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SIP Communication Log for 192.168.8.101

The document contains a detailed SIP (Session Initiation Protocol) communication log involving an INVITE request for a call to a specific SIP address. It includes various SIP headers, audio codec information, and responses from both the Asterisk PBX and a provider's server. Additionally, it captures the process of establishing a call and handling presence information with PUBLISH and SUBSCRIBE methods.

Uploaded by

Alaya Naqvi
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© © All Rights Reserved
We take content rights seriously. If you suspect this is your content, claim it here.
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Download as TXT, PDF, TXT or read online on Scribd
28 views17 pages

SIP Communication Log for 192.168.8.101

The document contains a detailed SIP (Session Initiation Protocol) communication log involving an INVITE request for a call to a specific SIP address. It includes various SIP headers, audio codec information, and responses from both the Asterisk PBX and a provider's server. Additionally, it captures the process of establishing a call and handling presence information with PUBLISH and SUBSCRIBE methods.

Uploaded by

Alaya Naqvi
Copyright
© © All Rights Reserved
We take content rights seriously. If you suspect this is your content, claim it here.
Available Formats
Download as TXT, PDF, TXT or read online on Scribd

<--- SIP read from UDP:103.255.4.

17:15136 --->
INVITE sip:0060123718121@[Link];transport=UDP SIP/2.0
Via: SIP/2.0/UDP [Link]:44591;branch=z9hG4bK-d8754z-5b4430db9a8615b8-1---
d8754z-
Max-Forwards: 70
Contact: <sip:5020@[Link]:44591;transport=UDP>
To: <sip:0060123718121@[Link];transport=UDP>
From: <sip:5020@[Link];transport=UDP>;tag=a566fb5c
Call-ID: NTgxZjYxNDE2NjRhYWNmNTEzZWVlMTNlOGI2YjE3ZjQ.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/sdp
Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri
User-Agent: Z 3.6.25251 r25476
Allow-Events: presence, kpml
Content-Length: 241

v=0
o=Z 0 0 IN IP4 [Link]
s=Z
c=IN IP4 [Link]
t=0 0
m=audio 8000 RTP/AVP 98 3 110 8 0 101
a=rtpmap:98 iLBC/8000
a=fmtp:98 mode=20
a=rtpmap:110 speex/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->
--- (14 headers 12 lines) ---
Sending to [Link]:15136 (NAT)
Sending to [Link]:15136 (NAT)
Using INVITE request as basis request -
NTgxZjYxNDE2NjRhYWNmNTEzZWVlMTNlOGI2YjE3ZjQ.
Found peer '5020' for '5020' from [Link]:15136
== Using SIP RTP CoS mark 5
Found RTP audio format 98
Found RTP audio format 3
Found RTP audio format 110
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 101
Found audio description format iLBC for ID 98
Found audio description format speex for ID 110
Found audio description format telephone-event for ID 101
Capabilities: us - (gsm|ilbc), peer -
audio=(gsm|ulaw|alaw|speex|ilbc)/video=(nothing)/text=(nothing), combined - (gsm|
ilbc)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-
event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port [Link]:8000
Looking for 0060123718121 in vtiger_inbound (domain [Link])
list_route: hop: <sip:5020@[Link]:44591;transport=UDP>

<--- Transmitting (NAT) to [Link]:15136 --->


SIP/2.0 100 Trying
Via: SIP/2.0/UDP [Link]:44591;branch=z9hG4bK-d8754z-5b4430db9a8615b8-1---
d8754z-;received=[Link];rport=15136
From: <sip:5020@[Link];transport=UDP>;tag=a566fb5c
To: <sip:0060123718121@[Link];transport=UDP>
Call-ID: NTgxZjYxNDE2NjRhYWNmNTEzZWVlMTNlOGI2YjE3ZjQ.
CSeq: 1 INVITE
Server: Asterisk PBX 11.13.1~dfsg-2+deb8u1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH,
MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:0060123718121@[Link]>
Content-Length: 0

<------------>
-- Executing [0060123718121@vtiger_inbound:1] Dial("SIP/5020-0000001a",
"SIP/provider/0060123718121,60") in new stack
== Using SIP RTP CoS mark 5
Audio is at 13036
Adding codec 100010 (ilbc) to SDP
Adding codec 100002 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to [Link]:
INVITE sip:0060123718121@[Link] SIP/2.0
Via: SIP/2.0/UDP [Link];branch=z9hG4bK00fe2662;rport
Max-Forwards: 70
From: <sip:605605001@[Link]>;tag=as1c55888d
To: <sip:0060123718121@[Link]>
Contact: <sip:605605001@[Link]>
Call-ID: 731eb3b30a08e30c5a2faed25f671a91@[Link]
CSeq: 102 INVITE
User-Agent: Asterisk PBX 11.13.1~dfsg-2+deb8u1
Date: Fri, 12 May 2017 [Link] GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH,
MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 298

v=0
o=root 1472336577 1472336577 IN IP4 [Link]
s=Asterisk PBX 11.13.1~dfsg-2+deb8u1
c=IN IP4 [Link]
t=0 0
m=audio 13036 RTP/AVP 97 3 101
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
-- Called SIP/provider/0060123718121

