<--- SIP read from UDP://10.10.10.240:5060 --->
INVITE sip:XXXYYYZZZZ@10.10.10.180 SIP/2.0
Via: SIP/2.0/UDP 10.10.10.240:5060;branch=z9hG4bK5e26a7fbc032688a0.8a43a92d69716a3bb
Max-Forwards: 70
From: "169" <sip:169@10.10.10.180>;tag=d8b0963f02
To: "XXXYYYZZZZ" <sip:XXXYYYZZZZ@10.10.10.180>
Call-ID: 497f44a1ebd18259
CSeq: 15616 INVITE
Allow:  INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO
Allow-Events: talk, hold, conference, LocalModeStatus
Contact: "169" <sip:169@10.10.10.240:5060;transport=udp>;+sip.instance="<urn:uuid:00000000-0000-1000-8000-00085D1992F0>"
Supported: gruu, path, timer, 100rel, replaces
User-Agent: Aastra 57i/2.5.1.2000
Content-Type: application/sdp
Content-Length: 594

v=0
o=MxSIP 0 0 IN IP4 10.10.10.240
s=SIP Call
c=IN IP4 10.10.10.240
t=0 0
m=audio 3000 RTP/AVP 0 18 106 107 113 110 111 112 98 97 115 96 9 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:106 BV16/8000
a=rtpmap:107 BV32/16000
a=rtpmap:113 L16/16000
a=rtpmap:110 PCMU/16000
a=rtpmap:111 PCMA/16000
a=rtpmap:112 L16/8000
a=rtpmap:98 G726-16/8000
a=rtpmap:97 G726-24/8000
a=rtpmap:115 G726-32/8000
a=rtpmap:96 G726-40/8000
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=silenceSupp:on - - - -
a=fmtp:101 0-15
a=ptime:30
a=sendrecv

<------------->
[Nov  9 12:30:41] VERBOSE[4267] logger.c: --- (14 headers 25 lines) ---
[Nov  9 12:30:41] VERBOSE[4267] logger.c:   == Using SIP RTP TOS bits 184
[Nov  9 12:30:41] VERBOSE[4267] logger.c:   == Using SIP RTP CoS mark 5
[Nov  9 12:30:41] VERBOSE[4267] logger.c:   == Using SIP VRTP CoS mark 6
[Nov  9 12:30:41] VERBOSE[4267] logger.c:   == Using UDPTL TOS bits 184
[Nov  9 12:30:41] VERBOSE[4267] logger.c:   == Using UDPTL CoS mark 5
[Nov  9 12:30:41] VERBOSE[4267] logger.c: Sending to 10.10.10.240 : 5060 (no NAT)
[Nov  9 12:30:41] VERBOSE[4267] logger.c: Using INVITE request as basis request - 497f44a1ebd18259
[Nov  9 12:30:41] VERBOSE[4267] logger.c: Found user '169' for '169'
[Nov  9 12:30:41] VERBOSE[4267] logger.c: 
<--- Reliably Transmitting (no NAT) to 10.10.10.240:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.10.10.240:5060;branch=z9hG4bK5e26a7fbc032688a0.8a43a92d69716a3bb;received=10.10.10.240
From: "169" <sip:169@10.10.10.180>;tag=d8b0963f02
To: "XXXYYYZZZZ" <sip:XXXYYYZZZZ@10.10.10.180>;tag=as61b62212
Call-ID: 497f44a1ebd18259
CSeq: 15616 INVITE
User-Agent: Asterisk PBX 1.6.0.18-rc1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1cd07594"
Content-Length: 0


<------------>
[Nov  9 12:30:41] VERBOSE[4267] logger.c: Scheduling destruction of SIP dialog '497f44a1ebd18259' in 32000 ms (Method: INVITE)
[Nov  9 12:30:42] VERBOSE[4267] logger.c: 
<--- SIP read from UDP://10.10.10.240:5060 --->
ACK sip:XXXYYYZZZZ@10.10.10.180 SIP/2.0
Via: SIP/2.0/UDP 10.10.10.240:5060;branch=z9hG4bK5e26a7fbc032688a0.8a43a92d69716a3bb
Max-Forwards: 70
From: "169" <sip:169@10.10.10.180>;tag=d8b0963f02
To: "XXXYYYZZZZ" <sip:XXXYYYZZZZ@10.10.10.180>;tag=as61b62212
Call-ID: 497f44a1ebd18259
CSeq: 15616 ACK
User-Agent: Aastra 57i/2.5.1.2000
Content-Length: 0


