<-- SIP read from 203.82.162.40:5060:
INVITE sip:97131806@203.82.162.33:5060 SIP/2.0
Via: SIP/2.0/UDP 203.82.162.40:5060;rport;branch=z9hG4bK8814f8f100008
From: 97131800<sip:97131800@203.82.162.33:5060>;tag=c9612e05
To: <sip:97131806@203.82.162.33:5060>
Call-ID: f3ba6eac-2@203.82.162.40
CSeq: 165143708 INVITE
Max-Forwards: 70
Contact: 97131800<sip:97131800@203.82.162.40:5060>
User-Agent: Soundwin_ATA
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, SUBSCRIBE, REFER, NOTIFY, PRACK, UPDATE, MESSAGE, SERVICE, INFO
Allow-Events: refer
Supported: 100rel, replaces
Content-Length: 514
Content-Type: application/sdp

v=0
o=- 1681692777 1714636915 IN IP4 203.82.162.40
s=Voice call
e=you@yourcompany.com
c=IN IP4 203.82.162.40
t=0 0
a=sendrecv
m=audio 5000 RTP/AVP 8 0 18 98 96 97 4 99 104 100
a=ptime:20
a=maxptime:30
a=sendrecv
a=rtpmap:8 pcma/8000
a=rtpmap:0 pcmu/8000
a=rtpmap:18 g729/8000
a=rtpmap:98 g726-32/8000
a=rtpmap:96 g726-16/8000
a=rtpmap:97 g726-24/8000
a=rtpmap:4 g723/8000
a=fmtp:4 bitrate=6.3
a=rtpmap:99 g726-40/8000
a=rtpmap:104 ilbc/8000
a=rtpmap:100 telephone-event/8000
a=fmtp:100 0-15

--- (14 headers 23 lines)---
Sending to 203.82.162.40 : 5060 (NAT)
Using INVITE request as basis request - f3ba6eac-2@203.82.162.40
Reliably Transmitting (NAT) to 203.82.162.40:5060:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 203.82.162.40:5060;branch=z9hG4bK8814f8f100008;received=203.82.162.40;rport=5060
From: 97131800<sip:97131800@203.82.162.33:5060>;tag=c9612e05
To: <sip:97131806@203.82.162.33:5060>;tag=as62cd93f1
Call-ID: f3ba6eac-2@203.82.162.40
CSeq: 165143708 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:97131806@203.82.162.33>
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="4fd0b919"
Content-Length: 0


---
Scheduling destruction of SIP dialog 'f3ba6eac-2@203.82.162.40' in 32000 ms (Method: INVITE)
Found user '97131800'
thirtythree*CLI>
<-- SIP read from 203.82.162.40:5060:
ACK sip:97131806@203.82.162.33:5060 SIP/2.0
Via: SIP/2.0/UDP 203.82.162.40:5060;rport;branch=z9hG4bK8814f8f100008
From: 97131800<sip:97131800@203.82.162.33:5060>;tag=c9612e05
To: <sip:97131806@203.82.162.33:5060>;tag=as62cd93f1
Call-ID: f3ba6eac-2@203.82.162.40
CSeq: 165143708 ACK
Max-Forwards: 70
User-Agent: Soundwin_ATA
Content-Length: 0


--- (9 headers 0 lines)---
thirtythree*CLI>
<-- SIP read from 203.82.162.40:5060:
INVITE sip:97131806@203.82.162.33:5060 SIP/2.0
Via: SIP/2.0/UDP 203.82.162.40:5060;rport;branch=z9hG4bK7344a2f500009
From: 97131800<sip:97131800@203.82.162.33:5060>;tag=c9612e05
To: <sip:97131806@203.82.162.33:5060>
Call-ID: f3ba6eac-2@203.82.162.40
CSeq: 165143709 INVITE
Max-Forwards: 70
Contact: 97131800<sip:97131800@203.82.162.40:5060>
User-Agent: Soundwin_ATA
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, SUBSCRIBE, REFER, NOTIFY, PRACK, UPDATE, MESSAGE, SERVICE, INFO
Allow-Events: refer
Supported: 100rel, replaces
Proxy-Authorization: Digest username="97131800", realm="asterisk", uri="sip:97131806@203.82.162.33", nonce="4fd0b919", response="02ce76176299f4be0447ebee46e111ea"
Content-Length: 514
Content-Type: application/sdp

