All pastes #507517 Raw Edit

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public text v1 · immutable
·published 2007-05-24 16:42 UTC
rendered paste body
DIAL PLAN:
[voxbone-in]
;
exten => 16477245778,1,Answer()
exten => 16477245778,2,Set(TIMEOUT(digit)=59)
exten => 16477245778,3,Set(FAXFILE=/tmp/incomingfax.tif)
exten => 16477245778,4,RxFAX(${FAXFILE})
;
exten => t,1,Hangup()
;
;
################################################# LOG ###########################################
atpserver104*CLI> 
<-- SIP read from 81.201.84.26:5060: 
INVITE sip:16477245778@66.165.164.45 SIP/2.0
Via: SIP/2.0/UDP 81.201.84.26:5060;branch=z9hG4bK06d22c1d;rport
From: "32" <sip:32@81.201.84.26>;tag=as2792e8a5
To: <sip:16477245778@66.165.164.45>
Contact: <sip:32@81.201.84.26>
Call-ID: 3921247c019a4e4d3abcb58b36fb7940@81.201.84.26
CSeq: 102 INVITE
User-Agent: Vox Callcontrol
Max-Forwards: 70
Date: Thu, 24 May 2007 15:02:16 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 311

v=0
o=root 25204 25204 IN IP4 81.201.84.26
s=session
c=IN IP4 81.201.84.26
t=0 0
m=audio 18370 RTP/AVP 8 0 18 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

--- (14 headers 15 lines) ---
Using INVITE request as basis request - 3921247c019a4e4d3abcb58b36fb7940@81.201.84.26
Sending to 81.201.84.26 : 5060 (NAT)
Found peer '81.201.84.26'
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 18
Found RTP audio format 101
Peer audio RTP is at port 81.201.84.26:18370
Found description format PCMA
Found description format PCMU
Found description format G729
Found description format telephone-event
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Looking for 16477245778 in voxbone-in (domain 66.165.164.45)
list_route: hop: <sip:32@81.201.84.26>
Transmitting (NAT) to 81.201.84.26:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 81.201.84.26:5060;branch=z9hG4bK06d22c1d;rport;received=81.201.84.26
From: "32" <sip:32@81.201.84.26>;tag=as2792e8a5
To: <sip:16477245778@66.165.164.45>
Call-ID: 3921247c019a4e4d3abcb58b36fb7940@81.201.84.26
CSeq: 102 INVITE
User-Agent: CallWeaver
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:16477245778@66.165.164.45>
Content-Length: 0


---
We're at 66.165.164.45 port 11276
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 81.201.84.26:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 81.201.84.26:5060;branch=z9hG4bK06d22c1d;rport;received=81.201.84.26
From: "32" <sip:32@81.201.84.26>;tag=as2792e8a5
To: <sip:16477245778@66.165.164.45>;tag=as743ecf13
Call-ID: 3921247c019a4e4d3abcb58b36fb7940@81.201.84.26
CSeq: 102 INVITE
User-Agent: CallWeaver
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:16477245778@66.165.164.45>
Content-Type: application/sdp
Content-Length: 269

v=0
o=root 25483 25483 IN IP4 66.165.164.45
s=session
c=IN IP4 66.165.164.45
t=0 0
m=audio 11276 RTP/AVP 0 8 122 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:122 t38/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
atpserver104*CLI> 
<-- SIP read from 81.201.84.26:5060: 
ACK sip:16477245778@66.165.164.45 SIP/2.0
Via: SIP/2.0/UDP 81.201.84.26:5060;branch=z9hG4bK67b86ff6;rport
From: "32" <sip:32@81.201.84.26>;tag=as2792e8a5
To: <sip:16477245778@66.165.164.45>;tag=as743ecf13
Contact: <sip:32@81.201.84.26>
Call-ID: 3921247c019a4e4d3abcb58b36fb7940@81.201.84.26
CSeq: 102 ACK
User-Agent: Vox Callcontrol
Max-Forwards: 70
Content-Length: 0


--- (10 headers 0 lines) ---
atpserver104*CLI> 
<-- SIP read from 81.201.84.26:5060: 
BYE sip:16477245778@66.165.164.45 SIP/2.0
Via: SIP/2.0/UDP 81.201.84.26:5060;branch=z9hG4bK1a8dc6e2;rport
From: "32" <sip:32@81.201.84.26>;tag=as2792e8a5
To: <sip:16477245778@66.165.164.45>;tag=as743ecf13
Call-ID: 3921247c019a4e4d3abcb58b36fb7940@81.201.84.26
CSeq: 103 BYE
User-Agent: Vox Callcontrol
Max-Forwards: 70
Content-Length: 0