<------------->
[Nov  9 12:30:42] VERBOSE[4267] logger.c: --- (9 headers 0 lines) ---
[Nov  9 12:30:42] VERBOSE[4267] logger.c: 
<--- SIP read from UDP://10.10.10.240:5060 --->
INVITE sip:XXXYYYZZZZ@10.10.10.180 SIP/2.0
Via: SIP/2.0/UDP 10.10.10.240:5060;branch=z9hG4bK8185a8922c55c776d.2bab1d9dbd39e9477
Max-Forwards: 70
From: "169" <sip:169@10.10.10.180>;tag=d8b0963f02
To: "XXXYYYZZZZ" <sip:XXXYYYZZZZ@10.10.10.180>
Call-ID: 497f44a1ebd18259
CSeq: 15617 INVITE
Allow:  INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO
Allow-Events: talk, hold, conference, LocalModeStatus
Authorization: Digest username="169",realm="asterisk",nonce="1cd07594",uri="sip:XXXYYYZZZZ@10.10.10.180",response="eb122c0055a8426e2655f0d9c2bd70d9",algorithm=MD5
Contact: "169" <sip:169@10.10.10.240:5060;transport=udp>;+sip.instance="<urn:uuid:00000000-0000-1000-8000-00085D1992F0>"
Supported: gruu, path, timer, 100rel, replaces
User-Agent: Aastra 57i/2.5.1.2000
Content-Type: application/sdp
Content-Length: 594

v=0
o=MxSIP 0 0 IN IP4 10.10.10.240
s=SIP Call
c=IN IP4 10.10.10.240
t=0 0
m=audio 3000 RTP/AVP 0 18 106 107 113 110 111 112 98 97 115 96 9 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:106 BV16/8000
a=rtpmap:107 BV32/16000
a=rtpmap:113 L16/16000
a=rtpmap:110 PCMU/16000
a=rtpmap:111 PCMA/16000
a=rtpmap:112 L16/8000
a=rtpmap:98 G726-16/8000
a=rtpmap:97 G726-24/8000
a=rtpmap:115 G726-32/8000
a=rtpmap:96 G726-40/8000
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=silenceSupp:on - - - -
a=fmtp:101 0-15
a=ptime:30
a=sendrecv

<------------->
[Nov  9 12:30:42] VERBOSE[4267] logger.c: --- (15 headers 25 lines) ---
[Nov  9 12:30:42] VERBOSE[4267] logger.c: Sending to 10.10.10.240 : 5060 (no NAT)
[Nov  9 12:30:42] VERBOSE[4267] logger.c: Using INVITE request as basis request - 497f44a1ebd18259
[Nov  9 12:30:42] VERBOSE[4267] logger.c: Found user '169' for '169'
[Nov  9 12:30:42] VERBOSE[4267] logger.c: Found RTP audio format 0
[Nov  9 12:30:42] VERBOSE[4267] logger.c: Found RTP audio format 18
[Nov  9 12:30:42] VERBOSE[4267] logger.c: Found RTP audio format 106
[Nov  9 12:30:42] VERBOSE[4267] logger.c: Found RTP audio format 107
[Nov  9 12:30:42] VERBOSE[4267] logger.c: Found RTP audio format 113
[Nov  9 12:30:42] VERBOSE[4267] logger.c: Found RTP audio format 110
[Nov  9 12:30:42] VERBOSE[4267] logger.c: Found RTP audio format 111
[Nov  9 12:30:42] VERBOSE[4267] logger.c: Found RTP audio format 112
[Nov  9 12:30:42] VERBOSE[4267] logger.c: Found RTP audio format 98
[Nov  9 12:30:42] VERBOSE[4267] logger.c: Found RTP audio format 97
[Nov  9 12:30:42] VERBOSE[4267] logger.c: Found RTP audio format 115
[Nov  9 12:30:42] VERBOSE[4267] logger.c: Found RTP audio format 96
[Nov  9 12:30:42] VERBOSE[4267] logger.c: Found RTP audio format 9
[Nov  9 12:30:42] VERBOSE[4267] logger.c: Found RTP audio format 8
[Nov  9 12:30:42] VERBOSE[4267] logger.c: Found RTP audio format 101
[Nov  9 12:30:42] VERBOSE[4267] logger.c: Found audio description format PCMU for ID 0
[Nov  9 12:30:42] VERBOSE[4267] logger.c: Found audio description format G729 for ID 18
[Nov  9 12:30:42] VERBOSE[4267] logger.c: Found unknown media description format BV16 for ID 106
[Nov  9 12:30:42] VERBOSE[4267] logger.c: Found unknown media description format BV32 for ID 107
[Nov  9 12:30:42] VERBOSE[4267] logger.c: Found audio description format L16 for ID 113
[Nov  9 12:30:42] VERBOSE[4267] logger.c: Found audio description format PCMU for ID 110
[Nov  9 12:30:42] VERBOSE[4267] logger.c: Found audio description format PCMA for ID 111
[Nov  9 12:30:42] VERBOSE[4267] logger.c: Found audio description format L16 for ID 112
[Nov  9 12:30:42] VERBOSE[4267] logger.c: Found unknown media description format G726-16 for ID 98
[Nov  9 12:30:42] VERBOSE[4267] logger.c: Found unknown media description format G726-24 for ID 97
[Nov  9 12:30:42] VERBOSE[4267] logger.c: Found audio description format G726-32 for ID 115
[Nov  9 12:30:42] VERBOSE[4267] logger.c: Found unknown media description format G726-40 for ID 96
[Nov  9 12:30:42] VERBOSE[4267] logger.c: Found audio description format G722 for ID 9
[Nov  9 12:30:42] VERBOSE[4267] logger.c: Found audio description format PCMA for ID 8
[Nov  9 12:30:42] VERBOSE[4267] logger.c: Found audio description format telephone-event for ID 101
[Nov  9 12:30:42] VERBOSE[4267] logger.c: Capabilities: us - 0x27f9fff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc|g726aal2|g722|slin16|jpeg|png|h261|h263|h263p|h264|mpeg4|t140), peer - audio=0x194c (ulaw|alaw|g726|slin|g729|g722)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x194c (ulaw|alaw|g726|slin|g729|g722)
[Nov  9 12:30:42] VERBOSE[4267] logger.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
[Nov  9 12:30:42] VERBOSE[4267] logger.c: Peer audio RTP is at port 10.10.10.240:3000
[Nov  9 12:30:42] VERBOSE[4267] logger.c: Peer doesn't provide video
[Nov  9 12:30:42] VERBOSE[4267] logger.c: Looking for XXXYYYZZZZ in longdistance (domain 10.10.10.180)
[Nov  9 12:30:42] VERBOSE[4267] logger.c: list_route: hop: <sip:169@10.10.10.240:5060;transport=udp>
[Nov  9 12:30:42] VERBOSE[4267] logger.c: 
<--- Transmitting (no NAT) to 10.10.10.240:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.10.10.240:5060;branch=z9hG4bK8185a8922c55c776d.2bab1d9dbd39e9477;received=10.10.10.240
From: "169" <sip:169@10.10.10.180>;tag=d8b0963f02
To: "XXXYYYZZZZ" <sip:XXXYYYZZZZ@10.10.10.180>
Call-ID: 497f44a1ebd18259
CSeq: 15617 INVITE
User-Agent: Asterisk PBX 1.6.0.18-rc1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Require: timer
Session-Expires: 1800;refresher=uas
Contact: <sip:XXXYYYZZZZ@10.10.10.180>
Content-Length: 0


