All pastes #242964 Raw Edit

Mine

public text v1 · immutable
·published 2006-11-07 22:15 UTC
rendered paste body

localhost*CLI> sip debug
SIP Debugging enabled
localhost*CLI>
<-- SIP read from 192.168.1.153:5060:
INVITE sip:3391788@192.168.1.125 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.153;branch=z9hG4bKbb4c8d3f9
Content-Length: 275
To: sip:3391788@192.168.1.125
From: 1000x029 <sip:1000x029@192.168.1.125>;tag=55e9615a9547584
Call-ID: d0f06e761e3e6c007fa41361d968826f@192.168.1.125
CSeq: 1346830124 INVITE
Supported: timer
Min-SE: 1800
Session-Expires: 3600
Allow: INVITE, ACK, BYE, CANCEL, REFER, NOTIFY
Content-Type: application/sdp
Contact: 1000x029 <sip:1000x029@192.168.1.153>
Supported: replaces
User-Agent: MxSipApp/5.0.15.86 MxSF/v3.2.1.1

v=0
o=MxSIP 2139887654189754277 1968665488102052864 IN IP4 192.168.1.153
s=-
c=IN IP4 192.168.1.153
t=0 0
a=sendrecv
m=audio 5004 RTP/AVP 18 0 101 13
a=rtpmap:18 G729/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:18 annexb=no
a=fmtp:101 0-15

--- (15 headers 12 lines) ---
Using INVITE request as basis request - d0f06e761e3e6c007fa41361d968826f@192.168.1.125
Sending to 192.168.1.153 : 5060 (non-NAT)
Found user '1000x029'
Found RTP audio format 18
Found RTP audio format 0
Found RTP audio format 101
Found RTP audio format 13
Peer audio RTP is at port 192.168.1.153:5004
Found description format G729
Found description format PCMU
Found description format telephone-event
Capabilities: us - 0x4 (ulaw), peer - audio=0x104 (ulaw|g729)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x3 (telephone-event|CN), combined - 0x1 (telephone-event)
Looking for 3391788 in faxing (domain 192.168.1.125)
list_route: hop: <sip:1000x029@192.168.1.153>
Transmitting (NAT) to 192.168.1.153:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.153;branch=z9hG4bKbb4c8d3f9;received=192.168.1.153
From: 1000x029 <sip:1000x029@192.168.1.125>;tag=55e9615a9547584
To: sip:3391788@192.168.1.125
Call-ID: d0f06e761e3e6c007fa41361d968826f@192.168.1.125
CSeq: 1346830124 INVITE
User-Agent: OpenPBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:3391788@192.168.1.125>
Content-Length: 0


---
    -- Executing Dial("SIP/1000x029-b63c", "SIP/3391788@17.216.11.25") in new stack
We're at 20.11.35.28 port 10724
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
13 headers, 10 lines
Reliably Transmitting (NAT) to 17.216.11.25:5060:
INVITE sip:3391788@17.216.11.25 SIP/2.0
Via: SIP/2.0/UDP 20.11.35.28:5060;branch=z9hG4bK56097349;rport
From: "1000x029" <sip:1000x029@20.11.35.28>;tag=as44cc2f57
To: <sip:3391788@17.216.11.25>
Contact: <sip:1000x029@20.11.35.28>
Call-ID: 149f525b487e5a374ed7e84518f5a4da@20.11.35.28
CSeq: 102 INVITE
User-Agent: OpenPBX
Max-Forwards: 70
Date: Tue, 07 Nov 2006 22:06:55 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 218

v=0
o=root 31966 31966 IN IP4 20.11.35.28
s=session
c=IN IP4 20.11.35.28
t=0 0
m=audio 10724 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
    -- Called 3391788@17.216.11.25
localhost*CLI>
<-- SIP read from 17.216.11.25:5060:
SIP/2.0 100 Trying
CSeq: 102 INVITE
Call-ID: 149f525b487e5a374ed7e84518f5a4da@20.11.35.28
From: "1000x029"<sip:1000x029@20.11.35.28>;tag=as44cc2f57
To: <sip:3391788@17.216.11.25>;tag=47d872fc-3fd2
Via: SIP/2.0/UDP 20.11.35.28:5060;branch=z9hG4bK56097349;rport
User-Agent: Quintum/1.0.0
Quintum: 0b053233313431