--- (9 headers 0 lines) ---
Sending to 81.201.84.26 : 5060 (NAT)
Transmitting (NAT) to 81.201.84.26:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 81.201.84.26:5060;branch=z9hG4bK1a8dc6e2;received=81.201.84.26;rport=5060
From: "32" <sip:32@81.201.84.26>;tag=as2792e8a5
To: <sip:16477245778@66.165.164.45>;tag=as743ecf13
Call-ID: 3921247c019a4e4d3abcb58b36fb7940@81.201.84.26
CSeq: 103 BYE
User-Agent: CallWeaver
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:16477245778@66.165.164.45>
Content-Length: 0


---
Destroying call '3921247c019a4e4d3abcb58b36fb7940@81.201.84.26'
atpserver104*CLI> 
<-- SIP read from 81.201.84.27:5060: 
INVITE sip:16477245778@66.165.164.45 SIP/2.0
Via: SIP/2.0/UDP 81.201.84.27:5060;branch=z9hG4bK008e843c;rport
From: "13054337646" <sip:13054337646@81.201.84.27>;tag=as2a216b77
To: <sip:16477245778@66.165.164.45>
Contact: <sip:13054337646@81.201.84.27>
Call-ID: 49d2d0610650fb167fd0a4852e052c17@81.201.84.27
CSeq: 102 INVITE
User-Agent: Vox Callcontrol
Max-Forwards: 70
Date: Thu, 24 May 2007 15:04:23 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 309

v=0
o=root 8351 8351 IN IP4 81.201.84.27
s=session
c=IN IP4 81.201.84.27
t=0 0
m=audio 15582 RTP/AVP 8 0 18 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

--- (14 headers 15 lines) ---
Using INVITE request as basis request - 49d2d0610650fb167fd0a4852e052c17@81.201.84.27
Sending to 81.201.84.27 : 5060 (NAT)
Found peer '81.201.84.27'
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 18
Found RTP audio format 101
Peer audio RTP is at port 81.201.84.27:15582
Found description format PCMA
Found description format PCMU
Found description format G729
Found description format telephone-event
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Looking for 16477245778 in voxbone-in (domain 66.165.164.45)
list_route: hop: <sip:13054337646@81.201.84.27>
Transmitting (NAT) to 81.201.84.27:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 81.201.84.27:5060;branch=z9hG4bK008e843c;rport;received=81.201.84.27
From: "13054337646" <sip:13054337646@81.201.84.27>;tag=as2a216b77
To: <sip:16477245778@66.165.164.45>
Call-ID: 49d2d0610650fb167fd0a4852e052c17@81.201.84.27
CSeq: 102 INVITE
User-Agent: CallWeaver
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:16477245778@66.165.164.45>
Content-Length: 0


---
We're at 66.165.164.45 port 18696
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 81.201.84.27:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 81.201.84.27:5060;branch=z9hG4bK008e843c;rport;received=81.201.84.27
From: "13054337646" <sip:13054337646@81.201.84.27>;tag=as2a216b77
To: <sip:16477245778@66.165.164.45>;tag=as48190407
Call-ID: 49d2d0610650fb167fd0a4852e052c17@81.201.84.27
CSeq: 102 INVITE
User-Agent: CallWeaver
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:16477245778@66.165.164.45>
Content-Type: application/sdp
Content-Length: 269

v=0
o=root 25483 25483 IN IP4 66.165.164.45
s=session
c=IN IP4 66.165.164.45
t=0 0
m=audio 18696 RTP/AVP 0 8 122 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:122 t38/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
atpserver104*CLI> 
<-- SIP read from 81.201.84.27:5060: 
ACK sip:16477245778@66.165.164.45 SIP/2.0
Via: SIP/2.0/UDP 81.201.84.27:5060;branch=z9hG4bK31509113;rport
From: "13054337646" <sip:13054337646@81.201.84.27>;tag=as2a216b77
To: <sip:16477245778@66.165.164.45>;tag=as48190407
Contact: <sip:13054337646@81.201.84.27>
Call-ID: 49d2d0610650fb167fd0a4852e052c17@81.201.84.27
CSeq: 102 ACK
User-Agent: Vox Callcontrol
Max-Forwards: 70
Content-Length: 0