<--- SIP read from UDP:[Link] --->


SIP/2.0 100 Giving a try
Via: SIP/2.0/UDP
[Link];received=[Link];branch=z9hG4bK00fe2662;rport=5261
From: <sip:605605001@[Link]>;tag=as1c55888d
To: <sip:0060123718121@[Link]>
Call-ID: 731eb3b30a08e30c5a2faed25f671a91@[Link]
CSeq: 102 INVITE
Server: OpenSIPS (1.10.0beta-notls (x86_64/linux))
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
> 0x7f003400bf50 -- Probation passed - setting RTP source address to
[Link]:11522

<--- SIP read from UDP:[Link]:15136 --->


PUBLISH sip:5020@[Link];transport=UDP SIP/2.0
Via: SIP/2.0/UDP [Link]:44591;branch=z9hG4bK-d8754z-fca8f6117ebfa171-1---
d8754z-
Max-Forwards: 70
Contact: <sip:5020@[Link]:44591;transport=UDP>
To: <sip:5020@[Link];transport=UDP>
From: <sip:5020@[Link];transport=UDP>;tag=df31695b
Call-ID: MTRkMTE3NzVlZmMxMWY1ZTk3Zjk2ZDVmYmNlMTY2NWY.
CSeq: 1 PUBLISH
Expires: 600
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/pidf+xml
Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri
User-Agent: Z 3.6.25251 r25476
Event: presence
Allow-Events: presence, kpml
Content-Length: 273

<?xml version="1.0" encoding="UTF-8"?>


<presence xmlns="urn:ietf:params:xml:ns:pidf"
entity="sip:5020@[Link];transport=UDP"> <tuple id="5020" >
<status><basic>open</basic></status> <note>On the phone</note> </tuple>
</presence>
<------------->
--- (16 headers 3 lines) ---
Sending to [Link]:15136 (NAT)

<--- Transmitting (NAT) to [Link]:15136 --->


SIP/2.0 489 Bad Event
Via: SIP/2.0/UDP [Link]:44591;branch=z9hG4bK-d8754z-fca8f6117ebfa171-1---
d8754z-;received=[Link];rport=15136
From: <sip:5020@[Link];transport=UDP>;tag=df31695b
To: <sip:5020@[Link];transport=UDP>;tag=as1a9149c7
Call-ID: MTRkMTE3NzVlZmMxMWY1ZTk3Zjk2ZDVmYmNlMTY2NWY.
CSeq: 1 PUBLISH
Server: Asterisk PBX 11.13.1~dfsg-2+deb8u1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH,
MESSAGE
Supported: replaces, timer
Content-Length: 0

<------------>
Really destroying SIP dialog 'MTRkMTE3NzVlZmMxMWY1ZTk3Zjk2ZDVmYmNlMTY2NWY.' Method:
PUBLISH
<--- SIP read from UDP:[Link] --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP
[Link];received=[Link];branch=z9hG4bK00fe2662;rport=5261
Record-Route: <sip:[Link];lr;ftag=as1c55888d;did=05a.23915b14>
From: <sip:605605001@[Link]>;tag=as1c55888d
To: <sip:0060123718121@[Link]>;tag=as636c8a82
Call-ID: 731eb3b30a08e30c5a2faed25f671a91@[Link]
CSeq: 102 INVITE
User-Agent: SoftSwitch
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: <sip:0060123718121@[Link]>
Content-Type: application/sdp
Content-Length: 288

v=0
o=root 31107 31107 IN IP4 [Link]
s=session
c=IN IP4 [Link]
t=0 0
m=audio 11522 RTP/AVP 97 3 101
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
<------------->
--- (13 headers 14 lines) ---
Found RTP audio format 97
Found RTP audio format 3
Found RTP audio format 101
Found audio description format iLBC for ID 97
Found audio description format GSM for ID 3
Found audio description format telephone-event for ID 101
Capabilities: us - (gsm|ilbc), peer -
audio=(gsm|ilbc)/video=(nothing)/text=(nothing), combined - (gsm|ilbc)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-
event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port [Link]:11522
list_route: hop: <sip:[Link];lr;ftag=as1c55888d;did=05a.23915b14>
set_destination: Parsing <sip:[Link];lr;ftag=as1c55888d;did=05a.23915b14>
for address/port to send to
set_destination: set destination to [Link]
Transmitting (NAT) to [Link]:
ACK sip:0060123718121@[Link] SIP/2.0
Via: SIP/2.0/UDP [Link];branch=z9hG4bK4910089e;rport
Route: <sip:[Link];lr;ftag=as1c55888d;did=05a.23915b14>
Max-Forwards: 70
From: <sip:605605001@[Link]>;tag=as1c55888d
To: <sip:0060123718121@[Link]>;tag=as636c8a82
Contact: <sip:605605001@[Link]>
Call-ID: 731eb3b30a08e30c5a2faed25f671a91@[Link]
CSeq: 102 ACK
User-Agent: Asterisk PBX 11.13.1~dfsg-2+deb8u1
Content-Length: 0

---

<--- SIP read from UDP:[Link]:15136 --->


SUBSCRIBE sip:5020@[Link];transport=UDP SIP/2.0
Via: SIP/2.0/UDP [Link]:44591;branch=z9hG4bK-d8754z-069200ee8b145c71-1---
d8754z-
Max-Forwards: 70
Contact: <sip:5020@[Link]:44591;transport=UDP>
To: <sip:5020@[Link];transport=UDP>
From: <sip:5020@[Link];transport=UDP>;tag=b7022909
Call-ID: MDJmN2FhNjk2ZGFjMmQ3OTkyNjk2ODc5NTQ5MDU5OTE.
CSeq: 1 SUBSCRIBE
Expires: 600
Accept: application/watcherinfo+xml
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri
User-Agent: Z 3.6.25251 r25476
Event: [Link]
Allow-Events: presence, kpml
Content-Length: 0

