rendered paste body<--- SIP read from 10.0.0.20:50998 --->
INVITE sip:1800652xxx@10.0.0.1 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.20:5060;branch=z9hG4bK2d9d9561
From: "6000" <sip:6000@10.0.0.1>;tag=000d29e76ed40025726eecb3-69388e37
To: <sip:1800652xxx@10.0.0.1>
Call-ID: 000d29e7-6ed40009-3419ff94-3ac8a41f@10.0.0.20
Max-Forwards: 70
Date: Wed, 05 Mar 2008 04:40:35 GMT
CSeq: 101 INVITE
User-Agent: Cisco-CP7940G/8.0
Contact: <sip:6000@10.0.0.20:5060;transport=udp>
Expires: 180
Accept: application/sdp
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE
Supported: replaces,join,norefersub
Content-Length: 270
Content-Type: application/sdp
Content-Disposition: session;handling=optional
v=0
o=Cisco-SIPUA 22668 0 IN IP4 10.0.0.20
s=SIP Call
t=0 0
m=audio 22554 RTP/AVP 8 0 18 101
c=IN IP4 10.0.0.20
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->
--- (17 headers 13 lines) ---
Sending to 10.0.0.20 : 5060 (no NAT)
Using INVITE request as basis request - 000d29e7-6ed40009-3419ff94-3ac8a41f@10.0.0.20
Found peer '6000'
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 18
Found RTP audio format 101
Peer audio RTP is at port 10.0.0.20:22554
Found description format PCMA for ID 8
Found description format PCMU for ID 0
Found description format G729 for ID 18
Found description format telephone-event for ID 101
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x10c (ulaw|alaw|g729)/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 10.0.0.20:22554
Looking for 1800652xxx in numberplan-custom-2 (domain 10.0.0.1)
list_route: hop: <sip:6000@10.0.0.20:5060;transport=udp>
supermarket*CLI>
<--- Transmitting (no NAT) to 10.0.0.20:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.0.0.20:5060;branch=z9hG4bK2d9d9561;received=10.0.0.20
From: "6000" <sip:6000@10.0.0.1>;tag=000d29e76ed40025726eecb3-69388e37
To: <sip:1800652xxx@10.0.0.1>
Call-ID: 000d29e7-6ed40009-3419ff94-3ac8a41f@10.0.0.20
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:1800652xxx@10.0.0.1>
Content-Length: 0
<------------>
-- Executing [1800652xxx@numberplan-custom-2:1] Macro("SIP/6000-083ffec0", "trunkdial|SIP/trunk_1/1800652xxx") in new stack
-- Executing [s@macro-trunkdial:1] Dial("SIP/6000-083ffec0", "SIP/trunk_1/1800652xxx") in new stack
Audio is at 210.8.131.201 port 9710
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x10 (g726aal2) to SDP
Adding codec 0x20 (adpcm) to SDP
Adding codec 0x40 (slin) to SDP
Adding codec 0x80 (lpc10) to SDP
Adding codec 0x200 (speex) to SDP
Adding codec 0x400 (ilbc) to SDP
Adding codec 0x800 (g726) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 210.8.38.129:5060:
INVITE sip:1800652xxx@210.8.38.129 SIP/2.0
Via: SIP/2.0/UDP 210.8.131.201:5060;branch=z9hG4bK1b80f465;rport
From: "Computer Room" <sip:038414xxxx@aapt.com.au>;tag=as40ac18e3
To: <sip:1800652xxx@210.8.38.129>
Contact: <sip:038414xxxx@210.8.131.201>
Call-ID: 63c1c4d00262339e02142bef5439be8c@aapt.com.au
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 05 Mar 2008 04:40:35 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 501
v=0
o=root 14641 14641 IN IP4 210.8.131.201
s=session
c=IN IP4 210.8.131.201
t=0 0
m=audio 9710 RTP/AVP 0 3 8 112 5 10 7 110 97 111 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:112 AAL2-G726-32/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:10 L16/8000
a=rtpmap:7 LPC/8000
a=rtpmap:110 speex/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:111 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