--- (8 headers 0 lines) ---
localhost*CLI>
<-- SIP read from 17.216.11.25:5060:
SIP/2.0 183 Session Progress
CSeq: 102 INVITE
Call-ID: 149f525b487e5a374ed7e84518f5a4da@20.11.35.28
Content-Type: application/sdp
From: "1000x029"<sip:1000x029@20.11.35.28>;tag=as44cc2f57
To: <sip:3391788@17.216.11.25>;tag=47d872fc-3fd2
Via: SIP/2.0/UDP 20.11.35.28:5060;branch=z9hG4bK56097349;rport
Content-Length: 209
User-Agent: Quintum/1.0.0
Quintum: 07120100010000bf0f06001e038082881e028288

v=0
o=Quintum 12902 31966 IN IP4 17.216.11.25
s=VoipCall
c=IN IP4 17.216.11.25
t=0 0
m=audio 11024 RTP/AVP 0 101
c=IN IP4 17.216.11.25
a=rtpmap:0 pcmu/8000/1
a=rtpmap:101 telephone-event/8000/1

--- (10 headers 9 lines) ---
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port 17.216.11.25:11024
Found description format pcmu
Found description format telephone-event
Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
    -- SIP/17.216.11.25-6a54 is making progress passing it to SIP/1000x029-b63c
We're at 192.168.1.125 port 11836
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Transmitting (NAT) to 192.168.1.153:5060:
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.1.153;branch=z9hG4bKbb4c8d3f9;received=192.168.1.153
From: 1000x029 <sip:1000x029@192.168.1.125>;tag=55e9615a9547584
To: sip:3391788@192.168.1.125;tag=as21c8fc76
Call-ID: d0f06e761e3e6c007fa41361d968826f@192.168.1.125
CSeq: 1346830124 INVITE
User-Agent: OpenPBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:3391788@192.168.1.125>
Content-Type: application/sdp
Content-Length: 218

v=0
o=root 31966 31966 IN IP4 192.168.1.125
s=session
c=IN IP4 192.168.1.125
t=0 0
m=audio 11836 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
localhost*CLI>
<-- SIP read from 17.216.11.25:5060:
SIP/2.0 200 OK
CSeq: 102 INVITE
Call-ID: 149f525b487e5a374ed7e84518f5a4da@20.11.35.28
Contact: <sip:3391788@17.216.11.25:5060>
Content-Type: application/sdp
From: "1000x029"<sip:1000x029@20.11.35.28>;tag=as44cc2f57
To: <sip:3391788@17.216.11.25>;tag=47d872fc-3fd2
Via: SIP/2.0/UDP 20.11.35.28:5060;branch=z9hG4bK56097349;rport
Content-Length: 209
User-Agent: Quintum/1.0.0

v=0
o=Quintum 12903 31966 IN IP4 17.216.11.25
s=VoipCall
c=IN IP4 17.216.11.25
t=0 0
m=audio 11024 RTP/AVP 0 101
c=IN IP4 17.216.11.25
a=rtpmap:0 pcmu/8000/1
a=rtpmap:101 telephone-event/8000/1

--- (10 headers 9 lines) ---
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port 17.216.11.25:11024
Found description format pcmu
Found description format telephone-event
Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
list_route: hop: <sip:3391788@17.216.11.25:5060>
set_destination: Parsing <sip:3391788@17.216.11.25:5060> for address/port to send to
set_destination: set destination to 17.216.11.25, port 5060
Transmitting (NAT) to 17.216.11.25:5060:
ACK sip:3391788@17.216.11.25:5060 SIP/2.0
Via: SIP/2.0/UDP 20.11.35.28:5060;branch=z9hG4bK74b46948;rport
From: "1000x029" <sip:1000x029@20.11.35.28>;tag=as44cc2f57
To: <sip:3391788@17.216.11.25>;tag=47d872fc-3fd2
Contact: <sip:1000x029@20.11.35.28>
Call-ID: 149f525b487e5a374ed7e84518f5a4da@20.11.35.28
CSeq: 102 ACK
User-Agent: OpenPBX
Max-Forwards: 70
Content-Length: 0


