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Receiving Unknown calls on extension

http://forum.eflo.net/VICIDIALforum/viewtopic.php?f=4&t=36145

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Receiving Unknown calls on extension

PostPosted: Mon Sep 19, 2016 9:43 am
by andrewmarba
Hi,

This is the first time that I am seeking help on this forum. I have been using GoAutoDial CE 2.1 only for outbound calling since Jan 2015 without any problem. And also my knowledge to linux is very limited.

Recently very strange things are happening on the dialer. Let me explain. This started to happen 2 weeks ago. When an agent is logged in suddenly receives inbound from 101 extension and when agent try to answer it is blank. Now the concerned issue is that I dont have any 101 extension configured in goautodial and all extension are in 8001,8002 series. And even if the agent logout from the dialer still he receives 101 call. I also dont have any inbound configured as this dialer is solely for outbound purpose only to UK.

Tried to google around but with no luck...

Here are the details:
Linux go.goautodial.org 2.6.18-238.9.1.el5.goPAE #1 SMP Thu Apr 28 05:24:24 EDT 2011 i686 i686 i386 GNU/Linux
Asterisk : Asterisk 1.4.39.1-vici RPM by demian@goautodial.com

CPU G2010 @ 2.80GHz
RAM 4GB
Intel H61 MotherBoard

No changes in Codecs.

Using IP based Voip.
And Scratch Install.


Dial plan:
[TataVoip]
disallow=all
allow=gsm
allow=ulaw
type=friend
dtmfmode=rfc2833
context=trunkinbound
host=202.54.112.194
qualify=yes
insecure=no
nat=no
allow=alaw
allow=g729

exten => _X.,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _X.,2,Dial(SIP/${EXTEN:1}@TataVoip,,tTo)
exten => _X.,3,Hangup

please let me know if i am missing anything.

Waiting for some to help.......

Re: Receiving Unknown calls on extension

PostPosted: Mon Sep 19, 2016 10:16 am
by ambiorixg12
Start by disallowing guest calls on your sip.conf file

[general]

allowguest=no ; Allow or reject guest calls (default is yes)

Re: Receiving Unknown calls on extension

PostPosted: Mon Sep 19, 2016 2:42 pm
by Vince-0
Show us the Asterisk console logs for this call event.

Re: Receiving Unknown calls on extension

PostPosted: Tue Sep 20, 2016 3:48 am
by andrewmarba
ambiorixg12 wrote:Start by disallowing guest calls on your sip.conf file

[general]

allowguest=no ; Allow or reject guest calls (default is yes)


Thanks for the effort.
I have made the changes and no unwanted calls so far..... Will monitor it and keep you posted.
Thank you once again

Re: Receiving Unknown calls on extension

PostPosted: Thu Sep 22, 2016 4:35 am
by andrewmarba
Hi,

Nightmares are back!!!!!

Today one of the agent started to receive call on his extension (8001) again. I tried to check on asterisk but found nothing. Below is the log from the moment the call came in to the moment the call was rejected by the agent. The log is with the sip debug on. The number displayed on the zoiper during the incoming was 1000


<------------->
[Sep 22 14:58:12] --- (17 headers 0 lines) ---
[Sep 22 14:58:12] Creating new subscription
[Sep 22 14:58:12] Sending to 192.168.1.21 : 5060 (NAT)
[Sep 22 14:58:12] Found peer '8001'
[Sep 22 14:58:12] Looking for 8001 in default (domain 192.168.1.6)
[Sep 22 14:58:12]
<--- Transmitting (NAT) to 192.168.1.21:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 114.143.71.27:5060;branch=z9hG4bK-524287-1---9cb98a3efa2f9863;received=192.168.1.21
From: <sip:8001@192.168.1.6;transport=UDP>;tag=87787b36
To: <sip:8001@192.168.1.6;transport=UDP>;tag=as36448ce9
Call-ID: MvPpRXXeh3xm33VjdjhYJw..
CSeq: 2 SUBSCRIBE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0