<------------->
--- (16 headers 0 lines) ---
Sending to [Link]:15136 (NAT)
Creating new subscription
Sending to [Link]:15136 (NAT)
list_route: hop: <sip:5020@[Link]:44591;transport=UDP>
Found peer '5020' for '5020' from [Link]:15136

<--- Transmitting (NAT) to [Link]:15136 --->


SIP/2.0 489 Bad Event
Via: SIP/2.0/UDP [Link]:44591;branch=z9hG4bK-d8754z-069200ee8b145c71-1---
d8754z-;received=[Link];rport=15136
From: <sip:5020@[Link];transport=UDP>;tag=b7022909
To: <sip:5020@[Link];transport=UDP>;tag=as01c4ca37
Call-ID: MDJmN2FhNjk2ZGFjMmQ3OTkyNjk2ODc5NTQ5MDU5OTE.
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX 11.13.1~dfsg-2+deb8u1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH,
MESSAGE
Supported: replaces, timer
Content-Length: 0

<------------>
Really destroying SIP dialog 'MDJmN2FhNjk2ZGFjMmQ3OTkyNjk2ODc5NTQ5MDU5OTE.' Method:
SUBSCRIBE
-- SIP/provider-0000001b answered SIP/5020-0000001a

<--- SIP read from UDP:[Link]:15136 --->


PUBLISH sip:testung@[Link];transport=UDP SIP/2.0
Via: SIP/2.0/UDP [Link]:44591;branch=z9hG4bK-d8754z-ef1908b14806db5f-1---
d8754z-
Max-Forwards: 70
Contact: <sip:testung@[Link]:44591;transport=UDP>
To: <sip:testung@[Link];transport=UDP>
From: <sip:testung@[Link];transport=UDP>;tag=cb41925b
Call-ID: ODFiZmI1NjYxMzliMWIxMDE5NGZiOTZmNTllMDYyYmM.
CSeq: 1 PUBLISH
Expires: 600
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/pidf+xml
Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri
User-Agent: Z 3.6.25251 r25476
Event: presence
Allow-Events: presence, kpml
Content-Length: 274

<?xml version="1.0" encoding="UTF-8"?>


<presence xmlns="urn:ietf:params:xml:ns:pidf"
entity="sip:testung@[Link];transport=UDP"> <tuple id="testung" >
<status><basic>open</basic></status> <note>On the phone</note> </tuple>
</presence>
<------------->
--- (16 headers 3 lines) ---
Sending to [Link]:15136 (NAT)

<--- Transmitting (NAT) to [Link]:15136 --->


SIP/2.0 489 Bad Event
Via: SIP/2.0/UDP [Link]:44591;branch=z9hG4bK-d8754z-ef1908b14806db5f-1---
d8754z-;received=[Link];rport=15136
From: <sip:testung@[Link];transport=UDP>;tag=cb41925b
To: <sip:testung@[Link];transport=UDP>;tag=as629927d6
Call-ID: ODFiZmI1NjYxMzliMWIxMDE5NGZiOTZmNTllMDYyYmM.
CSeq: 1 PUBLISH
Server: Asterisk PBX 11.13.1~dfsg-2+deb8u1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH,
MESSAGE
Supported: replaces, timer
Content-Length: 0

<------------>
Really destroying SIP dialog 'ODFiZmI1NjYxMzliMWIxMDE5NGZiOTZmNTllMDYyYmM.' Method:
PUBLISH

<--- SIP read from UDP:[Link]:15136 --->


SUBSCRIBE sip:testung@[Link];transport=UDP SIP/2.0
Via: SIP/2.0/UDP [Link]:44591;branch=z9hG4bK-d8754z-3d37067894b09ad9-1---
d8754z-
Max-Forwards: 70
Contact: <sip:testung@[Link]:44591;transport=UDP>
To: <sip:testung@[Link];transport=UDP>
From: <sip:testung@[Link];transport=UDP>;tag=604bce69
Call-ID: YTQzYTViN2RmYzRiNjMwOTU2ZDU5ZTU0ZDAxMDY1ZDE.
CSeq: 1 SUBSCRIBE
Expires: 600
Accept: application/watcherinfo+xml
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri
User-Agent: Z 3.6.25251 r25476
Event: [Link]
Allow-Events: presence, kpml
Content-Length: 0
<------------->
--- (16 headers 0 lines) ---
Sending to [Link]:15136 (NAT)
Creating new subscription
Sending to [Link]:15136 (NAT)
list_route: hop: <sip:testung@[Link]:44591;transport=UDP>
Found peer 'testung' for 'testung' from [Link]:15136

<--- Transmitting (NAT) to [Link]:15136 --->


SIP/2.0 489 Bad Event
Via: SIP/2.0/UDP [Link]:44591;branch=z9hG4bK-d8754z-3d37067894b09ad9-1---
d8754z-;received=[Link];rport=15136
From: <sip:testung@[Link];transport=UDP>;tag=604bce69
To: <sip:testung@[Link];transport=UDP>;tag=as3e61e734
Call-ID: YTQzYTViN2RmYzRiNjMwOTU2ZDU5ZTU0ZDAxMDY1ZDE.
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX 11.13.1~dfsg-2+deb8u1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH,
MESSAGE
Supported: replaces, timer
Content-Length: 0