-- Called trunk_1/1800652xxx
supermarket*CLI>
<--- SIP read from 210.8.38.129:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 210.8.131.201:5060;branch=z9hG4bK1b80f465;rport=5060
From: "Computer Room" <sip:038414xxxx@aapt.com.au>;tag=as40ac18e3
To: <sip:1800652xxx@210.8.38.129>
Call-ID: 63c1c4d00262339e02142bef5439be8c@aapt.com.au
CSeq: 102 INVITE
<------------->
--- (6 headers 0 lines) ---
supermarket*CLI>
<--- SIP read from 210.8.38.129:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 210.8.131.201:5060;branch=z9hG4bK1b80f465;rport=5060
From: "Computer Room" <sip:038414xxxx@aapt.com.au>;tag=as40ac18e3
To: <sip:1800652xxx@210.8.38.129>;tag=SDpmp6799-2121892385-1204692033314
Call-ID: 63c1c4d00262339e02142bef5439be8c@aapt.com.au
CSeq: 102 INVITE
WWW-Authenticate: DIGEST realm="aapt.com.au",qop="auth",algorithm=MD5,nonce="BroadWorksXfdfej601T2vggsiBW"
Content-Length: 0
<------------->
--- (8 headers 0 lines) ---
Transmitting (no NAT) to 210.8.38.129:5060:
ACK sip:1800652xxx@210.8.38.129 SIP/2.0
Via: SIP/2.0/UDP 210.8.131.201:5060;branch=z9hG4bK1b80f465;rport
From: "Computer Room" <sip:038414xxxx@aapt.com.au>;tag=as40ac18e3
To: <sip:1800652xxx@210.8.38.129>;tag=SDpmp6799-2121892385-1204692033314
Contact: <sip:038414xxxx@210.8.131.201>
Call-ID: 63c1c4d00262339e02142bef5439be8c@aapt.com.au
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
Audio is at 210.8.131.201 port 9710
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x10 (g726aal2) to SDP
Adding codec 0x20 (adpcm) to SDP
Adding codec 0x40 (slin) to SDP
Adding codec 0x80 (lpc10) to SDP
Adding codec 0x200 (speex) to SDP
Adding codec 0x400 (ilbc) to SDP
Adding codec 0x800 (g726) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 210.8.38.129:5060:
INVITE sip:1800652xxx@210.8.38.129 SIP/2.0
Via: SIP/2.0/UDP 210.8.131.201:5060;branch=z9hG4bK74502c09;rport
From: "Computer Room" <sip:038414xxxx@aapt.com.au>;tag=as40ac18e3
To: <sip:1800652xxx@210.8.38.129>
Contact: <sip:038414xxxx@210.8.131.201>
Call-ID: 63c1c4d00262339e02142bef5439be8c@aapt.com.au
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Authorization: Digest username="cjenkins", realm="aapt.com.au", algorithm=MD5, uri="sip:1800652xxx@210.8.38.129", nonce="BroadWorksXfdfej601T2vggsiBW", response="41d439b9bd4e32445ac613d60ea1472f", opaque="", qop=auth, cnonce="4f8fbe97", nc=00000001
Date: Wed, 05 Mar 2008 04:40:35 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 501
v=0
o=root 14641 14642 IN IP4 210.8.131.201
s=session
c=IN IP4 210.8.131.201
t=0 0
m=audio 9710 RTP/AVP 0 3 8 112 5 10 7 110 97 111 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:112 AAL2-G726-32/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:10 L16/8000
a=rtpmap:7 LPC/8000
a=rtpmap:110 speex/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:111 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
---
supermarket*CLI>
<--- SIP read from 210.8.38.129:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 210.8.131.201:5060;branch=z9hG4bK74502c09;rport=5060
From: "Computer Room" <sip:038414xxxx@aapt.com.au>;tag=as40ac18e3
To: <sip:1800652xxx@210.8.38.129>
Call-ID: 63c1c4d00262339e02142bef5439be8c@aapt.com.au
CSeq: 103 INVITE
<------------->
--- (6 headers 0 lines) ---
supermarket*CLI>
<--- SIP read from 210.8.38.129:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 210.8.131.201:5060;branch=z9hG4bK74502c09;rport=5060
From: "Computer Room" <sip:038414xxxx@aapt.com.au>;tag=as40ac18e3