<------------>
[Sep 22 14:58:12] Really destroying SIP dialog 'MvPpRXXeh3xm33VjdjhYJw..' Method : SUBSCRIBE
[Sep 22 14:58:13] == Parsing '/etc/asterisk/manager.conf': [Sep 22 14:58:13] Found
[Sep 22 14:58:13] == Manager 'sendcron' logged on from 127.0.0.1
[Sep 22 14:58:13] -- Executing [8600053@default:1] MeetMe("Local/8600053@default-665d,2", "8600053|F") in new stack
[Sep 22 14:58:13] > Channel Local/8600053@default-665d,1 was answered.
[Sep 22 14:58:13] -- Executing [900441243771354@default:1] AGI("Local/8600053@default-665d,1", "agi://127.0.0.1:4577/call_log") in new stack
[Sep 22 14:58:13] -- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
[Sep 22 14:58:13] -- Executing [900441243771354@default:2] Dial("Local/8600053@default-665d,1", "SIP/00441243771354@TataVoip||tTo") in new stack
[Sep 22 14:58:13] Audio is at 192.168.1.6 port 14570
[Sep 22 14:58:13] Adding codec 0x2 (gsm) to SDP
[Sep 22 14:58:13] Adding codec 0x4 (ulaw) to SDP
[Sep 22 14:58:13] Adding codec 0x8 (alaw) to SDP
[Sep 22 14:58:13] Adding non-codec 0x1 (telephone-event) to SDP
[Sep 22 14:58:13] Reliably Transmitting (no NAT) to 202.54.112.194:5060:
INVITE sip:00441243771354@202.54.112.194;cpd=on SIP/2.0
Via: SIP/2.0/UDP 192.168.1.6:5060;branch=z9hG4bK159e284d;rport
From: "M9221458130000082364" <sip:442036032149@192.168.1.6>;tag=as49eedaeb
To: <sip:00441243771354@202.54.112.194;cpd=on>
Contact: <sip:442036032149@192.168.1.6>
Call-ID: 22862bf755a05b4740f6d9ea490496e4@192.168.1.6
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "M9221458130000082364" <sip:442036032149@192.168.1.6>;privacy=off;screen=no
Date: Thu, 22 Sep 2016 09:28:13 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Type: application/sdp
Content-Length: 256

v=0
o=root 2938 2938 IN IP4 192.168.1.6
s=session
c=IN IP4 192.168.1.6
t=0 0
m=audio 14570 RTP/AVP 3 0 8 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
[Sep 22 14:58:13] -- Called 00441243771354@TataVoip
[Sep 22 14:58:13]
<--- SIP read from 202.54.112.194:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.6:5060;rport=5060;received=123.252.131.2;branch=z9hG4bK159e284d
From: "M9221458130000082364" <sip:442036032149@192.168.1.6:5060>;tag=as49eedaeb
To: <sip:00441243771354@202.54.112.194:5060;cpd=on>
Call-ID: 22862bf755a05b4740f6d9ea490496e4@192.168.1.6
CSeq: 102 INVITE
Content-Length: 0


<------------->
[Sep 22 14:58:13] --- (7 headers 0 lines) ---
[Sep 22 14:58:14] == Parsing '/etc/asterisk/manager.conf': [Sep 22 14:58:14] Found
[Sep 22 14:58:14] == Manager 'sendcron' logged on from 127.0.0.1
[Sep 22 14:58:14] -- Executing [128600055@default:1] MeetMeAdmin("Local/128600055@default-5b62,2", "8600055|m|1") in new stack
[Sep 22 14:58:14] -- Executing [128600055@default:2] Hangup("Local/128600055@default-5b62,2", "") in new stack
[Sep 22 14:58:14] == Spawn extension (default, 128600055, 2) exited non-zero on 'Local/128600055@default-5b62,2'
[Sep 22 14:58:14] -- Executing [h@default:1] DeadAGI("Local/128600055@default-5b62,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16----- ----------") in new stack
[Sep 22 14:58:14] -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
[Sep 22 14:58:15]
<--- SIP read from 202.54.112.194:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.1.6:5060;rport=5060;received=123.252.131.2;branch=z9hG4bK159e284d
To: <sip:00441243771354@202.54.112.194:5060;cpd=on>;tag=3683545128-11562
From: "M9221458130000082364" <sip:442036032149@192.168.1.6:5060>;tag=as49eedaeb
Call-ID: 22862bf755a05b4740f6d9ea490496e4@192.168.1.6
CSeq: 102 INVITE
Allow: PUBLISH,MESSAGE,UPDATE,PRACK,SUBSCRIBE,REFER,INFO,NOTIFY,REGISTER,OPTIONS ,BYE,INVITE,ACK,CANCEL
Contact: <sip:00441243771354@202.54.112.194:5060>
Content-Type: application/sdp
Accept: application/sdp
Content-Length: 218

v=0
o=tcl-ent-01 6746 661 IN IP4 202.54.112.194
s=sip call
c=IN IP4 80.231.94.6
t=0 0
m=audio 12508 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=ptime:20

<------------->
[Sep 22 14:58:15] --- (11 headers 11 lines) ---
[Sep 22 14:58:15] Found RTP audio format 0
[Sep 22 14:58:15] Found RTP audio format 101
[Sep 22 14:58:15] Found audio description format PCMU for ID 0
[Sep 22 14:58:15] Found audio description format telephone-event for ID 101
[Sep 22 14:58:15] Capabilities: us - 0x10e (gsm|ulaw|alaw|g729), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
[Sep 22 14:58:15] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
[Sep 22 14:58:15] Peer audio RTP is at port 80.231.94.6:12508
[Sep 22 14:58:15] -- SIP/TataVoip-00001653 is making progress passing it to Local/8600053@default-665d,1
go*CLI>
Disconnected from Asterisk server
Executing last minute cleanups
You have new mail in /var/spool/mail/root

