Skip to content

Instantly share code, notes, and snippets.

/spa112_pstn.log

Created June 3, 2013 20:39
Show Gist options
  • Star 0 You must be signed in to star a gist
  • Fork 0 You must be signed in to fork a gist
  • Save anonymous/5701207 to your computer and use it in GitHub Desktop.
Save anonymous/5701207 to your computer and use it in GitHub Desktop.
Download ZIP
Raw
spa112_pstn.log
-- Executing [74952150687@office:2] Dial("SIP/297-00000167", "SIP/mtt/74952150687,60,tTM(ondialanswer)") in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
Audio is at 18350
Adding codec 0x8 (alaw) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 80.75.130.136:5060:
INVITE sip:74952150687@80.75.130.136 SIP/2.0
Via: SIP/2.0/UDP 82.200.7.184:5060;branch=z9hG4bK0e118c4a;rport
Max-Forwards: 70
From: "4397" <sip:74957777777@82.200.7.184>;tag=as0e2e11a8
To: <sip:74952150687@80.75.130.136>
Contact: <sip:74957777777@82.200.7.184:5060>
Call-ID: 55c3594f5206005e3196bea907e88021@82.200.7.184:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
Date: Mon, 03 Jun 2013 20:34:37 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 275
v=0
o=root 1256665056 1256665056 IN IP4 82.200.7.184
s=Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
c=IN IP4 82.200.7.184
t=0 0
m=audio 18350 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
---
-- Called SIP/mtt/74952150687
<--- SIP read from UDP:80.75.130.136:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 82.200.7.184:5060;received=82.200.7.184;branch=z9hG4bK0e118c4a;rport=5060
From: "4397" <sip:74957777777@82.200.7.184>;tag=as0e2e11a8
To: <sip:74952150687@80.75.130.136>
Call-ID: 55c3594f5206005e3196bea907e88021@82.200.7.184:5060
CSeq: 102 INVITE
<------------->
--- (6 headers 0 lines) ---
<--- SIP read from UDP:80.75.130.136:5060 --->
SIP/2.0 183 Progress
Via: SIP/2.0/UDP 82.200.7.184:5060;received=82.200.7.184;branch=z9hG4bK0e118c4a;rport=5060
From: "4397" <sip:74957777777@82.200.7.184>;tag=as0e2e11a8
To: <sip:74952150687@80.75.130.136>;tag=1249863938-3792801228-380372658-4943405
Call-ID: 55c3594f5206005e3196bea907e88021@82.200.7.184:5060
CSeq: 102 INVITE
Contact: <sip:74952150687@80.75.130.136:5060;transport=udp>
Content-Type: application/sdp
Server: TS-v4.5.1-08b
Content-Length: 243
=0
o=- 1370291681 1370291681 IN IP4 80.75.130.136
s=-
c=IN IP4 80.75.130.136
t=0 0
m=audio 37756 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
a=silenceSupp:off - - - -
<------------->
--- (10 headers 12 lines) ---
list_route: hop: <sip:74952150687@80.75.130.136:5060;transport=udp>
Found RTP audio format 8
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=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 80.75.130.136:37756
-- SIP/mtt-00000168 is making progress passing it to SIP/297-00000167
Audio is at 13056
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
<--- Transmitting (NAT) to 192.168.33.200:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.33.200:5060;branch=z9hG4bK-40a28734;received=192.168.33.200;rport=5060
From: "297" <sip:297@192.168.5.253>;tag=ea48e7d0bf8d379bo0
To: <sip:74952150687@192.168.5.253>;tag=as49ceebd7
Call-ID: 1957afb4-53c87107@192.168.33.200
CSeq: 102 INVITE
Server: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:74952150687@192.168.5.253:5060>
Content-Type: application/sdp
Content-Length: 251
v=0
o=root 462103832 462103832 IN IP4 192.168.5.253
s=Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
c=IN IP4 192.168.5.253
t=0 0
m=audio 13056 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------>
<--- SIP read from UDP:80.75.130.136:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 82.200.7.184:5060;received=82.200.7.184;branch=z9hG4bK0e118c4a;rport=5060
From: "4397" <sip:74957777777@82.200.7.184>;tag=as0e2e11a8
To: <sip:74952150687@80.75.130.136>;tag=1249863938-3792801228-380372658-4943405
Call-ID: 55c3594f5206005e3196bea907e88021@82.200.7.184:5060
CSeq: 102 INVITE
Contact: <sip:74952150687@80.75.130.136:5060;transport=udp>
Content-Type: application/sdp
Allow: ACK, BYE, CANCEL, INFO, INVITE, OPTIONS, REFER, REGISTER, SUBSCRIBE, UPDATE
Server: TS-v4.5.1-08b
X-mera-expires: 86460
Content-Length: 243
v=0
o=- 1370291681 1370291681 IN IP4 80.75.130.136
s=-
c=IN IP4 80.75.130.136
t=0 0
m=audio 37756 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
a=silenceSupp:off - - - -
<------------->
--- (12 headers 12 lines) ---
Found RTP audio format 8
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=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 80.75.130.136:37756
list_route: hop: <sip:74952150687@80.75.130.136:5060;transport=udp>
set_destination: Parsing <sip:74952150687@80.75.130.136:5060;transport=udp> for address/port to send to
set_destination: set destination to 80.75.130.136:5060
Transmitting (NAT) to 80.75.130.136:5060:
ACK sip:74952150687@80.75.130.136:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 82.200.7.184:5060;branch=z9hG4bK0e8eb527;rport
Max-Forwards: 70
From: "4397" <sip:74957777777@82.200.7.184>;tag=as0e2e11a8
To: <sip:74952150687@80.75.130.136>;tag=1249863938-3792801228-380372658-4943405
Contact: <sip:74957777777@82.200.7.184:5060>
Call-ID: 55c3594f5206005e3196bea907e88021@82.200.7.184:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
Content-Length: 0
---
-- SIP/mtt-00000168 answered SIP/297-00000167
-- Executing [s@macro-ondialanswer:1] Macro("SIP/mtt-00000168", "onanswer,mtt/74952150687") in new stack
-- Executing [s@macro-onanswer:1] Macro("SIP/mtt-00000168", "setvar,operator,mtt/74952150687") in new stack
-- Executing [s@macro-setvar:1] Set("SIP/mtt-00000168", "MASTER_CHANNEL(CDR(operator))=mtt/74952150687") in new stack
-- Executing [s@macro-setvar:2] Set("SIP/mtt-00000168", "REALTIME_STORE(var_log,uniqueid,name,value)=1370291677.443,operator,mtt/74952150687") in new stack
-- Executing [s@macro-onanswer:2] Set("SIP/mtt-00000168", "MASTER_CHANNEL(CDR(wait_duration))=3") in new stack
-- Executing [s@macro-onanswer:3] Set("SIP/mtt-00000168", "MASTER_CHANNEL(CDR(real_disposition))=ANSWERED") in new stack
Audio is at 13056
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
<--- Reliably Transmitting (NAT) to 192.168.33.200:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.33.200:5060;branch=z9hG4bK-40a28734;received=192.168.33.200;rport=5060
From: "297" <sip:297@192.168.5.253>;tag=ea48e7d0bf8d379bo0
To: <sip:74952150687@192.168.5.253>;tag=as49ceebd7
Call-ID: 1957afb4-53c87107@192.168.33.200
CSeq: 102 INVITE
Server: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:74952150687@192.168.5.253:5060>
Content-Type: application/sdp
Content-Length: 251
v=0
o=root 462103832 462103833 IN IP4 192.168.5.253
s=Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
c=IN IP4 192.168.5.253
t=0 0
m=audio 13056 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------>
Retransmitting #1 (NAT) to 192.168.33.200:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.33.200:5060;branch=z9hG4bK-40a28734;received=192.168.33.200;rport=5060
From: "297" <sip:297@192.168.5.253>;tag=ea48e7d0bf8d379bo0
To: <sip:74952150687@192.168.5.253>;tag=as49ceebd7
Call-ID: 1957afb4-53c87107@192.168.33.200
CSeq: 102 INVITE
Server: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:74952150687@192.168.5.253:5060>
Content-Type: application/sdp
Content-Length: 251
v=0
o=root 462103832 462103833 IN IP4 192.168.5.253
s=Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
c=IN IP4 192.168.5.253
t=0 0
m=audio 13056 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
---
<--- SIP read from UDP:192.168.33.200:5060 --->
ACK sip:74952150687@192.168.5.253:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.33.200:5060;branch=z9hG4bK-376f80fd
From: "297" <sip:297@192.168.5.253>;tag=ea48e7d0bf8d379bo0
To: <sip:74952150687@192.168.5.253>;tag=as49ceebd7
Call-ID: 1957afb4-53c87107@192.168.33.200
CSeq: 102 ACK
Max-Forwards: 70
Authorization: Digest username="297",realm="asterisk",nonce="6b8984f9",uri="sip:74952150687@192.168.5.253",algorithm=MD5,response="e4432e3c6a0cb32da99c53bba2a6d814"
Contact: "297" <sip:297@192.168.33.200:5060;ref=297>
User-Agent: Cisco/SPA112-1.3.2(014)
Content-Length: 0
<------------->
--- (11 headers 0 lines) ---
<--- SIP read from UDP:192.168.33.200:5060 --->
ACK sip:74952150687@192.168.5.253:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.33.200:5060;branch=z9hG4bK-376f80fd
From: "297" <sip:297@192.168.5.253>;tag=ea48e7d0bf8d379bo0
To: <sip:74952150687@192.168.5.253>;tag=as49ceebd7
Call-ID: 1957afb4-53c87107@192.168.33.200
CSeq: 102 ACK
Max-Forwards: 70
Authorization: Digest username="297",realm="asterisk",nonce="6b8984f9",uri="sip:74952150687@192.168.5.253",algorithm=MD5,response="e4432e3c6a0cb32da99c53bba2a6d814"
Contact: "297" <sip:297@192.168.33.200:5060;ref=297>
User-Agent: Cisco/SPA112-1.3.2(014)
Content-Length: 0
<------------->
--- (11 headers 0 lines) ---
<--- SIP read from UDP:192.168.33.200:5060 --->
INVITE sip:74952150687@192.168.5.253:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.33.200:5060;branch=z9hG4bK-179929e2
From: "297" <sip:297@192.168.5.253>;tag=ea48e7d0bf8d379bo0
To: <sip:74952150687@192.168.5.253>;tag=as49ceebd7
Remote-Party-ID: "297" <sip:297@192.168.5.253>;screen=yes;party=calling
Call-ID: 1957afb4-53c87107@192.168.33.200
CSeq: 103 INVITE
Max-Forwards: 70
Authorization: Digest username="297",realm="asterisk",nonce="6b8984f9",uri="sip:74952150687@192.168.5.253:5060",algorithm=MD5,response="b4fe3c2bf1bcc7f8a85e468d00202155"
Contact: "297" <sip:297@192.168.33.200:5060;ref=297>
Expires: 30
User-Agent: Cisco/SPA112-1.3.2(014)
Content-Length: 271
Content-Type: application/sdp
v=0
o=- 2640556 2640557 IN IP4 192.168.33.200
s=-
c=IN IP4 192.168.33.200
t=0 0
m=image 16426 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:200
a=T38FaxMaxDatagram:200
a=T38FaxUdpEC:t38UDPRedundancy
<------------->
--- (14 headers 12 lines) ---
Sending to 192.168.33.200:5060 (NAT)
== Using UDPTL TOS bits 184
== Using UDPTL CoS mark 5
Got T.38 offer in SDP in dialog 1957afb4-53c87107@192.168.33.200
Capabilities: us - 0x8 (alaw), peer - audio=0x0 (nothing)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x0 (nothing)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x0 (nothing), combined - 0x0 (nothing)
Got T.38 Re-invite without audio. Keeping RTP active during T.38 session.
