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Sip Debug1

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
26 views17 pages

Sip Debug1

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
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<--- SIP read from UDP:103.255.4.

17:15136 --->
INVITE sip:0060123718121@XXX.XXX.XXX.XXX:5062;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.8.101:44591;branch=z9hG4bK-d8754z-5b4430db9a8615b8-1---
d8754z-
Max-Forwards: 70
Contact: <sip:5020@192.168.8.101:44591;transport=UDP>
To: <sip:0060123718121@XXX.XXX.XXX.XXX:5062;transport=UDP>
From: <sip:5020@XXX.XXX.XXX.XXX:5062;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 192.168.8.101
s=Z
c=IN IP4 192.168.8.101
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 103.255.4.17:15136 (NAT)
Sending to 103.255.4.17:15136 (NAT)
Using INVITE request as basis request -
NTgxZjYxNDE2NjRhYWNmNTEzZWVlMTNlOGI2YjE3ZjQ.
Found peer '5020' for '5020' from 103.255.4.17:15136
== 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 192.168.8.101:8000
Looking for 0060123718121 in vtiger_inbound (domain 103.25.202.195)
list_route: hop: <sip:5020@192.168.8.101:44591;transport=UDP>

<--- Transmitting (NAT) to 103.255.4.17:15136 --->


SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.8.101:44591;branch=z9hG4bK-d8754z-5b4430db9a8615b8-1---
d8754z-;received=103.255.4.17;rport=15136
From: <sip:5020@XXX.XXX.XXX.XXX:5062;transport=UDP>;tag=a566fb5c
To: <sip:0060123718121@XXX.XXX.XXX.XXX:5062;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@XXX.XXX.XXX.XXX:5062>
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 yyy.yyy.yyy.yyy:5060:
INVITE sip:0060123718121@sip3.providervoip.com:5060 SIP/2.0
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5062;branch=z9hG4bK00fe2662;rport
Max-Forwards: 70
From: <sip:605605001@XXX.XXX.XXX.XXX:5062>;tag=as1c55888d
To: <sip:0060123718121@sip3.providervoip.com:5060>
Contact: <sip:605605001@XXX.XXX.XXX.XXX:5062>
Call-ID: 731eb3b30a08e30c5a2faed25f671a91@XXX.XXX.XXX.XXX:5062
CSeq: 102 INVITE
User-Agent: Asterisk PBX 11.13.1~dfsg-2+deb8u1
Date: Fri, 12 May 2017 05:58:20 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 103.25.202.195
s=Asterisk PBX 11.13.1~dfsg-2+deb8u1
c=IN IP4 103.25.202.195
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:yyy.yyy.yyy.yyy:5060 --->


SIP/2.0 100 Giving a try
Via: SIP/2.0/UDP
XXX.XXX.XXX.XXX:5062;received=103.25.202.195;branch=z9hG4bK00fe2662;rport=5261
From: <sip:605605001@XXX.XXX.XXX.XXX:5062>;tag=as1c55888d
To: <sip:0060123718121@sip3.providervoip.com:5060>
Call-ID: 731eb3b30a08e30c5a2faed25f671a91@XXX.XXX.XXX.XXX:5062
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
103.246.89.131:11522

<--- SIP read from UDP:103.255.4.17:15136 --->


PUBLISH sip:5020@XXX.XXX.XXX.XXX:5062;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.8.101:44591;branch=z9hG4bK-d8754z-fca8f6117ebfa171-1---
d8754z-
Max-Forwards: 70
Contact: <sip:5020@192.168.8.101:44591;transport=UDP>
To: <sip:5020@XXX.XXX.XXX.XXX:5062;transport=UDP>
From: <sip:5020@XXX.XXX.XXX.XXX:5062;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@XXX.XXX.XXX.XXX:5062;transport=UDP"> <tuple id="5020" >
<status><basic>open</basic></status> <note>On the phone</note> </tuple>
</presence>
<------------->
--- (16 headers 3 lines) ---
Sending to 103.255.4.17:15136 (NAT)