Re: Receiving Unknown calls on extension

PostPosted: Thu Sep 22, 2016 5:08 am
by andrewmarba
Code: Select all
And here is the one with the call accepted Log. The Ip address of the agent is 192.168.1.21, extension is 8001

<------------>
[Sep 22 15:12:52] Scheduling destruction of SIP dialog 'M2RjNzgwNWI0NjkxODJlN2UxMGZkMTU1ZmYwYzJiMzA.' in 6400 ms (Method: SUBSCRIBE)
[Sep 22 15:12:52]
<--- SIP read from 192.168.1.36:49146 --->
SUBSCRIBE sip:8016@192.168.1.6;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 114.143.71.27:15813;branch=z9hG4bK-d8754z-125974c1ed210399-1---d8754z-
Max-Forwards: 70
Contact: <sip:8016@114.143.71.27:15813;transport=UDP>
To: <sip:8016@192.168.1.6;transport=UDP>
From: <sip:8016@192.168.1.6;transport=UDP>;tag=04252a6c
Call-ID: M2RjNzgwNWI0NjkxODJlN2UxMGZkMTU1ZmYwYzJiMzA.
CSeq: 2 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
Authorization: Digest username="8016",realm="asterisk",nonce="5a9a0dc6",uri="sip:8016@192.168.1.6;transport=UDP",response="b9b69d121814230f0033f0fbb41c7e47",algorithm=MD5
Event: presence.winfo
Allow-Events: presence, kpml
Content-Length: 0
<------------->
[Sep 22 15:12:55] --- (14 headers 0 lines) ---
[Sep 22 15:12:55] Really destroying SIP dialog '5df40c91425ca182660336fd04136b11@192.168.1.6' Method: OPTIONS
[Sep 22 15:12:56] Reliably Transmitting (NAT) to 192.168.1.140:40362:
OPTIONS sip:9001@114.143.71.27:10008;rinstance=c3a3e2e7e203fcae;transport=UDP;cpd=on SIP/2.0
Via: SIP/2.0/UDP 192.168.1.6:5060;branch=z9hG4bK1ef27799;rport
From: "asterisk" <sip:asterisk@192.168.1.6>;tag=as22dd74a5
To: <sip:9001@114.143.71.27:10008;rinstance=c3a3e2e7e203fcae;transport=UDP;cpd=on>
Contact: <sip:asterisk@192.168.1.6>
Call-ID: 75427d8b65ecdd9e4a542ad92d0d7f69@192.168.1.6
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
ax-Forwards: 70
Date: Thu, 22 Sep 2016 09:42:56 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0

<------------->
[Sep 22 15:12:56] --- (14 headers 0 lines) ---
[Sep 22 15:12:56] Really destroying SIP dialog '75427d8b65ecdd9e4a542ad92d0d7f69@192.168.1.6' Method: OPTIONS
[Sep 22 15:12:57]   == Parsing '/etc/asterisk/manager.conf': [Sep 22 15:12:57] Found
[Sep 22 15:12:57]   == Manager 'sendcron' logged on from 127.0.0.1
[Sep 22 15:12:57]     -- Executing [h@default:1] DeadAGI("Local/8600051@default-cf2c,1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----43-----30") in new stack
[Sep 22 15:12:57]     -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----43-----30 completed, returning 0
[Sep 22 15:12:57] Scheduling destruction of SIP dialog '4533c0dd34c3c4965bf51e022b8897ff@192.168.1.6' in 6400 ms (Method: INVITE)
[Sep 22 15:12:57] set_destination: Parsing <sip:00442392780643@202.54.112.194:5060> for address/port to send to
[Sep 22 15:12:57] set_destination: set destination to 202.54.112.194, port 5060
[Sep 22 15:12:57] Reliably Transmitting (no NAT) to 202.54.112.194:5060:
BYE sip:00442392780643@202.54.112.194:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.6:5060;branch=z9hG4bK2d654c6d;rport
From: "M9221512110000033220" <sip:442036032149@192.168.1.6>;tag=as72a5aedd
To: <sip:00442392780643@202.54.112.194;cpd=on>;tag=3683545972-211771
Call-ID: 4533c0dd34c3c4965bf51e022b8897ff@192.168.1.6
CSeq: 103 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "M9221512110000033220" <sip:442036032149@192.168.1.6>;privacy=off;screen=no
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0

