<------------->
[Jan  4 13:27:22] VERBOSE[18565] chan_sip.c: 
<--- SIP read from UDP:[2601:8:9181:4800:215:65ff:fe27:ac8e]:5062 --->
INVITE sip:9190000000@sip.ferree-clark.org:5060 SIP/2.0
Via: SIP/2.0/UDP [2601:8:9181:4800:215:65ff:fe27:ac8e]:5062;branch=z9hG4bK242854787
From: "Desk" <sip:yealinkphone@sip.ferree-clark.org>;tag=731755919
To: <sip:9190000000@sip.ferree-clark.org>
Call-ID: 2090487130@2601:8:9181:4800:215:65ff:fe27:ac8e
CSeq: 1 INVITE
Contact: <sip:yealinkphone@[2601:8:9181:4800:215:65ff:fe27:ac8e]:5062>
Content-Type: application/sdp
Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE
Max-Forwards: 70
User-Agent: Yealink SIP-T20P 9.73.0.40
Supported: replaces
Allow-Events: talk,hold,conference,refer,check-sync
Content-Length: 352

v=0
o=- 20006 20006 IN IP6 2601:8:9181:4800:215:65ff:fe27:ac8e
s=SDP data
c=IN IP6 2601:8:9181:4800:215:65ff:fe27:ac8e
t=0 0
m=audio 11792 RTP/AVP 0 8 18 9 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:9 G722/8000
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=ptime:20
a=sendrecv
<------------->
[Jan  4 13:27:22] VERBOSE[18565] chan_sip.c: --- (14 headers 15 lines) ---
[Jan  4 13:27:22] VERBOSE[18565] chan_sip.c: Sending to [2601:8:9181:4800:215:65ff:fe27:ac8e]:5062 (no NAT)
[Jan  4 13:27:22] VERBOSE[18565][C-00000000] chan_sip.c: Sending to [2601:8:9181:4800:215:65ff:fe27:ac8e]:5062 (no NAT)
[Jan  4 13:27:22] VERBOSE[18565][C-00000000] chan_sip.c: Using INVITE request as basis request - 2090487130@2601:8:9181:4800:215:65ff:fe27:ac8e
[Jan  4 13:27:22] VERBOSE[18565][C-00000000] chan_sip.c: Found peer 'yealinkphone' for 'yealinkphone' from [2601:8:9181:4800:215:65ff:fe27:ac8e]:5062
[Jan  4 13:27:22] VERBOSE[18565][C-00000000] chan_sip.c: Found RTP audio format 0
[Jan  4 13:27:22] VERBOSE[18565][C-00000000] chan_sip.c: Found RTP audio format 8
[Jan  4 13:27:22] VERBOSE[18565][C-00000000] chan_sip.c: Found RTP audio format 18
[Jan  4 13:27:22] VERBOSE[18565][C-00000000] chan_sip.c: Found RTP audio format 9
[Jan  4 13:27:22] VERBOSE[18565][C-00000000] chan_sip.c: Found RTP audio format 101
[Jan  4 13:27:22] VERBOSE[18565][C-00000000] chan_sip.c: Found audio description format PCMU for ID 0
[Jan  4 13:27:22] VERBOSE[18565][C-00000000] chan_sip.c: Found audio description format PCMA for ID 8
[Jan  4 13:27:22] VERBOSE[18565][C-00000000] chan_sip.c: Found audio description format G729 for ID 18
[Jan  4 13:27:22] VERBOSE[18565][C-00000000] chan_sip.c: Found audio description format G722 for ID 9
[Jan  4 13:27:22] VERBOSE[18565][C-00000000] chan_sip.c: Found audio description format telephone-event for ID 101
[Jan  4 13:27:22] VERBOSE[18565][C-00000000] chan_sip.c: Capabilities: us - (g729|ulaw|alaw), peer - audio=(ulaw|alaw|g722|g729)/video=(nothing)/text=(nothing), combined - (g729|ulaw|alaw)
[Jan  4 13:27:22] VERBOSE[18565][C-00000000] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
[Jan  4 13:27:22] VERBOSE[18565][C-00000000] chan_sip.c: Peer audio RTP is at port [2601:8:9181:4800:215:65ff:fe27:ac8e]:11792
[Jan  4 13:27:22] VERBOSE[18565][C-00000000] chan_sip.c: Looking for 9190000000 in outgoing (domain sip.ferree-clark.org)
[Jan  4 13:27:22] VERBOSE[18565][C-00000000] sip/route.c: sip_route_dump: route/path hop: <sip:yealinkphone@[2601:8:9181:4800:215:65ff:fe27:ac8e]:5062>
[Jan  4 13:27:22] VERBOSE[18565][C-00000000] chan_sip.c: 
<--- Transmitting (no NAT) to [2601:8:9181:4800:215:65ff:fe27:ac8e]:5062 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP [2601:8:9181:4800:215:65ff:fe27:ac8e]:5062;branch=z9hG4bK242854787;received=2601:8:9181:4800:215:65ff:fe27:ac8e
From: "Desk" <sip:yealinkphone@sip.ferree-clark.org>;tag=731755919
To: <sip:9190000000@sip.ferree-clark.org>
Call-ID: 2090487130@2601:8:9181:4800:215:65ff:fe27:ac8e
CSeq: 1 INVITE
Server: Asterisk PBX 13.1.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:9190000000@[2601:8:9181:4800:225:90ff:fe87:21f9]:5060>
Content-Length: 0