---
    -- SIP/17.216.11.25-6a54 answered SIP/1000x029-b63c
We're at 192.168.1.125 port 11836
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 192.168.1.153:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.153;branch=z9hG4bKbb4c8d3f9;received=192.168.1.153
From: 1000x029 <sip:1000x029@192.168.1.125>;tag=55e9615a9547584
To: sip:3391788@192.168.1.125;tag=as21c8fc76
Call-ID: d0f06e761e3e6c007fa41361d968826f@192.168.1.125
CSeq: 1346830124 INVITE
User-Agent: OpenPBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:3391788@192.168.1.125>
Content-Type: application/sdp
Content-Length: 218

v=0
o=root 31966 31967 IN IP4 192.168.1.125
s=session
c=IN IP4 192.168.1.125
t=0 0
m=audio 11836 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
    -- Attempting native bridge of SIP/1000x029-b63c and SIP/17.216.11.25-6a54
localhost*CLI>
<-- SIP read from 192.168.1.153:5060:
ACK sip:3391788@192.168.1.125 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.153;branch=z9hG4bK179cc1913
Content-Length: 0
To: sip:3391788@192.168.1.125;tag=as21c8fc76
From: 1000x029 <sip:1000x029@192.168.1.125>;tag=55e9615a9547584
Call-ID: d0f06e761e3e6c007fa41361d968826f@192.168.1.125
CSeq: 1346830124 ACK
Contact: 1000x029 <sip:1000x029@192.168.1.153>
User-Agent: MxSipApp/5.0.15.86 MxSF/v3.2.1.1


--- (9 headers 0 lines) ---
localhost*CLI>
<-- SIP read from 192.168.1.153:5060:
INVITE sip:3391788@192.168.1.125 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.153;branch=z9hG4bK27aa86940
Content-Length: 327
To: sip:3391788@192.168.1.125;tag=as21c8fc76
From: 1000x029 <sip:1000x029@192.168.1.125>;tag=55e9615a9547584
Call-ID: d0f06e761e3e6c007fa41361d968826f@192.168.1.125
CSeq: 1346830125 INVITE
Supported: timer
Min-SE: 1800
Session-Expires: 3600
Allow: INVITE, ACK, BYE, CANCEL, REFER, NOTIFY
Content-Type: application/sdp
Contact: 1000x029 <sip:1000x029@192.168.1.153>
Supported: replaces
User-Agent: MxSipApp/5.0.15.86 MxSF/v3.2.1.1

v=0
o=MxSIP 2139887654189754277 1968665488102052865 IN IP4 192.168.1.153
s=-
c=IN IP4 192.168.1.153
t=0 0
m=image 6004 udptl t38
a=T38MaxBitRate:14400
a=T38FaxRateManagement:transferredTCF
a=T38FaxFillBitRemoval:0
a=T38FaxTranscodingMMR:0
a=T38FaxTranscodingJBIG:0
a=T38FaxVersion:0
a=T38FaxUdpEC:t38UDPRedundancy

