bK759c42ed;rport
From: "4113999999" <sip:4113999999@172.30.3.118:5069>;tag=as0993a0e0
To: <sip:84112420008@172.30.3.119>;tag=1E3053F0-46D
Contact: <sip:4113999999@172.30.3.118:5069>
Call-ID: 021f0c8957a27161116586045875b092@172.30.3.118
CSeq: 103 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
wsh*CLI>
<--- SIP read from REALIP-1:61016 --->
ACK sip:74112420008@REALIP-2:5069 SIP/2.0
Via: SIP/2.0/UDP REALIP-1:5060;branch=z9hG4bK2052075
From: <sip:4113999999@REALIP-1>;tag=14FA5264-65F
To: <sip:74112420008@REALIP-2>;tag=as0a6ffa18
Date: Mon, 26 Jan 2009 04:12:52 GMT
Call-ID: 687504BE-EA9611DD-9C91FD15-F90032C4@REALIP-1
Max-Forwards: 70
CSeq: 102 ACK
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
Reliably Transmitting (NAT) to 172.30.3.119:5060:
OPTIONS sip:172.30.3.119 SIP/2.0
Via: SIP/2.0/UDP 172.30.3.118:5069;branch=z9hG4bK492a9b10;rport
From: "asterisk" <sip:asterisk@172.30.3.118:5069>;tag=as55dabcf9
To: <sip:172.30.3.119>
Contact: <sip:asterisk@172.30.3.118:5069>
Call-ID: 3ecc19b33eb5e9940c0d50a845be371c@172.30.3.118
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 26 Jan 2009 04:09:52 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
---
wsh*CLI>
<--- SIP read from 172.30.3.119:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.3.118:5069;branch=z9hG4bK492a9b10;rport
From: "asterisk" <sip:asterisk@172.30.3.118:5069>;tag=as55dabcf9
To: <sip:172.30.3.119>;tag=1E30D3BC-4F5
Date: Mon, 26 Jan 2009 04:15:08 GMT
Call-ID: 3ecc19b33eb5e9940c0d50a845be371c@172.30.3.118
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 OPTIONS
Supported: 100rel,replaces
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER
Accept: application/sdp
Allow-Events: telephone-event
Content-Length: 449
Content-Type: application/sdp
v=0
o=CiscoSystemsSIP-GW-UserAgent 9705 7832 IN IP4 172.30.3.119
s=SIP Call
c=IN IP4 172.30.3.119
t=0 0
m=audio 0 RTP/AVP 18 0 8 4 2 15 3
c=IN IP4 172.30.3.119
m=image 0 udptl t38
c=IN IP4 172.30.3.119
a=T38FaxVersion:0
a=T38MaxBitRate:9600
a=T38FaxFillBitRemoval:0
a=T38FaxTranscodingMMR:0
a=T38FaxTranscodingJBIG:0
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:200
a=T38FaxMaxDatagram:72
a=T38FaxUdpEC:t38UDPRedundancy
<------------->
--- (14 headers 18 lines) ---
Really destroying SIP dialog '3ecc19b33eb5e9940c0d50a845be371c@172.30.3.118' Method: OPTIONS
Reliably Transmitting (no NAT) to REALIP-1:5060:
OPTIONS sip:REALIP-1 SIP/2.0
Via: SIP/2.0/UDP REALIP-2:5069;branch=z9hG4bK25b30b5c;rport
From: "asterisk" <sip:asterisk@REALIP-2:5069>;tag=as417b72a3
To: <sip:REALIP-1>
Contact: <sip:asterisk@REALIP-2:5069>
Call-ID: 1d39bbe558305b1b23324599022a3420@REALIP-2
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 26 Jan 2009 04:09:52 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
---
wsh*CLI>
<--- SIP read from REALIP-1:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP REALIP-2:5069;branch=z9hG4bK25b30b5c;rport