<------------>
[Jan  4 13:27:22] WARNING[18551] func_cdr.c: Using the CDR function to set 'accountcode' is deprecated. Please use the CHANNEL function instead.
[Jan  4 13:27:22] VERBOSE[18654][C-00000000] chan_sip.c: Audio is at 10560
[Jan  4 13:27:22] VERBOSE[18654][C-00000000] chan_sip.c: Video is at 74.95.126.33:10114
[Jan  4 13:27:22] VERBOSE[18654][C-00000000] chan_sip.c: Adding codec ulaw to SDP
[Jan  4 13:27:22] VERBOSE[18654][C-00000000] chan_sip.c: Adding codec alaw to SDP
[Jan  4 13:27:22] VERBOSE[18654][C-00000000] chan_sip.c: Adding codec gsm to SDP
[Jan  4 13:27:22] VERBOSE[18654][C-00000000] chan_sip.c: Adding video codec h263 to SDP
[Jan  4 13:27:22] VERBOSE[18654][C-00000000] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
[Jan  4 13:27:22] VERBOSE[18654][C-00000000] chan_sip.c: Reliably Transmitting (no NAT) to 204.74.213.5:5060:
INVITE sip:19190000000@sip.callwithus.com SIP/2.0
Via: SIP/2.0/UDP 74.95.126.33:5060;branch=z9hG4bK61e60399
Max-Forwards: 70
From: "9191000000" <sip:9191000000@74.95.126.33>;tag=as129a6005
To: <sip:19190000000@sip.callwithus.com>
Contact: <sip:9191000000@74.95.126.33:5060>
Call-ID: 722f85362c3ef1df5e60d3e025456c82@74.95.126.33:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 13.1.0
Date: Sun, 04 Jan 2015 21:27:22 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 356

v=0
o=root 991698651 991698651 IN IP4 74.95.126.33
s=Asterisk PBX 13.1.0
c=IN IP4 74.95.126.33
b=CT:384
t=0 0
m=audio 10560 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:150
a=sendrecv
m=video 10114 RTP/AVP 34
a=rtpmap:34 H263/90000
a=sendrecv

---
[Jan  4 13:27:22] VERBOSE[18565] chan_sip.c: 
<--- SIP read from UDP:204.74.213.5:5060 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 74.95.126.33:5060;branch=z9hG4bK61e60399
From: "9191000000" <sip:9191000000@74.95.126.33>;tag=as129a6005
To: <sip:19190000000@sip.callwithus.com>;tag=05fa965d41f3adc51e16f9a7acf1c273.e96f
Call-ID: 722f85362c3ef1df5e60d3e025456c82@74.95.126.33:5060
CSeq: 102 INVITE
Proxy-Authenticate: Digest realm="sip.callwithus.com", nonce=""
Server: CWU SIP GW
Content-Length: 0

<------------->
[Jan  4 13:27:22] VERBOSE[18565] chan_sip.c: --- (9 headers 0 lines) ---
[Jan  4 13:27:22] VERBOSE[18565][C-00000000] chan_sip.c: Transmitting (no NAT) to 204.74.213.5:5060:
ACK sip:19190000000@sip.callwithus.com SIP/2.0
Via: SIP/2.0/UDP 74.95.126.33:5060;branch=z9hG4bK61e60399
Max-Forwards: 70
From: "9191000000" <sip:9191000000@74.95.126.33>;tag=as129a6005
To: <sip:19190000000@sip.callwithus.com>;tag=05fa965d41f3adc51e16f9a7acf1c273.e96f
Contact: <sip:9191000000@74.95.126.33:5060>
Call-ID: 722f85362c3ef1df5e60d3e025456c82@74.95.126.33:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 13.1.0
Content-Length: 0


---
[Jan  4 13:27:22] VERBOSE[18565][C-00000000] chan_sip.c: Audio is at 10560
[Jan  4 13:27:22] VERBOSE[18565][C-00000000] chan_sip.c: Video is at 74.95.126.33:10114
[Jan  4 13:27:22] VERBOSE[18565][C-00000000] chan_sip.c: Adding codec ulaw to SDP
[Jan  4 13:27:22] VERBOSE[18565][C-00000000] chan_sip.c: Adding codec alaw to SDP
[Jan  4 13:27:22] VERBOSE[18565][C-00000000] chan_sip.c: Adding codec gsm to SDP
[Jan  4 13:27:22] VERBOSE[18565][C-00000000] chan_sip.c: Adding video codec h263 to SDP
[Jan  4 13:27:22] VERBOSE[18565][C-00000000] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
[Jan  4 13:27:22] VERBOSE[18565][C-00000000] chan_sip.c: Reliably Transmitting (no NAT) to 204.74.213.5:5060:
INVITE sip:19190000000@sip.callwithus.com SIP/2.0
Via: SIP/2.0/UDP 74.95.126.33:5060;branch=z9hG4bK545f8324
Max-Forwards: 70
From: "9191000000" <sip:9191000000@74.95.126.33>;tag=as129a6005
To: <sip:19190000000@sip.callwithus.com>
Contact: <sip:9191000000@74.95.126.33:5060>
Call-ID: 722f85362c3ef1df5e60d3e025456c82@74.95.126.33:5060
CSeq: 103 INVITE
User-Agent: Asterisk PBX 13.1.0
Proxy-Authorization: Digest username="yyyyyyyy", realm="sip.callwithus.com", algorithm=MD5, uri="sip:19190000000@sip.callwithus.com", nonce="", response=""
Date: Sun, 04 Jan 2015 21:27:22 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 356

v=0
o=root 991698651 991698652 IN IP4 74.95.126.33
s=Asterisk PBX 13.1.0
c=IN IP4 74.95.126.33
b=CT:384
t=0 0
m=audio 10560 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:150
a=sendrecv
m=video 10114 RTP/AVP 34
a=rtpmap:34 H263/90000
a=sendrecv