--- (15 headers 13 lines) ---
Using INVITE request as basis request - d0f06e761e3e6c007fa41361d968826f@192.168.1.125
Sending to 192.168.1.153 : 5060 (non-NAT)
Got T.38 offer in SDP
Peer T.38 UDPTL is at port 192.168.1.153:6004
Capabilities: us - 0x4 (ulaw), peer - audio=0x104 (ulaw|g729)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x3 (telephone-event|CN), combined - 0x1 (telephone-event)
set_destination: Parsing <sip:3391788@17.216.11.25:5060> for address/port to send to
set_destination: set destination to 17.216.11.25, port 5060
T.38 UDPTL is at port 20.11.35.28:4034
13 headers, 15 lines
Reliably Transmitting (NAT) to 17.216.11.25:5060:
INVITE sip:3391788@17.216.11.25:5060 SIP/2.0
Via: SIP/2.0/UDP 20.11.35.28:5060;branch=z9hG4bK15dc39ba;rport
From: "1000x029" <sip:1000x029@20.11.35.28>;tag=as44cc2f57
To: <sip:3391788@17.216.11.25>;tag=47d872fc-3fd2
Contact: <sip:1000x029@20.11.35.28>
Call-ID: 149f525b487e5a374ed7e84518f5a4da@20.11.35.28
CSeq: 103 INVITE
User-Agent: OpenPBX
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
X-openpbx-info: SIP re-invite (T38 switchover)
Content-Type: application/sdp
Content-Length: 351

v=0
o=root 31966 31967 IN IP4 20.11.35.28
s=session
c=IN IP4 20.11.35.28
t=0 0
m=image 4034 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:9600
a=T38FaxFillBitRemoval:0
a=T38FaxTranscodingMMR:0
a=T38FaxTranscodingJBIG:0
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:400
a=T38FaxMaxDatagram:400
a=T38FaxUdpEC:t38UDPRedundancy

---
localhost*CLI>
<-- SIP read from 17.216.11.25:5060:
SIP/2.0 200 OK
CSeq: 103 INVITE
Call-ID: 149f525b487e5a374ed7e84518f5a4da@20.11.35.28
Contact: <sip:3391788@17.216.11.25:5060>
Content-Type: application/sdp
From: "1000x029"<sip:1000x029@20.11.35.28>;tag=as44cc2f57
To: <sip:3391788@17.216.11.25>;tag=47d872fc-3fd2
Via: SIP/2.0/UDP 20.11.35.28:5060;branch=z9hG4bK15dc39ba;rport
Content-Length: 264
User-Agent: Quintum/1.0.0

v=0
o=Quintum 12904 31967 IN IP4 17.216.11.25
s=VoipCall
c=IN IP4 17.216.11.25
t=0 0
m=image 11024 udptl t38
c=IN IP4 17.216.11.25
a=T38FaxVersion:0
a=T38MaxBitRate:9600
a=T38FaxFillBitRemoval:0
a=T38FaxTranscodingMMR:0
a=T38FaxTranscodingJBIG:0

--- (10 headers 12 lines) ---
Got T.38 offer in SDP
Peer T.38 UDPTL is at port 17.216.11.25:11024
Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
T.38 UDPTL is at port 192.168.1.125:4454
Reliably Transmitting (NAT) to 192.168.1.153:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.153;branch=z9hG4bK27aa86940;received=192.168.1.153
From: 1000x029 <sip:1000x029@192.168.1.125>;tag=55e9615a9547584
To: sip:3391788@192.168.1.125;tag=as21c8fc76
Call-ID: d0f06e761e3e6c007fa41361d968826f@192.168.1.125
CSeq: 1346830125 INVITE
User-Agent: OpenPBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:3391788@192.168.1.125>
Content-Type: application/sdp
Content-Length: 344

v=0
o=root 31966 31968 IN IP4 192.168.1.125
s=session
c=IN IP4 192.168.1.125
t=0 0
m=image 4454 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:9600
a=T38FaxFillBitRemoval:0
a=T38FaxTranscodingMMR:0
a=T38FaxTranscodingJBIG:0
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:400
a=T38FaxMaxDatagram:400
a=T38FaxUdpEC:t38UDPFEC

---
set_destination: Parsing <sip:3391788@17.216.11.25:5060> for address/port to send to
set_destination: set destination to 17.216.11.25, port 5060
Transmitting (NAT) to 17.216.11.25:5060:
ACK sip:3391788@17.216.11.25:5060 SIP/2.0
Via: SIP/2.0/UDP 20.11.35.28:5060;branch=z9hG4bK3e389047;rport
From: "1000x029" <sip:1000x029@20.11.35.28>;tag=as44cc2f57
To: <sip:3391788@17.216.11.25>;tag=47d872fc-3fd2
Contact: <sip:1000x029@20.11.35.28>
Call-ID: 149f525b487e5a374ed7e84518f5a4da@20.11.35.28
CSeq: 103 ACK
User-Agent: OpenPBX
Max-Forwards: 70
Content-Length: 0