<------------>
[Nov  9 12:30:42] VERBOSE[5341] logger.c:     -- Executing [XXXYYYZZZZ@longdistance:1] [1;36;40mAnswer[0;37;40m("[1;35;40mSIP/169-0000003d[0;37;40m", "[1;35;40m[0;37;40m") in new stack
[Nov  9 12:30:42] VERBOSE[5341] logger.c: Audio is at 10.10.10.180 port 19816
[Nov  9 12:30:42] VERBOSE[5341] logger.c: Adding codec 0x4 (ulaw) to SDP
[Nov  9 12:30:42] VERBOSE[5341] logger.c: Adding codec 0x8 (alaw) to SDP
[Nov  9 12:30:42] VERBOSE[5341] logger.c: Adding codec 0x40 (slin) to SDP
[Nov  9 12:30:42] VERBOSE[5341] logger.c: Adding codec 0x100 (g729) to SDP
[Nov  9 12:30:42] VERBOSE[5341] logger.c: Adding codec 0x800 (g726) to SDP
[Nov  9 12:30:42] VERBOSE[5341] logger.c: Adding codec 0x1000 (g722) to SDP
[Nov  9 12:30:42] VERBOSE[5341] logger.c: Adding non-codec 0x1 (telephone-event) to SDP
[Nov  9 12:30:42] VERBOSE[5341] logger.c: 
<--- Reliably Transmitting (no NAT) to 10.10.10.240:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.10.10.240:5060;branch=z9hG4bK8185a8922c55c776d.2bab1d9dbd39e9477;received=10.10.10.240
From: "169" <sip:169@10.10.10.180>;tag=d8b0963f02
To: "XXXYYYZZZZ" <sip:XXXYYYZZZZ@10.10.10.180>;tag=as69b05822
Call-ID: 497f44a1ebd18259
CSeq: 15617 INVITE
User-Agent: Asterisk PBX 1.6.0.18-rc1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Require: timer
Session-Expires: 1800;refresher=uas
Contact: <sip:XXXYYYZZZZ@10.10.10.180>
Content-Type: application/sdp
Content-Length: 421

v=0
o=root 1290194795 1290194795 IN IP4 10.10.10.180
s=Asterisk PBX 1.6.0.18-rc1
c=IN IP4 10.10.10.180
t=0 0
m=audio 19816 RTP/AVP 0 8 112 18 115 9 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:112 L16/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:115 G726-32/8000
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------>
[Nov  9 12:30:42] VERBOSE[4267] logger.c: 
<--- SIP read from UDP://10.10.10.240:5060 --->
ACK sip:XXXYYYZZZZ@10.10.10.180 SIP/2.0
Via: SIP/2.0/UDP 10.10.10.240:5060;branch=z9hG4bK6395f37c8649a577b.2d382e7b5643431e5
Max-Forwards: 70
From: "169" <sip:169@10.10.10.180>;tag=d8b0963f02
To: "XXXYYYZZZZ" <sip:XXXYYYZZZZ@10.10.10.180>;tag=as69b05822
Call-ID: 497f44a1ebd18259
CSeq: 15617 ACK
Authorization: Digest username="169",realm="asterisk",nonce="1cd07594",uri="sip:XXXYYYZZZZ@10.10.10.180",response="eb122c0055a8426e2655f0d9c2bd70d9",algorithm=MD5
User-Agent: Aastra 57i/2.5.1.2000
Content-Length: 0