---
[Sep 22 15:12:57]   == Spawn extension (default, 900442392780643, 2) exited non-zero on 'Local/8600051@default-cf2c,1'
[Sep 22 15:12:57]   == Spawn extension (default, 8600051, 1) exited non-zero on 'Local/8600051@default-cf2c,2'
[Sep 22 15:12:57]     -- Executing [h@default:1] DeadAGI("Local/8600051@default-cf2c,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
[Sep 22 15:12:57]     -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0--------------- completed, returning 0
[Sep 22 15:12:57]   == Parsing '/etc/asterisk/manager.conf': [Sep 22 15:12:57] Found
[Sep 22 15:12:57]   == Manager 'sendcron' logged on from 127.0.0.1
[Sep 22 15:12:57]   == Spawn extension (default, 58600051, 1) exited non-zero on 'Local/58600051@default-0812,2'
[Sep 22 15:12:57]     -- Executing [h@default:1] DeadAGI("Local/58600051@default-0812,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
[Sep 22 15:12:57]   == Parsing '/etc/asterisk/manager.conf': [Sep 22 15:12:57] Found
[Sep 22 15:12:57]   == Manager 'sendcron' logged on from 127.0.0.1
[Sep 22 15:12:57]     -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0--------------- completed, returning 0
[Sep 22 15:12:57]   == Spawn extension (default, 8309, 3) exited non-zero on 'Local/58600051@default-0812,1'
[Sep 22 15:12:57]     -- Executing [h@default:1] DeadAGI("Local/58600051@default-0812,1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
[Sep 22 15:12:57]     -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0--------------- completed, returning 0
[Sep 22 15:12:57]   == Parsing '/etc/asterisk/manager.conf': [Sep 22 15:12:57] Found
[Sep 22 15:12:57]   == Manager 'sendcron' logged on from 127.0.0.1
[Sep 22 15:12:57] Retransmitting #1 (no NAT) to 202.54.112.194:5060:
BYE sip:00442392780643@202.54.112.194:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.6:5060;branch=z9hG4bK2d654c6d;rport
From: "M9221512110000033220" <sip:442036032149@192.168.1.6>;tag=as72a5aedd
To: <sip:00442392780643@202.54.112.194;cpd=on>;tag=3683545972-211771
Call-ID: 4533c0dd34c3c4965bf51e022b8897ff@192.168.1.6
CSeq: 103 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "M9221512110000033220" <sip:442036032149@192.168.1.6>;privacy=off;screen=no
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0

---
[Sep 22 15:12:57]
<--- SIP read from 202.54.112.194:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.6:5060;rport=5060;received=123.252.131.2;branch=z9hG4bK2d654c6d
To: <sip:00442392780643@202.54.112.194:5060;cpd=on>;tag=3683545972-211771
From: "M9221512110000033220" <sip:442036032149@192.168.1.6:5060>;tag=as72a5aedd
Call-ID: 4533c0dd34c3c4965bf51e022b8897ff@192.168.1.6
CSeq: 103 BYE
Allow: PUBLISH,MESSAGE,UPDATE,PRACK,SUBSCRIBE,REFER,INFO,NOTIFY,REGISTER,OPTIONS,BYE,INVITE,ACK,CANCEL
Contact: <sip:00442392780643@202.54.112.194:5060>
Content-Length: 0

<------------->
[Sep 22 15:12:58] --- (13 headers 14 lines) ---
[Sep 22 15:12:58] Sending to 202.54.112.194 : 5060 (no NAT)
[Sep 22 15:12:58] Found RTP audio format 8
[Sep 22 15:12:58] Found RTP audio format 0
[Sep 22 15:12:58] Found RTP audio format 18
[Sep 22 15:12:58] Found RTP audio format 101
[Sep 22 15:12:58] Found audio description format PCMA for ID 8
[Sep 22 15:12:58] Found audio description format PCMU for ID 0
[Sep 22 15:12:58] Found audio description format G729 for ID 18
[Sep 22 15:12:58] Found audio description format telephone-event for ID 101
[Sep 22 15:12:58] Capabilities: us - 0x10e (gsm|ulaw|alaw|g729), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x10c (ulaw|alaw|g729)
[Sep 22 15:12:58] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
[Sep 22 15:12:58] Peer audio RTP is at port 209.58.46.7:27352
[Sep 22 15:12:58]
<--- Transmitting (no NAT) to 202.54.112.194:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 202.54.112.194:5060;branch=z9hG4bKc616902564f0a3044c9dd6d01a00c3f2;received=202.54.112.194
From: <sip:00441202876013@202.54.112.194:5060;cpd=on>;tag=3683545978-750949
To: "M9221512230000127709" <sip:442036032149@192.168.1.6:5060>;tag=as547a7dd1
Call-ID: 37c1696f6c852e06487a4c087402e407@192.168.1.6
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: <sip:442036032149@192.168.1.6>
Content-Length: 0