<--- Transmitting (NAT) to 103.255.4.17:15136 --->


SIP/2.0 489 Bad Event
Via: SIP/2.0/UDP 192.168.8.101:44591;branch=z9hG4bK-d8754z-fca8f6117ebfa171-1---
d8754z-;received=103.255.4.17;rport=15136
From: <sip:5020@XXX.XXX.XXX.XXX:5062;transport=UDP>;tag=df31695b
To: <sip:5020@XXX.XXX.XXX.XXX:5062;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:yyy.yyy.yyy.yyy:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP
XXX.XXX.XXX.XXX:5062;received=103.25.202.195;branch=z9hG4bK00fe2662;rport=5261
Record-Route: <sip:yyy.yyy.yyy.yyy;lr;ftag=as1c55888d;did=05a.23915b14>
From: <sip:605605001@XXX.XXX.XXX.XXX:5062>;tag=as1c55888d
To: <sip:0060123718121@sip3.providervoip.com:5060>;tag=as636c8a82
Call-ID: 731eb3b30a08e30c5a2faed25f671a91@XXX.XXX.XXX.XXX:5062
CSeq: 102 INVITE
User-Agent: SoftSwitch
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: <sip:0060123718121@103.246.89.131>
Content-Type: application/sdp
Content-Length: 288

v=0
o=root 31107 31107 IN IP4 103.246.89.131
s=session
c=IN IP4 103.246.89.131
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 103.246.89.131:11522
list_route: hop: <sip:yyy.yyy.yyy.yyy;lr;ftag=as1c55888d;did=05a.23915b14>
set_destination: Parsing <sip:yyy.yyy.yyy.yyy;lr;ftag=as1c55888d;did=05a.23915b14>
for address/port to send to
set_destination: set destination to yyy.yyy.yyy.yyy:5060
Transmitting (NAT) to yyy.yyy.yyy.yyy:5060:
ACK sip:0060123718121@103.246.89.131 SIP/2.0
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5062;branch=z9hG4bK4910089e;rport
Route: <sip:yyy.yyy.yyy.yyy;lr;ftag=as1c55888d;did=05a.23915b14>
Max-Forwards: 70
From: <sip:605605001@XXX.XXX.XXX.XXX:5062>;tag=as1c55888d
To: <sip:0060123718121@sip3.providervoip.com:5060>;tag=as636c8a82
Contact: <sip:605605001@XXX.XXX.XXX.XXX:5062>
Call-ID: 731eb3b30a08e30c5a2faed25f671a91@XXX.XXX.XXX.XXX:5062
CSeq: 102 ACK
User-Agent: Asterisk PBX 11.13.1~dfsg-2+deb8u1
Content-Length: 0

---

<--- SIP read from UDP:103.255.4.17:15136 --->


SUBSCRIBE sip:5020@XXX.XXX.XXX.XXX:5062;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.8.101:44591;branch=z9hG4bK-d8754z-069200ee8b145c71-1---
d8754z-
Max-Forwards: 70
Contact: <sip:5020@192.168.8.101:44591;transport=UDP>
To: <sip:5020@XXX.XXX.XXX.XXX:5062;transport=UDP>
From: <sip:5020@XXX.XXX.XXX.XXX:5062;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: presence.winfo
Allow-Events: presence, kpml
Content-Length: 0

<------------->
--- (16 headers 0 lines) ---
Sending to 103.255.4.17:15136 (NAT)
Creating new subscription
Sending to 103.255.4.17:15136 (NAT)
list_route: hop: <sip:5020@192.168.8.101:44591;transport=UDP>
Found peer '5020' for '5020' from 103.255.4.17:15136