<------------->
[Nov  9 12:30:42] VERBOSE[4267] logger.c: --- (10 headers 0 lines) ---
[Nov  9 12:30:42] VERBOSE[5341] logger.c:     -- Executing [XXXYYYZZZZ@longdistance:2] [1;36;40mGosub[0;37;40m("[1;35;40mSIP/169-0000003d[0;37;40m", "[1;35;40mDialOut,s,1(XXXYYYZZZZ,SIP/teliax-sip)[0;37;40m") in new stack
[Nov  9 12:30:42] VERBOSE[5341] logger.c:     -- Executing [s@DialOut:1] [1;36;40mNoOp[0;37;40m("[1;35;40mSIP/169-0000003d[0;37;40m", "[1;35;40mContext: DialOut called with outgoing number XXXYYYZZZZ, and Preferred-Out=SIP/teliax-sip[0;37;40m") in new stack
[Nov  9 12:30:42] VERBOSE[5341] logger.c:     -- Executing [s@DialOut:2] [1;36;40mNoOp[0;37;40m("[1;35;40mSIP/169-0000003d[0;37;40m", "[1;35;40m169=the local extension[0;37;40m") in new stack
[Nov  9 12:30:42] VERBOSE[5341] logger.c:     -- Executing [s@DialOut:3] [1;36;40mGotoIf[0;37;40m("[1;35;40mSIP/169-0000003d[0;37;40m", "[1;35;40m0?dial-out[0;37;40m") in new stack
[Nov  9 12:30:42] VERBOSE[5341] logger.c:     -- Executing [s@DialOut:4] [1;36;40mSet[0;37;40m("[1;35;40mSIP/169-0000003d[0;37;40m", "[1;35;40mCALLERID(num)=212AAABBBB[0;37;40m") in new stack
[Nov  9 12:30:42] VERBOSE[5341] logger.c:     -- Executing [s@DialOut:5] [1;36;40mDial[0;37;40m("[1;35;40mSIP/169-0000003d[0;37;40m", "[1;35;40mSIP/teliax-sip/XXXYYYZZZZ[0;37;40m") in new stack
[Nov  9 12:30:42] VERBOSE[5341] logger.c:   == Using SIP RTP TOS bits 184
[Nov  9 12:30:42] VERBOSE[5341] logger.c:   == Using SIP RTP CoS mark 5
[Nov  9 12:30:42] VERBOSE[5341] logger.c:   == Using SIP VRTP CoS mark 6
[Nov  9 12:30:42] VERBOSE[5341] logger.c:   == Using UDPTL TOS bits 184
[Nov  9 12:30:42] VERBOSE[5341] logger.c:   == Using UDPTL CoS mark 5
[Nov  9 12:30:42] VERBOSE[5341] logger.c: Audio is at AA.BB.CCC.DDD port 16884
[Nov  9 12:30:42] VERBOSE[5341] logger.c: Adding codec 0x4 (ulaw) to SDP
[Nov  9 12:30:42] VERBOSE[5341] logger.c: Adding codec 0x2 (gsm) to SDP
[Nov  9 12:30:42] VERBOSE[5341] logger.c: Adding non-codec 0x1 (telephone-event) to SDP
[Nov  9 12:30:42] VERBOSE[5341] logger.c: Reliably Transmitting (NAT) to 8.14.120.23:5060:
INVITE sip:XXXYYYZZZZ@nyc.teliax.net SIP/2.0
Via: SIP/2.0/UDP AA.BB.CCC.DDD:5060;branch=z9hG4bK69fbe370;rport
Max-Forwards: 70
From: "Andy Louie" <sip:212AAABBBB@AA.BB.CCC.DDD>;tag=as02acfb24
To: <sip:XXXYYYZZZZ@nyc.teliax.net>
Contact: <sip:212AAABBBB@AA.BB.CCC.DDD>
Call-ID: 5dc397ae328b13e66c3f974646dca77e@AA.BB.CCC.DDD
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.0.18-rc1
Date: Mon, 09 Nov 2009 17:30:42 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 291

v=0
o=root 743895692 743895692 IN IP4 AA.BB.CCC.DDD
s=Asterisk PBX 1.6.0.18-rc1
c=IN IP4 AA.BB.CCC.DDD
t=0 0
m=audio 16884 RTP/AVP 0 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
[Nov  9 12:30:42] VERBOSE[5341] logger.c:     -- Called teliax-sip/XXXYYYZZZZ
[Nov  9 12:30:42] VERBOSE[4267] logger.c: 
<--- SIP read from UDP://8.14.120.23:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.10.10.180:5060;branch=z9hG4bK69fbe370;rport=5060
From: "Andy Louie" <sip:212AAABBBB@10.10.10.180:5060>;tag=as02acfb24
To: <sip:XXXYYYZZZZ@nyc.teliax.net>
Call-ID: 5dc397ae328b13e66c3f974646dca77e@AA.BB.CCC.DDD
CSeq: 102 INVITE
User-Agent: Teliaware2.0.2
Content-Length: 0