<--- Transmitting (NAT) to 192.168.33.200:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.33.200:5060;branch=z9hG4bK-179929e2;received=192.168.33.200;rport=5060
From: "297" <sip:297@192.168.5.253>;tag=ea48e7d0bf8d379bo0
To: <sip:74952150687@192.168.5.253>;tag=as49ceebd7
Call-ID: 1957afb4-53c87107@192.168.33.200
CSeq: 103 INVITE
Server: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:74952150687@192.168.5.253:5060>
Content-Length: 0
<------------>
== Using UDPTL TOS bits 184
== Using UDPTL CoS mark 5
set_destination: Parsing <sip:74952150687@80.75.130.136:5060;transport=udp> for address/port to send to
set_destination: set destination to 80.75.130.136:5060
Reliably Transmitting (NAT) to 80.75.130.136:5060:
INVITE sip:74952150687@80.75.130.136:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 82.200.7.184:5060;branch=z9hG4bK06864b0c;rport
Max-Forwards: 70
From: "4397" <sip:74957777777@82.200.7.184>;tag=as0e2e11a8
To: <sip:74952150687@80.75.130.136>;tag=1249863938-3792801228-380372658-4943405
Contact: <sip:74957777777@82.200.7.184:5060>
Call-ID: 55c3594f5206005e3196bea907e88021@82.200.7.184:5060
CSeq: 103 INVITE
User-Agent: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 286
v=0
o=root 1256665056 1256665057 IN IP4 82.200.7.184
s=Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
c=IN IP4 82.200.7.184
t=0 0
m=image 4022 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxDatagram:389
a=T38FaxUdpEC:t38UDPRedundancy
---
Retransmitting #1 (NAT) to 80.75.130.136:5060:
INVITE sip:74952150687@80.75.130.136:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 82.200.7.184:5060;branch=z9hG4bK06864b0c;rport
Max-Forwards: 70
From: "4397" <sip:74957777777@82.200.7.184>;tag=as0e2e11a8
To: <sip:74952150687@80.75.130.136>;tag=1249863938-3792801228-380372658-4943405
Contact: <sip:74957777777@82.200.7.184:5060>
Call-ID: 55c3594f5206005e3196bea907e88021@82.200.7.184:5060
CSeq: 103 INVITE
User-Agent: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 286
v=0
o=root 1256665056 1256665057 IN IP4 82.200.7.184
s=Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
c=IN IP4 82.200.7.184
t=0 0
m=image 4022 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxDatagram:389
a=T38FaxUdpEC:t38UDPRedundancy
---
<--- SIP read from UDP:80.75.130.136:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 82.200.7.184:5060;received=82.200.7.184;branch=z9hG4bK06864b0c;rport=5060
From: "4397" <sip:74957777777@82.200.7.184>;tag=as0e2e11a8
To: <sip:74952150687@80.75.130.136>;tag=1249863938-3792801228-380372658-4943405
Call-ID: 55c3594f5206005e3196bea907e88021@82.200.7.184:5060
CSeq: 103 INVITE
Contact: <sip:74952150687@80.75.130.136:5060;transport=udp>
Content-Type: application/sdp
Allow: ACK, BYE, CANCEL, INFO, INVITE, OPTIONS, REFER, REGISTER, SUBSCRIBE, UPDATE
Server: TS-v4.5.1-08b
Content-Length: 197
v=0
o=- 1370291681 1370291681 IN IP4 80.75.130.136
s=-
c=IN IP4 80.75.130.136
t=0 0
m=image 37756 udptl t38
a=T38FaxRateManagement:transferredTCF
a=T38FaxUdpEC:t38UDPRedundancy
a=sendrecv
<------------->
--- (11 headers 9 lines) ---
Got T.38 offer in SDP in dialog 55c3594f5206005e3196bea907e88021@82.200.7.184:5060
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x0 (nothing)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x0 (nothing)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x0 (nothing), combined - 0x0 (nothing)
Got T.38 Re-invite without audio. Keeping RTP active during T.38 session.
set_destination: Parsing <sip:74952150687@80.75.130.136:5060;transport=udp> for address/port to send to
set_destination: set destination to 80.75.130.136:5060
Transmitting (NAT) to 80.75.130.136:5060:
ACK sip:74952150687@80.75.130.136:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 82.200.7.184:5060;branch=z9hG4bK1641d2fd;rport
Max-Forwards: 70
From: "4397" <sip:74957777777@82.200.7.184>;tag=as0e2e11a8
To: <sip:74952150687@80.75.130.136>;tag=1249863938-3792801228-380372658-4943405
Contact: <sip:74957777777@82.200.7.184:5060>
Call-ID: 55c3594f5206005e3196bea907e88021@82.200.7.184:5060
CSeq: 103 ACK
User-Agent: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
Content-Length: 0
---
<--- Reliably Transmitting (NAT) to 192.168.33.200:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.33.200:5060;branch=z9hG4bK-179929e2;received=192.168.33.200;rport=5060
From: "297" <sip:297@192.168.5.253>;tag=ea48e7d0bf8d379bo0
To: <sip:74952150687@192.168.5.253>;tag=as49ceebd7
Call-ID: 1957afb4-53c87107@192.168.33.200
CSeq: 103 INVITE
Server: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:74952150687@192.168.5.253:5060>
Content-Type: application/sdp
Content-Length: 285
v=0
o=root 462103832 462103834 IN IP4 192.168.5.253
s=Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
c=IN IP4 192.168.5.253
t=0 0
m=image 4884 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:2400
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxDatagram:397
a=T38FaxUdpEC:t38UDPRedundancy
<------------>
Retransmitting #1 (NAT) to 192.168.33.200:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.33.200:5060;branch=z9hG4bK-179929e2;received=192.168.33.200;rport=5060
From: "297" <sip:297@192.168.5.253>;tag=ea48e7d0bf8d379bo0
To: <sip:74952150687@192.168.5.253>;tag=as49ceebd7
Call-ID: 1957afb4-53c87107@192.168.33.200
CSeq: 103 INVITE
Server: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:74952150687@192.168.5.253:5060>
Content-Type: application/sdp
Content-Length: 285
v=0
o=root 462103832 462103834 IN IP4 192.168.5.253
s=Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
c=IN IP4 192.168.5.253
t=0 0
m=image 4884 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:2400
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxDatagram:397
a=T38FaxUdpEC:t38UDPRedundancy
---
<--- SIP read from UDP:192.168.33.200:5060 --->
ACK sip:74952150687@192.168.5.253:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.33.200:5060;branch=z9hG4bK-71ba8e50
From: "297" <sip:297@192.168.5.253>;tag=ea48e7d0bf8d379bo0
To: <sip:74952150687@192.168.5.253>;tag=as49ceebd7
Call-ID: 1957afb4-53c87107@192.168.33.200
CSeq: 103 ACK
Max-Forwards: 70
Authorization: Digest username="297",realm="asterisk",nonce="6b8984f9",uri="sip:74952150687@192.168.5.253:5060",algorithm=MD5,response="b4fe3c2bf1bcc7f8a85e468d00202155"
Contact: "297" <sip:297@192.168.33.200:5060;ref=297>
User-Agent: Cisco/SPA112-1.3.2(014)
Content-Length: 0
<------------->
--- (11 headers 0 lines) ---
<--- SIP read from UDP:192.168.33.200:5060 --->
ACK sip:74952150687@192.168.5.253:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.33.200:5060;branch=z9hG4bK-71ba8e50
From: "297" <sip:297@192.168.5.253>;tag=ea48e7d0bf8d379bo0
To: <sip:74952150687@192.168.5.253>;tag=as49ceebd7
Call-ID: 1957afb4-53c87107@192.168.33.200
CSeq: 103 ACK
Max-Forwards: 70
Authorization: Digest username="297",realm="asterisk",nonce="6b8984f9",uri="sip:74952150687@192.168.5.253:5060",algorithm=MD5,response="b4fe3c2bf1bcc7f8a85e468d00202155"
Contact: "297" <sip:297@192.168.33.200:5060;ref=297>
User-Agent: Cisco/SPA112-1.3.2(014)
Content-Length: 0
<------------->
--- (11 headers 0 lines) ---
UDPTL (SIP/297-00000167): packet from 192.168.33.200:16426 (seq 0, len 6)
UDPTL (SIP/mtt-00000168): packet to 80.75.130.136:37756 (seq 0, len 6)
UDPTL (SIP/mtt-00000168): packet from 80.75.130.136:37756 (seq 0, len 6)
UDPTL (SIP/297-00000167): packet to 192.168.33.200:16426 (seq 0, len 6)
UDPTL (SIP/mtt-00000168): packet from 80.75.130.136:37756 (seq 0, len 6)