To: <sip:1800652xxx@210.8.38.129>;tag=SDpmp6799-1528787319-1204692036089
Call-ID: 63c1c4d00262339e02142bef5439be8c@aapt.com.au
CSeq: 103 INVITE
Supported:
Contact: <sip:1800652xxx@210.8.38.129:5060;transport=udp>
Session: Media
Remote-Party-ID: <sip:1800652xxx@10.8.180.65;user=phone>;screen=yes;party=called;privacy=off;id-type=subscriber
Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,NOTIFY,UPDATE
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 248
v=0
o=BroadWorks 47058 1 IN IP4 210.8.38.129
s=-
c=IN IP4 210.8.38.129
t=0 0
m=audio 10770 RTP/AVP 0 101
c=IN IP4 210.8.38.129
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=bsoft: 1 image udptl t38
<------------->
--- (14 headers 12 lines) ---
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port 210.8.38.129:10770
Found description format PCMU for ID 0
Found description format telephone-event for ID 101
Capabilities: us - 0x3f1fff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc|g726aal2|g722|jpeg|png|h261|h263|h263p|h264), 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 210.8.38.129:10770
-- SIP/trunk_1-084058c8 is making progress passing it to SIP/6000-083ffec0
Audio is at 10.0.0.1 port 9318
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
<--- Transmitting (no NAT) to 10.0.0.20:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 10.0.0.20:5060;branch=z9hG4bK2d9d9561;received=10.0.0.20
From: "6000" <sip:6000@10.0.0.1>;tag=000d29e76ed40025726eecb3-69388e37
To: <sip:1800652xxx@10.0.0.1>;tag=as0a81775a
Call-ID: 000d29e7-6ed40009-3419ff94-3ac8a41f@10.0.0.20
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:1800652xxx@10.0.0.1>
Content-Type: application/sdp
Content-Length: 255
v=0
o=root 14641 14641 IN IP4 10.0.0.1
s=session
c=IN IP4 10.0.0.1
t=0 0
m=audio 9318 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=ptime:20
a=sendrecv
<------------>
supermarket*CLI>
<--- SIP read from 210.8.38.129:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 210.8.131.201:5060;branch=z9hG4bK74502c09;rport=5060
From: "Computer Room" <sip:038414xxxx@aapt.com.au>;tag=as40ac18e3
To: <sip:1800652xxx@210.8.38.129>;tag=SDpmp6799-1528787319-1204692036089
Call-ID: 63c1c4d00262339e02142bef5439be8c@aapt.com.au
CSeq: 103 INVITE
Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,UPDATE,NOTIFY
Supported:
Accept: multipart/mixed,application/media_control+xml,application/sdp
Contact: <sip:1800652xxx@210.8.38.129:5060;transport=udp>
Remote-Party-ID: <sip:1800652xxx@10.8.180.65;user=phone>;screen=yes;party=called;privacy=off;id-type=subscriber
Content-Type: application/sdp
Content-Length: 248
v=0
o=BroadWorks 47058 1 IN IP4 210.8.38.129
s=-
c=IN IP4 210.8.38.129
t=0 0
m=audio 10770 RTP/AVP 0 101
c=IN IP4 210.8.38.129
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=bsoft: 1 image udptl t38
<------------->
--- (13 headers 12 lines) ---
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port 210.8.38.129:10770
Found description format PCMU for ID 0
Found description format telephone-event for ID 101
Capabilities: us - 0x3f1fff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc|g726aal2|g722|jpeg|png|h261|h263|h263p|h264), 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 210.8.38.129:10770
list_route: hop: <sip:1800652xxx@210.8.38.129:5060;transport=udp>
set_destination: Parsing <sip:1800652xxx@210.8.38.129:5060;transport=udp> for address/port to send to
set_destination: set destination to 210.8.38.129, port 5060
Transmitting (no NAT) to 210.8.38.129:5060:
ACK sip:1800652xxx@210.8.38.129:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 210.8.131.201:5060;branch=z9hG4bK3728275d;rport
From: "Computer Room" <sip:038414xxxx@aapt.com.au>;tag=as40ac18e3
To: <sip:1800652xxx@210.8.38.129>;tag=SDpmp6799-1528787319-1204692036089
Contact: <sip:038414xxxx@210.8.131.201>
Call-ID: 63c1c4d00262339e02142bef5439be8c@aapt.com.au
CSeq: 103 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
-- SIP/trunk_1-084058c8 answered SIP/6000-083ffec0
Audio is at 10.0.0.1 port 9318
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
<--- Reliably Transmitting (no NAT) to 10.0.0.20:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.0.20:5060;branch=z9hG4bK2d9d9561;received=10.0.0.20
From: "6000" <sip:6000@10.0.0.1>;tag=000d29e76ed40025726eecb3-69388e37
To: <sip:1800652xxx@10.0.0.1>;tag=as0a81775a
Call-ID: 000d29e7-6ed40009-3419ff94-3ac8a41f@10.0.0.20
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:1800652xxx@10.0.0.1>
Content-Type: application/sdp
Content-Length: 255
v=0
o=root 14641 14642 IN IP4 10.0.0.1
s=session
c=IN IP4 10.0.0.1
t=0 0
m=audio 9318 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=ptime:20
a=sendrecv
<------------>
-- Packet2Packet bridging SIP/6000-083ffec0 and SIP/trunk_1-084058c8
Sent RTP P2P packet to 10.0.0.20:22554 (type 00, len 000160)
Sent RTP P2P packet to 10.0.0.20:22554 (type 00, len 000160)
Sent RTP P2P packet to 10.0.0.20:22554 (type 00, len 000160)
Sent RTP P2P packet to 210.8.38.129:10770 (type 08, len 000160)
Sent RTP P2P packet to 10.0.0.20:22554 (type 00, len 000160)
Sent RTP P2P packet to 210.8.38.129:10770 (type 08, len 000160)
Sent RTP P2P packet to 10.0.0.20:22554 (type 00, len 000160)
Sent RTP P2P packet to 210.8.38.129:10770 (type 08, len 000160)
Sent RTP P2P packet to 10.0.0.20:22554 (type 00, len 000160)
Sent RTP P2P packet to 210.8.38.129:10770 (type 08, len 000160)
Sent RTP P2P packet to 10.0.0.20:22554 (type 00, len 000160)
Sent RTP P2P packet to 210.8.38.129:10770 (type 08, len 000160)
Sent RTP P2P packet to 10.0.0.20:22554 (type 00, len 000160)
Sent RTP P2P packet to 210.8.38.129:10770 (type 08, len 000160)
Sent RTP P2P packet to 10.0.0.20:22554 (type 00, len 000160)
Sent RTP P2P packet to 210.8.38.129:10770 (type 08, len 000160)
Sent RTP P2P packet to 10.0.0.20:22554 (type 00, len 000160)
Sent RTP P2P packet to 210.8.38.129:10770 (type 08, len 000160)
Sent RTP P2P packet to 10.0.0.20:22554 (type 00, len 000160)
Sent RTP P2P packet to 210.8.38.129:10770 (type 08, len 000160)
Sent RTP P2P packet to 10.0.0.20:22554 (type 00, len 000160)
Sent RTP P2P packet to 210.8.38.129:10770 (type 08, len 000160)
Sent RTP P2P packet to 10.0.0.20:22554 (type 00, len 000160)
Sent RTP P2P packet to 210.8.38.129:10770 (type 08, len 000160)
Retransmitting #1 (no NAT) to 10.0.0.20:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.0.20:5060;branch=z9hG4bK2d9d9561;received=10.0.0.20
From: "6000" <sip:6000@10.0.0.1>;tag=000d29e76ed40025726eecb3-69388e37
To: <sip:1800652xxx@10.0.0.1>;tag=as0a81775a
Call-ID: 000d29e7-6ed40009-3419ff94-3ac8a41f@10.0.0.20
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:1800652xxx@10.0.0.1>
Content-Type: application/sdp
Content-Length: 255
v=0
o=root 14641 14642 IN IP4 10.0.0.1
s=session
c=IN IP4 10.0.0.1
t=0 0
m=audio 9318 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=ptime:20
a=sendrecv
---
Sent RTP P2P packet to 10.0.0.20:22554 (type 00, len 000160)
Sent RTP P2P packet to 210.8.38.129:10770 (type 08, len 000160)