--- (10 headers 0 lines) ---
atpserver104*CLI> 
<-- SIP read from 81.201.84.27:5060: 
BYE sip:16477245778@66.165.164.45 SIP/2.0
Via: SIP/2.0/UDP 81.201.84.27:5060;branch=z9hG4bK0a6169a2;rport
From: "13054337646" <sip:13054337646@81.201.84.27>;tag=as2a216b77
To: <sip:16477245778@66.165.164.45>;tag=as48190407
Call-ID: 49d2d0610650fb167fd0a4852e052c17@81.201.84.27
CSeq: 103 BYE
User-Agent: Vox Callcontrol
Max-Forwards: 70
Content-Length: 0


--- (9 headers 0 lines) ---
Sending to 81.201.84.27 : 5060 (NAT)
Transmitting (NAT) to 81.201.84.27:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 81.201.84.27:5060;branch=z9hG4bK0a6169a2;received=81.201.84.27;rport=5060
From: "13054337646" <sip:13054337646@81.201.84.27>;tag=as2a216b77
To: <sip:16477245778@66.165.164.45>;tag=as48190407
Call-ID: 49d2d0610650fb167fd0a4852e052c17@81.201.84.27
CSeq: 103 BYE
User-Agent: CallWeaver
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:16477245778@66.165.164.45>
Content-Length: 0


---
Destroying call '49d2d0610650fb167fd0a4852e052c17@81.201.84.27'
atpserver104*CLI> 
<-- SIP read from 81.201.84.26:5060: 
INVITE sip:16477245778@66.165.164.45 SIP/2.0
Via: SIP/2.0/UDP 81.201.84.26:5060;branch=z9hG4bK3f87916b;rport
From: "17868667762" <sip:17868667762@81.201.84.26>;tag=as5684bd40
To: <sip:16477245778@66.165.164.45>
Contact: <sip:17868667762@81.201.84.26>
Call-ID: 275022702403d37275677f4513807370@81.201.84.26
CSeq: 102 INVITE
User-Agent: Vox Callcontrol
Max-Forwards: 70
Date: Thu, 24 May 2007 15:05:08 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 311

v=0
o=root 25697 25697 IN IP4 81.201.84.26
s=session
c=IN IP4 81.201.84.26
t=0 0
m=audio 12150 RTP/AVP 8 0 18 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

--- (14 headers 15 lines) ---
Using INVITE request as basis request - 275022702403d37275677f4513807370@81.201.84.26
Sending to 81.201.84.26 : 5060 (NAT)
Found peer '81.201.84.26'
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 18
Found RTP audio format 101
Peer audio RTP is at port 81.201.84.26:12150
Found description format PCMA
Found description format PCMU
Found description format G729
Found description format telephone-event
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Looking for 16477245778 in voxbone-in (domain 66.165.164.45)
list_route: hop: <sip:17868667762@81.201.84.26>
Transmitting (NAT) to 81.201.84.26:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 81.201.84.26:5060;branch=z9hG4bK3f87916b;rport;received=81.201.84.26
From: "17868667762" <sip:17868667762@81.201.84.26>;tag=as5684bd40
To: <sip:16477245778@66.165.164.45>
Call-ID: 275022702403d37275677f4513807370@81.201.84.26
CSeq: 102 INVITE
User-Agent: CallWeaver
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:16477245778@66.165.164.45>
Content-Length: 0


---
We're at 66.165.164.45 port 11396
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 81.201.84.26:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 81.201.84.26:5060;branch=z9hG4bK3f87916b;rport;received=81.201.84.26
From: "17868667762" <sip:17868667762@81.201.84.26>;tag=as5684bd40
To: <sip:16477245778@66.165.164.45>;tag=as6eb23e61
Call-ID: 275022702403d37275677f4513807370@81.201.84.26
CSeq: 102 INVITE
User-Agent: CallWeaver
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:16477245778@66.165.164.45>
Content-Type: application/sdp
Content-Length: 269

v=0
o=root 25483 25483 IN IP4 66.165.164.45
s=session
c=IN IP4 66.165.164.45
t=0 0
m=audio 11396 RTP/AVP 0 8 122 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:122 t38/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
atpserver104*CLI> 
<-- SIP read from 81.201.84.26:5060: 
ACK sip:16477245778@66.165.164.45 SIP/2.0
Via: SIP/2.0/UDP 81.201.84.26:5060;branch=z9hG4bK35c64c71;rport
From: "17868667762" <sip:17868667762@81.201.84.26>;tag=as5684bd40
To: <sip:16477245778@66.165.164.45>;tag=as6eb23e61
Contact: <sip:17868667762@81.201.84.26>
Call-ID: 275022702403d37275677f4513807370@81.201.84.26
CSeq: 102 ACK
User-Agent: Vox Callcontrol
Max-Forwards: 70
Content-Length: 0