UDPTL (SIP/mtt-00000168): packet from 80.75.130.136:37756 (seq 0, len 6)
UDPTL (SIP/mtt-00000168): packet from 80.75.130.136:37756 (seq 0, len 6)
UDPTL (SIP/mtt-00000168): packet from 80.75.130.136:37756 (seq 0, len 6)
UDPTL (SIP/mtt-00000168): packet from 80.75.130.136:37756 (seq 0, len 6)
UDPTL (SIP/mtt-00000168): packet from 80.75.130.136:37756 (seq 0, len 6)
UDPTL (SIP/mtt-00000168): packet from 80.75.130.136:37756 (seq 0, len 6)
UDPTL (SIP/mtt-00000168): packet from 80.75.130.136:37756 (seq 1, len 8)
UDPTL (SIP/297-00000167): packet to 192.168.33.200:16426 (seq 1, len 8)
UDPTL (SIP/mtt-00000168): packet from 80.75.130.136:37756 (seq 1, len 8)
UDPTL (SIP/mtt-00000168): packet from 80.75.130.136:37756 (seq 1, len 8)
UDPTL (SIP/mtt-00000168): packet from 80.75.130.136:37756 (seq 1, len 8)
UDPTL (SIP/mtt-00000168): packet from 80.75.130.136:37756 (seq 1, len 8)
UDPTL (SIP/mtt-00000168): packet from 80.75.130.136:37756 (seq 1, len 8)
UDPTL (SIP/mtt-00000168): packet from 80.75.130.136:37756 (seq 1, len 8)
UDPTL (SIP/mtt-00000168): packet from 80.75.130.136:37756 (seq 1, len 8)
UDPTL (SIP/297-00000167): packet from 192.168.33.200:16426 (seq 1, len 8)
UDPTL (SIP/mtt-00000168): packet to 80.75.130.136:37756 (seq 1, len 8)
UDPTL (SIP/mtt-00000168): packet from 80.75.130.136:37756 (seq 2, len 15)
UDPTL (SIP/297-00000167): packet to 192.168.33.200:16426 (seq 2, len 13)
UDPTL (SIP/mtt-00000168): packet from 80.75.130.136:37756 (seq 3, len 22)
UDPTL (SIP/297-00000167): packet to 192.168.33.200:16426 (seq 3, len 18)
UDPTL (SIP/mtt-00000168): packet from 80.75.130.136:37756 (seq 4, len 29)
UDPTL (SIP/297-00000167): packet to 192.168.33.200:16426 (seq 4, len 18)
UDPTL (SIP/mtt-00000168): packet from 80.75.130.136:37756 (seq 5, len 36)
UDPTL (SIP/297-00000167): packet to 192.168.33.200:16426 (seq 5, len 18)
UDPTL (SIP/mtt-00000168): packet from 80.75.130.136:37756 (seq 6, len 43)
UDPTL (SIP/297-00000167): packet to 192.168.33.200:16426 (seq 6, len 18)
UDPTL (SIP/mtt-00000168): packet from 80.75.130.136:37756 (seq 7, len 50)
UDPTL (SIP/297-00000167): packet to 192.168.33.200:16426 (seq 7, len 18)
UDPTL (SIP/mtt-00000168): packet from 80.75.130.136:37756 (seq 8, len 55)
UDPTL (SIP/297-00000167): packet to 192.168.33.200:16426 (seq 8, len 18)
UDPTL (SIP/mtt-00000168): packet from 80.75.130.136:37756 (seq 9, len 60)
UDPTL (SIP/297-00000167): packet to 192.168.33.200:16426 (seq 9, len 18)
UDPTL (SIP/mtt-00000168): packet from 80.75.130.136:37756 (seq 10, len 60)
UDPTL (SIP/297-00000167): packet to 192.168.33.200:16426 (seq 10, len 18)
UDPTL (SIP/mtt-00000168): packet from 80.75.130.136:37756 (seq 11, len 60)
UDPTL (SIP/297-00000167): packet to 192.168.33.200:16426 (seq 11, len 18)
UDPTL (SIP/mtt-00000168): packet from 80.75.130.136:37756 (seq 12, len 60)
UDPTL (SIP/297-00000167): packet to 192.168.33.200:16426 (seq 12, len 18)
UDPTL (SIP/mtt-00000168): packet from 80.75.130.136:37756 (seq 13, len 60)
UDPTL (SIP/297-00000167): packet to 192.168.33.200:16426 (seq 13, len 18)
UDPTL (SIP/mtt-00000168): packet from 80.75.130.136:37756 (seq 14, len 60)
UDPTL (SIP/297-00000167): packet to 192.168.33.200:16426 (seq 14, len 18)
UDPTL (SIP/mtt-00000168): packet from 80.75.130.136:37756 (seq 15, len 60)
UDPTL (SIP/297-00000167): packet to 192.168.33.200:16426 (seq 15, len 18)
UDPTL (SIP/mtt-00000168): packet from 80.75.130.136:37756 (seq 16, len 60)
UDPTL (SIP/297-00000167): packet to 192.168.33.200:16426 (seq 16, len 18)
UDPTL (SIP/mtt-00000168): packet from 80.75.130.136:37756 (seq 17, len 60)
UDPTL (SIP/297-00000167): packet to 192.168.33.200:16426 (seq 17, len 18)
UDPTL (SIP/mtt-00000168): packet from 80.75.130.136:37756 (seq 18, len 60)
UDPTL (SIP/297-00000167): packet to 192.168.33.200:16426 (seq 18, len 18)
UDPTL (SIP/mtt-00000168): packet from 80.75.130.136:37756 (seq 19, len 60)
UDPTL (SIP/297-00000167): packet to 192.168.33.200:16426 (seq 19, len 18)
UDPTL (SIP/mtt-00000168): packet from 80.75.130.136:37756 (seq 20, len 60)
UDPTL (SIP/297-00000167): packet to 192.168.33.200:16426 (seq 20, len 18)
UDPTL (SIP/mtt-00000168): packet from 80.75.130.136:37756 (seq 21, len 60)
UDPTL (SIP/297-00000167): packet to 192.168.33.200:16426 (seq 21, len 18)
UDPTL (SIP/mtt-00000168): packet from 80.75.130.136:37756 (seq 22, len 60)
UDPTL (SIP/297-00000167): packet to 192.168.33.200:16426 (seq 22, len 18)
UDPTL (SIP/mtt-00000168): packet from 80.75.130.136:37756 (seq 23, len 60)
UDPTL (SIP/297-00000167): packet to 192.168.33.200:16426 (seq 23, len 18)
UDPTL (SIP/mtt-00000168): packet from 80.75.130.136:37756 (seq 24, len 60)
UDPTL (SIP/297-00000167): packet to 192.168.33.200:16426 (seq 24, len 18)
UDPTL (SIP/mtt-00000168): packet from 80.75.130.136:37756 (seq 25, len 57)
UDPTL (SIP/297-00000167): packet to 192.168.33.200:16426 (seq 25, len 15)
UDPTL (SIP/mtt-00000168): packet from 80.75.130.136:37756 (seq 25, len 57)
UDPTL (SIP/mtt-00000168): packet from 80.75.130.136:37756 (seq 25, len 57)
UDPTL (SIP/mtt-00000168): packet from 80.75.130.136:37756 (seq 25, len 57)
UDPTL (SIP/mtt-00000168): packet from 80.75.130.136:37756 (seq 25, len 57)
UDPTL (SIP/mtt-00000168): packet from 80.75.130.136:37756 (seq 25, len 57)
UDPTL (SIP/mtt-00000168): packet from 80.75.130.136:37756 (seq 25, len 57)
UDPTL (SIP/mtt-00000168): packet from 80.75.130.136:37756 (seq 25, len 57)
UDPTL (SIP/mtt-00000168): packet from 80.75.130.136:37756 (seq 26, len 57)
UDPTL (SIP/297-00000167): packet to 192.168.33.200:16426 (seq 26, len 15)
UDPTL (SIP/mtt-00000168): packet from 80.75.130.136:37756 (seq 27, len 57)
UDPTL (SIP/297-00000167): packet to 192.168.33.200:16426 (seq 27, len 18)
UDPTL (SIP/mtt-00000168): packet from 80.75.130.136:37756 (seq 28, len 57)
UDPTL (SIP/297-00000167): packet to 192.168.33.200:16426 (seq 28, len 18)
UDPTL (SIP/mtt-00000168): packet from 80.75.130.136:37756 (seq 29, len 57)
UDPTL (SIP/297-00000167): packet to 192.168.33.200:16426 (seq 29, len 18)
UDPTL (SIP/mtt-00000168): packet from 80.75.130.136:37756 (seq 30, len 57)
UDPTL (SIP/297-00000167): packet to 192.168.33.200:16426 (seq 30, len 18)
UDPTL (SIP/mtt-00000168): packet from 80.75.130.136:37756 (seq 31, len 57)
UDPTL (SIP/297-00000167): packet to 192.168.33.200:16426 (seq 31, len 18)
UDPTL (SIP/mtt-00000168): packet from 80.75.130.136:37756 (seq 32, len 57)
UDPTL (SIP/297-00000167): packet to 192.168.33.200:16426 (seq 32, len 18)
UDPTL (SIP/mtt-00000168): packet from 80.75.130.136:37756 (seq 33, len 57)
UDPTL (SIP/297-00000167): packet to 192.168.33.200:16426 (seq 33, len 15)
UDPTL (SIP/mtt-00000168): packet from 80.75.130.136:37756 (seq 33, len 57)
UDPTL (SIP/mtt-00000168): packet from 80.75.130.136:37756 (seq 33, len 57)
UDPTL (SIP/mtt-00000168): packet from 80.75.130.136:37756 (seq 33, len 57)
UDPTL (SIP/mtt-00000168): packet from 80.75.130.136:37756 (seq 33, len 57)
UDPTL (SIP/mtt-00000168): packet from 80.75.130.136:37756 (seq 33, len 57)
UDPTL (SIP/mtt-00000168): packet from 80.75.130.136:37756 (seq 33, len 57)
UDPTL (SIP/mtt-00000168): packet from 80.75.130.136:37756 (seq 33, len 57)
UDPTL (SIP/mtt-00000168): packet from 80.75.130.136:37756 (seq 34, len 54)
UDPTL (SIP/297-00000167): packet to 192.168.33.200:16426 (seq 34, len 12)
UDPTL (SIP/mtt-00000168): packet from 80.75.130.136:37756 (seq 34, len 54)
UDPTL (SIP/mtt-00000168): packet from 80.75.130.136:37756 (seq 34, len 54)
UDPTL (SIP/mtt-00000168): packet from 80.75.130.136:37756 (seq 34, len 54)