Sent RTP P2P packet to 10.0.0.20:22554 (type 00, len 000160)
Sent RTP P2P packet to 210.8.38.129:10770 (type 08, len 000160)
Sent RTP P2P packet to 10.0.0.20:22554 (type 00, len 000160)
Sent RTP P2P packet to 210.8.38.129:10770 (type 08, len 000160)
supermarket*CLI>
<--- SIP read from 10.0.0.20:50998 --->
ACK sip:1800652xxx@10.0.0.1:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.20:5060;branch=z9hG4bK03ad3fd0
From: "6000" <sip:6000@10.0.0.1>;tag=000d29e76ed40025726eecb3-69388e37
To: <sip:1800652xxx@10.0.0.1>;tag=as0a81775a
Call-ID: 000d29e7-6ed40009-3419ff94-3ac8a41f@10.0.0.20
Max-Forwards: 70
Date: Wed, 05 Mar 2008 04:40:38 GMT
CSeq: 101 ACK
User-Agent: Cisco-CP7940G/8.0
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
Sent RTP P2P packet to 10.0.0.20:22554 (type 00, len 000160)
Sent RTP P2P packet to 210.8.38.129:10770 (type 08, len 000160)
Sent RTP P2P packet to 10.0.0.20:22554 (type 00, len 000160)
Sent RTP P2P packet to 210.8.38.129:10770 (type 08, len 000160)
Sent RTP P2P packet to 10.0.0.20:22554 (type 00, len 000160)
Sent RTP P2P packet to 210.8.38.129:10770 (type 08, len 000160)
Sent RTP P2P packet to 10.0.0.20:22554 (type 00, len 000160)
Sent RTP P2P packet to 210.8.38.129:10770 (type 08, len 000160)
Sent RTP P2P packet to 10.0.0.20:22554 (type 00, len 000160)
supermarket*CLI>
<--- SIP read from 10.0.0.20:51037 --->
ACK sip:1800652xxx@10.0.0.1:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.20:5060;branch=z9hG4bK03ad3fd0
From: "6000" <sip:6000@10.0.0.1>;tag=000d29e76ed40025726eecb3-69388e37
To: <sip:1800652xxx@10.0.0.1>;tag=as0a81775a
Call-ID: 000d29e7-6ed40009-3419ff94-3ac8a41f@10.0.0.20
Max-Forwards: 70
Date: Wed, 05 Mar 2008 04:40:38 GMT
CSeq: 101 ACK
User-Agent: Cisco-CP7940G/8.0
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
Sent RTP P2P packet to 210.8.38.129:10770 (type 08, len 000160)
Sent RTP P2P packet to 10.0.0.20:22554 (type 00, len 000160)
Sent RTP P2P packet to 210.8.38.129:10770 (type 08, len 000160)
Sent RTP P2P packet to 10.0.0.20:22554 (type 00, len 000160)
Sent RTP P2P packet to 210.8.38.129:10770 (type 08, len 000160)
Sent RTP P2P packet to 10.0.0.20:22554 (type 00, len 000160)
Sent RTP P2P packet to 210.8.38.129:10770 (type 08, len 000160)
Sent RTP P2P packet to 10.0.0.20:22554 (type 00, len 000160)
Sent RTP P2P packet to 210.8.38.129:10770 (type 08, len 000160)
Sent RTP P2P packet to 10.0.0.20:22554 (type 00, len 000160)
Sent RTP P2P packet to 210.8.38.129:10770 (type 08, len 000160)
Sent RTP P2P packet to 10.0.0.20:22554 (type 00, len 000160)
Sent RTP P2P packet to 210.8.38.129:10770 (type 08, len 000160)
Sent RTP P2P packet to 10.0.0.20:22554 (type 00, len 000160)
Sent RTP P2P packet to 210.8.38.129:10770 (type 08, len 000160)
Sent RTP P2P packet to 10.0.0.20:22554 (type 00, len 000160)
Sent RTP P2P packet to 210.8.38.129:10770 (type 08, len 000160)
Sent RTP P2P packet to 10.0.0.20:22554 (type 00, len 000160)
Sent RTP P2P packet to 210.8.38.129:10770 (type 08, len 000160)
Sent RTP P2P packet to 10.0.0.20:22554 (type 00, len 000160)
Sent RTP P2P packet to 210.8.38.129:10770 (type 08, len 000160)
Sent RTP P2P packet to 10.0.0.20:22554 (type 00, len 000160)
Sent RTP P2P packet to 210.8.38.129:10770 (type 08, len 000160)
Sent RTP P2P packet to 10.0.0.20:22554 (type 00, len 000160)
Sent RTP P2P packet to 210.8.38.129:10770 (type 08, len 000160)
Sent RTP P2P packet to 10.0.0.20:22554 (type 00, len 000160)
Sent RTP P2P packet to 10.0.0.20:22554 (type 00, len 000160)