--- (10 headers 0 lines) ---
set_destination: Parsing <sip:17868667762@81.201.84.26> for address/port to send to
set_destination: set destination to 81.201.84.26, port 5060
T.38 UDPTL is at port 66.165.164.45:11396
13 headers, 15 lines
Reliably Transmitting (NAT) to 81.201.84.26:5060:
INVITE sip:17868667762@81.201.84.26 SIP/2.0
Via: SIP/2.0/UDP 66.165.164.45:5060;branch=z9hG4bK573868d2;rport
From: <sip:16477245778@66.165.164.45>;tag=as6eb23e61
To: "17868667762" <sip:17868667762@81.201.84.26>;tag=as5684bd40
Contact: <sip:16477245778@66.165.164.45>
Call-ID: 275022702403d37275677f4513807370@81.201.84.26
CSeq: 102 INVITE
User-Agent: CallWeaver
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
X-callweaver-info: SIP re-invite (T38 switchover)
Content-Type: application/sdp
Content-Length: 346

v=0
o=root 25483 25484 IN IP4 66.165.164.45
s=session
c=IN IP4 66.165.164.45
t=0 0
m=image 11396 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxFillBitRemoval:0
a=T38FaxTranscodingMMR:0
a=T38FaxTranscodingJBIG:0
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:400
a=T38FaxMaxDatagram:400
a=T38FaxUdpEC:t38UDPFEC

---
May 24 11:33:03 NOTICE[3062512560]: channel.c:1651 opbx_read: Dropping incompatible voice frame on SIP/81.201.84.26-b6703008 of format slin since our native format has changed to alaw
atpserver104*CLI> 
<-- SIP read from 81.201.84.26:5060: 
SIP/2.0 488 Not acceptable here
Via: SIP/2.0/UDP 66.165.164.45:5060;branch=z9hG4bK573868d2;received=66.165.164.45;rport=5060
From: <sip:16477245778@66.165.164.45>;tag=as6eb23e61
To: "17868667762" <sip:17868667762@81.201.84.26>;tag=as5684bd40
Call-ID: 275022702403d37275677f4513807370@81.201.84.26
CSeq: 102 INVITE
User-Agent: Vox Callcontrol
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16


--- (12 headers 0 lines) ---
set_destination: Parsing <sip:17868667762@81.201.84.26> for address/port to send to
set_destination: set destination to 81.201.84.26, port 5060
Transmitting (NAT) to 81.201.84.26:5060:
ACK sip:17868667762@81.201.84.26 SIP/2.0
Via: SIP/2.0/UDP 66.165.164.45:5060;branch=z9hG4bK573868d2;rport
From: <sip:16477245778@66.165.164.45>;tag=as6eb23e61
To: "17868667762" <sip:17868667762@81.201.84.26>;tag=as5684bd40
Contact: <sip:16477245778@66.165.164.45>
Call-ID: 275022702403d37275677f4513807370@81.201.84.26
CSeq: 102 ACK
User-Agent: CallWeaver
Max-Forwards: 70
Content-Length: 0


---
atpserver104*CLI> 
<-- SIP read from 81.201.84.26:5060: 
BYE sip:16477245778@66.165.164.45 SIP/2.0
Via: SIP/2.0/UDP 81.201.84.26:5060;branch=z9hG4bK3e390820;rport
From: "17868667762" <sip:17868667762@81.201.84.26>;tag=as5684bd40
To: <sip:16477245778@66.165.164.45>;tag=as6eb23e61
Call-ID: 275022702403d37275677f4513807370@81.201.84.26
CSeq: 103 BYE
User-Agent: Vox Callcontrol
Max-Forwards: 70
Content-Length: 0