<------------>
Really destroying SIP dialog 'YTQzYTViN2RmYzRiNjMwOTU2ZDU5ZTU0ZDAxMDY1ZDE.' Method:
SUBSCRIBE
Audio is at 16204
Adding codec 100010 (ilbc) to SDP
Adding codec 100002 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (NAT) to [Link]:15136 --->


SIP/2.0 200 OK
Via: SIP/2.0/UDP [Link]:44591;branch=z9hG4bK-d8754z-5b4430db9a8615b8-1---
d8754z-;received=[Link];rport=15136
From: <sip:5020@[Link];transport=UDP>;tag=a566fb5c
To: <sip:0060123718121@[Link];transport=UDP>;tag=as4ab9447f
Call-ID: NTgxZjYxNDE2NjRhYWNmNTEzZWVlMTNlOGI2YjE3ZjQ.
CSeq: 1 INVITE
Server: Asterisk PBX 11.13.1~dfsg-2+deb8u1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH,
MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:0060123718121@[Link]>
Content-Type: application/sdp
Require: timer
Content-Length: 298

v=0
o=root 1436607589 1436607589 IN IP4 [Link]
s=Asterisk PBX 11.13.1~dfsg-2+deb8u1
c=IN IP4 [Link]
t=0 0
m=audio 16204 RTP/AVP 98 3 101
a=rtpmap:98 iLBC/8000
a=fmtp:98 mode=30
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

<------------>
-- Locally bridging SIP/5020-0000001a and SIP/provider-0000001b
> 0x7f003400bf50 -- Probation passed - setting RTP source address to
[Link]:11522
Retransmitting #1 (NAT) to [Link]:15136:
SIP/2.0 200 OK
Via: SIP/2.0/UDP [Link]:44591;branch=z9hG4bK-d8754z-5b4430db9a8615b8-1---
d8754z-;received=[Link];rport=15136
From: <sip:5020@[Link];transport=UDP>;tag=a566fb5c
To: <sip:0060123718121@[Link];transport=UDP>;tag=as4ab9447f
Call-ID: NTgxZjYxNDE2NjRhYWNmNTEzZWVlMTNlOGI2YjE3ZjQ.
CSeq: 1 INVITE
Server: Asterisk PBX 11.13.1~dfsg-2+deb8u1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH,
MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:0060123718121@[Link]>
Content-Type: application/sdp
Require: timer
Content-Length: 298

v=0
o=root 1436607589 1436607589 IN IP4 [Link]
s=Asterisk PBX 11.13.1~dfsg-2+deb8u1
c=IN IP4 [Link]
t=0 0
m=audio 16204 RTP/AVP 98 3 101
a=rtpmap:98 iLBC/8000
a=fmtp:98 mode=30
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---

<--- SIP read from UDP:[Link]:15136 --->


ACK sip:0060123718121@[Link] SIP/2.0
Via: SIP/2.0/UDP [Link]:44591;branch=z9hG4bK-d8754z-c6484b546ba07e06-1---
d8754z-
Max-Forwards: 70
Contact: <sip:5020@[Link]:44591;transport=UDP>
To: <sip:0060123718121@[Link];transport=UDP>;tag=as4ab9447f
From: <sip:5020@[Link];transport=UDP>;tag=a566fb5c
Call-ID: NTgxZjYxNDE2NjRhYWNmNTEzZWVlMTNlOGI2YjE3ZjQ.
CSeq: 1 ACK
User-Agent: Z 3.6.25251 r25476
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
> 0x7f00400e95d0 -- Probation passed - setting RTP source address to
[Link]:17975
> 0x7f00400e95d0 -- Probation passed - setting RTP source address to
[Link]:17975

<--- SIP read from UDP:[Link]:15136 --->


PUBLISH sip:5020@[Link];transport=UDP SIP/2.0
Via: SIP/2.0/UDP [Link]:44591;branch=z9hG4bK-d8754z-b5e85164634a2e91-1---
d8754z-
Max-Forwards: 70
Contact: <sip:5020@[Link]:44591;transport=UDP>
To: <sip:5020@[Link];transport=UDP>
From: <sip:5020@[Link];transport=UDP>;tag=ad2d6060
Call-ID: Mjc5MWQ2MTFiZGRlMTVlOTQxMjE0YmJiZTUxYTc5YTM.
CSeq: 1 PUBLISH
Expires: 600
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/pidf+xml
Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri
User-Agent: Z 3.6.25251 r25476
Event: presence
Allow-Events: presence, kpml
Content-Length: 273

<?xml version="1.0" encoding="UTF-8"?>


<presence xmlns="urn:ietf:params:xml:ns:pidf"
entity="sip:5020@[Link];transport=UDP"> <tuple id="5020" >
<status><basic>open</basic></status> <note>On the phone</note> </tuple>
</presence>
<------------->
--- (16 headers 3 lines) ---
Sending to [Link]:15136 (NAT)

<--- Transmitting (NAT) to [Link]:15136 --->


SIP/2.0 489 Bad Event
Via: SIP/2.0/UDP [Link]:44591;branch=z9hG4bK-d8754z-b5e85164634a2e91-1---
d8754z-;received=[Link];rport=15136
From: <sip:5020@[Link];transport=UDP>;tag=ad2d6060
To: <sip:5020@[Link];transport=UDP>;tag=as174fb810
Call-ID: Mjc5MWQ2MTFiZGRlMTVlOTQxMjE0YmJiZTUxYTc5YTM.
CSeq: 1 PUBLISH
Server: Asterisk PBX 11.13.1~dfsg-2+deb8u1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH,
MESSAGE
Supported: replaces, timer
Content-Length: 0