---
[Jan  4 13:27:22] VERBOSE[18565] chan_sip.c: 
<--- SIP read from UDP:204.74.213.5:5060 --->
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP 74.95.126.33:5060;branch=z9hG4bK545f8324
From: "9191000000" <sip:9191000000@74.95.126.33>;tag=as129a6005
To: <sip:19190000000@sip.callwithus.com>
Call-ID: 722f85362c3ef1df5e60d3e025456c82@74.95.126.33:5060
CSeq: 103 INVITE
Server: CWU SIP GW
Content-Length: 0

<------------->
[Jan  4 13:27:22] VERBOSE[18565] chan_sip.c: --- (8 headers 0 lines) ---
[Jan  4 13:27:26] VERBOSE[18565] chan_sip.c: 
<--- SIP read from UDP:204.74.213.5:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 74.95.126.33:5060;branch=z9hG4bK545f8324
Record-Route: <sip:204.74.213.5;lr=on;ftag=as129a6005>
From: "9191000000" <sip:9191000000@74.95.126.33>;tag=as129a6005
To: <sip:19190000000@sip.callwithus.com>;tag=806U5yKKmrByc
Call-ID: 722f85362c3ef1df5e60d3e025456c82@74.95.126.33:5060
CSeq: 103 INVITE
Contact: <sip:19190000000@37.187.29.231:5062;transport=udp>
User-Agent: CWU SIP GW
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, NOTIFY
Supported: timer, path, replaces
Allow-Events: talk, hold, conference, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 244

v=0
o=Sonus_UAC 9997 30030 IN IP4 69.85.185.222
s=SIP Media Capabilities
c=IN IP4 69.85.185.222
t=0 0
m=audio 28246 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
m=video 0 RTP/AVP 34
<------------->
[Jan  4 13:27:26] VERBOSE[18565] chan_sip.c: --- (16 headers 11 lines) ---
[Jan  4 13:27:26] VERBOSE[18565][C-00000000] sip/route.c: sip_route_dump: route/path hop: <sip:204.74.213.5;lr=on;ftag=as129a6005>
[Jan  4 13:27:26] VERBOSE[18565][C-00000000] chan_sip.c: Found RTP audio format 0
[Jan  4 13:27:26] VERBOSE[18565][C-00000000] chan_sip.c: Found RTP audio format 101
[Jan  4 13:27:26] VERBOSE[18565][C-00000000] chan_sip.c: Found audio description format PCMU for ID 0
[Jan  4 13:27:26] VERBOSE[18565][C-00000000] chan_sip.c: Found audio description format telephone-event for ID 101
[Jan  4 13:27:26] VERBOSE[18565][C-00000000] chan_sip.c: Capabilities: us - (ulaw|alaw|gsm|h263), peer - audio=(ulaw)/video=(nothing)/text=(nothing), combined - (ulaw)
[Jan  4 13:27:26] VERBOSE[18565][C-00000000] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
[Jan  4 13:27:26] VERBOSE[18565][C-00000000] chan_sip.c: Peer audio RTP is at port 69.85.185.222:28246
[Jan  4 13:27:26] VERBOSE[18565][C-00000000] chan_sip.c: Peer doesn't provide video
[Jan  4 13:27:26] VERBOSE[18654][C-00000000] chan_sip.c: 
<--- Transmitting (no NAT) to [2601:8:9181:4800:215:65ff:fe27:ac8e]:5062 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP [2601:8:9181:4800:215:65ff:fe27:ac8e]:5062;branch=z9hG4bK242854787;received=2601:8:9181:4800:215:65ff:fe27:ac8e
From: "Desk" <sip:yealinkphone@sip.ferree-clark.org>;tag=731755919
To: <sip:9190000000@sip.ferree-clark.org>;tag=as75cfe7e8
Call-ID: 2090487130@2601:8:9181:4800:215:65ff:fe27:ac8e
CSeq: 1 INVITE
Server: Asterisk PBX 13.1.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:9190000000@[2601:8:9181:4800:225:90ff:fe87:21f9]:5060>
Content-Length: 0