Sent RTP P2P packet to 210.8.38.129:10770 (type 08, len 000160)
Sent RTP P2P packet to 210.8.38.129:10770 (type 08, len 000160)
Sent RTP P2P packet to 10.0.0.20:22554 (type 00, len 000160)
Sent RTP P2P packet to 210.8.38.129:10770 (type 08, len 000160)
Sent RTP P2P packet to 10.0.0.20:22554 (type 00, len 000160)
Sent RTP P2P packet to 210.8.38.129:10770 (type 08, len 000160)
Sent RTP P2P packet to 10.0.0.20:22554 (type 00, len 000160)
Sent RTP P2P packet to 210.8.38.129:10770 (type 08, len 000160)
Sent RTP P2P packet to 10.0.0.20:22554 (type 00, len 000160)
Sent RTP P2P packet to 10.0.0.20:22554 (type 00, len 000160)
Sent RTP P2P packet to 10.0.0.20:22554 (type 00, len 000160)
supermarket*CLI>
<--- SIP read from 10.0.0.20:50998 --->
BYE sip:1800652xxx@10.0.0.1:5060 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.20:5060;branch=z9hG4bK66af7688
From: "6000" <sip:6000@10.0.0.1>;tag=000d29e76ed40025726eecb3-69388e37
To: <sip:1800652xxx@10.0.0.1>;tag=as0a81775a
Call-ID: 000d29e7-6ed40009-3419ff94-3ac8a41f@10.0.0.20
Max-Forwards: 70
Date: Wed, 05 Mar 2008 04:40:39 GMT
CSeq: 102 BYE
User-Agent: Cisco-CP7940G/8.0
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
Sending to 10.0.0.20 : 5060 (no NAT)
<--- Transmitting (no NAT) to 10.0.0.20:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.0.0.20:5060;branch=z9hG4bK66af7688;received=10.0.0.20
From: "6000" <sip:6000@10.0.0.1>;tag=000d29e76ed40025726eecb3-69388e37
To: <sip:1800652xxx@10.0.0.1>;tag=as0a81775a
Call-ID: 000d29e7-6ed40009-3419ff94-3ac8a41f@10.0.0.20
CSeq: 102 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:1800652xxx@10.0.0.1>
Content-Length: 0
supermarket*CLI>
<------------>
Scheduling destruction of SIP dialog '63c1c4d00262339e02142bef5439be8c@aapt.com.au' in 32000 ms (Method: INVITE)
set_destination: Parsing <sip:1800652xxx@210.8.38.129:5060;transport=udp> for address/port to send to
set_destination: set destination to 210.8.38.129, port 5060
Reliably Transmitting (no NAT) to 210.8.38.129:5060:
BYE sip:1800652xxx@210.8.38.129:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 210.8.131.201:5060;branch=z9hG4bK56c36209;rport
From: "Computer Room" <sip:038414xxxx@aapt.com.au>;tag=as40ac18e3
To: <sip:1800652xxx@210.8.38.129>;tag=SDpmp6799-1528787319-1204692036089
Call-ID: 63c1c4d00262339e02142bef5439be8c@aapt.com.au
CSeq: 104 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Authorization: Digest username="cjenkins", realm="aapt.com.au", algorithm=MD5, uri="sip:1800652xxx@210.8.38.129:5060", nonce="BroadWorksXfdfej601T2vggsiBW", response="3cf443c5c3e4a4a0bf527889e9fe101e", opaque="", qop=auth, cnonce="1b1783f7", nc=00000002
Content-Length: 0
---
== Spawn extension (macro-trunkdial, s, 1) exited non-zero on 'SIP/6000-083ffec0' in macro 'trunkdial'
== Spawn extension (macro-trunkdial, s, 1) exited non-zero on 'SIP/6000-083ffec0'
Really destroying SIP dialog '000d29e7-6ed40009-3419ff94-3ac8a41f@10.0.0.20' Method: BYE
supermarket*CLI>
<--- SIP read from 210.8.38.129:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 210.8.131.201:5060;branch=z9hG4bK56c36209;rport=5060
From: "Computer Room" <sip:038414xxxx@aapt.com.au>;tag=as40ac18e3
To: <sip:1800652xxx@210.8.38.129>;tag=SDpmp6799-1528787319-1204692036089
Call-ID: 63c1c4d00262339e02142bef5439be8c@aapt.com.au
CSeq: 104 BYE
Content-Length: 0
<------------->
--- (7 headers 0 lines) ---
Really destroying SIP dialog '63c1c4d00262339e02142bef5439be8c@aapt.com.au' Method: INVITE