<------------>
Really destroying SIP dialog 'Mjc5MWQ2MTFiZGRlMTVlOTQxMjE0YmJiZTUxYTc5YTM.' Method:
PUBLISH

<--- SIP read from UDP:[Link]:15136 --->


SUBSCRIBE sip:5020@[Link];transport=UDP SIP/2.0
Via: SIP/2.0/UDP [Link]:44591;branch=z9hG4bK-d8754z-44d10166b47d3bbe-1---
d8754z-
Max-Forwards: 70
Contact: <sip:5020@[Link]:44591;transport=UDP>
To: <sip:5020@[Link];transport=UDP>
From: <sip:5020@[Link];transport=UDP>;tag=0d698b15
Call-ID: MzZiMDY2NTgzOWE5ZWJjNTVjZDZkOGNlMTM0YTA1MWI.
CSeq: 1 SUBSCRIBE
Expires: 600
Accept: application/watcherinfo+xml
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri
User-Agent: Z 3.6.25251 r25476
Event: [Link]
Allow-Events: presence, kpml
Content-Length: 0

<------------->
--- (16 headers 0 lines) ---
Sending to [Link]:15136 (NAT)
Creating new subscription
Sending to [Link]:15136 (NAT)
list_route: hop: <sip:5020@[Link]:44591;transport=UDP>
Found peer '5020' for '5020' from [Link]:15136

<--- Transmitting (NAT) to [Link]:15136 --->


SIP/2.0 489 Bad Event
Via: SIP/2.0/UDP [Link]:44591;branch=z9hG4bK-d8754z-44d10166b47d3bbe-1---
d8754z-;received=[Link];rport=15136
From: <sip:5020@[Link];transport=UDP>;tag=0d698b15
To: <sip:5020@[Link];transport=UDP>;tag=as0f6b01eb
Call-ID: MzZiMDY2NTgzOWE5ZWJjNTVjZDZkOGNlMTM0YTA1MWI.
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX 11.13.1~dfsg-2+deb8u1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH,
MESSAGE
Supported: replaces, timer
Content-Length: 0

<------------>
Really destroying SIP dialog 'MzZiMDY2NTgzOWE5ZWJjNTVjZDZkOGNlMTM0YTA1MWI.' Method:
SUBSCRIBE

<--- SIP read from UDP:[Link]:15136 --->


PUBLISH sip:testung@[Link];transport=UDP SIP/2.0
Via: SIP/2.0/UDP [Link]:44591;branch=z9hG4bK-d8754z-5171dc804981270e-1---
d8754z-
Max-Forwards: 70
Contact: <sip:testung@[Link]:44591;transport=UDP>
To: <sip:testung@[Link];transport=UDP>
From: <sip:testung@[Link];transport=UDP>;tag=3e02410e
Call-ID: ODQyMTdiN2Y2NDA2YmE2YmJmMjQzNzdhZTkxYWY3NTU.
CSeq: 1 PUBLISH
Expires: 600
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/pidf+xml
Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri
User-Agent: Z 3.6.25251 r25476
Event: presence
Allow-Events: presence, kpml
Content-Length: 274

<?xml version="1.0" encoding="UTF-8"?>


<presence xmlns="urn:ietf:params:xml:ns:pidf"
entity="sip:testung@[Link];transport=UDP"> <tuple id="testung" >
<status><basic>open</basic></status> <note>On the phone</note> </tuple>
</presence>
<------------->
--- (16 headers 3 lines) ---
Sending to [Link]:15136 (NAT)

<--- Transmitting (NAT) to [Link]:15136 --->


SIP/2.0 489 Bad Event
Via: SIP/2.0/UDP [Link]:44591;branch=z9hG4bK-d8754z-5171dc804981270e-1---
d8754z-;received=[Link];rport=15136
From: <sip:testung@[Link];transport=UDP>;tag=3e02410e
To: <sip:testung@[Link];transport=UDP>;tag=as3b09b227
Call-ID: ODQyMTdiN2Y2NDA2YmE2YmJmMjQzNzdhZTkxYWY3NTU.
CSeq: 1 PUBLISH
Server: Asterisk PBX 11.13.1~dfsg-2+deb8u1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH,
MESSAGE
Supported: replaces, timer
Content-Length: 0

<------------>
Really destroying SIP dialog 'ODQyMTdiN2Y2NDA2YmE2YmJmMjQzNzdhZTkxYWY3NTU.' Method:
PUBLISH

<--- SIP read from UDP:[Link]:15136 --->


SUBSCRIBE sip:testung@[Link];transport=UDP SIP/2.0
Via: SIP/2.0/UDP [Link]:44591;branch=z9hG4bK-d8754z-950f7c2b0ba4a0ee-1---
d8754z-
Max-Forwards: 70
Contact: <sip:testung@[Link]:44591;transport=UDP>
To: <sip:testung@[Link];transport=UDP>
From: <sip:testung@[Link];transport=UDP>;tag=f568e02b
Call-ID: MWU5MGZhMWUxYjBkY2Q4Mjc3ZGI5YzlkNWYyYzhmOTA.
CSeq: 1 SUBSCRIBE
Expires: 600
Accept: application/watcherinfo+xml
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri
User-Agent: Z 3.6.25251 r25476
Event: [Link]
Allow-Events: presence, kpml
Content-Length: 0