v=0
o=- 1681692777 1714636915 IN IP4 203.82.162.40
s=Voice call
e=you@yourcompany.com
c=IN IP4 203.82.162.40
t=0 0
a=sendrecv
m=audio 5000 RTP/AVP 8 0 18 98 96 97 4 99 104 100
a=ptime:20
a=maxptime:30
a=sendrecv
a=rtpmap:8 pcma/8000
a=rtpmap:0 pcmu/8000
a=rtpmap:18 g729/8000
a=rtpmap:98 g726-32/8000
a=rtpmap:96 g726-16/8000
a=rtpmap:97 g726-24/8000
a=rtpmap:4 g723/8000
a=fmtp:4 bitrate=6.3
a=rtpmap:99 g726-40/8000
a=rtpmap:104 ilbc/8000
a=rtpmap:100 telephone-event/8000
a=fmtp:100 0-15

--- (15 headers 23 lines)---
Sending to 203.82.162.40 : 5060 (NAT)
Using INVITE request as basis request - f3ba6eac-2@203.82.162.40
Found user '97131800'
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 18
Found RTP audio format 98
Found RTP audio format 96
Found RTP audio format 97
Found RTP audio format 4
Found RTP audio format 99
Found RTP audio format 104
Found RTP audio format 100
Peer audio RTP is at port 203.82.162.40:5000
Found description format pcma for ID 8
Found description format pcmu for ID 0
Found description format g729 for ID 18
Found description format g726-32 for ID 98
Found description format g726-16 for ID 96
Found description format g726-24 for ID 97
Found description format g723 for ID 4
Found description format g726-40 for ID 99
Found description format ilbc for ID 104
Found description format telephone-event for ID 100
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x200d0d (g723|ulaw|alaw|g726|g729|ilbc|h264)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 203.82.162.40:5000
Peer video RTP is at port 203.82.162.40:38920
Looking for 97131806 in from-internal (domain 203.82.162.33)
list_route: hop: <sip:97131800@203.82.162.40:5060>
Transmitting (NAT) to 203.82.162.40:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 203.82.162.40:5060;branch=z9hG4bK7344a2f500009;received=203.82.162.40;rport=5060
From: 97131800<sip:97131800@203.82.162.33:5060>;tag=c9612e05
To: <sip:97131806@203.82.162.33:5060>
Call-ID: f3ba6eac-2@203.82.162.40
CSeq: 165143709 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:97131806@203.82.162.33>
Content-Length: 0


---
Audio is at 203.82.162.33 port 11428
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 203.82.162.41:5060:
INVITE sip:97131806@203.82.162.41:5060 SIP/2.0
Via: SIP/2.0/UDP 203.82.162.33:5060;branch=z9hG4bK7ef84bdb;rport
From: "97131800" <sip:97131800@203.82.162.33>;tag=as003b0f10
To: <sip:97131806@203.82.162.41:5060>
Contact: <sip:97131800@203.82.162.33>
Call-ID: 1af2fa69406f74a621272d7063e55188@203.82.162.33
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 28 Jul 2006 05:25:31 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 251

v=0
o=root 8992 8992 IN IP4 203.82.162.40
s=session
c=IN IP4 203.82.162.40
t=0 0
m=audio 5000 RTP/AVP 0 8 100
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:100 telephone-event/8000
a=fmtp:100 0-16
a=silenceSupp:off - - - -
a=sendrecv

---
thirtythree*CLI>
<-- SIP read from 203.82.162.41:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 203.82.162.33:5060;rport=5060;received=203.82.162.33;branch=z9hG4bK7ef84bdb
From: "97131800"<sip:97131800@203.82.162.33>;tag=as003b0f10
To: <sip:97131806@203.82.162.41:5060>
Call-ID: 1af2fa69406f74a621272d7063e55188@203.82.162.33
CSeq: 102 INVITE
Max-Forwards: 70
User-Agent: Soundwin_ATA
Content-Length: 0