<--- Transmitting (NAT) to 103.255.4.17:15136 --->


SIP/2.0 489 Bad Event
Via: SIP/2.0/UDP 192.168.8.101:44591;branch=z9hG4bK-d8754z-069200ee8b145c71-1---
d8754z-;received=103.255.4.17;rport=15136
From: <sip:5020@XXX.XXX.XXX.XXX:5062;transport=UDP>;tag=b7022909
To: <sip:5020@XXX.XXX.XXX.XXX:5062;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:103.255.4.17:15136 --->


PUBLISH sip:testung@103.25.202.195;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.8.101:44591;branch=z9hG4bK-d8754z-ef1908b14806db5f-1---
d8754z-
Max-Forwards: 70
Contact: <sip:testung@192.168.8.101:44591;transport=UDP>
To: <sip:testung@103.25.202.195;transport=UDP>
From: <sip:testung@103.25.202.195;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@103.25.202.195;transport=UDP"> <tuple id="testung" >
<status><basic>open</basic></status> <note>On the phone</note> </tuple>
</presence>
<------------->
--- (16 headers 3 lines) ---
Sending to 103.255.4.17:15136 (NAT)

<--- Transmitting (NAT) to 103.255.4.17:15136 --->


SIP/2.0 489 Bad Event
Via: SIP/2.0/UDP 192.168.8.101:44591;branch=z9hG4bK-d8754z-ef1908b14806db5f-1---
d8754z-;received=103.255.4.17;rport=15136
From: <sip:testung@103.25.202.195;transport=UDP>;tag=cb41925b
To: <sip:testung@103.25.202.195;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:103.255.4.17:15136 --->


SUBSCRIBE sip:testung@103.25.202.195;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.8.101:44591;branch=z9hG4bK-d8754z-3d37067894b09ad9-1---
d8754z-
Max-Forwards: 70
Contact: <sip:testung@192.168.8.101:44591;transport=UDP>
To: <sip:testung@103.25.202.195;transport=UDP>
From: <sip:testung@103.25.202.195;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: presence.winfo
Allow-Events: presence, kpml
Content-Length: 0
<------------->
--- (16 headers 0 lines) ---
Sending to 103.255.4.17:15136 (NAT)
Creating new subscription
Sending to 103.255.4.17:15136 (NAT)
list_route: hop: <sip:testung@192.168.8.101:44591;transport=UDP>
Found peer 'testung' for 'testung' from 103.255.4.17:15136

<--- Transmitting (NAT) to 103.255.4.17:15136 --->


SIP/2.0 489 Bad Event
Via: SIP/2.0/UDP 192.168.8.101:44591;branch=z9hG4bK-d8754z-3d37067894b09ad9-1---
d8754z-;received=103.255.4.17;rport=15136
From: <sip:testung@103.25.202.195;transport=UDP>;tag=604bce69
To: <sip:testung@103.25.202.195;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 103.255.4.17:15136 --->


SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.8.101:44591;branch=z9hG4bK-d8754z-5b4430db9a8615b8-1---
d8754z-;received=103.255.4.17;rport=15136
From: <sip:5020@XXX.XXX.XXX.XXX:5062;transport=UDP>;tag=a566fb5c
To: <sip:0060123718121@XXX.XXX.XXX.XXX:5062;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@XXX.XXX.XXX.XXX:5062>
Content-Type: application/sdp
Require: timer
Content-Length: 298

v=0
o=root 1436607589 1436607589 IN IP4 103.25.202.195
s=Asterisk PBX 11.13.1~dfsg-2+deb8u1
c=IN IP4 103.25.202.195
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
103.246.89.131:11522
Retransmitting #1 (NAT) to 103.255.4.17:15136:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.8.101:44591;branch=z9hG4bK-d8754z-5b4430db9a8615b8-1---
d8754z-;received=103.255.4.17;rport=15136
From: <sip:5020@XXX.XXX.XXX.XXX:5062;transport=UDP>;tag=a566fb5c
To: <sip:0060123718121@XXX.XXX.XXX.XXX:5062;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@XXX.XXX.XXX.XXX:5062>
Content-Type: application/sdp
Require: timer
Content-Length: 298

v=0
o=root 1436607589 1436607589 IN IP4 103.25.202.195
s=Asterisk PBX 11.13.1~dfsg-2+deb8u1
c=IN IP4 103.25.202.195
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:103.255.4.17:15136 --->