<------------>
[Jan  4 13:27:26] VERBOSE[18565] chan_sip.c: Reliably Transmitting (no NAT) to 217.14.138.127:5060:
OPTIONS sip:draytel.org SIP/2.0
Via: SIP/2.0/UDP 74.95.126.33:5060;branch=z9hG4bK5407cfec
Max-Forwards: 70
From: "asterisk" <sip:wwwwwww@74.95.126.33>;tag=as51535b94
To: <sip:draytel.org>
Contact: <sip:wwwwwww@74.95.126.33:5060>
Call-ID: 060615385d5201e53a6ff72e58b24a96@74.95.126.33:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 13.1.0
Date: Sun, 04 Jan 2015 21:27:26 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---
[Jan  4 13:27:26] VERBOSE[18565] chan_sip.c: Reliably Transmitting (no NAT) to [2601:8:9181:4800:215:65ff:fe27:ac8e]:5062:
OPTIONS sip:yealinkphone@[2601:8:9181:4800:215:65ff:fe27:ac8e]:5062 SIP/2.0
Via: SIP/2.0/UDP [2601:8:9181:4800:225:90ff:fe87:21f9]:5060;branch=z9hG4bK40629ea0
Max-Forwards: 70
From: "asterisk" <sip:asterisk@[2601:8:9181:4800:225:90ff:fe87:21f9]>;tag=as3c5dd84a
To: <sip:yealinkphone@[2601:8:9181:4800:215:65ff:fe27:ac8e]:5062>
Contact: <sip:asterisk@[2601:8:9181:4800:225:90ff:fe87:21f9]:5060>
Call-ID: 21d0b01816c96ee87f5cb4fc6f59e15e@[2601:8:9181:4800:225:90ff:fe87:21f9]:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 13.1.0
Date: Sun, 04 Jan 2015 21:27:26 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---
[Jan  4 13:27:26] VERBOSE[18565] chan_sip.c: Reliably Transmitting (no NAT) to 204.74.213.5:5060:
OPTIONS sip:sip.callwithus.com SIP/2.0
Via: SIP/2.0/UDP 74.95.126.33:5060;branch=z9hG4bK1060c1cd
Max-Forwards: 70
From: "asterisk" <sip:asterisk@74.95.126.33>;tag=as4bd9d6b7
To: <sip:sip.callwithus.com>
Contact: <sip:asterisk@74.95.126.33:5060>
Call-ID: 344802473216fc6b0959e2d4167f681c@74.95.126.33:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 13.1.0
Date: Sun, 04 Jan 2015 21:27:26 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---
[Jan  4 13:27:26] VERBOSE[18565] chan_sip.c: Reliably Transmitting (no NAT) to 64.2.142.28:5060:
OPTIONS sip:inbound3.vitelity.net SIP/2.0
Via: SIP/2.0/UDP 74.95.126.33:5060;branch=z9hG4bK2b72c4ac
Max-Forwards: 70
From: "asterisk" <sip:asterisk@74.95.126.33>;tag=as5b013e8c
To: <sip:inbound3.vitelity.net>
Contact: <sip:asterisk@74.95.126.33:5060>
Call-ID: 7b65923f05ecb47e44fb39475593faf4@74.95.126.33:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 13.1.0
Date: Sun, 04 Jan 2015 21:27:26 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---
[Jan  4 13:27:26] VERBOSE[18565] chan_sip.c: 
<--- SIP read from UDP:[2601:8:9181:4800:215:65ff:fe27:ac8e]:5062 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP [2601:8:9181:4800:225:90ff:fe87:21f9]:5060;branch=z9hG4bK40629ea0
From: "asterisk" <sip:asterisk@[2601:8:9181:4800:225:90ff:fe87:21f9]>;tag=as3c5dd84a
To: <sip:yealinkphone@[2601:8:9181:4800:215:65ff:fe27:ac8e]:5062>;tag=2502215134
Call-ID: 21d0b01816c96ee87f5cb4fc6f59e15e@[2601:8:9181:4800:225:90ff:fe87:21f9]:5060
CSeq: 102 OPTIONS
User-Agent: Yealink SIP-T20P 9.73.0.40
Content-Length: 0

<------------->
[Jan  4 13:27:26] VERBOSE[18565] chan_sip.c: --- (8 headers 0 lines) ---
[Jan  4 13:27:26] VERBOSE[18565] chan_sip.c: Really destroying SIP dialog '21d0b01816c96ee87f5cb4fc6f59e15e@[2601:8:9181:4800:225:90ff:fe87:21f9]:5060' Method: OPTIONS
[Jan  4 13:27:26] VERBOSE[18565] chan_sip.c: 
<--- SIP read from UDP:204.74.213.5:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 74.95.126.33:5060;branch=z9hG4bK1060c1cd
From: "asterisk" <sip:asterisk@74.95.126.33>;tag=as4bd9d6b7
To: <sip:sip.callwithus.com>;tag=05fa965d41f3adc51e16f9a7acf1c273.5406
Call-ID: 344802473216fc6b0959e2d4167f681c@74.95.126.33:5060
CSeq: 102 OPTIONS
Server: CWU SIP GW
Content-Length: 0

<------------->
[Jan  4 13:27:26] VERBOSE[18565] chan_sip.c: --- (8 headers 0 lines) ---
[Jan  4 13:27:26] VERBOSE[18565] chan_sip.c: Really destroying SIP dialog '344802473216fc6b0959e2d4167f681c@74.95.126.33:5060' Method: OPTIONS
[Jan  4 13:27:26] VERBOSE[18565] chan_sip.c: 
<--- SIP read from UDP:64.2.142.28:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 74.95.126.33:5060;branch=z9hG4bK2b72c4ac;received=74.95.126.33
From: "asterisk" <sip:asterisk@74.95.126.33>;tag=as5b013e8c
To: <sip:inbound3.vitelity.net>;tag=as2577333a
Call-ID: 7b65923f05ecb47e44fb39475593faf4@74.95.126.33:5060
CSeq: 102 OPTIONS
User-Agent: packetrino
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Accept: application/sdp
Content-Length: 0

<------------->
[Jan  4 13:27:26] VERBOSE[18565] chan_sip.c: --- (11 headers 0 lines) ---
[Jan  4 13:27:26] VERBOSE[18565] chan_sip.c: Really destroying SIP dialog '7b65923f05ecb47e44fb39475593faf4@74.95.126.33:5060' Method: OPTIONS
[Jan  4 13:27:26] VERBOSE[18565] chan_sip.c: Reliably Transmitting (no NAT) to 217.10.79.23:5060:
OPTIONS sip:sipgate.co.uk SIP/2.0
Via: SIP/2.0/UDP 74.95.126.33:5060;branch=z9hG4bK51360f40
Max-Forwards: 70
From: "asterisk" <sip:zzzzzzz@74.95.126.33>;tag=as3ae6d9ba
To: <sip:sipgate.co.uk>
Contact: <sip:zzzzzzz@74.95.126.33:5060>
Call-ID: 0e61df024e55627c30e2df14730d9966@74.95.126.33:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 13.1.0
Date: Sun, 04 Jan 2015 21:27:26 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---
[Jan  4 13:27:26] VERBOSE[18565] chan_sip.c: 
<--- SIP read from UDP:217.14.138.127:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 74.95.126.33:5060;branch=z9hG4bK5407cfec
From: "asterisk" <sip:wwwwwww@74.95.126.33>;tag=as51535b94
To: <sip:draytel.org>;tag=dea7ace6b651c43aa467a5e170703289.24e1
Call-ID: 060615385d5201e53a6ff72e58b24a96@74.95.126.33:5060
CSeq: 102 OPTIONS
Server: OpenSIPS (1.5.3-notls (x86_64/linux))
Content-Length: 0

