rendered paste body<-- SIP read from 202.171.182.120:5060:
INVITE sip:97131806@203.82.162.33:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.51:5060;rport;branch=z9hG4bK28f8d27000084
From: 97131800<sip:97131800@203.82.162.33:5060>;tag=8c113248
To: <sip:97131806@203.82.162.33:5060>
Call-ID: b7242eeb-13@192.168.1.51
CSeq: 248108771 INVITE
Max-Forwards: 70
Contact: 97131800<sip:97131800@192.168.1.51:5060>
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: 309
Content-Type: application/sdp
v=0
o=- 1369133069 1125898167 IN IP4 192.168.1.51
s=Voice call
e=you@yourcompany.com
c=IN IP4 192.168.1.51
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
--- (14 headers 14 lines)---
Sending to 202.171.182.120 : 5060 (NAT)
Using INVITE request as basis request - b7242eeb-13@192.168.1.51
Reliably Transmitting (NAT) to 202.171.182.120:5060:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.1.51:5060;branch=z9hG4bK28f8d27000084;received=202.171.182.120;rport=5060
From: 97131800<sip:97131800@203.82.162.33:5060>;tag=8c113248
To: <sip:97131806@203.82.162.33:5060>;tag=as0b37710d
Call-ID: b7242eeb-13@192.168.1.51
CSeq: 248108771 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="4f417525"
Content-Length: 0
---
Scheduling destruction of SIP dialog 'b7242eeb-13@192.168.1.51' in 32000 ms (Method: INVITE)
Found user '97131800'
thirtythree*CLI>
<-- SIP read from 202.171.182.120:5060:
ACK sip:97131806@203.82.162.33:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.51:5060;rport;branch=z9hG4bK28f8d27000084
From: 97131800<sip:97131800@203.82.162.33:5060>;tag=8c113248
To: <sip:97131806@203.82.162.33:5060>;tag=as0b37710d
Call-ID: b7242eeb-13@192.168.1.51
CSeq: 248108771 ACK
Max-Forwards: 70
User-Agent: Soundwin_ATA
Content-Length: 0
--- (9 headers 0 lines)---
thirtythree*CLI>
<-- SIP read from 202.171.182.120:5060:
INVITE sip:97131806@203.82.162.33:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.51:5060;rport;branch=z9hG4bK1396c49000085
From: 97131800<sip:97131800@203.82.162.33:5060>;tag=8c113248
To: <sip:97131806@203.82.162.33:5060>
Call-ID: b7242eeb-13@192.168.1.51
CSeq: 248108772 INVITE
Max-Forwards: 70
Contact: 97131800<sip:97131800@192.168.1.51:5060>
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="4f417525", response="895606bdec8a11eb0d84e112ec4f1c92"
Content-Length: 309
Content-Type: application/sdp
v=0
o=- 1369133069 1125898167 IN IP4 192.168.1.51
s=Voice call
e=you@yourcompany.com
c=IN IP4 192.168.1.51
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
--- (15 headers 14 lines)---
Sending to 202.171.182.120 : 5060 (NAT)
Using INVITE request as basis request - b7242eeb-13@192.168.1.51
Found user '97131800'
Got T.38 offer in SDP in dialog b7242eeb-13@192.168.1.51
Peer doesn't provide audio
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x3d0fff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc|g726aal2|jpeg|h261|h263|h263p|h264)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x7 (telephone-event|CN), combined - 0x1 (telephone-event)
Peer audio RTP is at port 192.168.1.51:5000
Peer video RTP is at port 192.168.1.51:24585
Looking for 97131806 in from-internal (domain 203.82.162.33)
list_route: hop: <sip:97131800@192.168.1.51:5060>
Transmitting (NAT) to 202.171.182.120:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.51:5060;branch=z9hG4bK1396c49000085;received=202.171.182.120;rport=5060