<------------>
[Sep 22 15:12:58]     -- Started music on hold, class 'default', on Local/8600058@default-d249,1
[Sep 22 15:12:58]   == Parsing '/etc/asterisk/manager.conf': [Sep 22 15:12:58] Found
[Sep 22 15:12:58]   == Manager 'sendcron' logged on from 127.0.0.1
[Sep 22 15:12:58]     -- Stopped music on hold on Local/8600058@default-d249,1
[Sep 22 15:12:58]     -- Executing [h@default:1] DeadAGI("Local/8600058@default-d249,1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----35-----18") in new stack
[Sep 22 15:12:58]   == Parsing '/etc/asterisk/manager.conf': [Sep 22 15:12:58] Found
[Sep 22 15:12:58]   == Manager 'sendcron' logged on from 127.0.0.1
[Sep 22 15:12:58]   == Spawn extension (default, 58600058, 1) exited non-zero on 'Local/58600058@default-a8a8,2'
[Sep 22 15:12:58]     -- Executing [h@default:1] DeadAGI("Local/58600058@default-a8a8,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
[Sep 22 15:12:58]   == Parsing '/etc/asterisk/manager.conf': [Sep 22 15:12:58] Found
[Sep 22 15:12:58]   == Manager 'sendcron' logged on from 127.0.0.1
[Sep 22 15:12:58]     -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----ANSWER-----35-----18 completed, returning 0
[Sep 22 15:12:58] Scheduling destruction of SIP dialog '37c1696f6c852e06487a4c087402e407@192.168.1.6' in 6400 ms (Method: INVITE)
[Sep 22 15:12:58]   == Spawn extension (default, 900441202876013, 2) exited non-zero on 'Local/8600058@default-d249,1'
[Sep 22 15:12:58]   == Spawn extension (default, 8600058, 1) exited non-zero on 'Local/8600058@default-d249,2'
[Sep 22 15:12:58]     -- Executing [h@default:1] DeadAGI("Local/8600058@default-d249,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
[Sep 22 15:12:58]     -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0--------------- completed, returning 0
[Sep 22 15:12:58]   == Spawn extension (default, 8309, 3) exited non-zero on 'Local/58600058@default-a8a8,1'
[Sep 22 15:12:58]     -- Executing [h@default:1] DeadAGI("Local/58600058@default-a8a8,1", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0---------------") in new stack
[Sep 22 15:12:58]     -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0--------------- completed, returning 0
[Sep 22 15:12:58]     -- AGI Script agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0--------------- completed, returning 0
[Sep 22 15:12:58]   == Parsing '/etc/asterisk/manager.conf': [Sep 22 15:12:58] Found
[Sep 22 15:12:58]   == Manager 'sendcron' logged on from 127.0.0.1
[Sep 22 15:12:58] Retransmitting #1 (no NAT) to 202.54.112.194:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 202.54.112.194:5060;branch=z9hG4bKc616902564f0a3044c9dd6d01a00c3f2;received=202.54.112.194
From: <sip:00441202876013@202.54.112.194:5060;cpd=on>;tag=3683545978-750949
To: "M9221512230000127709" <sip:442036032149@192.168.1.6:5060>;tag=as547a7dd1
Call-ID: 37c1696f6c852e06487a4c087402e407@192.168.1.6
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: <sip:442036032149@192.168.1.6>
Content-Type: application/sdp
Content-Length: 280

v=0
o=root 2938 2939 IN IP4 192.168.1.6
s=session
c=IN IP4 192.168.1.6
t=0 0
m=audio 18796 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=recvonly

---
[Sep 22 15:12:58]
<--- SIP read from 202.54.112.194:5060 --->
ACK sip:442036032149@192.168.1.6:5060 SIP/2.0
Max-Forwards: 69
To: "M9221512230000127709" <sip:442036032149@192.168.1.6:5060>;tag=as547a7dd1
From: <sip:00441202876013@202.54.112.194:5060;cpd=on>;tag=3683545978-750949
Call-ID: 37c1696f6c852e06487a4c087402e407@192.168.1.6
CSeq: 2 ACK
Allow: PUBLISH,MESSAGE,UPDATE,SUBSCRIBE,REFER,INFO,NOTIFY,REGISTER,OPTIONS,BYE,INVITE,ACK,CANCEL
Via: SIP/2.0/UDP 202.54.112.194:5060;branch=z9hG4bKb9c48611d8fcfa53fd9c51bc0605be81
Contact: <sip:00441202876013@202.54.112.194:5060>
Content-Length: 0

---
[Sep 22 15:12:58] Scheduling destruction of SIP dialog '37c1696f6c852e06487a4c087402e407@192.168.1.6' in 6400 ms (Method: ACK)
[Sep 22 15:12:59] Retransmitting #1 (no NAT) to 202.54.112.194:5060:
BYE sip:00441202876013@202.54.112.194:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.6:5060;branch=z9hG4bK01b03599;rport
From: "M9221512230000127709" <sip:442036032149@192.168.1.6:5060>;tag=as547a7dd1
To: <sip:00441202876013@202.54.112.194:5060;cpd=on>;tag=3683545978-750949
Call-ID: 37c1696f6c852e06487a4c087402e407@192.168.1.6
CSeq: 103 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "M9221512230000127709" <sip:442036032149@192.168.1.6>;privacy=off;screen=no
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0