<------------->
[Nov  9 12:30:42] VERBOSE[4267] logger.c: --- (8 headers 0 lines) ---
[Nov  9 12:30:42] VERBOSE[4267] logger.c: 
<--- SIP read from UDP://8.14.120.23:5060 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 10.10.10.180:5060;branch=z9hG4bK69fbe370;rport=5060
From: "Andy Louie" <sip:212AAABBBB@10.10.10.180:5060>;tag=as02acfb24
To: <sip:XXXYYYZZZZ@nyc.teliax.net>;tag=8DHtU8rmS4r3D
Call-ID: 5dc397ae328b13e66c3f974646dca77e@AA.BB.CCC.DDD
CSeq: 102 INVITE
User-Agent: Teliaware2.0.2
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
Supported: timer, precondition, path, replaces
Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer
Proxy-Authenticate: Digest realm="nyc.teliax.net", nonce="2eddfb9a-55cd-de11-beeb-00114336b0d5", algorithm=MD5, qop="auth"
Content-Length: 0


<------------->
[Nov  9 12:30:42] VERBOSE[4267] logger.c: --- (13 headers 0 lines) ---
[Nov  9 12:30:42] VERBOSE[4267] logger.c: Transmitting (NAT) to 8.14.120.23:5060:
ACK sip:XXXYYYZZZZ@nyc.teliax.net SIP/2.0
Via: SIP/2.0/UDP AA.BB.CCC.DDD:5060;branch=z9hG4bK69fbe370;rport
Max-Forwards: 70
From: "Andy Louie" <sip:212AAABBBB@AA.BB.CCC.DDD>;tag=as02acfb24
To: <sip:XXXYYYZZZZ@nyc.teliax.net>;tag=8DHtU8rmS4r3D
Contact: <sip:212AAABBBB@AA.BB.CCC.DDD>
Call-ID: 5dc397ae328b13e66c3f974646dca77e@AA.BB.CCC.DDD
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.6.0.18-rc1
Content-Length: 0


---
[Nov  9 12:30:42] VERBOSE[4267] logger.c: Audio is at AA.BB.CCC.DDD port 16884
[Nov  9 12:30:42] VERBOSE[4267] logger.c: Adding codec 0x4 (ulaw) to SDP
[Nov  9 12:30:42] VERBOSE[4267] logger.c: Adding codec 0x2 (gsm) to SDP
[Nov  9 12:30:42] VERBOSE[4267] logger.c: Adding non-codec 0x1 (telephone-event) to SDP
[Nov  9 12:30:42] VERBOSE[4267] logger.c: Reliably Transmitting (NAT) to 8.14.120.23:5060:
INVITE sip:XXXYYYZZZZ@nyc.teliax.net SIP/2.0
Via: SIP/2.0/UDP AA.BB.CCC.DDD:5060;branch=z9hG4bK40a13008;rport
Max-Forwards: 70
From: "Andy Louie" <sip:212AAABBBB@AA.BB.CCC.DDD>;tag=as02acfb24
To: <sip:XXXYYYZZZZ@nyc.teliax.net>
Contact: <sip:212AAABBBB@AA.BB.CCC.DDD>
Call-ID: 5dc397ae328b13e66c3f974646dca77e@AA.BB.CCC.DDD
CSeq: 103 INVITE
User-Agent: Asterisk PBX 1.6.0.18-rc1
Proxy-Authorization: Digest username="USERNAME(Sip)", realm="nyc.teliax.net", algorithm=MD5, uri="sip:XXXYYYZZZZ@nyc.teliax.net", nonce="2eddfb9a-55cd-de11-beeb-00114336b0d5", response="390a59a00bfc17499ab4348920b716c9", qop=auth, cnonce="58d6a2f7", nc=00000001
Date: Mon, 09 Nov 2009 17:30:42 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 291

v=0
o=root 743895692 743895693 IN IP4 AA.BB.CCC.DDD
s=Asterisk PBX 1.6.0.18-rc1
c=IN IP4 AA.BB.CCC.DDD
t=0 0
m=audio 16884 RTP/AVP 0 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
[Nov  9 12:30:42] VERBOSE[4267] logger.c: 
<--- SIP read from UDP://8.14.120.23:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.10.10.180:5060;branch=z9hG4bK40a13008;rport=5060
From: "Andy Louie" <sip:212AAABBBB@10.10.10.180:5060>;tag=as02acfb24
To: <sip:XXXYYYZZZZ@nyc.teliax.net>
Call-ID: 5dc397ae328b13e66c3f974646dca77e@AA.BB.CCC.DDD
CSeq: 103 INVITE
User-Agent: Teliaware2.0.2
Content-Length: 0