<------------->
[Jan  4 13:27:26] VERBOSE[18565] chan_sip.c: --- (8 headers 0 lines) ---
[Jan  4 13:27:26] VERBOSE[18565] chan_sip.c: Really destroying SIP dialog '060615385d5201e53a6ff72e58b24a96@74.95.126.33:5060' Method: OPTIONS
[Jan  4 13:27:26] VERBOSE[18565] chan_sip.c: Reliably Transmitting (no NAT) to 64.2.142.106:5060:
OPTIONS sip:outbound1.vitelity.net SIP/2.0
Via: SIP/2.0/UDP 74.95.126.33:5060;branch=z9hG4bK004765a1
Max-Forwards: 70
From: "asterisk" <sip:user@74.95.126.33>;tag=as193929b2
To: <sip:outbound1.vitelity.net>
Contact: <sip:user@74.95.126.33:5060>
Call-ID: 13434e5955793ee40865c3c20c970d2f@74.95.126.33:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 13.1.0
Date: Sun, 04 Jan 2015 21:27:26 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---
[Jan  4 13:27:26] VERBOSE[18565] chan_sip.c: 
<--- SIP read from UDP:64.2.142.106:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 74.95.126.33:5060;branch=z9hG4bK004765a1;received=74.95.126.33
From: "asterisk" <sip:user@74.95.126.33>;tag=as193929b2
To: <sip:outbound1.vitelity.net>;tag=as0b30662c
Call-ID: 13434e5955793ee40865c3c20c970d2f@74.95.126.33:5060
CSeq: 102 OPTIONS
User-Agent: packetrino
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Accept: application/sdp
Content-Length: 0

<------------->
[Jan  4 13:27:26] VERBOSE[18565] chan_sip.c: --- (11 headers 0 lines) ---
[Jan  4 13:27:26] VERBOSE[18565] chan_sip.c: Really destroying SIP dialog '13434e5955793ee40865c3c20c970d2f@74.95.126.33:5060' Method: OPTIONS
[Jan  4 13:27:26] VERBOSE[18565] chan_sip.c: 
<--- SIP read from UDP:217.10.79.23:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 74.95.126.33:5060;branch=z9hG4bK51360f40
From: "asterisk" <sip:zzzzzzz@74.95.126.33>;tag=as3ae6d9ba
To: <sip:sipgate.co.uk>;tag=6557515e56873518037b90874afb064e.233b
Call-ID: 0e61df024e55627c30e2df14730d9966@74.95.126.33:5060
CSeq: 102 OPTIONS
Accept: */*
Accept-Encoding: 
Accept-Language: en
Supported: 
Content-Length: 0

<------------->
[Jan  4 13:27:26] VERBOSE[18565] chan_sip.c: --- (11 headers 0 lines) ---
[Jan  4 13:27:26] VERBOSE[18565] chan_sip.c: Really destroying SIP dialog '0e61df024e55627c30e2df14730d9966@74.95.126.33:5060' Method: OPTIONS
[Jan  4 13:27:31] VERBOSE[18565] chan_sip.c: 
<--- SIP read from UDP:204.74.213.5:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 74.95.126.33:5060;branch=z9hG4bK545f8324
Record-Route: <sip:204.74.213.5;lr=on;ftag=as129a6005>
From: "9191000000" <sip:9191000000@74.95.126.33>;tag=as129a6005
To: <sip:19190000000@sip.callwithus.com>;tag=806U5yKKmrByc
Call-ID: 722f85362c3ef1df5e60d3e025456c82@74.95.126.33:5060
CSeq: 103 INVITE
Contact: <sip:19190000000@37.187.29.231:5062;transport=udp>
User-Agent: CWU SIP GW
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, NOTIFY
Require: timer
Supported: timer, path, replaces
Allow-Events: talk, hold, conference, refer
Session-Expires: 1800;refresher=uac
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 244

v=0
o=Sonus_UAC 9997 30030 IN IP4 69.85.185.222
s=SIP Media Capabilities
c=IN IP4 69.85.185.222
t=0 0
m=audio 28246 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
m=video 0 RTP/AVP 34
<------------->
[Jan  4 13:27:31] VERBOSE[18565] chan_sip.c: --- (17 headers 11 lines) ---
[Jan  4 13:27:31] VERBOSE[18565][C-00000000] sip/route.c: sip_route_dump: route/path hop: <sip:204.74.213.5;lr=on;ftag=as129a6005>
[Jan  4 13:27:31] VERBOSE[18565][C-00000000] chan_sip.c: set_destination: Parsing <sip:204.74.213.5;lr=on;ftag=as129a6005> for address/port to send to
[Jan  4 13:27:31] VERBOSE[18565][C-00000000] chan_sip.c: set_destination: set destination to 204.74.213.5:5060
[Jan  4 13:27:31] VERBOSE[18565][C-00000000] chan_sip.c: Transmitting (no NAT) to 204.74.213.5:5060:
ACK sip:19190000000@37.187.29.231:5062;transport=udp SIP/2.0
Via: SIP/2.0/UDP 74.95.126.33:5060;branch=z9hG4bK716e1731
Route: <sip:204.74.213.5;lr=on;ftag=as129a6005>
Max-Forwards: 70
From: "9191000000" <sip:9191000000@74.95.126.33>;tag=as129a6005
To: <sip:19190000000@sip.callwithus.com>;tag=806U5yKKmrByc
Contact: <sip:9191000000@74.95.126.33:5060>
Call-ID: 722f85362c3ef1df5e60d3e025456c82@74.95.126.33:5060
CSeq: 103 ACK
User-Agent: Asterisk PBX 13.1.0
Content-Length: 0