---
[Sep 22 15:12:59]
<--- SIP read from 202.54.112.194:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.6:5060;rport=5060;received=123.252.131.2;branch=z9hG4bK01b03599
To: <sip:00441202876013@202.54.112.194:5060;cpd=on>;tag=3683545978-750949
From: "M9221512230000127709" <sip:442036032149@192.168.1.6:5060>;tag=as547a7dd1
Call-ID: 37c1696f6c852e06487a4c087402e407@192.168.1.6
CSeq: 103 BYE
Allow: PUBLISH,MESSAGE,UPDATE,PRACK,SUBSCRIBE,REFER,INFO,NOTIFY,REGISTER,OPTIONS,BYE,INVITE,ACK,CANCEL
Contact: <sip:00441202876013@202.54.112.194:5060>
Content-Length: 0

<------------->
[Sep 22 15:12:59] --- (9 headers 0 lines) ---
[Sep 22 15:12:59] SIP Response message for INCOMING dialog BYE arrived
[Sep 22 15:12:59] Really destroying SIP dialog '37c1696f6c852e06487a4c087402e407@192.168.1.6' Method: ACK
[Sep 22 15:12:59]   == Manager 'sendcron' logged off from 127.0.0.1
[Sep 22 15:12:59]   == Manager 'sendcron' logged off from 127.0.0.1
[Sep 22 15:12:59]   == Manager 'sendcron' logged off from 127.0.0.1
[Sep 22 15:12:59]   == Manager 'sendcron' logged off from 127.0.0.1
[Sep 22 15:13:00]
<--- SIP read from 202.54.112.194:5060 --->
INVITE sip:442036032149@192.168.1.6:5060 SIP/2.0
Max-Forwards: 69
Supported: 100rel
To: "M9221511220000125121" <sip:442036032149@192.168.1.6:5060>;tag=as74a3a0a6
From: <sip:00441406380812@202.54.112.194:5060;cpd=on>;tag=3683545917-959183
Call-ID: 1fcbad2337b410b0496d0ebd329d6338@192.168.1.6
CSeq: 2 INVITE
Allow: PUBLISH,MESSAGE,UPDATE,PRACK,SUBSCRIBE,REFER,INFO,NOTIFY,REGISTER,OPTIONS,BYE,INVITE,ACK,CANCEL
Via: SIP/2.0/UDP 202.54.112.194:5060;branch=z9hG4bKc4009f36b3e9e283db69886ea8da613a
Contact: <sip:00441406380812@202.54.112.194:5060>
Content-Type: application/sdp
Accept: application/sdp
Content-Length: 291

v=0
o=tcl-ent-01 2125 29089 IN IP4 202.54.112.194
s=sip call
c=IN IP4 209.58.46.6
t=0 0
m=audio 13498 RTP/AVP 8 0 18 101
a=sendonly
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20

<------------->
[Sep 22 15:13:00] --- (13 headers 14 lines) ---
[Sep 22 15:13:00] Sending to 202.54.112.194 : 5060 (no NAT)
[Sep 22 15:13:00] Found RTP audio format 8
[Sep 22 15:13:00] Found RTP audio format 0
[Sep 22 15:13:00] Found RTP audio format 18
[Sep 22 15:13:00] Found RTP audio format 101
[Sep 22 15:13:00] Found audio description format PCMA for ID 8
[Sep 22 15:13:00] Found audio description format PCMU for ID 0
[Sep 22 15:13:00] Found audio description format G729 for ID 18
[Sep 22 15:13:00] Found audio description format telephone-event for ID 101
[Sep 22 15:13:00] Capabilities: us - 0x10e (gsm|ulaw|alaw|g729), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x10c (ulaw|alaw|g729)
[Sep 22 15:13:00] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
[Sep 22 15:13:00] Peer audio RTP is at port 209.58.46.6:13498
[Sep 22 15:13:00]
<--- Transmitting (no NAT) to 202.54.112.194:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 202.54.112.194:5060;branch=z9hG4bKc4009f36b3e9e283db69886ea8da613a;received=202.54.112.194
From: <sip:00441406380812@202.54.112.194:5060;cpd=on>;tag=3683545917-959183
To: "M9221511220000125121" <sip:442036032149@192.168.1.6:5060>;tag=as74a3a0a6
Call-ID: 1fcbad2337b410b0496d0ebd329d6338@192.168.1.6
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: <sip:442036032149@192.168.1.6>
Content-Length: 0

<------------>
[Sep 22 15:13:00] Audio is at 192.168.1.6 port 13600
[Sep 22 15:13:00] Adding codec 0x4 (ulaw) to SDP
[Sep 22 15:13:00] Adding codec 0x8 (alaw) to SDP
[Sep 22 15:13:00] Adding codec 0x100 (g729) to SDP
[Sep 22 15:13:00] Adding non-codec 0x1 (telephone-event) to SDP
[Sep 22 15:13:00]
<--- Reliably Transmitting (no NAT) to 202.54.112.194:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 202.54.112.194:5060;branch=z9hG4bKc4009f36b3e9e283db69886ea8da613a;received=202.54.112.194
From: <sip:00441406380812@202.54.112.194:5060;cpd=on>;tag=3683545917-959183
To: "M9221511220000125121" <sip:442036032149@192.168.1.6:5060>;tag=as74a3a0a6
Call-ID: 1fcbad2337b410b0496d0ebd329d6338@192.168.1.6
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: <sip:442036032149@192.168.1.6>
Content-Type: application/sdp
Content-Length: 280

v=0
o=root 2938 2939 IN IP4 192.168.1.6
s=session
c=IN IP4 192.168.1.6
t=0 0
m=audio 13600 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=recvonly