--- (9 headers 0 lines) ---
Sending to 81.201.84.26 : 5060 (NAT)
Transmitting (NAT) to 81.201.84.26:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 81.201.84.26:5060;branch=z9hG4bK3e390820;received=81.201.84.26;rport=5060
From: "17868667762" <sip:17868667762@81.201.84.26>;tag=as5684bd40
To: <sip:16477245778@66.165.164.45>;tag=as6eb23e61
Call-ID: 275022702403d37275677f4513807370@81.201.84.26
CSeq: 103 BYE
User-Agent: CallWeaver
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:16477245778@66.165.164.45>
Content-Length: 0
X-CallWeaver-HangupCause: Bearer capability not available


---
Destroying call '275022702403d37275677f4513807370@81.201.84.26'
atpserver104*CLI> 
<-- SIP read from 81.201.84.27:5060: 
INVITE sip:16477245778@66.165.164.45 SIP/2.0
Via: SIP/2.0/UDP 81.201.84.27:5060;branch=z9hG4bK55a6bdc9;rport
From: "17868667762" <sip:17868667762@81.201.84.27>;tag=as1cc96db7
To: <sip:16477245778@66.165.164.45>
Contact: <sip:17868667762@81.201.84.27>
Call-ID: 1bf9c0e031362984645b01ec75a99ade@81.201.84.27
CSeq: 102 INVITE
User-Agent: Vox Callcontrol
Max-Forwards: 70
Date: Thu, 24 May 2007 15:07:23 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 309

v=0
o=root 8947 8947 IN IP4 81.201.84.27
s=session
c=IN IP4 81.201.84.27
t=0 0
m=audio 18150 RTP/AVP 8 0 18 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

--- (14 headers 15 lines) ---
Using INVITE request as basis request - 1bf9c0e031362984645b01ec75a99ade@81.201.84.27
Sending to 81.201.84.27 : 5060 (NAT)
Found peer '81.201.84.27'
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 18
Found RTP audio format 101
Peer audio RTP is at port 81.201.84.27:18150
Found description format PCMA
Found description format PCMU
Found description format G729
Found description format telephone-event
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Looking for 16477245778 in voxbone-in (domain 66.165.164.45)
list_route: hop: <sip:17868667762@81.201.84.27>
Transmitting (NAT) to 81.201.84.27:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 81.201.84.27:5060;branch=z9hG4bK55a6bdc9;rport;received=81.201.84.27
From: "17868667762" <sip:17868667762@81.201.84.27>;tag=as1cc96db7
To: <sip:16477245778@66.165.164.45>
Call-ID: 1bf9c0e031362984645b01ec75a99ade@81.201.84.27
CSeq: 102 INVITE
User-Agent: CallWeaver
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:16477245778@66.165.164.45>
Content-Length: 0


---
We're at 66.165.164.45 port 17188
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 81.201.84.27:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 81.201.84.27:5060;branch=z9hG4bK55a6bdc9;rport;received=81.201.84.27
From: "17868667762" <sip:17868667762@81.201.84.27>;tag=as1cc96db7
To: <sip:16477245778@66.165.164.45>;tag=as4ae3ba28
Call-ID: 1bf9c0e031362984645b01ec75a99ade@81.201.84.27
CSeq: 102 INVITE
User-Agent: CallWeaver
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:16477245778@66.165.164.45>
Content-Type: application/sdp
Content-Length: 269

v=0
o=root 25483 25483 IN IP4 66.165.164.45
s=session
c=IN IP4 66.165.164.45
t=0 0
m=audio 17188 RTP/AVP 0 8 122 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:122 t38/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
atpserver104*CLI> 
<-- SIP read from 81.201.84.27:5060: 
ACK sip:16477245778@66.165.164.45 SIP/2.0
Via: SIP/2.0/UDP 81.201.84.27:5060;branch=z9hG4bK1ba829c6;rport
From: "17868667762" <sip:17868667762@81.201.84.27>;tag=as1cc96db7
To: <sip:16477245778@66.165.164.45>;tag=as4ae3ba28
Contact: <sip:17868667762@81.201.84.27>
Call-ID: 1bf9c0e031362984645b01ec75a99ade@81.201.84.27
CSeq: 102 ACK
User-Agent: Vox Callcontrol
Max-Forwards: 70
Content-Length: 0