<------------->
[Nov  9 12:30:42] VERBOSE[4267] logger.c: --- (8 headers 0 lines) ---
[Nov  9 12:30:43] VERBOSE[4267] logger.c: Really destroying SIP dialog 'cd03c439483b856e8bd7576b2e32db84@10.10.10.210' Method: REGISTER
[Nov  9 12:30:44] VERBOSE[4267] logger.c: 
<--- SIP read from UDP://8.14.120.23:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 10.10.10.180:5060;branch=z9hG4bK40a13008;rport=5060
From: "Andy Louie" <sip:212AAABBBB@10.10.10.180:5060>;tag=as02acfb24
To: <sip:XXXYYYZZZZ@nyc.teliax.net>;tag=a03BZytUKp58m
Call-ID: 5dc397ae328b13e66c3f974646dca77e@AA.BB.CCC.DDD
CSeq: 103 INVITE
Contact: <sip:mod_sofia@8.14.120.23:5060;transport=udp>
User-Agent: Teliaware2.0.2
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
Supported: timer, precondition, path, replaces
Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 245

v=0
o=FreeSWITCH 1257763792 1257763793 IN IP4 8.14.120.23
s=FreeSWITCH
c=IN IP4 8.14.120.23
t=0 0
m=audio 24052 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20

<------------->
[Nov  9 12:30:44] VERBOSE[4267] logger.c: --- (15 headers 11 lines) ---
[Nov  9 12:30:44] VERBOSE[4267] logger.c: Found RTP audio format 0
[Nov  9 12:30:44] VERBOSE[4267] logger.c: Found RTP audio format 101
[Nov  9 12:30:44] VERBOSE[4267] logger.c: Found audio description format PCMU for ID 0
[Nov  9 12:30:44] VERBOSE[4267] logger.c: Found audio description format telephone-event for ID 101
[Nov  9 12:30:44] VERBOSE[4267] logger.c: Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
[Nov  9 12:30:44] VERBOSE[4267] logger.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
[Nov  9 12:30:44] VERBOSE[4267] logger.c: Peer audio RTP is at port 8.14.120.23:24052
[Nov  9 12:30:44] VERBOSE[5341] logger.c:     -- SIP/teliax-sip-0000003e is making progress passing it to SIP/169-0000003d
[Nov  9 12:30:46] VERBOSE[4267] logger.c: Really destroying SIP dialog '2b0b230d19917df671f2c0947a4babba@76.248.146.19' Method: OPTIONS
[Nov  9 12:30:46] VERBOSE[4267] logger.c: Really destroying SIP dialog 'b6a3745994239a067a7b7e8f7a230474@10.10.10.133' Method: REGISTER
...............


[Nov  9 12:30:55] VERBOSE[4267] logger.c: 
<--- SIP read from UDP://8.14.120.23:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.10.10.180:5060;branch=z9hG4bK40a13008;rport=5060
From: "Andy Louie" <sip:212AAABBBB@10.10.10.180:5060>;tag=as02acfb24
To: <sip:XXXYYYZZZZ@nyc.teliax.net>;tag=a03BZytUKp58m
Call-ID: 5dc397ae328b13e66c3f974646dca77e@AA.BB.CCC.DDD
CSeq: 103 INVITE
Contact: <sip:mod_sofia@8.14.120.23:5060;transport=udp>
User-Agent: Teliaware2.0.2
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
Require: timer
Supported: timer, precondition, path, replaces
Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer
Session-Expires: 120;refresher=uac
Min-SE: 120
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 245

v=0
o=FreeSWITCH 1257763792 1257763793 IN IP4 8.14.120.23
s=FreeSWITCH
c=IN IP4 8.14.120.23
t=0 0
m=audio 24052 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20

<------------->
[Nov  9 12:30:55] VERBOSE[4267] logger.c: --- (17 headers 11 lines) ---
[Nov  9 12:30:55] VERBOSE[4267] logger.c: list_route: hop: <sip:mod_sofia@8.14.120.23:5060;transport=udp>
[Nov  9 12:30:55] VERBOSE[4267] logger.c: set_destination: Parsing <sip:mod_sofia@8.14.120.23:5060;transport=udp> for address/port to send to
[Nov  9 12:30:55] VERBOSE[4267] logger.c: set_destination: set destination to 8.14.120.23, port 5060
[Nov  9 12:30:55] VERBOSE[4267] logger.c: Transmitting (NAT) to 8.14.120.23:5060:
ACK sip:mod_sofia@8.14.120.23:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP AA.BB.CCC.DDD:5060;branch=z9hG4bK6a333f9b;rport
Max-Forwards: 70
From: "Andy Louie" <sip:212AAABBBB@AA.BB.CCC.DDD>;tag=as02acfb24
To: <sip:XXXYYYZZZZ@nyc.teliax.net>;tag=a03BZytUKp58m
Contact: <sip:212AAABBBB@AA.BB.CCC.DDD>
Call-ID: 5dc397ae328b13e66c3f974646dca77e@AA.BB.CCC.DDD
CSeq: 103 ACK
User-Agent: Asterisk PBX 1.6.0.18-rc1
Content-Length: 0