From: "asterisk" <sip:asterisk@REALIP-2:5069>;tag=as417b72a3
To: <sip:REALIP-1>;tag=14FAD298-1AE6
Date: Mon, 26 Jan 2009 04:13:13 GMT
Call-ID: 1d39bbe558305b1b23324599022a3420@REALIP-2
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 OPTIONS
Supported: 100rel,replaces
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER
Accept: application/sdp
Allow-Events: telephone-event
Content-Length: 449
Content-Type: application/sdp
v=0
o=CiscoSystemsSIP-GW-UserAgent 8573 7247 IN IP4 REALIP-1
s=SIP Call
c=IN IP4 REALIP-1
t=0 0
m=audio 0 RTP/AVP 18 0 8 4 2 15 3
c=IN IP4 REALIP-1
m=image 0 udptl t38
c=IN IP4 REALIP-1
a=T38FaxVersion:0
a=T38MaxBitRate:9600
a=T38FaxFillBitRemoval:0
a=T38FaxTranscodingMMR:0
a=T38FaxTranscodingJBIG:0
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:200
a=T38FaxMaxDatagram:72
a=T38FaxUdpEC:t38UDPRedundancy
<------------->
--- (14 headers 18 lines) ---
Really destroying SIP dialog '1d39bbe558305b1b23324599022a3420@REALIP-2' Method: OPTIONS
<--- SIP read from REALIP-1:61016 --->
INVITE sip:74112420008@REALIP-2:5069 SIP/2.0
Via: SIP/2.0/UDP REALIP-1:5060;branch=z9hG4bK2064EF
From: <sip:4113999999@REALIP-1>;tag=14FA5264-65F
To: <sip:74112420008@REALIP-2>;tag=as0a6ffa18
Date: Mon, 26 Jan 2009 04:13:57 GMT
Call-ID: 687504BE-EA9611DD-9C91FD15-F90032C4@REALIP-1
Supported: 100rel,timer,replaces
Min-SE: 1800
Cisco-Guid: 1752339486-3935703517-2166554647-1493462536
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER
CSeq: 103 INVITE
Max-Forwards: 70
Remote-Party-ID: <sip:4113999999@REALIP-1>;party=calling;screen=yes;privacy=off
Timestamp: 1232943237
Contact: <sip:4113999999@REALIP-1:5060>
Expires: 300
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Length: 270
v=0
o=CiscoSystemsSIP-GW-UserAgent 499 7956 IN IP4 REALIP-1
s=SIP Call
c=IN IP4 REALIP-1
t=0 0
m=audio 19116 RTP/AVP 8 101 19
c=IN IP4 REALIP-1
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:19 CN/8000
a=ptime:20
<------------->
--- (20 headers 12 lines) ---
Sending to REALIP-1 : 5060 (no NAT)
Found RTP audio format 8
Found RTP audio format 101
Found RTP audio format 19
Peer audio RTP is at port REALIP-1:19116
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Found audio description format CN for ID 19
Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x3 (telephone-event|CN), combined - 0x1 (telephone-event)
Peer audio RTP is at port REALIP-1:19116
<--- Transmitting (no NAT) to REALIP-1:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP REALIP-1:5060;branch=z9hG4bK2064EF;received=REALIP-1
From: <sip:4113999999@REALIP-1>;tag=14FA5264-65F
To: <sip:74112420008@REALIP-2>;tag=as0a6ffa18
Call-ID: 687504BE-EA9611DD-9C91FD15-F90032C4@REALIP-1
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:74112420008@REALIP-2:5069>
Content-Length: 0
<------------>
[Jan 26 13:10:36] WARNING[4753]: chan_sip.c:14583 handle_request_invite: RTP re-invite after T38 session not handled yet !