--- (9 headers 0 lines)---
thirtythree*CLI>
<-- SIP read from 203.82.162.41:5060:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 203.82.162.33:5060;rport=5060;received=203.82.162.33;branch=z9hG4bK7ef84bdb
From: "97131800"<sip:97131800@203.82.162.33>;tag=as003b0f10
To: <sip:97131806@203.82.162.41:5060>;tag=b3f48d5
Call-ID: 1af2fa69406f74a621272d7063e55188@203.82.162.33
CSeq: 102 INVITE
Max-Forwards: 70
Contact: <sip:97131806@203.82.162.41:5060>
User-Agent: Soundwin_ATA
Content-Length: 0


--- (10 headers 0 lines)---
Transmitting (NAT) to 203.82.162.40:5060:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 203.82.162.40:5060;branch=z9hG4bK7344a2f500009;received=203.82.162.40;rport=5060
From: 97131800<sip:97131800@203.82.162.33:5060>;tag=c9612e05
To: <sip:97131806@203.82.162.33:5060>;tag=as633ebc1b
Call-ID: f3ba6eac-2@203.82.162.40
CSeq: 165143709 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:97131806@203.82.162.33>
Content-Length: 0
thirtythree*CLI>

---
thirtythree*CLI>
<-- SIP read from 203.82.162.41:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 203.82.162.33:5060;rport=5060;received=203.82.162.33;branch=z9hG4bK7ef84bdb
From: "97131800"<sip:97131800@203.82.162.33>;tag=as003b0f10
To: <sip:97131806@203.82.162.41:5060>;tag=b3f48d5
Call-ID: 1af2fa69406f74a621272d7063e55188@203.82.162.33
CSeq: 102 INVITE
Max-Forwards: 70
Contact: <sip:97131806@203.82.162.41:5060>
User-Agent: Soundwin_ATA
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, SUBSCRIBE, REFER, NOTIFY, PRACK, UPDATE, MESSAGE, SERVICE, INFO
Allow-Events: refer
Supported: 100rel, replaces
Content-Length: 220
Content-Type: application/sdp

v=0
o=- 596516649 1189641421 IN IP4 203.82.162.41
s=session
c=IN IP4 203.82.162.41
t=0 0
a=sendrecv
m=audio 5000 RTP/AVP 0 100
a=sendrecv
a=rtpmap:0 pcmu/8000
a=rtpmap:100 telephone-event/8000
a=fmtp:100 0-16

--- (14 headers 11 lines)---
Found RTP audio format 0
Found RTP audio format 100
Peer audio RTP is at port 203.82.162.41:5000
Found description format pcmu for ID 0
Found description format telephone-event for ID 100
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=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 203.82.162.41:5000
list_route: hop: <sip:97131806@203.82.162.41:5060>
set_destination: Parsing <sip:97131806@203.82.162.41:5060> for address/port to send to
set_destination: set destination to 203.82.162.41, port 5060
Transmitting (NAT) to 203.82.162.41:5060:
ACK sip:97131806@203.82.162.41:5060 SIP/2.0
Via: SIP/2.0/UDP 203.82.162.33:5060;branch=z9hG4bK1196b1b4;rport
From: "97131800" <sip:97131800@203.82.162.33>;tag=as003b0f10
To: <sip:97131806@203.82.162.41:5060>;tag=b3f48d5
Contact: <sip:97131800@203.82.162.33>
Call-ID: 1af2fa69406f74a621272d7063e55188@203.82.162.33
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
Audio is at 203.82.162.33 port 11872
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 203.82.162.40:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 203.82.162.40:5060;branch=z9hG4bK7344a2f500009;received=203.82.162.40;rport=5060
From: 97131800<sip:97131800@203.82.162.33:5060>;tag=c9612e05
To: <sip:97131806@203.82.162.33:5060>;tag=as633ebc1b
Call-ID: f3ba6eac-2@203.82.162.40
CSeq: 165143709 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:97131806@203.82.162.33>
Content-Type: application/sdp
Content-Length: 251

v=0
o=root 8992 8992 IN IP4 203.82.162.41
s=session
c=IN IP4 203.82.162.41
t=0 0
m=audio 5000 RTP/AVP 0 8 100
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:100 telephone-event/8000
a=fmtp:100 0-16
a=silenceSupp:off - - - -
a=sendrecv