<------------->
--- (16 headers 0 lines) ---
Sending to [Link]:15136 (NAT)
Creating new subscription
Sending to [Link]:15136 (NAT)
list_route: hop: <sip:testung@[Link]:44591;transport=UDP>
Found peer 'testung' for 'testung' from [Link]:15136

<--- Transmitting (NAT) to [Link]:15136 --->


SIP/2.0 489 Bad Event
Via: SIP/2.0/UDP [Link]:44591;branch=z9hG4bK-d8754z-950f7c2b0ba4a0ee-1---
d8754z-;received=[Link];rport=15136
From: <sip:testung@[Link];transport=UDP>;tag=f568e02b
To: <sip:testung@[Link];transport=UDP>;tag=as05e533f5
Call-ID: MWU5MGZhMWUxYjBkY2Q4Mjc3ZGI5YzlkNWYyYzhmOTA.
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX 11.13.1~dfsg-2+deb8u1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH,
MESSAGE
Supported: replaces, timer
Content-Length: 0

<------------>
Really destroying SIP dialog 'MWU5MGZhMWUxYjBkY2Q4Mjc3ZGI5YzlkNWYyYzhmOTA.' Method:
SUBSCRIBE

<--- SIP read from UDP:[Link]:15136 --->


ACK sip:0060123718121@[Link] SIP/2.0
Via: SIP/2.0/UDP [Link]:44591;branch=z9hG4bK-d8754z-c6484b546ba07e06-1---
d8754z-
Max-Forwards: 70
Contact: <sip:5020@[Link]:44591;transport=UDP>
To: <sip:0060123718121@[Link];transport=UDP>;tag=as4ab9447f
From: <sip:5020@[Link];transport=UDP>;tag=a566fb5c
Call-ID: NTgxZjYxNDE2NjRhYWNmNTEzZWVlMTNlOGI2YjE3ZjQ.
CSeq: 1 ACK
User-Agent: Z 3.6.25251 r25476
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---

<--- SIP read from UDP:[Link]:1028 --->

<------------->

<--- SIP read from UDP:[Link]:54935 --->

<------------->

<--- SIP read from UDP:[Link] --->


BYE sip:605605001@[Link] SIP/2.0
Via: SIP/2.0/UDP [Link];branch=z9hG4bKf191.bb4b6c11.0
Via: SIP/2.0/UDP
[Link]:5060;received=[Link];branch=z9hG4bK17dee670;rport=5060
From: <sip:0060123718121@[Link]>;tag=as636c8a82
To: <sip:605605001@[Link]>;tag=as1c55888d
Call-ID: 731eb3b30a08e30c5a2faed25f671a91@[Link]
CSeq: 102 BYE
User-Agent: SoftSwitch
Max-Forwards: 69
X-Asterisk-HangupCause: Unknown
X-Asterisk-HangupCauseCode: 0
Content-Length: 0

<------------->
--- (12 headers 0 lines) ---
Sending to [Link] (NAT)
Scheduling destruction of SIP dialog
'731eb3b30a08e30c5a2faed25f671a91@[Link]' in 6400 ms (Method: BYE)
<--- Transmitting (NAT) to [Link] --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP
[Link];branch=z9hG4bKf191.bb4b6c11.0;received=[Link];rport=5
060
Via: SIP/2.0/UDP
[Link]:5060;received=[Link];branch=z9hG4bK17dee670;rport=5060
From: <sip:0060123718121@[Link]>;tag=as636c8a82
To: <sip:605605001@[Link]>;tag=as1c55888d
Call-ID: 731eb3b30a08e30c5a2faed25f671a91@[Link]
CSeq: 102 BYE
Server: Asterisk PBX 11.13.1~dfsg-2+deb8u1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH,
MESSAGE
Supported: replaces, timer
Content-Length: 0

<------------>
== Spawn extension (vtiger_inbound, 0060123718121, 1) exited non-zero on
'SIP/5020-0000001a'
Scheduling destruction of SIP dialog 'NTgxZjYxNDE2NjRhYWNmNTEzZWVlMTNlOGI2YjE3ZjQ.'
in 23424 ms (Method: ACK)
set_destination: Parsing <sip:5020@[Link]:44591;transport=UDP> for
address/port to send to
set_destination: set destination to [Link]:44591
Reliably Transmitting (NAT) to [Link]:15136:
BYE sip:5020@[Link]:44591;transport=UDP SIP/2.0
Via: SIP/2.0/UDP [Link];branch=z9hG4bK672a0b72;rport
Max-Forwards: 70
From: <sip:0060123718121@[Link];transport=UDP>;tag=as4ab9447f
To: <sip:5020@[Link];transport=UDP>;tag=a566fb5c
Call-ID: NTgxZjYxNDE2NjRhYWNmNTEzZWVlMTNlOGI2YjE3ZjQ.
CSeq: 102 BYE
User-Agent: Asterisk PBX 11.13.1~dfsg-2+deb8u1
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0