<------------>
[Sep 22 15:13:00]     -- Started music on hold, class 'default', on Local/8600059@default-6bde,1
[Sep 22 15:13:00] Retransmitting #1 (no NAT) to 202.54.112.194:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 202.54.112.194:5060;branch=z9hG4bKc4009f36b3e9e283db69886ea8da613a;received=202.54.112.194
From: <sip:00441406380812@202.54.112.194:5060;cpd=on>;tag=3683545917-959183
To: "M9221511220000125121" <sip:442036032149@192.168.1.6:5060>;tag=as74a3a0a6
Call-ID: 1fcbad2337b410b0496d0ebd329d6338@192.168.1.6
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: <sip:442036032149@192.168.1.6>
Content-Type: application/sdp
Content-Length: 280

v=0
o=root 2938 2939 IN IP4 192.168.1.6
s=session
c=IN IP4 192.168.1.6
t=0 0
m=audio 13600 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=recvonly

---
[Sep 22 15:13:00]
<--- SIP read from 202.54.112.194:5060 --->
ACK sip:442036032149@192.168.1.6:5060 SIP/2.0
Max-Forwards: 69
To: "M9221511220000125121" <sip:442036032149@192.168.1.6:5060>;tag=as74a3a0a6
From: <sip:00441406380812@202.54.112.194:5060;cpd=on>;tag=3683545917-959183
Call-ID: 1fcbad2337b410b0496d0ebd329d6338@192.168.1.6
CSeq: 2 ACK
Allow: PUBLISH,MESSAGE,UPDATE,SUBSCRIBE,REFER,INFO,NOTIFY,REGISTER,OPTIONS,BYE,INVITE,ACK,CANCEL
Via: SIP/2.0/UDP 202.54.112.194:5060;branch=z9hG4bK6c02b94eaf3b8f6ecfca5b4cf5bcb1ec
Contact: <sip:00441406380812@202.54.112.194:5060>
Content-Length: 0

<------------->
[Sep 22 15:13:00] --- (10 headers 0 lines) ---
[Sep 22 15:13:00] NOTICE[15084]: channel.c:2616 __ast_read: Dropping incompatibl                                                                                        e voice frame on SIP/TataVoip-00001675 of format alaw since our native format ha                                                                                        s changed to 0x4 (ulaw)
[Sep 22 15:13:00]   == Manager 'sendcron' logged off from 127.0.0.1
[Sep 22 15:13:00]
<--- SIP read from 202.54.112.194:5060 --->
INVITE sip:442036032149@192.168.1.6:5060 SIP/2.0
Max-Forwards: 69
Supported: 100rel
To: "M9221511220000125121" <sip:442036032149@192.168.1.6:5060>;tag=as74a3a0a6
From: <sip:00441406380812@202.54.112.194:5060;cpd=on>;tag=3683545917-959183
Call-ID: 1fcbad2337b410b0496d0ebd329d6338@192.168.1.6
CSeq: 3 INVITE
Allow: PUBLISH,MESSAGE,UPDATE,PRACK,SUBSCRIBE,REFER,INFO,NOTIFY,REGISTER,OPTIONS                                                                                        ,BYE,INVITE,ACK,CANCEL
Via: SIP/2.0/UDP 202.54.112.194:5060;branch=z9hG4bK9fe1b7165fcbe162c219bb6d2d4d3                                                                                        4ab
Contact: <sip:00441406380812@202.54.112.194:5060>
Content-Type: application/sdp
Accept: application/sdp
Content-Length: 220

v=0
o=tcl-ent-01 2125 29090 IN IP4 202.54.112.194
s=sip call
c=IN IP4 209.58.46.6
t=0 0
m=audio 13498 RTP/AVP 8 101
a=sendonly
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20