--- (10 headers 0 lines) ---
set_destination: Parsing <sip:17868667762@81.201.84.27> for address/port to send to
set_destination: set destination to 81.201.84.27, port 5060
T.38 UDPTL is at port 66.165.164.45:17188
13 headers, 15 lines
Reliably Transmitting (NAT) to 81.201.84.27:5060:
INVITE sip:17868667762@81.201.84.27 SIP/2.0
Via: SIP/2.0/UDP 66.165.164.45:5060;branch=z9hG4bK460e5419;rport
From: <sip:16477245778@66.165.164.45>;tag=as4ae3ba28
To: "17868667762" <sip:17868667762@81.201.84.27>;tag=as1cc96db7
Contact: <sip:16477245778@66.165.164.45>
Call-ID: 1bf9c0e031362984645b01ec75a99ade@81.201.84.27
CSeq: 102 INVITE
User-Agent: CallWeaver
Max-Forwards: 70
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
X-callweaver-info: SIP re-invite (T38 switchover)
Content-Type: application/sdp
Content-Length: 346

v=0
o=root 25483 25484 IN IP4 66.165.164.45
s=session
c=IN IP4 66.165.164.45
t=0 0
m=image 17188 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxFillBitRemoval:0
a=T38FaxTranscodingMMR:0
a=T38FaxTranscodingJBIG:0
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:400
a=T38FaxMaxDatagram:400
a=T38FaxUdpEC:t38UDPFEC

---
May 24 11:35:18 NOTICE[3062512560]: channel.c:1651 opbx_read: Dropping incompatible voice frame on SIP/81.201.84.27-b6703008 of format slin since our native format has changed to alaw
atpserver104*CLI> 
<-- SIP read from 81.201.84.27:5060: 
SIP/2.0 488 Not acceptable here
Via: SIP/2.0/UDP 66.165.164.45:5060;branch=z9hG4bK460e5419;received=66.165.164.45;rport=5060
From: <sip:16477245778@66.165.164.45>;tag=as4ae3ba28
To: "17868667762" <sip:17868667762@81.201.84.27>;tag=as1cc96db7
Call-ID: 1bf9c0e031362984645b01ec75a99ade@81.201.84.27
CSeq: 102 INVITE
User-Agent: Vox Callcontrol
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16


--- (12 headers 0 lines) ---
set_destination: Parsing <sip:17868667762@81.201.84.27> for address/port to send to
set_destination: set destination to 81.201.84.27, port 5060
Transmitting (NAT) to 81.201.84.27:5060:
ACK sip:17868667762@81.201.84.27 SIP/2.0
Via: SIP/2.0/UDP 66.165.164.45:5060;branch=z9hG4bK460e5419;rport
From: <sip:16477245778@66.165.164.45>;tag=as4ae3ba28
To: "17868667762" <sip:17868667762@81.201.84.27>;tag=as1cc96db7
Contact: <sip:16477245778@66.165.164.45>
Call-ID: 1bf9c0e031362984645b01ec75a99ade@81.201.84.27
CSeq: 102 ACK
User-Agent: CallWeaver
Max-Forwards: 70
Content-Length: 0


---
atpserver104*CLI> 
<-- SIP read from 81.201.84.27:5060: 
BYE sip:16477245778@66.165.164.45 SIP/2.0
Via: SIP/2.0/UDP 81.201.84.27:5060;branch=z9hG4bK530a0398;rport
From: "17868667762" <sip:17868667762@81.201.84.27>;tag=as1cc96db7
To: <sip:16477245778@66.165.164.45>;tag=as4ae3ba28
Call-ID: 1bf9c0e031362984645b01ec75a99ade@81.201.84.27
CSeq: 103 BYE
User-Agent: Vox Callcontrol
Max-Forwards: 70
Content-Length: 0


--- (9 headers 0 lines) ---
Sending to 81.201.84.27 : 5060 (NAT)
Transmitting (NAT) to 81.201.84.27:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 81.201.84.27:5060;branch=z9hG4bK530a0398;received=81.201.84.27;rport=5060
From: "17868667762" <sip:17868667762@81.201.84.27>;tag=as1cc96db7
To: <sip:16477245778@66.165.164.45>;tag=as4ae3ba28
Call-ID: 1bf9c0e031362984645b01ec75a99ade@81.201.84.27
CSeq: 103 BYE
User-Agent: CallWeaver
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Max-Forwards: 70
Contact: <sip:16477245778@66.165.164.45>
Content-Length: 0
-CallWeaver-HangupCause: Bearer capability not available


---
Destroying call '1bf9c0e031362984645b01ec75a99ade@81.201.84.27'
atpserver104*CLI> exit