---
[Nov  9 12:30:55] VERBOSE[4267] logger.c: set_destination: Parsing <sip:mod_sofia@8.14.120.23:5060;transport=udp> for address/port to send to
[Nov  9 12:30:55] VERBOSE[4267] logger.c: set_destination: set destination to 8.14.120.23, port 5060
[Nov  9 12:30:55] VERBOSE[4267] logger.c: Reliably Transmitting (NAT) to 8.14.120.23:5060:
BYE sip:mod_sofia@8.14.120.23:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP AA.BB.CCC.DDD:5060;branch=z9hG4bK563779b7;rport
Max-Forwards: 70
From: "Andy Louie" <sip:212AAABBBB@AA.BB.CCC.DDD>;tag=as02acfb24
To: <sip:XXXYYYZZZZ@nyc.teliax.net>;tag=a03BZytUKp58m
Call-ID: 5dc397ae328b13e66c3f974646dca77e@AA.BB.CCC.DDD
CSeq: 104 BYE
User-Agent: Asterisk PBX 1.6.0.18-rc1
Proxy-Authorization: Digest username="USERNAME(Sip)", realm="nyc.teliax.net", algorithm=MD5, uri="sip:mod_sofia@8.14.120.23:5060", nonce="2eddfb9a-55cd-de11-beeb-00114336b0d5", response="58869d6a5f383924a82c44085e1d2260", qop=auth, cnonce="73fdb6d6", nc=00000002
X-Asterisk-HangupCause: Unknown
X-Asterisk-HangupCauseCode: 0
Content-Length: 0


---
[Nov  9 12:30:55] VERBOSE[4267] logger.c: Scheduling destruction of SIP dialog '5dc397ae328b13e66c3f974646dca77e@AA.BB.CCC.DDD' in 6400 ms (Method: INVITE)
[Nov  9 12:30:55] VERBOSE[5341] logger.c:     -- SIP/teliax-sip-0000003e answered SIP/169-0000003d
[Nov  9 12:30:55] VERBOSE[5341] logger.c:     -- Packet2Packet bridging SIP/169-0000003d and SIP/teliax-sip-0000003e
[Nov  9 12:30:55] VERBOSE[4267] logger.c: 
<--- SIP read from UDP://8.14.120.23:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.10.10.180:5060;branch=z9hG4bK563779b7;rport=5060
From: "Andy Louie" <sip:212AAABBBB@10.10.10.180:5060>;tag=as02acfb24
To: <sip:XXXYYYZZZZ@nyc.teliax.net>;tag=a03BZytUKp58m
Call-ID: 5dc397ae328b13e66c3f974646dca77e@AA.BB.CCC.DDD
CSeq: 104 BYE
User-Agent: Teliaware2.0.2
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
Supported: timer, precondition, path, replaces
Content-Length: 0

........................

[Nov  9 12:31:02] WARNING[4267] chan_sip.c: Autodestruct on dialog '5dc397ae328b13e66c3f974646dca77e@AA.BB.CCC.DDD' with owner in place (Method: INVITE)
[Nov  9 12:31:02] VERBOSE[5341] logger.c: Scheduling destruction of SIP dialog '5dc397ae328b13e66c3f974646dca77e@AA.BB.CCC.DDD' in 6400 ms (Method: INVITE)
[Nov  9 12:31:02] VERBOSE[5341] logger.c: set_destination: Parsing <sip:mod_sofia@8.14.120.23:5060;transport=udp> for address/port to send to
[Nov  9 12:31:02] VERBOSE[5341] logger.c: set_destination: set destination to 8.14.120.23, port 5060
[Nov  9 12:31:02] VERBOSE[5341] logger.c: Reliably Transmitting (NAT) to 8.14.120.23:5060:
BYE sip:mod_sofia@8.14.120.23:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP AA.BB.CCC.DDD:5060;branch=z9hG4bK380d54f6;rport
Max-Forwards: 70
From: "Andy Louie" <sip:212AAABBBB@AA.BB.CCC.DDD>;tag=as02acfb24
To: <sip:XXXYYYZZZZ@nyc.teliax.net>;tag=a03BZytUKp58m
Call-ID: 5dc397ae328b13e66c3f974646dca77e@AA.BB.CCC.DDD
CSeq: 105 BYE
User-Agent: Asterisk PBX 1.6.0.18-rc1
Proxy-Authorization: Digest username="USERNAME(Sip)", realm="nyc.teliax.net", algorithm=MD5, uri="sip:mod_sofia@8.14.120.23:5060", nonce="2eddfb9a-55cd-de11-beeb-00114336b0d5", response="3537abf0dc2aac9453f04e9536fadd3e", qop=auth, cnonce="741b19a3", nc=00000003
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