---
thirtythree*CLI>
<-- SIP read from 203.82.162.40:5060:
ACK sip:97131806@203.82.162.33:5060 SIP/2.0
Via: SIP/2.0/UDP 203.82.162.40:5060;rport;branch=z9hG4bK957a5a7d
From: 97131800<sip:97131800@203.82.162.33:5060>;tag=c9612e05
To: <sip:97131806@203.82.162.33:5060>;tag=as633ebc1b
Call-ID: f3ba6eac-2@203.82.162.40
CSeq: 165143709 ACK
Max-Forwards: 70
User-Agent: Soundwin_ATA
Proxy-Authorization: Digest username="97131800", realm="asterisk", uri="sip:97131806@203.82.162.33", nonce="4fd0b919", response="02ce76176299f4be0447ebee46e111ea"
Content-Length: 0


--- (10 headers 0 lines)---
thirtythree*CLI>
<-- SIP read from 203.82.162.41:5060:
INVITE sip:97131800@203.82.162.33:5060 SIP/2.0
Via: SIP/2.0/UDP 203.82.162.41:5060;rport;branch=z9hG4bK23d0b20600015
From: <sip:97131806@203.82.162.41:5060>;tag=b3f48d5
To: "97131800"<sip:97131800@203.82.162.33>;tag=as003b0f10
Call-ID: 1af2fa69406f74a621272d7063e55188@203.82.162.33
CSeq: 1302760466 INVITE
Max-Forwards: 70
Contact: <sip:97131806@203.82.162.41:5060>
User-Agent: Soundwin_ATA
Content-Length: 288
Content-Type: application/sdp

v=0
o=- 1025202362 1350490027 IN IP4 203.82.162.41
s=Voice call
c=IN IP4 203.82.162.41
t=0 0
a=sendrecv
m=image 5000 udptl t38
a=sendrecv
a=T38MaxBitRate:14400
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:280
a=T38FaxMaxDatagram:316
a=T38FaxUdpEC:t38UDPRedundancy

--- (11 headers 13 lines)---
Sending to 203.82.162.41 : 5060 (NAT)
Got T.38 offer in SDP in dialog 1af2fa69406f74a621272d7063e55188@203.82.162.33
Peer doesn't provide audio
Capabilities: us - 0xc (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)
Jul 28 15:25:44 NOTICE[9003]: chan_sip.c:4858 process_sdp: No compatible codecs, not accepting this offer!
Transmitting (NAT) to 203.82.162.41:5060:
SIP/2.0 488 Not acceptable here
Via: SIP/2.0/UDP 203.82.162.41:5060;branch=z9hG4bK23d0b20600015;received=203.82.162.41;rport=5060
From: <sip:97131806@203.82.162.41:5060>;tag=b3f48d5
To: "97131800"<sip:97131800@203.82.162.33>;tag=as003b0f10
Call-ID: 1af2fa69406f74a621272d7063e55188@203.82.162.33
CSeq: 1302760466 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:97131800@203.82.162.33>
Content-Length: 0
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16


---
thirtythree*CLI>
<-- SIP read from 203.82.162.41:5060:
ACK sip:97131800@203.82.162.33:5060 SIP/2.0
Via: SIP/2.0/UDP 203.82.162.41:5060;rport;branch=z9hG4bK23d0b20600015
From: <sip:97131806@203.82.162.41:5060>;tag=b3f48d5
To: "97131800"<sip:97131800@203.82.162.33>;tag=as003b0f10
Call-ID: 1af2fa69406f74a621272d7063e55188@203.82.162.33
CSeq: 1302760466 ACK
Max-Forwards: 70
User-Agent: Soundwin_ATA
Content-Length: 0