UDPTL (SIP/mtt-00000168): packet from 80.75.130.136:37756 (seq 34, len 54)
UDPTL (SIP/mtt-00000168): packet from 80.75.130.136:37756 (seq 35, len 49)
UDPTL (SIP/297-00000167): packet to 192.168.33.200:16426 (seq 35, len 10)
UDPTL (SIP/mtt-00000168): packet from 80.75.130.136:37756 (seq 35, len 49)
UDPTL (SIP/mtt-00000168): packet from 80.75.130.136:37756 (seq 35, len 49)
UDPTL (SIP/mtt-00000168): packet from 80.75.130.136:37756 (seq 35, len 49)
UDPTL (SIP/mtt-00000168): packet from 80.75.130.136:37756 (seq 34, len 54)
UDPTL (SIP/mtt-00000168): packet from 80.75.130.136:37756 (seq 35, len 49)
UDPTL (SIP/297-00000167): packet to 192.168.33.200:16426 (seq 36, len 8)
UDPTL (SIP/mtt-00000168): packet from 80.75.130.136:37756 (seq 34, len 54)
UDPTL (SIP/mtt-00000168): packet from 80.75.130.136:37756 (seq 34, len 54)
UDPTL (SIP/mtt-00000168): packet from 80.75.130.136:37756 (seq 35, len 49)
UDPTL (SIP/297-00000167): packet to 192.168.33.200:16426 (seq 37, len 8)
UDPTL (SIP/mtt-00000168): packet from 80.75.130.136:37756 (seq 35, len 49)
UDPTL (SIP/mtt-00000168): packet from 80.75.130.136:37756 (seq 35, len 49)
UDPTL (SIP/297-00000167): packet from 192.168.33.200:16426 (seq 2, len 10)
UDPTL (SIP/mtt-00000168): packet to 80.75.130.136:37756 (seq 2, len 10)
UDPTL (SIP/297-00000167): packet from 192.168.33.200:16426 (seq 3, len 10)
UDPTL (SIP/mtt-00000168): packet to 80.75.130.136:37756 (seq 3, len 12)
UDPTL (SIP/297-00000167): packet from 192.168.33.200:16426 (seq 3, len 10)
UDPTL (SIP/297-00000167): packet from 192.168.33.200:16426 (seq 3, len 10)
UDPTL (SIP/297-00000167): packet from 192.168.33.200:16426 (seq 4, len 38)
UDPTL (SIP/mtt-00000168): packet to 80.75.130.136:37756 (seq 4, len 40)
UDPTL (SIP/297-00000167): packet from 192.168.33.200:16426 (seq 5, len 49)
UDPTL (SIP/mtt-00000168): packet to 80.75.130.136:37756 (seq 5, len 51)
UDPTL (SIP/297-00000167): packet from 192.168.33.200:16426 (seq 6, len 49)
UDPTL (SIP/mtt-00000168): packet to 80.75.130.136:37756 (seq 6, len 51)
UDPTL (SIP/297-00000167): packet from 192.168.33.200:16426 (seq 7, len 21)
UDPTL (SIP/mtt-00000168): packet to 80.75.130.136:37756 (seq 7, len 51)
UDPTL (SIP/297-00000167): packet from 192.168.33.200:16426 (seq 8, len 10)
UDPTL (SIP/mtt-00000168): packet to 80.75.130.136:37756 (seq 8, len 23)
UDPTL (SIP/297-00000167): packet from 192.168.33.200:16426 (seq 9, len 60)
UDPTL (SIP/mtt-00000168): packet to 80.75.130.136:37756 (seq 9, len 64)
UDPTL (SIP/297-00000167): packet from 192.168.33.200:16426 (seq 10, len 112)
UDPTL (SIP/mtt-00000168): packet to 80.75.130.136:37756 (seq 10, len 116)
UDPTL (SIP/297-00000167): packet from 192.168.33.200:16426 (seq 11, len 112)
UDPTL (SIP/mtt-00000168): packet to 80.75.130.136:37756 (seq 11, len 168)
UDPTL (SIP/297-00000167): packet from 192.168.33.200:16426 (seq 12, len 112)
UDPTL (SIP/mtt-00000168): packet to 80.75.130.136:37756 (seq 12, len 220)
UDPTL (SIP/297-00000167): packet from 192.168.33.200:16426 (seq 13, len 112)
UDPTL (SIP/mtt-00000168): packet to 80.75.130.136:37756 (seq 13, len 220)
UDPTL (SIP/297-00000167): packet from 192.168.33.200:16426 (seq 14, len 112)
UDPTL (SIP/mtt-00000168): packet to 80.75.130.136:37756 (seq 14, len 220)
UDPTL (SIP/297-00000167): packet from 192.168.33.200:16426 (seq 15, len 112)
UDPTL (SIP/mtt-00000168): packet to 80.75.130.136:37756 (seq 15, len 220)
UDPTL (SIP/297-00000167): packet from 192.168.33.200:16426 (seq 16, len 112)
UDPTL (SIP/mtt-00000168): packet to 80.75.130.136:37756 (seq 16, len 220)
UDPTL (SIP/297-00000167): packet from 192.168.33.200:16426 (seq 17, len 112)
UDPTL (SIP/mtt-00000168): packet to 80.75.130.136:37756 (seq 17, len 220)
UDPTL (SIP/297-00000167): packet from 192.168.33.200:16426 (seq 18, len 112)
UDPTL (SIP/mtt-00000168): packet to 80.75.130.136:37756 (seq 18, len 220)
UDPTL (SIP/297-00000167): packet from 192.168.33.200:16426 (seq 19, len 112)
UDPTL (SIP/mtt-00000168): packet to 80.75.130.136:37756 (seq 19, len 220)
UDPTL (SIP/297-00000167): packet from 192.168.33.200:16426 (seq 20, len 112)
UDPTL (SIP/mtt-00000168): packet to 80.75.130.136:37756 (seq 20, len 220)
UDPTL (SIP/297-00000167): packet from 192.168.33.200:16426 (seq 21, len 112)
UDPTL (SIP/mtt-00000168): packet to 80.75.130.136:37756 (seq 21, len 220)
UDPTL (SIP/297-00000167): packet from 192.168.33.200:16426 (seq 22, len 112)
UDPTL (SIP/mtt-00000168): packet to 80.75.130.136:37756 (seq 22, len 220)
UDPTL (SIP/297-00000167): packet from 192.168.33.200:16426 (seq 23, len 112)
UDPTL (SIP/mtt-00000168): packet to 80.75.130.136:37756 (seq 23, len 220)
UDPTL (SIP/297-00000167): packet from 192.168.33.200:16426 (seq 24, len 112)
UDPTL (SIP/mtt-00000168): packet to 80.75.130.136:37756 (seq 24, len 220)
UDPTL (SIP/297-00000167): packet from 192.168.33.200:16426 (seq 25, len 112)
UDPTL (SIP/mtt-00000168): packet to 80.75.130.136:37756 (seq 25, len 220)
UDPTL (SIP/297-00000167): packet from 192.168.33.200:16426 (seq 26, len 112)
UDPTL (SIP/mtt-00000168): packet to 80.75.130.136:37756 (seq 26, len 220)
UDPTL (SIP/297-00000167): packet from 192.168.33.200:16426 (seq 27, len 112)
UDPTL (SIP/mtt-00000168): packet to 80.75.130.136:37756 (seq 27, len 220)
UDPTL (SIP/297-00000167): packet from 192.168.33.200:16426 (seq 28, len 112)
UDPTL (SIP/mtt-00000168): packet to 80.75.130.136:37756 (seq 28, len 220)
UDPTL (SIP/297-00000167): packet from 192.168.33.200:16426 (seq 29, len 112)
UDPTL (SIP/mtt-00000168): packet to 80.75.130.136:37756 (seq 29, len 220)
UDPTL (SIP/297-00000167): packet from 192.168.33.200:16426 (seq 30, len 112)
UDPTL (SIP/mtt-00000168): packet to 80.75.130.136:37756 (seq 30, len 220)
UDPTL (SIP/297-00000167): packet from 192.168.33.200:16426 (seq 31, len 112)
UDPTL (SIP/mtt-00000168): packet to 80.75.130.136:37756 (seq 31, len 220)
UDPTL (SIP/297-00000167): packet from 192.168.33.200:16426 (seq 32, len 112)
UDPTL (SIP/mtt-00000168): packet to 80.75.130.136:37756 (seq 32, len 220)
UDPTL (SIP/297-00000167): packet from 192.168.33.200:16426 (seq 33, len 112)
UDPTL (SIP/mtt-00000168): packet to 80.75.130.136:37756 (seq 33, len 220)
UDPTL (SIP/297-00000167): packet from 192.168.33.200:16426 (seq 34, len 112)
UDPTL (SIP/mtt-00000168): packet to 80.75.130.136:37756 (seq 34, len 220)
<--- SIP read from TCP:192.168.5.168:51018 --->
REGISTER sip:192.168.5.253 SIP/2.0
Via: SIP/2.0/TCP 192.168.5.168:51018;branch=z9hG4bK07ca719b
From: <sip:102@192.168.5.253>;tag=08cc683104ba2f8e0ca384a9-13b6ac29
To: <sip:102@192.168.5.253>
Call-ID: 08cc6831-04ba00e0-601cfbdf-74158f57@192.168.5.168
Max-Forwards: 70
Date: Mon, 03 Jun 2013 20:34:50 GMT
CSeq: 11941 REGISTER
User-Agent: Cisco-CP9971/9.1.1
Contact: <sip:102@192.168.5.168:51018;transport=tcp>;+sip.instance="<urn:uuid:00000000-0000-0000-0000-08cc683104ba>";+u.sip!devicename.ccm.cisco.com="SEP08CC683104BA";+u.sip!model.ccm.cisco.com="493"
Supported: replaces,join,sdp-anat,norefersub,extended-refer,X-cisco-callinfo,X-cisco-serviceuri,X-cisco-escapecodes,X-cisco-service-control,X-cisco-srtp-fallback,X-cisco-monrec,X-cisco-config,X-cisco-sis-5.1.0,X-cisco-xsi-8.0.1
Content-Length: 0
Expires: 60
<------------->
--- (13 headers 0 lines) ---
Sending to 192.168.5.168:51018 (NAT)
<--- Transmitting (NAT) to 192.168.5.168:51018 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/TCP 192.168.5.168:51018;branch=z9hG4bK07ca719b;received=192.168.5.168;rport=51018
From: <sip:102@192.168.5.253>;tag=08cc683104ba2f8e0ca384a9-13b6ac29
To: <sip:102@192.168.5.253>;tag=as42a851c9