ACK sip:0060123718121@XXX.XXX.XXX.XXX:5062 SIP/2.0
Via: SIP/2.0/UDP 192.168.8.101:44591;branch=z9hG4bK-d8754z-c6484b546ba07e06-1---
d8754z-
Max-Forwards: 70
Contact: <sip:5020@192.168.8.101:44591;transport=UDP>
To: <sip:0060123718121@XXX.XXX.XXX.XXX:5062;transport=UDP>;tag=as4ab9447f
From: <sip:5020@XXX.XXX.XXX.XXX:5062;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
103.255.4.17:17975
> 0x7f00400e95d0 -- Probation passed - setting RTP source address to
103.255.4.17:17975

<--- SIP read from UDP:103.255.4.17:15136 --->


PUBLISH sip:5020@XXX.XXX.XXX.XXX:5062;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.8.101:44591;branch=z9hG4bK-d8754z-b5e85164634a2e91-1---
d8754z-
Max-Forwards: 70
Contact: <sip:5020@192.168.8.101:44591;transport=UDP>
To: <sip:5020@XXX.XXX.XXX.XXX:5062;transport=UDP>
From: <sip:5020@XXX.XXX.XXX.XXX:5062;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@XXX.XXX.XXX.XXX:5062;transport=UDP"> <tuple id="5020" >
<status><basic>open</basic></status> <note>On the phone</note> </tuple>
</presence>
<------------->
--- (16 headers 3 lines) ---
Sending to 103.255.4.17:15136 (NAT)

<--- Transmitting (NAT) to 103.255.4.17:15136 --->


SIP/2.0 489 Bad Event
Via: SIP/2.0/UDP 192.168.8.101:44591;branch=z9hG4bK-d8754z-b5e85164634a2e91-1---
d8754z-;received=103.255.4.17;rport=15136
From: <sip:5020@XXX.XXX.XXX.XXX:5062;transport=UDP>;tag=ad2d6060
To: <sip:5020@XXX.XXX.XXX.XXX:5062;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:103.255.4.17:15136 --->


SUBSCRIBE sip:5020@XXX.XXX.XXX.XXX:5062;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.8.101:44591;branch=z9hG4bK-d8754z-44d10166b47d3bbe-1---
d8754z-
Max-Forwards: 70
Contact: <sip:5020@192.168.8.101:44591;transport=UDP>
To: <sip:5020@XXX.XXX.XXX.XXX:5062;transport=UDP>
From: <sip:5020@XXX.XXX.XXX.XXX:5062;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: presence.winfo
Allow-Events: presence, kpml
Content-Length: 0

<------------->
--- (16 headers 0 lines) ---
Sending to 103.255.4.17:15136 (NAT)
Creating new subscription
Sending to 103.255.4.17:15136 (NAT)
list_route: hop: <sip:5020@192.168.8.101:44591;transport=UDP>
Found peer '5020' for '5020' from 103.255.4.17:15136

<--- Transmitting (NAT) to 103.255.4.17:15136 --->


SIP/2.0 489 Bad Event
Via: SIP/2.0/UDP 192.168.8.101:44591;branch=z9hG4bK-d8754z-44d10166b47d3bbe-1---
d8754z-;received=103.255.4.17;rport=15136
From: <sip:5020@XXX.XXX.XXX.XXX:5062;transport=UDP>;tag=0d698b15
To: <sip:5020@XXX.XXX.XXX.XXX:5062;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:103.255.4.17:15136 --->