---
Retransmitting #1 (NAT) to [Link]:15136:
BYE sip:5020@[Link]:44591;transport=UDP SIP/2.0
Via: SIP/2.0/UDP [Link];branch=z9hG4bK672a0b72;rport
Max-Forwards: 70
From: <sip:0060123718121@[Link];transport=UDP>;tag=as4ab9447f
To: <sip:5020@[Link];transport=UDP>;tag=a566fb5c
Call-ID: NTgxZjYxNDE2NjRhYWNmNTEzZWVlMTNlOGI2YjE3ZjQ.
CSeq: 102 BYE
User-Agent: Asterisk PBX 11.13.1~dfsg-2+deb8u1
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0

---

<--- SIP read from UDP:[Link]:15136 --->


SIP/2.0 200 OK
Via: SIP/2.0/UDP [Link];branch=z9hG4bK672a0b72;rport=5261
Contact: <sip:5020@[Link]:44591;transport=UDP>
To: <sip:5020@[Link];transport=UDP>;tag=a566fb5c
From: <sip:0060123718121@[Link];transport=UDP>;tag=as4ab9447f
Call-ID: NTgxZjYxNDE2NjRhYWNmNTEzZWVlMTNlOGI2YjE3ZjQ.
CSeq: 102 BYE
User-Agent: Z 3.6.25251 r25476
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog 'NTgxZjYxNDE2NjRhYWNmNTEzZWVlMTNlOGI2YjE3ZjQ.' Method:
ACK

<--- SIP read from UDP:[Link]:15136 --->


PUBLISH sip:5020@[Link];transport=UDP SIP/2.0
Via: SIP/2.0/UDP [Link]:44591;branch=z9hG4bK-d8754z-2d82c36de1c5f8bf-1---
d8754z-
Max-Forwards: 70
Contact: <sip:5020@[Link]:44591;transport=UDP>
To: <sip:5020@[Link];transport=UDP>
From: <sip:5020@[Link];transport=UDP>;tag=414b3c62
Call-ID: ZjkzNjA4NmY1YjFjNDJiMDNiMGFkMTM1ODMzMWFhOGU.
CSeq: 1 PUBLISH
Expires: 600
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/pidf+xml
Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri
User-Agent: Z 3.6.25251 r25476
Event: presence
Allow-Events: presence, kpml
Content-Length: 267

<?xml version="1.0" encoding="UTF-8"?>


<presence xmlns="urn:ietf:params:xml:ns:pidf"
entity="sip:5020@[Link];transport=UDP"> <tuple id="5020" >
<status><basic>open</basic></status> <note>Online</note> </tuple>
</presence>
<------------->
--- (16 headers 3 lines) ---
Sending to [Link]:15136 (NAT)

<--- Transmitting (NAT) to [Link]:15136 --->


SIP/2.0 489 Bad Event
Via: SIP/2.0/UDP [Link]:44591;branch=z9hG4bK-d8754z-2d82c36de1c5f8bf-1---
d8754z-;received=[Link];rport=15136
From: <sip:5020@[Link];transport=UDP>;tag=414b3c62
To: <sip:5020@[Link];transport=UDP>;tag=as37c957bf
Call-ID: ZjkzNjA4NmY1YjFjNDJiMDNiMGFkMTM1ODMzMWFhOGU.
CSeq: 1 PUBLISH
Server: Asterisk PBX 11.13.1~dfsg-2+deb8u1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH,
MESSAGE
Supported: replaces, timer
Content-Length: 0

<------------>
Really destroying SIP dialog 'ZjkzNjA4NmY1YjFjNDJiMDNiMGFkMTM1ODMzMWFhOGU.' Method:
PUBLISH

<--- SIP read from UDP:[Link]:15136 --->


SUBSCRIBE sip:5020@[Link];transport=UDP SIP/2.0
Via: SIP/2.0/UDP [Link]:44591;branch=z9hG4bK-d8754z-2dce4186697d0bc1-1---
d8754z-
Max-Forwards: 70
Contact: <sip:5020@[Link]:44591;transport=UDP>
To: <sip:5020@[Link];transport=UDP>
From: <sip:5020@[Link];transport=UDP>;tag=776f483a
Call-ID: Njk4YjBiN2JhY2ViN2M0YmU0M2RmMmYwMDBmYTNiMjY.
CSeq: 1 SUBSCRIBE
Expires: 600
Accept: application/watcherinfo+xml
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri
User-Agent: Z 3.6.25251 r25476
Event: [Link]
Allow-Events: presence, kpml
Content-Length: 0

<------------->
--- (16 headers 0 lines) ---
Sending to [Link]:15136 (NAT)
Creating new subscription
Sending to [Link]:15136 (NAT)
list_route: hop: <sip:5020@[Link]:44591;transport=UDP>
Found peer '5020' for '5020' from [Link]:15136

<--- Transmitting (NAT) to [Link]:15136 --->


SIP/2.0 489 Bad Event
Via: SIP/2.0/UDP [Link]:44591;branch=z9hG4bK-d8754z-2dce4186697d0bc1-1---
d8754z-;received=[Link];rport=15136
From: <sip:5020@[Link];transport=UDP>;tag=776f483a
To: <sip:5020@[Link];transport=UDP>;tag=as4581566c
Call-ID: Njk4YjBiN2JhY2ViN2M0YmU0M2RmMmYwMDBmYTNiMjY.
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX 11.13.1~dfsg-2+deb8u1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH,
MESSAGE
Supported: replaces, timer
Content-Length: 0

<------------>
Really destroying SIP dialog 'Njk4YjBiN2JhY2ViN2M0YmU0M2RmMmYwMDBmYTNiMjY.' Method:
SUBSCRIBE