--- (9 headers 0 lines)---
thirtythree*CLI>
<-- SIP read from 203.82.162.41:5060:
INVITE sip:97131800@203.82.162.33:5060 SIP/2.0
Via: SIP/2.0/UDP 203.82.162.41:5060;rport;branch=z9hG4bK4978bbdc00016
From: <sip:97131806@203.82.162.41:5060>;tag=b3f48d5
To: "97131800"<sip:97131800@203.82.162.33>;tag=as003b0f10
Call-ID: 1af2fa69406f74a621272d7063e55188@203.82.162.33
CSeq: 1302760467 INVITE
Max-Forwards: 70
Contact: <sip:97131806@203.82.162.41:5060>
User-Agent: Soundwin_ATA
Content-Length: 442
Content-Type: application/sdp

v=0
o=- 783368690 1102520059 IN IP4 203.82.162.41
s=Voice call
e=you@yourcompany.com
c=IN IP4 203.82.162.41
t=0 0
a=sendrecv
m=audio 5000 RTP/AVP 121 0 100
a=ptime:20
a=maxptime:30
a=sendrecv
a=rtpmap:121 t38/8000
a=rtpmap:0 pcmu/8000
a=rtpmap:100 telephone-event/8000
a=fmtp:100 0-15
a=T38MaxBitRate:14400
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:280
a=T38FaxMaxDatagram:316
a=T38FaxUdpEC:t38UDPRedundancy

--- (11 headers 20 lines)---
Sending to 203.82.162.41 : 5060 (NAT)
Found RTP audio format 121
Found RTP audio format 0
Found RTP audio format 100
Peer audio RTP is at port 203.82.162.41:5000
Found description format t38 for ID 121
Found description format pcmu for ID 0
Found description format telephone-event for ID 100
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x5 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 203.82.162.41:5000
Audio is at 203.82.162.33 port 11428
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 203.82.162.41:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 203.82.162.41:5060;branch=z9hG4bK4978bbdc00016;received=203.82.162.41;rport=5060
From: <sip:97131806@203.82.162.41:5060>;tag=b3f48d5
To: "97131800"<sip:97131800@203.82.162.33>;tag=as003b0f10
Call-ID: 1af2fa69406f74a621272d7063e55188@203.82.162.33
CSeq: 1302760467 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:97131800@203.82.162.33>
Content-Type: application/sdp
Content-Length: 227

v=0
o=root 8992 8993 IN IP4 203.82.162.40
s=session
c=IN IP4 203.82.162.40
t=0 0
m=audio 5000 RTP/AVP 0 100
a=rtpmap:0 PCMU/8000
a=rtpmap:100 telephone-event/8000
a=fmtp:100 0-16
a=silenceSupp:off - - - -
a=sendrecv

---
thirtythree*CLI>
<-- SIP read from 203.82.162.41:5060:
ACK sip:97131800@203.82.162.33:5060 SIP/2.0
Via: SIP/2.0/UDP 203.82.162.41:5060;rport;branch=z9hG4bK39660f19
From: <sip:97131806@203.82.162.41:5060>;tag=b3f48d5
To: "97131800"<sip:97131800@203.82.162.33>;tag=as003b0f10
Call-ID: 1af2fa69406f74a621272d7063e55188@203.82.162.33
CSeq: 1302760467 ACK
Max-Forwards: 70
User-Agent: Soundwin_ATA
Content-Length: 0


--- (9 headers 0 lines)---
thirtythree*CLI>
<-- SIP read from 203.82.162.41:5060:
INVITE sip:97131800@203.82.162.33:5060 SIP/2.0
Via: SIP/2.0/UDP 203.82.162.41:5060;rport;branch=z9hG4bK9126a8c100017
From: <sip:97131806@203.82.162.41:5060>;tag=b3f48d5
To: "97131800"<sip:97131800@203.82.162.33>;tag=as003b0f10
Call-ID: 1af2fa69406f74a621272d7063e55188@203.82.162.33
CSeq: 1302760468 INVITE
Max-Forwards: 70
Contact: <sip:97131806@203.82.162.41:5060>
User-Agent: Soundwin_ATA
Content-Length: 288
Content-Type: application/sdp

v=0
o=- 2044897763 1967513926 IN IP4 203.82.162.41
s=Voice call
c=IN IP4 203.82.162.41
t=0 0
a=sendrecv
m=image 5000 udptl t38
a=sendrecv
a=T38MaxBitRate:14400
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:280
a=T38FaxMaxDatagram:316
a=T38FaxUdpEC:t38UDPRedundancy