---
[Jan  4 13:27:31] VERBOSE[18654][C-00000000] chan_sip.c: Audio is at 10722
[Jan  4 13:27:31] VERBOSE[18654][C-00000000] chan_sip.c: Adding codec g729 to SDP
[Jan  4 13:27:31] VERBOSE[18654][C-00000000] chan_sip.c: Adding codec ulaw to SDP
[Jan  4 13:27:31] VERBOSE[18654][C-00000000] chan_sip.c: Adding codec alaw to SDP
[Jan  4 13:27:31] VERBOSE[18654][C-00000000] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
[Jan  4 13:27:31] VERBOSE[18654][C-00000000] chan_sip.c: 
<--- Reliably Transmitting (no NAT) to [2601:8:9181:4800:215:65ff:fe27:ac8e]:5062 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP [2601:8:9181:4800:215:65ff:fe27:ac8e]:5062;branch=z9hG4bK242854787;received=2601:8:9181:4800:215:65ff:fe27:ac8e
From: "Desk" <sip:yealinkphone@sip.ferree-clark.org>;tag=731755919
To: <sip:9190000000@sip.ferree-clark.org>;tag=as75cfe7e8
Call-ID: 2090487130@2601:8:9181:4800:215:65ff:fe27:ac8e
CSeq: 1 INVITE
Server: Asterisk PBX 13.1.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:9190000000@[2601:8:9181:4800:225:90ff:fe87:21f9]:5060>
Content-Type: application/sdp
Content-Length: 356

v=0
o=root 1992844822 1992844822 IN IP6 2601:8:9181:4800:225:90ff:fe87:21f9
s=Asterisk PBX 13.1.0
c=IN IP6 2601:8:9181:4800:225:90ff:fe87:21f9
t=0 0
m=audio 10722 RTP/AVP 18 0 8 101
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:150
a=sendrecv

<------------>
[Jan  4 13:27:31] VERBOSE[18565] chan_sip.c: 
<--- SIP read from UDP:[2601:8:9181:4800:215:65ff:fe27:ac8e]:5062 --->
ACK sip:9190000000@[2601:8:9181:4800:225:90ff:fe87:21f9]:5060 SIP/2.0
Via: SIP/2.0/UDP [2601:8:9181:4800:215:65ff:fe27:ac8e]:5062;branch=z9hG4bK3837222386
From: "Desk" <sip:yealinkphone@sip.ferree-clark.org>;tag=731755919
To: <sip:9190000000@sip.ferree-clark.org>;tag=as75cfe7e8
Call-ID: 2090487130@2601:8:9181:4800:215:65ff:fe27:ac8e
CSeq: 1 ACK
Contact: <sip:yealinkphone@[2601:8:9181:4800:215:65ff:fe27:ac8e]:5062>
Max-Forwards: 70
User-Agent: Yealink SIP-T20P 9.73.0.40
Content-Length: 0

<------------->
[Jan  4 13:27:31] VERBOSE[18565] chan_sip.c: --- (10 headers 0 lines) ---
[Jan  4 13:27:34] VERBOSE[18565] chan_sip.c: Really destroying SIP dialog 'D0Kv3mSglUnxIfM3cK1OyaYh7ItYedXg' Method: REGISTER
[Jan  4 13:27:46] VERBOSE[18565] chan_sip.c: 
<--- SIP read from UDP:[2601:8:9181:4800:215:65ff:fe27:ac8e]:5062 --->


<------------->
[Jan  4 13:27:48] VERBOSE[18565] chan_sip.c: 
<--- SIP read from UDP:204.74.213.5:5060 --->
BYE sip:9191000000@74.95.126.33:5060 SIP/2.0
Via: SIP/2.0/UDP 204.74.213.5;branch=z9hG4bK0815.b0cf769944d4eb5b9c1826aec624a9fc.0
Via: SIP/2.0/UDP 37.187.29.231:5062;rport=5062;branch=z9hG4bK0etve86j8eaBS
Max-Forwards: 16
From: <sip:19190000000@sip.callwithus.com>;tag=806U5yKKmrByc
To: "9191000000" <sip:9191000000@74.95.126.33>;tag=as129a6005
Call-ID: 722f85362c3ef1df5e60d3e025456c82@74.95.126.33:5060
CSeq: 69867370 BYE
User-Agent: CWU SIP GW
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, NOTIFY
Supported: timer, path, replaces
Content-Length: 0

<------------->
[Jan  4 13:27:48] VERBOSE[18565] chan_sip.c: --- (12 headers 0 lines) ---
[Jan  4 13:27:48] VERBOSE[18565][C-00000000] chan_sip.c: Sending to 204.74.213.5:5060 (no NAT)
[Jan  4 13:27:48] VERBOSE[18565][C-00000000] chan_sip.c: Scheduling destruction of SIP dialog '722f85362c3ef1df5e60d3e025456c82@74.95.126.33:5060' in 6400 ms (Method: BYE)
[Jan  4 13:27:48] VERBOSE[18565][C-00000000] chan_sip.c: 
<--- Transmitting (no NAT) to 204.74.213.5:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 204.74.213.5;branch=z9hG4bK0815.b0cf769944d4eb5b9c1826aec624a9fc.0;received=204.74.213.5
Via: SIP/2.0/UDP 37.187.29.231:5062;rport=5062;branch=z9hG4bK0etve86j8eaBS
From: <sip:19190000000@sip.callwithus.com>;tag=806U5yKKmrByc
To: "9191000000" <sip:9191000000@74.95.126.33>;tag=as129a6005
Call-ID: 722f85362c3ef1df5e60d3e025456c82@74.95.126.33:5060
CSeq: 69867370 BYE
Server: Asterisk PBX 13.1.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