<--- SIP read from UDP:[Link]:15136 --->


PUBLISH sip:testung@[Link];transport=UDP SIP/2.0
Via: SIP/2.0/UDP [Link]:44591;branch=z9hG4bK-d8754z-51b29fcc1da89b15-1---
d8754z-
Max-Forwards: 70
Contact: <sip:testung@[Link]:44591;transport=UDP>
To: <sip:testung@[Link];transport=UDP>
From: <sip:testung@[Link];transport=UDP>;tag=f16cc056
Call-ID: ZTY5NjEyNTYxYzQ0Mzk2NDA4YjhhNGM3MzdkM2MzZjI.
CSeq: 1 PUBLISH
Expires: 600
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/pidf+xml
Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri
User-Agent: Z 3.6.25251 r25476
Event: presence
Allow-Events: presence, kpml
Content-Length: 268

<?xml version="1.0" encoding="UTF-8"?>


<presence xmlns="urn:ietf:params:xml:ns:pidf"
entity="sip:testung@[Link];transport=UDP"> <tuple id="testung" >
<status><basic>open</basic></status> <note>Online</note> </tuple>
</presence>
<------------->
--- (16 headers 3 lines) ---
Sending to [Link]:15136 (NAT)

<--- Transmitting (NAT) to [Link]:15136 --->


SIP/2.0 489 Bad Event
Via: SIP/2.0/UDP [Link]:44591;branch=z9hG4bK-d8754z-51b29fcc1da89b15-1---
d8754z-;received=[Link];rport=15136
From: <sip:testung@[Link];transport=UDP>;tag=f16cc056
To: <sip:testung@[Link];transport=UDP>;tag=as452b149c
Call-ID: ZTY5NjEyNTYxYzQ0Mzk2NDA4YjhhNGM3MzdkM2MzZjI.
CSeq: 1 PUBLISH
Server: Asterisk PBX 11.13.1~dfsg-2+deb8u1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH,
MESSAGE
Supported: replaces, timer
Content-Length: 0

<------------>
Really destroying SIP dialog 'ZTY5NjEyNTYxYzQ0Mzk2NDA4YjhhNGM3MzdkM2MzZjI.' Method:
PUBLISH

<--- SIP read from UDP:[Link]:15136 --->


SUBSCRIBE sip:testung@[Link];transport=UDP SIP/2.0
Via: SIP/2.0/UDP [Link]:44591;branch=z9hG4bK-d8754z-fe80cd049e5d9ad5-1---
d8754z-
Max-Forwards: 70
Contact: <sip:testung@[Link]:44591;transport=UDP>
To: <sip:testung@[Link];transport=UDP>
From: <sip:testung@[Link];transport=UDP>;tag=4a231410
Call-ID: NjE2N2Y5NTA0NmNiNzljZjFjN2U1NDgzYzQ2YzcwNTI.
CSeq: 1 SUBSCRIBE
Expires: 600
Accept: application/watcherinfo+xml
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri
User-Agent: Z 3.6.25251 r25476
Event: [Link]
Allow-Events: presence, kpml
Content-Length: 0

<------------->
--- (16 headers 0 lines) ---
Sending to [Link]:15136 (NAT)
Creating new subscription
Sending to [Link]:15136 (NAT)
list_route: hop: <sip:testung@[Link]:44591;transport=UDP>
Found peer 'testung' for 'testung' from [Link]:15136

<--- Transmitting (NAT) to [Link]:15136 --->


SIP/2.0 489 Bad Event
Via: SIP/2.0/UDP [Link]:44591;branch=z9hG4bK-d8754z-fe80cd049e5d9ad5-1---
d8754z-;received=[Link];rport=15136
From: <sip:testung@[Link];transport=UDP>;tag=4a231410
To: <sip:testung@[Link];transport=UDP>;tag=as19836346
Call-ID: NjE2N2Y5NTA0NmNiNzljZjFjN2U1NDgzYzQ2YzcwNTI.
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX 11.13.1~dfsg-2+deb8u1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH,
MESSAGE
Supported: replaces, timer
Content-Length: 0

<------------>
Really destroying SIP dialog 'NjE2N2Y5NTA0NmNiNzljZjFjN2U1NDgzYzQ2YzcwNTI.' Method:
SUBSCRIBE

<--- SIP read from UDP:[Link]:15136 --->


SIP/2.0 200 OK
Via: SIP/2.0/UDP [Link];branch=z9hG4bK672a0b72;rport=5261
Contact: <sip:5020@[Link]:44591;transport=UDP>
To: <sip:5020@[Link];transport=UDP>;tag=a566fb5c
From: <sip:0060123718121@[Link];transport=UDP>;tag=as4ab9447f
Call-ID: NTgxZjYxNDE2NjRhYWNmNTEzZWVlMTNlOGI2YjE3ZjQ.
CSeq: 102 BYE
User-Agent: Z 3.6.25251 r25476
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Debian-81-64b*CLI> core show ch
channel channels channeltypes channeltype
Debian-81-64b*CLI> core show channels
Channel Location State Application(Data)
0 active channels
0 active calls
17 calls processed
Really destroying SIP dialog
'731eb3b30a08e30c5a2faed25f671a91@[Link]' Method: BYE
Debian-81-64b*CLI> exit
Asterisk cleanly ending (0).
Executing last minute cleanups
root@Debian-81-64b:~# ~~

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