PUBLISH sip:testung@103.25.202.195;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.8.101:44591;branch=z9hG4bK-d8754z-5171dc804981270e-1---
d8754z-
Max-Forwards: 70
Contact: <sip:testung@192.168.8.101:44591;transport=UDP>
To: <sip:testung@103.25.202.195;transport=UDP>
From: <sip:testung@103.25.202.195;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@103.25.202.195;transport=UDP"> <tuple id="testung" >
<status><basic>open</basic></status> <note>On the phone</note> </tuple>
</presence>
<------------->
--- (16 headers 3 lines) ---
Sending to 103.255.4.17:15136 (NAT)

<--- Transmitting (NAT) to 103.255.4.17:15136 --->


SIP/2.0 489 Bad Event
Via: SIP/2.0/UDP 192.168.8.101:44591;branch=z9hG4bK-d8754z-5171dc804981270e-1---
d8754z-;received=103.255.4.17;rport=15136
From: <sip:testung@103.25.202.195;transport=UDP>;tag=3e02410e
To: <sip:testung@103.25.202.195;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:103.255.4.17:15136 --->


SUBSCRIBE sip:testung@103.25.202.195;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.8.101:44591;branch=z9hG4bK-d8754z-950f7c2b0ba4a0ee-1---
d8754z-
Max-Forwards: 70
Contact: <sip:testung@192.168.8.101:44591;transport=UDP>
To: <sip:testung@103.25.202.195;transport=UDP>
From: <sip:testung@103.25.202.195;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: presence.winfo
Allow-Events: presence, kpml
Content-Length: 0

<------------->
--- (16 headers 0 lines) ---
Sending to 103.255.4.17:15136 (NAT)
Creating new subscription
Sending to 103.255.4.17:15136 (NAT)
list_route: hop: <sip:testung@192.168.8.101:44591;transport=UDP>
Found peer 'testung' for 'testung' from 103.255.4.17:15136

<--- Transmitting (NAT) to 103.255.4.17:15136 --->


SIP/2.0 489 Bad Event
Via: SIP/2.0/UDP 192.168.8.101:44591;branch=z9hG4bK-d8754z-950f7c2b0ba4a0ee-1---
d8754z-;received=103.255.4.17;rport=15136
From: <sip:testung@103.25.202.195;transport=UDP>;tag=f568e02b
To: <sip:testung@103.25.202.195;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:103.255.4.17:15136 --->


ACK sip:0060123718121@XXX.XXX.XXX.XXX:5062 SIP/2.0
Via: SIP/2.0/UDP 192.168.8.101:44591;branch=z9hG4bK-d8754z-c6484b546ba07e06-1---
d8754z-
Max-Forwards: 70
Contact: <sip:5020@192.168.8.101:44591;transport=UDP>
To: <sip:0060123718121@XXX.XXX.XXX.XXX:5062;transport=UDP>;tag=as4ab9447f
From: <sip:5020@XXX.XXX.XXX.XXX:5062;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:115.134.150.151:1028 --->