<------------>
[Jan  4 13:27:48] VERBOSE[18654][C-00000000] chan_sip.c: Scheduling destruction of SIP dialog '2090487130@2601:8:9181:4800:215:65ff:fe27:ac8e' in 6400 ms (Method: ACK)
[Jan  4 13:27:48] VERBOSE[18654][C-00000000] chan_sip.c: set_destination: Parsing <sip:yealinkphone@[2601:8:9181:4800:215:65ff:fe27:ac8e]:5062> for address/port to send to
[Jan  4 13:27:48] VERBOSE[18654][C-00000000] chan_sip.c: set_destination: set destination to [2601:8:9181:4800:215:65ff:fe27:ac8e]:5062
[Jan  4 13:27:48] VERBOSE[18654][C-00000000] chan_sip.c: Reliably Transmitting (no NAT) to [2601:8:9181:4800:215:65ff:fe27:ac8e]:5062:
BYE sip:yealinkphone@[2601:8:9181:4800:215:65ff:fe27:ac8e]:5062 SIP/2.0
Via: SIP/2.0/UDP [2601:8:9181:4800:225:90ff:fe87:21f9]:5060;branch=z9hG4bK48f06251
Max-Forwards: 70
From: <sip:9190000000@sip.ferree-clark.org>;tag=as75cfe7e8
To: "Desk" <sip:yealinkphone@sip.ferree-clark.org>;tag=731755919
Call-ID: 2090487130@2601:8:9181:4800:215:65ff:fe27:ac8e
CSeq: 102 BYE
User-Agent: Asterisk PBX 13.1.0
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


---
[Jan  4 13:27:48] VERBOSE[18565] chan_sip.c: 
<--- SIP read from UDP:[2601:8:9181:4800:215:65ff:fe27:ac8e]:5062 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP [2601:8:9181:4800:225:90ff:fe87:21f9]:5060;branch=z9hG4bK48f06251
From: <sip:9190000000@sip.ferree-clark.org>;tag=as75cfe7e8
To: "Desk" <sip:yealinkphone@sip.ferree-clark.org>;tag=731755919
Call-ID: 2090487130@2601:8:9181:4800:215:65ff:fe27:ac8e
CSeq: 102 BYE
User-Agent: Yealink SIP-T20P 9.73.0.40
Content-Length: 0

<------------->
[Jan  4 13:27:48] VERBOSE[18565] chan_sip.c: --- (8 headers 0 lines) ---
[Jan  4 13:27:48] VERBOSE[18565][C-00000000] chan_sip.c: SIP Response message for INCOMING dialog BYE arrived
[Jan  4 13:27:48] VERBOSE[18565] chan_sip.c: Really destroying SIP dialog '2090487130@2601:8:9181:4800:215:65ff:fe27:ac8e' Method: ACK
[Jan  4 13:27:54] VERBOSE[18565] chan_sip.c: Really destroying SIP dialog '722f85362c3ef1df5e60d3e025456c82@74.95.126.33:5060' Method: BYE
[Jan  4 13:27:57] NOTICE[18565] chan_sip.c:    -- Re-registration for  user@inbound3.vitelity.net
[Jan  4 13:27:57] VERBOSE[18565] chan_sip.c: REGISTER 12 headers, 0 lines
[Jan  4 13:27:57] VERBOSE[18565] chan_sip.c: Reliably Transmitting (no NAT) to 64.2.142.28:5060:
REGISTER sip:inbound3.vitelity.net SIP/2.0
Via: SIP/2.0/UDP 74.95.126.33:5060;branch=z9hG4bK7414349e
Max-Forwards: 70
From: <sip:user@inbound3.vitelity.net>;tag=as213a2f69
To: <sip:user@inbound3.vitelity.net>
Call-ID: 73a7a2971b31631e69b3419d0ad92b1f@10.1.10.2
CSeq: 106 REGISTER
Supported: replaces, timer
User-Agent: Asterisk PBX 13.1.0
Authorization: Digest username="user", realm="asterisk", algorithm=MD5, uri="sip:inbound3.vitelity.net", nonce="", response=""
Expires: 120
Contact: <sip:s@74.95.126.33:5060>
Content-Length: 0


---
[Jan  4 13:27:57] VERBOSE[18565] chan_sip.c: 
<--- SIP read from UDP:64.2.142.28:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 74.95.126.33:5060;branch=z9hG4bK7414349e;received=74.95.126.33
From: <sip:user@inbound3.vitelity.net>;tag=as213a2f69
To: <sip:user@inbound3.vitelity.net>
Call-ID: 73a7a2971b31631e69b3419d0ad92b1f@10.1.10.2
CSeq: 106 REGISTER
User-Agent: packetrino
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0

<------------->
[Jan  4 13:27:57] VERBOSE[18565] chan_sip.c: --- (10 headers 0 lines) ---
[Jan  4 13:27:57] VERBOSE[18565] chan_sip.c: 
<--- SIP read from UDP:64.2.142.28:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 74.95.126.33:5060;branch=z9hG4bK7414349e;received=74.95.126.33
From: <sip:user@inbound3.vitelity.net>;tag=as213a2f69
To: <sip:user@inbound3.vitelity.net>;tag=as03e090b7
Call-ID: 73a7a2971b31631e69b3419d0ad92b1f@10.1.10.2
CSeq: 106 REGISTER
User-Agent: packetrino
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce=""
Content-Length: 0