Call-ID: 08cc6831-04ba00e0-601cfbdf-74158f57@192.168.5.168
CSeq: 11941 REGISTER
Server: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5e1a2f6f"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '08cc6831-04ba00e0-601cfbdf-74158f57@192.168.5.168' in 32000 ms (Method: REGISTER)
<--- SIP read from TCP:192.168.5.168:51018 --->
REGISTER sip:192.168.5.253 SIP/2.0
Via: SIP/2.0/TCP 192.168.5.168:51018;branch=z9hG4bK55e29837
From: <sip:102@192.168.5.253>;tag=08cc683104ba2f8e0ca384a9-13b6ac29
To: <sip:102@192.168.5.253>
Call-ID: 08cc6831-04ba00e0-601cfbdf-74158f57@192.168.5.168
Max-Forwards: 70
Date: Mon, 03 Jun 2013 20:34:50 GMT
CSeq: 11942 REGISTER
User-Agent: Cisco-CP9971/9.1.1
Contact: <sip:102@192.168.5.168:51018;transport=tcp>;+sip.instance="<urn:uuid:00000000-0000-0000-0000-08cc683104ba>";+u.sip!devicename.ccm.cisco.com="SEP08CC683104BA";+u.sip!model.ccm.cisco.com="493"
Authorization: Digest username="102",realm="asterisk",uri="sip:192.168.5.253",response="673b0e250be7e7ab817713122f16dae4",nonce="5e1a2f6f",algorithm=MD5
Supported: replaces,join,sdp-anat,norefersub,extended-refer,X-cisco-callinfo,X-cisco-serviceuri,X-cisco-escapecodes,X-cisco-service-control,X-cisco-srtp-fallback,X-cisco-monrec,X-cisco-config,X-cisco-sis-5.1.0,X-cisco-xsi-8.0.1
Content-Length: 0
Expires: 60
<------------->
--- (14 headers 0 lines) ---
Sending to 192.168.5.168:51018 (NAT)
Reliably Transmitting (NAT) to 192.168.5.168:51018:
OPTIONS sip:102@192.168.5.168:51018;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 192.168.5.253:5060;branch=z9hG4bK316ee4b9;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@192.168.5.253>;tag=as642e14eb
To: <sip:102@192.168.5.168:51018;transport=tcp>
Contact: <sip:asterisk@192.168.5.253:5060;transport=TCP>
Call-ID: 3670ee847af4644e27d6a5aa4970c27b@192.168.5.253:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
Date: Mon, 03 Jun 2013 20:34:51 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0
---
<--- Transmitting (NAT) to 192.168.5.168:51018 --->
SIP/2.0 200 OK
Via: SIP/2.0/TCP 192.168.5.168:51018;branch=z9hG4bK55e29837;received=192.168.5.168;rport=51018
From: <sip:102@192.168.5.253>;tag=08cc683104ba2f8e0ca384a9-13b6ac29
To: <sip:102@192.168.5.253>;tag=as42a851c9
Call-ID: 08cc6831-04ba00e0-601cfbdf-74158f57@192.168.5.168
CSeq: 11942 REGISTER
Server: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 60
Contact: <sip:102@192.168.5.168:51018;transport=tcp>;expires=60
Date: Mon, 03 Jun 2013 20:34:51 GMT
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '08cc6831-04ba00e0-601cfbdf-74158f57@192.168.5.168' in 32000 ms (Method: REGISTER)
<--- SIP read from TCP:192.168.5.168:51018 --->
SIP/2.0 200 OK
Via: SIP/2.0/TCP 192.168.5.253:5060;branch=z9hG4bK316ee4b9;rport
From: "asterisk" <sip:asterisk@192.168.5.253>;tag=as642e14eb
To: <sip:102@192.168.5.168:51018;transport=tcp>;tag=08cc683104ba2f8f15b18b9b-68d9b6c9
Call-ID: 3670ee847af4644e27d6a5aa4970c27b@192.168.5.253:5060
Date: Mon, 03 Jun 2013 20:34:50 GMT
CSeq: 102 OPTIONS
Server: Cisco-CP9971/9.1.1
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE
Allow-Events: kpml,dialog,refer
Accept: application/sdp,multipart/mixed,multipart/alternative
Accept-Encoding: identity
Accept-Language: en
Supported: replaces,join,sdp-anat,norefersub,extended-refer,X-cisco-callinfo,X-cisco-serviceuri,X-cisco-escapecodes,X-cisco-service-control,X-cisco-srtp-fallback,X-cisco-monrec,X-cisco-config,X-cisco-sis-5.1.0
Content-Length: 487
Content-Type: application/sdp
Content-Disposition: session;handling=optional
v=0
o=Cisco-SIPUA 26891 0 IN IP4 192.168.5.168
s=SIP Call
t=0 0
m=audio 0 RTP/AVP 0 8 18 102 116 124 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:102 L16/16000
a=rtpmap:116 iLBC/8000
a=fmtp:116 mode=20
a=rtpmap:124 ISAC/16000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
m=video 0 RTP/AVP 97
b=TIAS:1000000
a=rtpmap:97 H264/90000
a=fmtp:97 profile-level-id=42801E;packetization-mode=0;level-asymmetry-allowed=1
<------------->
--- (17 headers 19 lines) ---
UDPTL (SIP/297-00000167): packet from 192.168.33.200:16426 (seq 35, len 112)
UDPTL (SIP/mtt-00000168): packet to 80.75.130.136:37756 (seq 35, len 220)
UDPTL (SIP/297-00000167): packet from 192.168.33.200:16426 (seq 36, len 112)
UDPTL (SIP/mtt-00000168): packet to 80.75.130.136:37756 (seq 36, len 220)
UDPTL (SIP/297-00000167): packet from 192.168.33.200:16426 (seq 37, len 112)
UDPTL (SIP/mtt-00000168): packet to 80.75.130.136:37756 (seq 37, len 220)
UDPTL (SIP/297-00000167): packet from 192.168.33.200:16426 (seq 38, len 112)
UDPTL (SIP/mtt-00000168): packet to 80.75.130.136:37756 (seq 38, len 220)
UDPTL (SIP/297-00000167): packet from 192.168.33.200:16426 (seq 39, len 112)
UDPTL (SIP/mtt-00000168): packet to 80.75.130.136:37756 (seq 39, len 220)
Really destroying SIP dialog '3670ee847af4644e27d6a5aa4970c27b@192.168.5.253:5060' Method: OPTIONS
UDPTL (SIP/297-00000167): packet from 192.168.33.200:16426 (seq 40, len 112)
UDPTL (SIP/mtt-00000168): packet to 80.75.130.136:37756 (seq 40, len 220)
UDPTL (SIP/297-00000167): packet from 192.168.33.200:16426 (seq 41, len 112)
UDPTL (SIP/mtt-00000168): packet to 80.75.130.136:37756 (seq 41, len 220)
UDPTL (SIP/297-00000167): packet from 192.168.33.200:16426 (seq 42, len 112)
UDPTL (SIP/mtt-00000168): packet to 80.75.130.136:37756 (seq 42, len 220)
UDPTL (SIP/297-00000167): packet from 192.168.33.200:16426 (seq 43, len 112)
UDPTL (SIP/mtt-00000168): packet to 80.75.130.136:37756 (seq 43, len 220)
UDPTL (SIP/297-00000167): packet from 192.168.33.200:16426 (seq 44, len 112)
UDPTL (SIP/mtt-00000168): packet to 80.75.130.136:37756 (seq 44, len 220)
UDPTL (SIP/297-00000167): packet from 192.168.33.200:16426 (seq 45, len 112)
UDPTL (SIP/mtt-00000168): packet to 80.75.130.136:37756 (seq 45, len 220)
UDPTL (SIP/297-00000167): packet from 192.168.33.200:16426 (seq 46, len 109)
UDPTL (SIP/mtt-00000168): packet to 80.75.130.136:37756 (seq 46, len 217)
UDPTL (SIP/297-00000167): packet from 192.168.33.200:16426 (seq 47, len 57)
UDPTL (SIP/mtt-00000168): packet to 80.75.130.136:37756 (seq 47, len 165)
UDPTL (SIP/297-00000167): packet from 192.168.33.200:16426 (seq 48, len 8)
UDPTL (SIP/mtt-00000168): packet to 80.75.130.136:37756 (seq 48, len 113)
<--- SIP read from UDP:192.168.33.200:5060 --->
REGISTER sip:192.168.5.253 SIP/2.0
Via: SIP/2.0/UDP 192.168.33.200:5060;branch=z9hG4bK-1123c91b
From: "297" <sip:297@192.168.5.253>;tag=5ea8e566fa4825d1o0
To: "297" <sip:297@192.168.5.253>
Call-ID: 11d50af5-5adc2460@192.168.33.200
CSeq: 25980 REGISTER
Max-Forwards: 70
Authorization: Digest username="297",realm="asterisk",nonce="00ee470a",uri="sip:192.168.5.253",algorithm=MD5,response="1747f302a3ddeee393e4ddd36082c55c"
Contact: "297" <sip:297@192.168.33.200:5060>;expires=60
User-Agent: Cisco/SPA112-1.3.2(014)
P-Station-Name: ;mac=887556043ebe; display=""; sn=CCQ16320NVT
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: replaces
<------------->
--- (14 headers 0 lines) ---
Sending to 192.168.33.200:5060 (NAT)
<--- Transmitting (NAT) to 192.168.33.200:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.33.200:5060;branch=z9hG4bK-1123c91b;received=192.168.33.200;rport=5060
From: "297" <sip:297@192.168.5.253>;tag=5ea8e566fa4825d1o0
To: "297" <sip:297@192.168.5.253>;tag=as439dc200
Call-ID: 11d50af5-5adc2460@192.168.33.200
CSeq: 25980 REGISTER
Server: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="33ef6720"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '11d50af5-5adc2460@192.168.33.200' in 32000 ms (Method: REGISTER)