---
[Nov  9 12:31:02] VERBOSE[5341] logger.c:   == Spawn extension (DialOut, s, 5) exited non-zero on 'SIP/169-0000003d'
[Nov  9 12:31:02] VERBOSE[5341] logger.c: Scheduling destruction of SIP dialog '497f44a1ebd18259' in 32000 ms (Method: ACK)
[Nov  9 12:31:02] VERBOSE[5341] logger.c: set_destination: Parsing <sip:169@10.10.10.240:5060;transport=udp> for address/port to send to
[Nov  9 12:31:02] VERBOSE[5341] logger.c: set_destination: set destination to 10.10.10.240, port 5060
[Nov  9 12:31:02] VERBOSE[5341] logger.c: Reliably Transmitting (no NAT) to 10.10.10.240:5060:
BYE sip:169@10.10.10.240:5060;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.10.10.180:5060;branch=z9hG4bK0c53c92d;rport
Max-Forwards: 70
From: "XXXYYYZZZZ" <sip:XXXYYYZZZZ@10.10.10.180>;tag=as69b05822
To: "169" <sip:169@10.10.10.180>;tag=d8b0963f02
Call-ID: 497f44a1ebd18259
CSeq: 102 BYE
User-Agent: Asterisk PBX 1.6.0.18-rc1
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


---
[Nov  9 12:31:02] VERBOSE[4267] logger.c: 
<--- SIP read from UDP://8.14.120.23:5060 --->
SIP/2.0 481 Call Does Not Exist
Via: SIP/2.0/UDP 10.10.10.180:5060;branch=z9hG4bK380d54f6;rport=5060
From: "Andy Louie" <sip:212AAABBBB@10.10.10.180:5060>;tag=as02acfb24
To: <sip:XXXYYYZZZZ@nyc.teliax.net>;tag=a03BZytUKp58m
Call-ID: 5dc397ae328b13e66c3f974646dca77e@AA.BB.CCC.DDD
CSeq: 105 BYE
User-Agent: Teliaware2.0.2
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
Supported: timer, precondition, path, replaces
Content-Length: 0


<------------->
[Nov  9 12:31:02] VERBOSE[4267] logger.c: --- (10 headers 0 lines) ---
[Nov  9 12:31:02] VERBOSE[4267] logger.c: Really destroying SIP dialog '5dc397ae328b13e66c3f974646dca77e@AA.BB.CCC.DDD' Method: INVITE
[Nov  9 12:31:02] VERBOSE[4267] logger.c: 
<--- SIP read from UDP://10.10.10.240:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.10.10.180:5060;branch=z9hG4bK0c53c92d;rport=5060;received=10.10.10.180
From: "XXXYYYZZZZ" <sip:XXXYYYZZZZ@10.10.10.180>;tag=as69b05822
To: "169" <sip:169@10.10.10.180>;tag=d8b0963f02
Call-ID: 497f44a1ebd18259
CSeq: 102 BYE
Server: Aastra 57i/2.5.1.2000
Content-Length: 0


<------------->
[Nov  9 12:31:02] VERBOSE[4267] logger.c: --- (8 headers 0 lines) ---
[Nov  9 12:31:02] VERBOSE[4267] logger.c: SIP Response message for INCOMING dialog BYE arrived
[Nov  9 12:31:02] VERBOSE[4267] logger.c: Really destroying SIP dialog '497f44a1ebd18259' Method: ACK
[Nov  9 12:31:04] VERBOSE[4267] logger.c: Reliably Transmitting (NAT) to 8.14.120.23:5060:
OPTIONS sip:nyc.teliax.net SIP/2.0
Via: SIP/2.0/UDP AA.BB.CCC.DDD:5060;branch=z9hG4bK5f76b1a1;rport
Max-Forwards: 70
From: "newharbor" <sip:newharbor@AA.BB.CCC.DDD>;tag=as2de9e6de
To: <sip:nyc.teliax.net>
Contact: <sip:newharbor@AA.BB.CCC.DDD>
Call-ID: 643c589d6a4f1cad3f9ee5ec4ac12c46@AA.BB.CCC.DDD
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.6.0.18-rc1
Date: Mon, 09 Nov 2009 17:31:04 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


---
[Nov  9 12:31:04] VERBOSE[4267] logger.c: 
<--- SIP read from UDP://8.14.120.23:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.10.10.180:5060;branch=z9hG4bK5f76b1a1;rport=5060
From: "newharbor" <sip:newharbor@10.10.10.180:5060>;tag=as2de9e6de
To: <sip:nyc.teliax.net>;tag=6a7c55e54aFNN
Call-ID: 643c589d6a4f1cad3f9ee5ec4ac12c46@AA.BB.CCC.DDD
CSeq: 102 OPTIONS
Contact: <sip:8.14.120.23>
User-Agent: Teliaware2.0.2
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH
Supported: timer, precondition, path, replaces
Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer
Content-Length: 0