<------------->
[Jan  4 13:27:57] VERBOSE[18565] chan_sip.c: --- (11 headers 0 lines) ---
[Jan  4 13:27:57] VERBOSE[18565] chan_sip.c: Responding to challenge, registration to domain/host name inbound3.vitelity.net
[Jan  4 13:27:57] VERBOSE[18565] chan_sip.c: REGISTER 12 headers, 0 lines
[Jan  4 13:27:57] VERBOSE[18565] chan_sip.c: Reliably Transmitting (no NAT) to 64.2.142.28:5060:
REGISTER sip:inbound3.vitelity.net SIP/2.0
Via: SIP/2.0/UDP 74.95.126.33:5060;branch=z9hG4bK72b8ad64
Max-Forwards: 70
From: <sip:user@inbound3.vitelity.net>;tag=as213a2f69
To: <sip:user@inbound3.vitelity.net>
Call-ID: 73a7a2971b31631e69b3419d0ad92b1f@10.1.10.2
CSeq: 107 REGISTER
Supported: replaces, timer
User-Agent: Asterisk PBX 13.1.0
Authorization: Digest username="user", realm="asterisk", algorithm=MD5, uri="sip:inbound3.vitelity.net", nonce="", response=""
Expires: 120
Contact: <sip:s@74.95.126.33:5060>
Content-Length: 0


---
[Jan  4 13:27:57] VERBOSE[18565] chan_sip.c: 
<--- SIP read from UDP:64.2.142.28:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 74.95.126.33:5060;branch=z9hG4bK72b8ad64;received=74.95.126.33
From: <sip:user@inbound3.vitelity.net>;tag=as213a2f69
To: <sip:user@inbound3.vitelity.net>
Call-ID: 73a7a2971b31631e69b3419d0ad92b1f@10.1.10.2
CSeq: 107 REGISTER
User-Agent: packetrino
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0

<------------->
[Jan  4 13:27:57] VERBOSE[18565] chan_sip.c: --- (10 headers 0 lines) ---
[Jan  4 13:27:57] VERBOSE[18565] chan_sip.c: 
<--- SIP read from UDP:64.2.142.28:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 74.95.126.33:5060;branch=z9hG4bK72b8ad64;received=74.95.126.33
From: <sip:user@inbound3.vitelity.net>;tag=as213a2f69
To: <sip:user@inbound3.vitelity.net>;tag=as03e090b7
Call-ID: 73a7a2971b31631e69b3419d0ad92b1f@10.1.10.2
CSeq: 107 REGISTER
User-Agent: packetrino
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Expires: 60
Contact: <sip:s@74.95.126.33:5060>;expires=60
Date: Sun, 04 Jan 2015 21:27:56 GMT
Content-Length: 0

<------------->
[Jan  4 13:27:57] VERBOSE[18565] chan_sip.c: --- (13 headers 0 lines) ---
[Jan  4 13:27:57] NOTICE[18565] chan_sip.c: Outbound Registration: Expiry for inbound3.vitelity.net is 60 sec (Scheduling reregistration in 45 s)
[Jan  4 13:27:57] VERBOSE[18565] chan_sip.c: Really destroying SIP dialog '73a7a2971b31631e69b3419d0ad92b1f@10.1.10.2' Method: REGISTER
[Jan  4 13:27:57] NOTICE[18565] chan_sip.c:    -- Re-registration for  xxxxx@callcentric.com
[Jan  4 13:27:57] VERBOSE[18565] chan_sip.c: REGISTER 12 headers, 0 lines
[Jan  4 13:27:57] VERBOSE[18565] chan_sip.c: Reliably Transmitting (no NAT) to 204.11.192.171:5080:
REGISTER sip:callcentric.com:5080 SIP/2.0
Via: SIP/2.0/UDP 74.95.126.33:5060;branch=z9hG4bK5f6f12c8
Max-Forwards: 70
From: <sip:xxxxx@callcentric.com>;tag=as6901a220
To: <sip:xxxxx@callcentric.com>
Call-ID: 09af2c8612ae94451cfae9bb6dc4b35e@10.1.10.2
CSeq: 105 REGISTER
Supported: replaces, timer
User-Agent: Asterisk PBX 13.1.0
Authorization: Digest username="xxxxx", realm="callcentric.com", algorithm=MD5, uri="sip:sip:callcentric.com", nonce="", response=""
Expires: 120
Contact: <sip:19140000000@74.95.126.33:5060>
Content-Length: 0


---
[Jan  4 13:27:58] VERBOSE[18565] chan_sip.c: 
<--- SIP read from UDP:204.11.192.171:5080 --->
SIP/2.0 200 Ok
v: SIP/2.0/UDP 74.95.126.33:5060;branch=z9hG4bK5f6f12c8
f: <sip:xxxxx@callcentric.com>;tag=as6901a220
t: <sip:xxxxx@callcentric.com>
i: 09af2c8612ae94451cfae9bb6dc4b35e@10.1.10.2
CSeq: 105 REGISTER
m: <sip:19140000000@74.95.126.33:5060>;expires=62
l: 0

<------------->
[Jan  4 13:27:58] VERBOSE[18565] chan_sip.c: --- (8 headers 0 lines) ---
[Jan  4 13:27:58] NOTICE[18565] chan_sip.c: Outbound Registration: Expiry for callcentric.com is 62 sec (Scheduling reregistration in 47 s)
[Jan  4 13:27:58] VERBOSE[18565] chan_sip.c: Really destroying SIP dialog '09af2c8612ae94451cfae9bb6dc4b35e@10.1.10.2' Method: REGISTER
[Jan  4 13:28:11] VERBOSE[18532] asterisk.c: Asterisk cleanly ending (0).
[Jan  4 13:28:11] VERBOSE[18532] asterisk.c: Executing last minute cleanups