<--- SIP read from UDP:192.168.33.200:5060 --->
REGISTER sip:192.168.5.253 SIP/2.0
Via: SIP/2.0/UDP 192.168.33.200:5060;branch=z9hG4bK-3f0bb4a4
From: "297" <sip:297@192.168.5.253>;tag=5ea8e566fa4825d1o0
To: "297" <sip:297@192.168.5.253>
Call-ID: 11d50af5-5adc2460@192.168.33.200
CSeq: 25981 REGISTER
Max-Forwards: 70
Authorization: Digest username="297",realm="asterisk",nonce="33ef6720",uri="sip:192.168.5.253",algorithm=MD5,response="1144f44eae93dcf76fbb0e44e408dc56"
Contact: "297" <sip:297@192.168.33.200:5060>;expires=60
User-Agent: Cisco/SPA112-1.3.2(014)
P-Station-Name: ;mac=887556043ebe; display=""; sn=CCQ16320NVT
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: replaces
<------------->
--- (14 headers 0 lines) ---
Sending to 192.168.33.200:5060 (NAT)
Reliably Transmitting (NAT) to 192.168.33.200:5060:
OPTIONS sip:297@192.168.33.200:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.5.253:5060;branch=z9hG4bK665fa6dc;rport
Max-Forwards: 70
From: "asterisk" <sip:asterisk@192.168.5.253>;tag=as2315a5f1
To: <sip:297@192.168.33.200:5060>
Contact: <sip:asterisk@192.168.5.253:5060>
Call-ID: 4f9ea60d559b24fe20cee0f2496b2ab7@192.168.5.253:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
Date: Mon, 03 Jun 2013 20:34:53 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0
---
<--- Transmitting (NAT) to 192.168.33.200:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.33.200:5060;branch=z9hG4bK-3f0bb4a4;received=192.168.33.200;rport=5060
From: "297" <sip:297@192.168.5.253>;tag=5ea8e566fa4825d1o0
To: "297" <sip:297@192.168.5.253>;tag=as439dc200
Call-ID: 11d50af5-5adc2460@192.168.33.200
CSeq: 25981 REGISTER
Server: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 60
Contact: <sip:297@192.168.33.200:5060>;expires=60
Date: Mon, 03 Jun 2013 20:34:53 GMT
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog '11d50af5-5adc2460@192.168.33.200' in 32000 ms (Method: REGISTER)
<--- SIP read from UDP:192.168.33.200:5060 --->
SIP/2.0 200 OK
To: <sip:297@192.168.33.200:5060>;tag=6e94e8cabd1b59bdi0
From: "asterisk" <sip:asterisk@192.168.5.253>;tag=as2315a5f1
Call-ID: 4f9ea60d559b24fe20cee0f2496b2ab7@192.168.5.253:5060
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP 192.168.5.253:5060;branch=z9hG4bK665fa6dc
Server: Cisco/SPA112-1.3.2(014)
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: replaces
<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '4f9ea60d559b24fe20cee0f2496b2ab7@192.168.5.253:5060' Method: OPTIONS
<--- SIP read from UDP:80.75.130.136:5060 --->
INVITE sip:74957777777@82.200.7.184:5060 SIP/2.0
Via: SIP/2.0/UDP 80.75.130.136:5060;branch=z9hG4bKi2a81n307ooh8to310k1sb00000k1.1
From: <sip:74952150687@80.75.130.136>;tag=1249863938-3792801228-380372658-4943405
To: <sip:74957777777@82.200.7.184;realip=82.200.7.184>;tag=as0e2e11a8
Call-ID: 55c3594f5206005e3196bea907e88021@82.200.7.184:5060
CSeq: 104 INVITE
Contact: <sip:74952150687@80.75.130.136:5060;transport=udp>
Content-Type: application/sdp
Allow: ACK, BYE, CANCEL, INFO, INVITE, OPTIONS, REFER, REGISTER, SUBSCRIBE, UPDATE
Max-Forwards: 69
User-Agent: TS-v4.5.1-08b
Content-Length: 243
v=0
o=- 1370291703 1370291703 IN IP4 80.75.130.136
s=-
c=IN IP4 80.75.130.136
t=0 0
m=audio 37756 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
a=silenceSupp:off - - - -
<------------->
--- (12 headers 12 lines) ---
Sending to 80.75.130.136:5060 (NAT)
Found RTP audio format 8
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=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 80.75.130.136:37756
<--- Transmitting (NAT) to 80.75.130.136:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 80.75.130.136:5060;branch=z9hG4bKi2a81n307ooh8to310k1sb00000k1.1;received=80.75.130.136;rport=5060
From: <sip:74952150687@80.75.130.136>;tag=1249863938-3792801228-380372658-4943405
To: <sip:74957777777@82.200.7.184;realip=82.200.7.184>;tag=as0e2e11a8
Call-ID: 55c3594f5206005e3196bea907e88021@82.200.7.184:5060
CSeq: 104 INVITE
Server: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:74957777777@82.200.7.184:5060>
Content-Length: 0
<------------>
Audio is at 18350
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
<--- Reliably Transmitting (NAT) to 80.75.130.136:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 80.75.130.136:5060;branch=z9hG4bKi2a81n307ooh8to310k1sb00000k1.1;received=80.75.130.136;rport=5060
From: <sip:74952150687@80.75.130.136>;tag=1249863938-3792801228-380372658-4943405
To: <sip:74957777777@82.200.7.184;realip=82.200.7.184>;tag=as0e2e11a8
Call-ID: 55c3594f5206005e3196bea907e88021@82.200.7.184:5060
CSeq: 104 INVITE
Server: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:74957777777@82.200.7.184:5060>
Content-Type: application/sdp
Content-Length: 251
v=0
o=root 1256665056 1256665058 IN IP4 82.200.7.184
s=Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
c=IN IP4 82.200.7.184
t=0 0
m=audio 18350 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------>
set_destination: Parsing <sip:297@192.168.33.200:5060;ref=297> for address/port to send to
set_destination: set destination to 192.168.33.200:5060
Audio is at 13056
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 192.168.33.200:5060:
INVITE sip:297@192.168.33.200:5060;ref=297 SIP/2.0
Via: SIP/2.0/UDP 192.168.5.253:5060;branch=z9hG4bK7a476057;rport
Max-Forwards: 70
From: <sip:74952150687@192.168.5.253>;tag=as49ceebd7
To: "297" <sip:297@192.168.5.253>;tag=ea48e7d0bf8d379bo0
Contact: <sip:74952150687@192.168.5.253:5060>
Call-ID: 1957afb4-53c87107@192.168.33.200
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 251
v=0
o=root 462103832 462103835 IN IP4 192.168.5.253
s=Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
c=IN IP4 192.168.5.253
t=0 0
m=audio 13056 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
---
<--- SIP read from UDP:80.75.130.136:5060 --->
ACK sip:74957777777@82.200.7.184:5060 SIP/2.0
Via: SIP/2.0/UDP 80.75.130.136:5060;branch=z9hG4bKpldr4e00b04ggsgso0g0.1
From: <sip:74952150687@80.75.130.136>;tag=1249863938-3792801228-380372658-4943405
To: <sip:74957777777@82.200.7.184;realip=82.200.7.184>;tag=as0e2e11a8
Call-ID: 55c3594f5206005e3196bea907e88021@82.200.7.184:5060
CSeq: 104 ACK
Contact: <sip:74952150687@80.75.130.136:5060;transport=udp>
Max-Forwards: 69
User-Agent: TS-v4.5.1-08b
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
Retransmitting #1 (NAT) to 192.168.33.200:5060:
INVITE sip:297@192.168.33.200:5060;ref=297 SIP/2.0
Via: SIP/2.0/UDP 192.168.5.253:5060;branch=z9hG4bK7a476057;rport
Max-Forwards: 70
From: <sip:74952150687@192.168.5.253>;tag=as49ceebd7
To: "297" <sip:297@192.168.5.253>;tag=ea48e7d0bf8d379bo0
Contact: <sip:74952150687@192.168.5.253:5060>
Call-ID: 1957afb4-53c87107@192.168.33.200
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 251
v=0
o=root 462103832 462103835 IN IP4 192.168.5.253
s=Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
c=IN IP4 192.168.5.253
t=0 0
m=audio 13056 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
---
<--- SIP read from UDP:192.168.33.200:5060 --->
SIP/2.0 200 OK
To: "297" <sip:297@192.168.5.253>;tag=ea48e7d0bf8d379bo0
From: <sip:74952150687@192.168.5.253>;tag=as49ceebd7
Call-ID: 1957afb4-53c87107@192.168.33.200
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.5.253:5060;branch=z9hG4bK7a476057
Contact: "297" <sip:297@192.168.33.200:5060>
Server: Cisco/SPA112-1.3.2(014)
Content-Length: 259