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

<--- SIP read from UDP:115.134.150.151:54935 --->

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

<--- SIP read from UDP:yyy.yyy.yyy.yyy:5060 --->


BYE sip:605605001@XXX.XXX.XXX.XXX:5062 SIP/2.0
Via: SIP/2.0/UDP yyy.yyy.yyy.yyy:5060;branch=z9hG4bKf191.bb4b6c11.0
Via: SIP/2.0/UDP
103.246.89.131:5060;received=103.246.89.131;branch=z9hG4bK17dee670;rport=5060
From: <sip:0060123718121@sip3.providervoip.com:5060>;tag=as636c8a82
To: <sip:605605001@XXX.XXX.XXX.XXX:5062>;tag=as1c55888d
Call-ID: 731eb3b30a08e30c5a2faed25f671a91@XXX.XXX.XXX.XXX:5062
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 yyy.yyy.yyy.yyy:5060 (NAT)
Scheduling destruction of SIP dialog
'731eb3b30a08e30c5a2faed25f671a91@XXX.XXX.XXX.XXX:5062' in 6400 ms (Method: BYE)
<--- Transmitting (NAT) to yyy.yyy.yyy.yyy:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP
yyy.yyy.yyy.yyy:5060;branch=z9hG4bKf191.bb4b6c11.0;received=yyy.yyy.yyy.yyy;rport=5
060
Via: SIP/2.0/UDP
103.246.89.131:5060;received=103.246.89.131;branch=z9hG4bK17dee670;rport=5060
From: <sip:0060123718121@sip3.providervoip.com:5060>;tag=as636c8a82
To: <sip:605605001@XXX.XXX.XXX.XXX:5062>;tag=as1c55888d
Call-ID: 731eb3b30a08e30c5a2faed25f671a91@XXX.XXX.XXX.XXX:5062
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@192.168.8.101:44591;transport=UDP> for
address/port to send to
set_destination: set destination to 192.168.8.101:44591
Reliably Transmitting (NAT) to 103.255.4.17:15136:
BYE sip:5020@192.168.8.101:44591;transport=UDP SIP/2.0
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5062;branch=z9hG4bK672a0b72;rport
Max-Forwards: 70
From: <sip:0060123718121@XXX.XXX.XXX.XXX:5062;transport=UDP>;tag=as4ab9447f
To: <sip:5020@XXX.XXX.XXX.XXX:5062;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 103.255.4.17:15136:
BYE sip:5020@192.168.8.101:44591;transport=UDP SIP/2.0
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5062;branch=z9hG4bK672a0b72;rport
Max-Forwards: 70
From: <sip:0060123718121@XXX.XXX.XXX.XXX:5062;transport=UDP>;tag=as4ab9447f
To: <sip:5020@XXX.XXX.XXX.XXX:5062;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:103.255.4.17:15136 --->


SIP/2.0 200 OK
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5062;branch=z9hG4bK672a0b72;rport=5261
Contact: <sip:5020@192.168.8.101:44591;transport=UDP>
To: <sip:5020@XXX.XXX.XXX.XXX:5062;transport=UDP>;tag=a566fb5c
From: <sip:0060123718121@XXX.XXX.XXX.XXX:5062;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:103.255.4.17:15136 --->


PUBLISH sip:5020@XXX.XXX.XXX.XXX:5062;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.8.101:44591;branch=z9hG4bK-d8754z-2d82c36de1c5f8bf-1---
d8754z-
Max-Forwards: 70
Contact: <sip:5020@192.168.8.101:44591;transport=UDP>
To: <sip:5020@XXX.XXX.XXX.XXX:5062;transport=UDP>
From: <sip:5020@XXX.XXX.XXX.XXX:5062;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@XXX.XXX.XXX.XXX:5062;transport=UDP"> <tuple id="5020" >
<status><basic>open</basic></status> <note>Online</note> </tuple>
</presence>
<------------->
--- (16 headers 3 lines) ---
Sending to 103.255.4.17:15136 (NAT)

<--- Transmitting (NAT) to 103.255.4.17:15136 --->


SIP/2.0 489 Bad Event
Via: SIP/2.0/UDP 192.168.8.101:44591;branch=z9hG4bK-d8754z-2d82c36de1c5f8bf-1---
d8754z-;received=103.255.4.17;rport=15136
From: <sip:5020@XXX.XXX.XXX.XXX:5062;transport=UDP>;tag=414b3c62
To: <sip:5020@XXX.XXX.XXX.XXX:5062;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:103.255.4.17:15136 --->


SUBSCRIBE sip:5020@XXX.XXX.XXX.XXX:5062;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.8.101:44591;branch=z9hG4bK-d8754z-2dce4186697d0bc1-1---
d8754z-
Max-Forwards: 70
Contact: <sip:5020@192.168.8.101:44591;transport=UDP>
To: <sip:5020@XXX.XXX.XXX.XXX:5062;transport=UDP>
From: <sip:5020@XXX.XXX.XXX.XXX:5062;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: presence.winfo
Allow-Events: presence, kpml
Content-Length: 0