From: 97131800<sip:97131800@203.82.162.33:5060>;tag=8c113248
To: <sip:97131806@203.82.162.33:5060>
Call-ID: b7242eeb-13@192.168.1.51
CSeq: 248108772 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 11354
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 202.171.182.120:10800:
INVITE sip:97131806@192.168.1.54:5060 SIP/2.0
Via: SIP/2.0/UDP 203.82.162.33:5060;branch=z9hG4bK4883837d;rport
From: "97131800" <sip:97131800@203.82.162.33>;tag=as7cc2ac54
To: <sip:97131806@192.168.1.54:5060>
Contact: <sip:97131800@203.82.162.33>
Call-ID: 107dcb9d747564104f7a1dbf1aac0926@203.82.162.33
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 24 Jul 2006 02:46:00 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 254
v=0
o=root 19738 19738 IN IP4 203.82.162.33
s=session
c=IN IP4 203.82.162.33
t=0 0
m=audio 11354 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=sendrecv
---
thirtythree*CLI>
<-- SIP read from 202.171.182.120:10800:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 203.82.162.33:5060;rport=5060;received=203.82.162.33;branch=z9hG4bK4883837d
From: "97131800"<sip:97131800@203.82.162.33>;tag=as7cc2ac54
To: <sip:97131806@192.168.1.54:5060>
Call-ID: 107dcb9d747564104f7a1dbf1aac0926@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 202.171.182.120:10800:
SIP/2.0 488 Incompatible Media Format
Via: SIP/2.0/UDP 203.82.162.33:5060;rport=5060;received=203.82.162.33;branch=z9hG4bK4883837d
From: "97131800"<sip:97131800@203.82.162.33>;tag=as7cc2ac54
To: <sip:97131806@192.168.1.54:5060>;tag=15a4583
Call-ID: 107dcb9d747564104f7a1dbf1aac0926@203.82.162.33
CSeq: 102 INVITE
Max-Forwards: 70
User-Agent: Soundwin_ATA
Warning: 305 ATA.Soundwin "Incompatible Media Format"
Content-Length: 0
--- (10 headers 0 lines)---
Transmitting (NAT) to 202.171.182.120:10800:
ACK sip:97131806@192.168.1.54:5060 SIP/2.0
Via: SIP/2.0/UDP 203.82.162.33:5060;branch=z9hG4bK4883837d;rport
From: "97131800" <sip:97131800@203.82.162.33>;tag=as7cc2ac54
To: <sip:97131806@192.168.1.54:5060>;tag=15a4583
Contact: <sip:97131800@203.82.162.33>
Call-ID: 107dcb9d747564104f7a1dbf1aac0926@203.82.162.33
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
Transmitting (NAT) to 202.171.182.120:5060:
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 192.168.1.51:5060;branch=z9hG4bK1396c49000085;received=202.171.182.120;rport=5060
From: 97131800<sip:97131800@203.82.162.33:5060>;tag=8c113248
To: <sip:97131806@203.82.162.33:5060>;tag=as716bcd36
Call-ID: b7242eeb-13@192.168.1.51
CSeq: 248108772 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
X-Asterisk-HangupCause: Bearer capability not available
X-Asterisk-HangupCauseCode: 58
---
thirtythree*CLI>
<-- SIP read from 202.171.182.120:5060:
ACK sip:97131806@203.82.162.33:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.51:5060;rport;branch=z9hG4bK1396c49000085
From: 97131800<sip:97131800@203.82.162.33:5060>;tag=8c113248
To: <sip:97131806@203.82.162.33:5060>;tag=as716bcd36
Call-ID: b7242eeb-13@192.168.1.51
CSeq: 248108772 ACK
Max-Forwards: 70
User-Agent: Soundwin_ATA
Content-Length: 0
--- (9 headers 0 lines)---
Really destroying SIP dialog '107dcb9d747564104f7a1dbf1aac0926@203.82.162.33' Me
Really destroying SIP dialog 'b7242eeb-13@192.168.1.51' Method: ACK