<--- Reliably Transmitting (no NAT) to REALIP-1:5060 --->
SIP/2.0 488 Not Acceptable Here (unsupported)
Via: SIP/2.0/UDP REALIP-1:5060;branch=z9hG4bK2064EF;received=REALIP-1
From: <sip:4113999999@REALIP-1>;tag=14FA5264-65F
To: <sip:74112420008@REALIP-2>;tag=as0a6ffa18
Call-ID: 687504BE-EA9611DD-9C91FD15-F90032C4@REALIP-1
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
<------------>
wsh*CLI>
<--- SIP read from REALIP-1:5060 --->
ACK sip:74112420008@REALIP-2:5069 SIP/2.0
Via: SIP/2.0/UDP REALIP-1:5060;branch=z9hG4bK2064EF
From: <sip:4113999999@REALIP-1>;tag=14FA5264-65F
To: <sip:74112420008@REALIP-2>;tag=as0a6ffa18
Date: Mon, 26 Jan 2009 04:13:57 GMT
Call-ID: 687504BE-EA9611DD-9C91FD15-F90032C4@REALIP-1
Max-Forwards: 70
CSeq: 103 ACK
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
Reliably Transmitting (NAT) to 172.30.3.119:5060:
OPTIONS sip:172.30.3.119 SIP/2.0
Via: SIP/2.0/UDP 172.30.3.118:5069;branch=z9hG4bK14e9b687;rport
From: "asterisk" <sip:asterisk@172.30.3.118:5069>;tag=as6ae2f162
To: <sip:172.30.3.119>
Contact: <sip:asterisk@172.30.3.118:5069>
Call-ID: 612c32711ce662201f9506f821b41381@172.30.3.118
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 26 Jan 2009 04:10:52 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
---
wsh*CLI>
<--- SIP read from 172.30.3.119:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.3.118:5069;branch=z9hG4bK14e9b687;rport
From: "asterisk" <sip:asterisk@172.30.3.118:5069>;tag=as6ae2f162
To: <sip:172.30.3.119>;tag=1E31BE2C-1D12
Date: Mon, 26 Jan 2009 04:16:08 GMT
Call-ID: 612c32711ce662201f9506f821b41381@172.30.3.118
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 OPTIONS
Supported: 100rel,replaces
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER
Accept: application/sdp
Allow-Events: telephone-event
Content-Length: 448
Content-Type: application/sdp
v=0
o=CiscoSystemsSIP-GW-UserAgent 6908 653 IN IP4 172.30.3.119
s=SIP Call
c=IN IP4 172.30.3.119
t=0 0
m=audio 0 RTP/AVP 18 0 8 4 2 15 3
c=IN IP4 172.30.3.119
m=image 0 udptl t38
c=IN IP4 172.30.3.119
a=T38FaxVersion:0
a=T38MaxBitRate:9600
a=T38FaxFillBitRemoval:0
a=T38FaxTranscodingMMR:0
a=T38FaxTranscodingJBIG:0
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:200
a=T38FaxMaxDatagram:72
a=T38FaxUdpEC:t38UDPRedundancy
<------------->
--- (14 headers 18 lines) ---
Really destroying SIP dialog '612c32711ce662201f9506f821b41381@172.30.3.118' Method: OPTIONS
Reliably Transmitting (no NAT) to REALIP-1:5060:
OPTIONS sip:REALIP-1 SIP/2.0
Via: SIP/2.0/UDP REALIP-2:5069;branch=z9hG4bK0c4f9366;rport
From: "asterisk" <sip:asterisk@REALIP-2:5069>;tag=as6331bb78
To: <sip:REALIP-1>
Contact: <sip:asterisk@REALIP-2:5069>
Call-ID: 79a093954672c4e9221cee6f626ab2e9@REALIP-2
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 26 Jan 2009 04:10:52 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
---
wsh*CLI>
<--- SIP read from REALIP-1:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP REALIP-2:5069;branch=z9hG4bK0c4f9366;rport
From: "asterisk" <sip:asterisk@REALIP-2:5069>;tag=as6331bb78
To: <sip:REALIP-1>;tag=14FBBD00-1919
Date: Mon, 26 Jan 2009 04:14:13 GMT
Call-ID: 79a093954672c4e9221cee6f626ab2e9@REALIP-2
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 102 OPTIONS