Content-Type: application/sdp
v=0
o=- 2640556 2640558 IN IP4 192.168.33.200
s=-
c=IN IP4 192.168.33.200
t=0 0
m=audio 16426 RTP/AVP 8 100 101
a=rtpmap:8 PCMA/8000
a=rtpmap:100 NSE/8000
a=fmtp:100 192-193
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
<------------->
--- (10 headers 13 lines) ---
Found RTP audio format 8
Found RTP audio format 100
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found unknown media description format NSE for ID 100
Found audio description format telephone-event for ID 101
Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=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 192.168.33.200:16426
set_destination: Parsing <sip:297@192.168.33.200:5060;ref=297> for address/port to send to
set_destination: set destination to 192.168.33.200:5060
Transmitting (NAT) to 192.168.33.200:5060:
ACK sip:297@192.168.33.200:5060;ref=297 SIP/2.0
Via: SIP/2.0/UDP 192.168.5.253:5060;branch=z9hG4bK101e4896;rport
Max-Forwards: 70
From: <sip:74952150687@192.168.5.253>;tag=as49ceebd7
To: "297" <sip:297@192.168.5.253>;tag=ea48e7d0bf8d379bo0
Contact: <sip:74952150687@192.168.5.253:5060>
Call-ID: 1957afb4-53c87107@192.168.33.200
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
Content-Length: 0
---
<--- SIP read from UDP:192.168.33.200:5060 --->
SIP/2.0 200 OK
To: "297" <sip:297@192.168.5.253>;tag=ea48e7d0bf8d379bo0
From: <sip:74952150687@192.168.5.253>;tag=as49ceebd7
Call-ID: 1957afb4-53c87107@192.168.33.200
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.5.253:5060;branch=z9hG4bK7a476057
Contact: "297" <sip:297@192.168.33.200:5060>
Server: Cisco/SPA112-1.3.2(014)
Content-Length: 259
Content-Type: application/sdp
v=0
o=- 2640556 2640558 IN IP4 192.168.33.200
s=-
c=IN IP4 192.168.33.200
t=0 0
m=audio 16426 RTP/AVP 8 100 101
a=rtpmap:8 PCMA/8000
a=rtpmap:100 NSE/8000
a=fmtp:100 192-193
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
<------------->
--- (10 headers 13 lines) ---
set_destination: Parsing <sip:297@192.168.33.200:5060;ref=297> for address/port to send to
set_destination: set destination to 192.168.33.200:5060
Transmitting (NAT) to 192.168.33.200:5060:
ACK sip:297@192.168.33.200:5060;ref=297 SIP/2.0
Via: SIP/2.0/UDP 192.168.5.253:5060;branch=z9hG4bK17bbd9ba;rport
Max-Forwards: 70
From: <sip:74952150687@192.168.5.253>;tag=as49ceebd7
To: "297" <sip:297@192.168.5.253>;tag=ea48e7d0bf8d379bo0
Contact: <sip:74952150687@192.168.5.253:5060>
Call-ID: 1957afb4-53c87107@192.168.33.200
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
Content-Length: 0
---
<--- SIP read from UDP:192.168.33.200:5060 --->
INVITE sip:74952150687@192.168.5.253:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.33.200:5060;branch=z9hG4bK-68e2bbc2
From: "297" <sip:297@192.168.5.253>;tag=ea48e7d0bf8d379bo0
To: <sip:74952150687@192.168.5.253>;tag=as49ceebd7
Remote-Party-ID: "297" <sip:297@192.168.5.253>;screen=yes;party=calling
Call-ID: 1957afb4-53c87107@192.168.33.200
CSeq: 104 INVITE
Max-Forwards: 70
Authorization: Digest username="297",realm="asterisk",nonce="6b8984f9",uri="sip:74952150687@192.168.5.253:5060",algorithm=MD5,response="b4fe3c2bf1bcc7f8a85e468d00202155"
Contact: "297" <sip:297@192.168.33.200:5060;ref=297>
Expires: 30
User-Agent: Cisco/SPA112-1.3.2(014)
Content-Length: 271
Content-Type: application/sdp
v=0
o=- 2640556 2640559 IN IP4 192.168.33.200
s=-
c=IN IP4 192.168.33.200
t=0 0
m=image 16426 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxBuffer:200
a=T38FaxMaxDatagram:200
a=T38FaxUdpEC:t38UDPRedundancy
<------------->
--- (14 headers 12 lines) ---
Sending to 192.168.33.200:5060 (NAT)
Got T.38 offer in SDP in dialog 1957afb4-53c87107@192.168.33.200
Capabilities: us - 0x8 (alaw), peer - audio=0x0 (nothing)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x0 (nothing)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x0 (nothing), combined - 0x0 (nothing)
Got T.38 Re-invite without audio. Keeping RTP active during T.38 session.
<--- Transmitting (NAT) to 192.168.33.200:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.33.200:5060;branch=z9hG4bK-68e2bbc2;received=192.168.33.200;rport=5060
From: "297" <sip:297@192.168.5.253>;tag=ea48e7d0bf8d379bo0
To: <sip:74952150687@192.168.5.253>;tag=as49ceebd7
Call-ID: 1957afb4-53c87107@192.168.33.200
CSeq: 104 INVITE
Server: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:74952150687@192.168.5.253:5060>
Content-Length: 0
<------------>
set_destination: Parsing <sip:74952150687@80.75.130.136:5060;transport=udp> for address/port to send to
set_destination: set destination to 80.75.130.136:5060
Reliably Transmitting (NAT) to 80.75.130.136:5060:
INVITE sip:74952150687@80.75.130.136:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 82.200.7.184:5060;branch=z9hG4bK1fb000e7;rport
Max-Forwards: 70
From: <sip:74957777777@82.200.7.184;realip=82.200.7.184>;tag=as0e2e11a8
To: <sip:74952150687@80.75.130.136>;tag=1249863938-3792801228-380372658-4943405
Contact: <sip:74957777777@82.200.7.184:5060>
Call-ID: 55c3594f5206005e3196bea907e88021@82.200.7.184:5060
CSeq: 104 INVITE
User-Agent: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 286
v=0
o=root 1256665056 1256665059 IN IP4 82.200.7.184
s=Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
c=IN IP4 82.200.7.184
t=0 0
m=image 4022 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxDatagram:389
a=T38FaxUdpEC:t38UDPRedundancy
---
Retransmitting #1 (NAT) to 80.75.130.136:5060:
INVITE sip:74952150687@80.75.130.136:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 82.200.7.184:5060;branch=z9hG4bK1fb000e7;rport
Max-Forwards: 70
From: <sip:74957777777@82.200.7.184;realip=82.200.7.184>;tag=as0e2e11a8
To: <sip:74952150687@80.75.130.136>;tag=1249863938-3792801228-380372658-4943405
Contact: <sip:74957777777@82.200.7.184:5060>
Call-ID: 55c3594f5206005e3196bea907e88021@82.200.7.184:5060
CSeq: 104 INVITE
User-Agent: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 286
v=0
o=root 1256665056 1256665059 IN IP4 82.200.7.184
s=Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
c=IN IP4 82.200.7.184
t=0 0
m=image 4022 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:14400
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxDatagram:389
a=T38FaxUdpEC:t38UDPRedundancy
---
<--- SIP read from UDP:80.75.130.136:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 82.200.7.184:5060;received=82.200.7.184;branch=z9hG4bK1fb000e7;rport=5060
From: <sip:74957777777@82.200.7.184;realip=82.200.7.184>;tag=as0e2e11a8
To: <sip:74952150687@80.75.130.136>;tag=1249863938-3792801228-380372658-4943405
Call-ID: 55c3594f5206005e3196bea907e88021@82.200.7.184:5060
CSeq: 104 INVITE
Contact: <sip:74952150687@80.75.130.136:5060;transport=udp>
Content-Type: application/sdp
Allow: ACK, BYE, CANCEL, INFO, INVITE, OPTIONS, REFER, REGISTER, SUBSCRIBE, UPDATE
Server: TS-v4.5.1-08b
Content-Length: 197
v=0
o=- 1370291705 1370291705 IN IP4 80.75.130.136
s=-
c=IN IP4 80.75.130.136
t=0 0
m=image 37756 udptl t38
a=T38FaxRateManagement:transferredTCF
a=T38FaxUdpEC:t38UDPRedundancy
a=sendrecv
<------------->
--- (11 headers 9 lines) ---
Got T.38 offer in SDP in dialog 55c3594f5206005e3196bea907e88021@82.200.7.184:5060
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x0 (nothing)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x0 (nothing)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x0 (nothing), combined - 0x0 (nothing)
Got T.38 Re-invite without audio. Keeping RTP active during T.38 session.