---
localhost*CLI>
<-- SIP read from 192.168.1.153:5060:
ACK sip:3391788@192.168.1.125 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.153;branch=z9hG4bK44b9b03d3
Content-Length: 0
To: sip:3391788@192.168.1.125;tag=as21c8fc76
From: 1000x029 <sip:1000x029@192.168.1.125>;tag=55e9615a9547584
Call-ID: d0f06e761e3e6c007fa41361d968826f@192.168.1.125
CSeq: 1346830125 ACK
Contact: 1000x029 <sip:1000x029@192.168.1.153>
User-Agent: MxSipApp/5.0.15.86 MxSF/v3.2.1.1


--- (9 headers 0 lines) ---
localhost*CLI>
<-- SIP read from 192.168.1.153:5060:
BYE sip:3391788@192.168.1.125 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.153;branch=z9hG4bK338e2a7d9
Content-Length: 0
To: sip:3391788@192.168.1.125;tag=as21c8fc76
From: 1000x029 <sip:1000x029@192.168.1.125>;tag=55e9615a9547584
Call-ID: d0f06e761e3e6c007fa41361d968826f@192.168.1.125
CSeq: 1346830126 BYE
Supported: timer
Supported: replaces
User-Agent: MxSipApp/5.0.15.86 MxSF/v3.2.1.1


--- (10 headers 0 lines) ---
Sending to 192.168.1.153 : 5060 (non-NAT)
Transmitting (NAT) to 192.168.1.153:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.153;branch=z9hG4bK338e2a7d9;received=192.168.1.153
From: 1000x029 <sip:1000x029@192.168.1.125>;tag=55e9615a9547584
To: sip:3391788@192.168.1.125;tag=as21c8fc76
Call-ID: d0f06e761e3e6c007fa41361d968826f@192.168.1.125
CSeq: 1346830126 BYE
User-Agent: OpenPBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:3391788@192.168.1.125>
Content-Length: 0
X-OpenPBX-HangupCause: Normal Clearing


---
set_destination: Parsing <sip:3391788@17.216.11.25:5060> for address/port to send to
set_destination: set destination to 17.216.11.25, port 5060
Reliably Transmitting (NAT) to 17.216.11.25:5060:
BYE sip:3391788@17.216.11.25:5060 SIP/2.0
Via: SIP/2.0/UDP 20.11.35.28:5060;branch=z9hG4bK0fe5d431;rport
From: "1000x029" <sip:1000x029@20.11.35.28>;tag=as44cc2f57
To: <sip:3391788@17.216.11.25>;tag=47d872fc-3fd2
Contact: <sip:1000x029@20.11.35.28>
Call-ID: 149f525b487e5a374ed7e84518f5a4da@20.11.35.28
CSeq: 104 BYE
User-Agent: OpenPBX
Max-Forwards: 70
Content-Length: 0


---
  == Spawn extension (faxing, 3391788, 1) exited non-zero on 'SIP/1000x029-b63c'
localhost*CLI>
<-- SIP read from 17.216.11.25:5060:
SIP/2.0 200 OK
CSeq: 104 BYE
Call-ID: 149f525b487e5a374ed7e84518f5a4da@20.11.35.28
From: "1000x029"<sip:1000x029@20.11.35.28>;tag=as44cc2f57
To: <sip:3391788@17.216.11.25>;tag=47d872fc-3fd2
Via: SIP/2.0/UDP 20.11.35.28:5060;branch=z9hG4bK0fe5d431;rport


--- (6 headers 0 lines) ---
Destroying call '149f525b487e5a374ed7e84518f5a4da@20.11.35.28'
Destroying call 'd0f06e761e3e6c007fa41361d968826f@192.168.1.125'
localhost*CLI> sip no debug
SIP Debugging Disabled
localhost*CLI>