--- (11 headers 13 lines)---
Sending to 203.82.162.41 : 5060 (NAT)
Got T.38 offer in SDP in dialog 1af2fa69406f74a621272d7063e55188@203.82.162.33
Peer doesn't provide audio
Capabilities: us - 0xc (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)
Jul 28 15:25:44 NOTICE[9003]: chan_sip.c:4858 process_sdp: No compatible codecs, not accepting this offer!
Transmitting (NAT) to 203.82.162.41:5060:
SIP/2.0 488 Not acceptable here
Via: SIP/2.0/UDP 203.82.162.41:5060;branch=z9hG4bK9126a8c100017;received=203.82.162.41;rport=5060
From: <sip:97131806@203.82.162.41:5060>;tag=b3f48d5
To: "97131800"<sip:97131800@203.82.162.33>;tag=as003b0f10
Call-ID: 1af2fa69406f74a621272d7063e55188@203.82.162.33
CSeq: 1302760468 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:97131800@203.82.162.33>
Content-Length: 0
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16


---
thirtythree*CLI>
<-- SIP read from 203.82.162.41:5060:
ACK sip:97131800@203.82.162.33:5060 SIP/2.0
Via: SIP/2.0/UDP 203.82.162.41:5060;rport;branch=z9hG4bK9126a8c100017
From: <sip:97131806@203.82.162.41:5060>;tag=b3f48d5
To: "97131800"<sip:97131800@203.82.162.33>;tag=as003b0f10
Call-ID: 1af2fa69406f74a621272d7063e55188@203.82.162.33
CSeq: 1302760468 ACK
Max-Forwards: 70
User-Agent: Soundwin_ATA
Content-Length: 0


--- (9 headers 0 lines)---
thirtythree*CLI>
<-- SIP read from 203.82.162.41:5060:
BYE sip:97131800@203.82.162.33:5060 SIP/2.0
Via: SIP/2.0/UDP 203.82.162.41:5060;rport;branch=z9hG4bKfaf3777c00018
From: <sip:97131806@203.82.162.41:5060>;tag=b3f48d5
To: "97131800"<sip:97131800@203.82.162.33>;tag=as003b0f10
Call-ID: 1af2fa69406f74a621272d7063e55188@203.82.162.33
CSeq: 1302760469 BYE
Max-Forwards: 70
User-Agent: Soundwin_ATA
Content-Length: 0


--- (9 headers 0 lines)---
Sending to 203.82.162.41 : 5060 (NAT)
Transmitting (NAT) to 203.82.162.41:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 203.82.162.41:5060;branch=z9hG4bKfaf3777c00018;received=203.82.162.41;rport=5060
From: <sip:97131806@203.82.162.41:5060>;tag=b3f48d5
To: "97131800"<sip:97131800@203.82.162.33>;tag=as003b0f10
Call-ID: 1af2fa69406f74a621272d7063e55188@203.82.162.33
CSeq: 1302760469 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:97131800@203.82.162.33>
Content-Length: 0


---
set_destination: Parsing <sip:97131800@203.82.162.40:5060> for address/port to send to
set_destination: set destination to 203.82.162.40, port 5060
Reliably Transmitting (NAT) to 203.82.162.40:5060:
BYE sip:97131800@203.82.162.40:5060 SIP/2.0
Via: SIP/2.0/UDP 203.82.162.33:5060;branch=z9hG4bK3d4b652e;rport
From: <sip:97131806@203.82.162.33:5060>;tag=as633ebc1b
To: 97131800<sip:97131800@203.82.162.33:5060>;tag=c9612e05
Contact: <sip:97131806@203.82.162.33>
Call-ID: f3ba6eac-2@203.82.162.40
CSeq: 102 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
thirtythree*CLI>
<-- SIP read from 203.82.162.40:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 203.82.162.33:5060;rport=5060;received=203.82.162.33;branch=z9hG4bK3d4b652e
From: <sip:97131806@203.82.162.33:5060>;tag=as633ebc1b
To: 97131800<sip:97131800@203.82.162.33:5060>;tag=c9612e05
Call-ID: f3ba6eac-2@203.82.162.40
CSeq: 102 BYE
Max-Forwards: 70
User-Agent: Soundwin_ATA
Content-Length: 0


--- (9 headers 0 lines)---
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog '1af2fa69406f74a621272d7063e55188@203.82.162.33' Method: BYE
Really destroying SIP dialog 'f3ba6eac-2@203.82.162.40' Method: ACK