set_destination: Parsing <sip:74952150687@80.75.130.136:5060;transport=udp> for address/port to send to
set_destination: set destination to 80.75.130.136:5060
Transmitting (NAT) to 80.75.130.136:5060:
ACK sip:74952150687@80.75.130.136:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 82.200.7.184:5060;branch=z9hG4bK78cbf78a;rport
Max-Forwards: 70
From: <sip:74957777777@82.200.7.184;realip=82.200.7.184>;tag=as0e2e11a8
To: <sip:74952150687@80.75.130.136>;tag=1249863938-3792801228-380372658-4943405
Contact: <sip:74957777777@82.200.7.184:5060>
Call-ID: 55c3594f5206005e3196bea907e88021@82.200.7.184:5060
CSeq: 104 ACK
User-Agent: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
Content-Length: 0
---
<--- Reliably Transmitting (NAT) to 192.168.33.200:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.33.200:5060;branch=z9hG4bK-68e2bbc2;received=192.168.33.200;rport=5060
From: "297" <sip:297@192.168.5.253>;tag=ea48e7d0bf8d379bo0
To: <sip:74952150687@192.168.5.253>;tag=as49ceebd7
Call-ID: 1957afb4-53c87107@192.168.33.200
CSeq: 104 INVITE
Server: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:74952150687@192.168.5.253:5060>
Content-Type: application/sdp
Content-Length: 285
v=0
o=root 462103832 462103836 IN IP4 192.168.5.253
s=Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
c=IN IP4 192.168.5.253
t=0 0
m=image 4884 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:2400
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxDatagram:397
a=T38FaxUdpEC:t38UDPRedundancy
<------------>
Retransmitting #1 (NAT) to 192.168.33.200:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.33.200:5060;branch=z9hG4bK-68e2bbc2;received=192.168.33.200;rport=5060
From: "297" <sip:297@192.168.5.253>;tag=ea48e7d0bf8d379bo0
To: <sip:74952150687@192.168.5.253>;tag=as49ceebd7
Call-ID: 1957afb4-53c87107@192.168.33.200
CSeq: 104 INVITE
Server: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:74952150687@192.168.5.253:5060>
Content-Type: application/sdp
Content-Length: 285
v=0
o=root 462103832 462103836 IN IP4 192.168.5.253
s=Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
c=IN IP4 192.168.5.253
t=0 0
m=image 4884 udptl t38
a=T38FaxVersion:0
a=T38MaxBitRate:2400
a=T38FaxRateManagement:transferredTCF
a=T38FaxMaxDatagram:397
a=T38FaxUdpEC:t38UDPRedundancy
---
<--- SIP read from UDP:192.168.33.200:5060 --->
ACK sip:74952150687@192.168.5.253:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.33.200:5060;branch=z9hG4bK-103b2f4b
From: "297" <sip:297@192.168.5.253>;tag=ea48e7d0bf8d379bo0
To: <sip:74952150687@192.168.5.253>;tag=as49ceebd7
Call-ID: 1957afb4-53c87107@192.168.33.200
CSeq: 104 ACK
Max-Forwards: 70
Authorization: Digest username="297",realm="asterisk",nonce="6b8984f9",uri="sip:74952150687@192.168.5.253:5060",algorithm=MD5,response="b4fe3c2bf1bcc7f8a85e468d00202155"
Contact: "297" <sip:297@192.168.33.200:5060;ref=297>
User-Agent: Cisco/SPA112-1.3.2(014)
Content-Length: 0
<------------->
--- (11 headers 0 lines) ---
<--- SIP read from UDP:192.168.33.200:5060 --->
ACK sip:74952150687@192.168.5.253:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.33.200:5060;branch=z9hG4bK-103b2f4b
From: "297" <sip:297@192.168.5.253>;tag=ea48e7d0bf8d379bo0
To: <sip:74952150687@192.168.5.253>;tag=as49ceebd7
Call-ID: 1957afb4-53c87107@192.168.33.200
CSeq: 104 ACK
Max-Forwards: 70
Authorization: Digest username="297",realm="asterisk",nonce="6b8984f9",uri="sip:74952150687@192.168.5.253:5060",algorithm=MD5,response="b4fe3c2bf1bcc7f8a85e468d00202155"
Contact: "297" <sip:297@192.168.33.200:5060;ref=297>
User-Agent: Cisco/SPA112-1.3.2(014)
Content-Length: 0
<------------->
--- (11 headers 0 lines) ---
UDPTL (SIP/297-00000167): packet from 192.168.33.200:16426 (seq 0, len 6)
Reliably Transmitting (NAT) to 80.75.130.136:5060:
OPTIONS sip:80.75.130.136 SIP/2.0
Via: SIP/2.0/UDP 82.200.7.184:5060;branch=z9hG4bK65c9c6ea;rport
Max-Forwards: 70
From: "asterisk" <sip:74957777777@82.200.7.184>;tag=as7305f41b
To: <sip:80.75.130.136>
Contact: <sip:74957777777@82.200.7.184:5060>
Call-ID: 45020b150651aa1e10ba589c153dda37@82.200.7.184:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
Date: Mon, 03 Jun 2013 20:35:10 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0
---
<--- SIP read from UDP:80.75.130.136:5060 --->
SIP/2.0 480 No Routes Found
Via: SIP/2.0/UDP 82.200.7.184:5060;received=82.200.7.184;branch=z9hG4bK65c9c6ea;rport=5060
From: "asterisk" <sip:74957777777@82.200.7.184>;tag=as7305f41b
To: <sip:80.75.130.136>;tag=aprqngfrt-0mov5c30000c6
Call-ID: 45020b150651aa1e10ba589c153dda37@82.200.7.184:5060
CSeq: 102 OPTIONS
<------------->
--- (6 headers 0 lines) ---
Really destroying SIP dialog '45020b150651aa1e10ba589c153dda37@82.200.7.184:5060' Method: OPTIONS
UDPTL (SIP/mtt-00000168): packet from 80.75.130.136:37756 (seq 0, len 6)
UDPTL (SIP/mtt-00000168): packet from 80.75.130.136:37756 (seq 0, len 6)
UDPTL (SIP/mtt-00000168): packet from 80.75.130.136:37756 (seq 0, len 6)
UDPTL (SIP/mtt-00000168): packet from 80.75.130.136:37756 (seq 0, len 6)
UDPTL (SIP/mtt-00000168): packet from 80.75.130.136:37756 (seq 0, len 6)
UDPTL (SIP/mtt-00000168): packet from 80.75.130.136:37756 (seq 0, len 6)
UDPTL (SIP/mtt-00000168): packet from 80.75.130.136:37756 (seq 0, len 6)
UDPTL (SIP/mtt-00000168): packet from 80.75.130.136:37756 (seq 0, len 6)
UDPTL (SIP/mtt-00000168): packet from 80.75.130.136:37756 (seq 1, len 13)
UDPTL (SIP/297-00000167): packet to 192.168.33.200:16426 (seq 38, len 13)
<--- SIP read from UDP:192.168.33.200:5060 --->
BYE sip:74952150687@192.168.5.253:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.33.200:5060;branch=z9hG4bK-5cd9ac81
From: "297" <sip:297@192.168.5.253>;tag=ea48e7d0bf8d379bo0
To: <sip:74952150687@192.168.5.253>;tag=as49ceebd7
Call-ID: 1957afb4-53c87107@192.168.33.200
CSeq: 105 BYE
Max-Forwards: 70
Authorization: Digest username="297",realm="asterisk",nonce="6b8984f9",uri="sip:74952150687@192.168.5.253:5060",algorithm=MD5,response="762264144d8d087217a640ea1e40b26c"
User-Agent: Cisco/SPA112-1.3.2(014)
P-RTP-Stat: PS=0,OS=0,PR=0,OR=0,PL=0,JI=0,LA=0,DU=32,EN=G711a,DE=G711a
Content-Length: 0
<------------->
--- (11 headers 0 lines) ---
Sending to 192.168.33.200:5060 (NAT)
Scheduling destruction of SIP dialog '1957afb4-53c87107@192.168.33.200' in 6400 ms (Method: BYE)
<--- Transmitting (NAT) to 192.168.33.200:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.33.200:5060;branch=z9hG4bK-5cd9ac81;received=192.168.33.200;rport=5060
From: "297" <sip:297@192.168.5.253>;tag=ea48e7d0bf8d379bo0
To: <sip:74952150687@192.168.5.253>;tag=as49ceebd7
Call-ID: 1957afb4-53c87107@192.168.33.200
CSeq: 105 BYE
Server: Asterisk PBX 1.8.10.1~dfsg-1ubuntu1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0
<------------>
-- Executing [h@office:1] Macro("SIP/297-00000167", "onhangup") in new stack
Sign up for free to join this conversation on GitHub. Already have an account? Sign in to comment