Supported: 100rel,replaces
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER
Accept: application/sdp
Allow-Events: telephone-event
Content-Length: 449
Content-Type: application/sdp
v=0
o=CiscoSystemsSIP-GW-UserAgent 2586 6672 IN IP4 REALIP-1
s=SIP Call
c=IN IP4 REALIP-1
t=0 0
m=audio 0 RTP/AVP 18 0 8 4 2 15 3
c=IN IP4 REALIP-1
m=image 0 udptl t38
c=IN IP4 REALIP-1
a=T38FaxVersion:0
a=T38MaxBitRate:9600
a=T38FaxFillBitRemoval:0
a=T38FaxTranscodingMMR:0
a=T38FaxTranscodingJBIG:0
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:200
a=T38FaxMaxDatagram:72
a=T38FaxUdpEC:t38UDPRedundancy
<------------->
--- (14 headers 18 lines) ---
Really destroying SIP dialog '79a093954672c4e9221cee6f626ab2e9@REALIP-2' Method: OPTIONS
<--- SIP read from REALIP-1:61016 --->
BYE sip:74112420008@REALIP-2:5069 SIP/2.0
Via: SIP/2.0/UDP REALIP-1:5060;branch=z9hG4bK207771
From: <sip:4113999999@REALIP-1>;tag=14FA5264-65F
To: <sip:74112420008@REALIP-2>;tag=as0a6ffa18
Date: Mon, 26 Jan 2009 04:13:57 GMT
Call-ID: 687504BE-EA9611DD-9C91FD15-F90032C4@REALIP-1
User-Agent: Cisco-SIPGateway/IOS-12.x
Max-Forwards: 70
Timestamp: 1232943264
CSeq: 104 BYE
Reason: Q.850;cause=16
Content-Length: 0
<------------->
--- (12 headers 0 lines) ---
Sending to REALIP-1 : 5060 (no NAT)
<--- Transmitting (no NAT) to REALIP-1:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP REALIP-1:5060;branch=z9hG4bK207771;received=REALIP-1
From: <sip:4113999999@REALIP-1>;tag=14FA5264-65F
To: <sip:74112420008@REALIP-2>;tag=as0a6ffa18
Call-ID: 687504BE-EA9611DD-9C91FD15-F90032C4@REALIP-1
CSeq: 104 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:74112420008@REALIP-2:5069>
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '021f0c8957a27161116586045875b092@172.30.3.118' in 6400 ms (Method: INVITE)
set_destination: Parsing <sip:84112420008@172.30.3.119:5060> for address/port to send to
set_destination: set destination to 172.30.3.119, port 5060
Reliably Transmitting (NAT) to 172.30.3.119:5060:
BYE sip:84112420008@172.30.3.119:5060 SIP/2.0
Via: SIP/2.0/UDP 172.30.3.118:5069;branch=z9hG4bK09af4d22;rport
From: "4113999999" <sip:4113999999@172.30.3.118:5069>;tag=as0993a0e0
To: <sip:84112420008@172.30.3.119>;tag=1E3053F0-46D
Call-ID: 021f0c8957a27161116586045875b092@172.30.3.118
CSeq: 104 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
== Spawn extension (macro-incoming-from-test1, s, 8) exited non-zero on 'SIP/REALIP-1-081e7748' in macro 'incoming-from-test1'
== Spawn extension (macro-incoming-from-test1, s, 8) exited non-zero on 'SIP/REALIP-1-081e7748'
<--- SIP read from 172.30.3.119:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.30.3.118:5069;branch=z9hG4bK09af4d22;rport
From: "4113999999" <sip:4113999999@172.30.3.118:5069>;tag=as0993a0e0
To: <sip:84112420008@172.30.3.119>;tag=1E3053F0-46D
Date: Mon, 26 Jan 2009 04:16:19 GMT
Call-ID: 021f0c8957a27161116586045875b092@172.30.3.118
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0
CSeq: 104 BYE
<------------->
--- (9 headers 0 lines) ---
Really destroying SIP dialog '021f0c8957a27161116586045875b092@172.30.3.118' Method: INVITE
Really destroying SIP dialog '687504BE-EA9611DD-9C91FD15-F90032C4@REALIP-1' Method: BYE
wsh*CLI>
wsh:/asterisk#
|