<------------->
--- (16 headers 0 lines) ---
Sending to 103.255.4.17:15136 (NAT)
Creating new subscription
Sending to 103.255.4.17:15136 (NAT)
list_route: hop: <sip:5020@192.168.8.101:44591;transport=UDP>
Found peer '5020' for '5020' from 103.255.4.17:15136

<--- Transmitting (NAT) to 103.255.4.17:15136 --->


SIP/2.0 489 Bad Event
Via: SIP/2.0/UDP 192.168.8.101:44591;branch=z9hG4bK-d8754z-2dce4186697d0bc1-1---
d8754z-;received=103.255.4.17;rport=15136
From: <sip:5020@XXX.XXX.XXX.XXX:5062;transport=UDP>;tag=776f483a
To: <sip:5020@XXX.XXX.XXX.XXX:5062;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:103.255.4.17:15136 --->


PUBLISH sip:testung@103.25.202.195;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.8.101:44591;branch=z9hG4bK-d8754z-51b29fcc1da89b15-1---
d8754z-
Max-Forwards: 70
Contact: <sip:testung@192.168.8.101:44591;transport=UDP>
To: <sip:testung@103.25.202.195;transport=UDP>
From: <sip:testung@103.25.202.195;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@103.25.202.195;transport=UDP"> <tuple id="testung" >
<status><basic>open</basic></status> <note>Online</note> </tuple>
</presence>
<------------->
--- (16 headers 3 lines) ---
Sending to 103.255.4.17:15136 (NAT)

<--- Transmitting (NAT) to 103.255.4.17:15136 --->


SIP/2.0 489 Bad Event
Via: SIP/2.0/UDP 192.168.8.101:44591;branch=z9hG4bK-d8754z-51b29fcc1da89b15-1---
d8754z-;received=103.255.4.17;rport=15136
From: <sip:testung@103.25.202.195;transport=UDP>;tag=f16cc056
To: <sip:testung@103.25.202.195;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:103.255.4.17:15136 --->


SUBSCRIBE sip:testung@103.25.202.195;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.8.101:44591;branch=z9hG4bK-d8754z-fe80cd049e5d9ad5-1---
d8754z-
Max-Forwards: 70
Contact: <sip:testung@192.168.8.101:44591;transport=UDP>
To: <sip:testung@103.25.202.195;transport=UDP>
From: <sip:testung@103.25.202.195;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: presence.winfo
Allow-Events: presence, kpml
Content-Length: 0

<------------->
--- (16 headers 0 lines) ---
Sending to 103.255.4.17:15136 (NAT)
Creating new subscription
Sending to 103.255.4.17:15136 (NAT)
list_route: hop: <sip:testung@192.168.8.101:44591;transport=UDP>
Found peer 'testung' for 'testung' from 103.255.4.17:15136

<--- Transmitting (NAT) to 103.255.4.17:15136 --->


SIP/2.0 489 Bad Event
Via: SIP/2.0/UDP 192.168.8.101:44591;branch=z9hG4bK-d8754z-fe80cd049e5d9ad5-1---
d8754z-;received=103.255.4.17;rport=15136
From: <sip:testung@103.25.202.195;transport=UDP>;tag=4a231410
To: <sip:testung@103.25.202.195;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:103.255.4.17:15136 --->


SIP/2.0 200 OK
Via: SIP/2.0/UDP XXX.XXX.XXX.XXX:5062;branch=z9hG4bK672a0b72;rport=5261
Contact: <sip:5020@192.168.8.101:44591;transport=UDP>
To: <sip:5020@XXX.XXX.XXX.XXX:5062;transport=UDP>;tag=a566fb5c
From: <sip:0060123718121@XXX.XXX.XXX.XXX:5062;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@XXX.XXX.XXX.XXX:5062' Method: BYE
Debian-81-64b*CLI> exit
Asterisk cleanly ending (0).
Executing last minute cleanups
root@Debian-81-64b:~# ~~

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