<------------->
[Sep 22 15:13:00] --- (13 headers 11 lines) ---
[Sep 22 15:13:00] Sending to 202.54.112.194 : 5060 (no NAT)
[Sep 22 15:13:00] Found RTP audio format 8
[Sep 22 15:13:00] Found RTP audio format 101
[Sep 22 15:13:00] Found audio description format PCMA for ID 8
[Sep 22 15:13:00] Found audio description format telephone-event for ID 101
[Sep 22 15:13:00] Capabilities: us - 0x10e (gsm|ulaw|alaw|g729), peer - audio=0x                                                                                        8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw)
[Sep 22 15:13:00] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), pee                                                                                        r - 0x1 (telephone-event), combined - 0x1 (telephone-event)
[Sep 22 15:13:00] Peer audio RTP is at port 209.58.46.6:13498
[Sep 22 15:13:00]
<--- Transmitting (no NAT) to 202.54.112.194:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 202.54.112.194:5060;branch=z9hG4bK9fe1b7165fcbe162c219bb6d2d4d3                                                                                        4ab;received=202.54.112.194
From: <sip:00441406380812@202.54.112.194:5060;cpd=on>;tag=3683545917-959183
To: "M9221511220000125121" <sip:442036032149@192.168.1.6:5060>;tag=as74a3a0a6
Call-ID: 1fcbad2337b410b0496d0ebd329d6338@192.168.1.6
CSeq: 3 INVITE
ser-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: <sip:442036032149@192.168.1.6>
Content-Length: 0

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

[Sep 22 15:13:10]
<--- SIP read from 202.54.112.194:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.1.6:5060;rport=5060;received=123.252.131.2;branch=z9hG4                                                                                        bK22acfa35
To: <sip:00441772316415@202.54.112.194:5060;cpd=on>;tag=3683546023-85149
From: "M9221513070000086461" <sip:442036032149@192.168.1.6:5060>;tag=as619625c8
Call-ID: 54ebf92f7cedb8467a6526bd2f271389@192.168.1.6
CSeq: 102 INVITE
Allow: PUBLISH,MESSAGE,UPDATE,PRACK,SUBSCRIBE,REFER,INFO,NOTIFY,REGISTER,OPTIONS                                                                                        ,BYE,INVITE,ACK,CANCEL
Contact: <sip:00441772316415@202.54.112.194:5060>
Content-Type: application/sdp
Accept: application/sdp
Content-Length: 221

v=0
o=tcl-ent-01 26972 18527 IN IP4 202.54.112.194
s=sip call
c=IN IP4 209.58.46.6
t=0 0
m=audio 25908 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=ptime:20

<------------->
[Sep 22 15:13:10] --- (11 headers 11 lines) ---
[Sep 22 15:13:10] Found RTP audio format 0
[Sep 22 15:13:10] Found RTP audio format 101
[Sep 22 15:13:10] Found audio description format PCMU for ID 0
[Sep 22 15:13:10] Found audio description format telephone-event for ID 101
[Sep 22 15:13:10] Capabilities: us - 0x10e (gsm|ulaw|alaw|g729), peer - audio=0x                                                                                        4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
[Sep 22 15:13:10] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), pee                                                                                        r - 0x1 (telephone-event), combined - 0x1 (telephone-event)
[Sep 22 15:13:10] Peer audio RTP is at port 209.58.46.6:25908
[Sep 22 15:13:10]     -- SIP/TataVoip-00001679 is making progress passing it to                                                                                         Local/8600061@default-3e8a,1
[Sep 22 15:13:10]
<--- SIP read from 192.168.1.21:5060 --->
INVITE sip:442036032149@192.168.1.6 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.21:5060;branch=z9hG4bK-524287-1---94ffeecff90d333c
Max-Forwards: 70
Contact: <sip:8001@114.143.71.27:5060>
To: "S1609221429458600057" <sip:442036032149@192.168.1.6>;tag=as70598a1a
From: <sip:8001@114.143.71.27:5060;rinstance=be557f71884d5872;transport=UDP;cpd=                                                                                        on>;tag=da573746
Call-ID: 19da0e7b62dd1e0d4889afe502ffae7d@192.168.1.6
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIB                                                                                        E
Content-Type: application/sdp
Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-                                                                                        serviceuri
User-Agent: Z 3.7.30891 r30851
Allow-Events: presence, kpml
Content-Length: 242

v=0
o=Z 0 4 IN IP4 114.143.71.27
s=Z
c=IN IP4 114.143.71.27
t=0 0
m=audio 16384 RTP/AVP 0 3 110 8 97 101
a=rtpmap:110 speex/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=inactive

<------------->
[Sep 22 15:13:10] --- (14 headers 12 lines) ---
[Sep 22 15:13:10] Sending to 192.168.1.21 : 5060 (NAT)
[Sep 22 15:13:10] Found RTP audio format 0
[Sep 22 15:13:10] Found RTP audio format 3
[Sep 22 15:13:10] Found RTP audio format 110
[Sep 22 15:13:10] Found RTP audio format 8
[Sep 22 15:13:10] Found RTP audio format 97
[Sep 22 15:13:10] Found RTP audio format 101
[Sep 22 15:13:10] Found audio description format speex for ID 110
[Sep 22 15:13:10] Found audio description format iLBC for ID 97
[Sep 22 15:13:10] Found audio description format telephone-event for ID 101
[Sep 22 15:13:10] Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x60e (gsm|ula                                                                                        w|alaw|speex|ilbc)/video=0x0 (nothing), combined - 0x6 (gsm|ulaw)
[Sep 22 15:13:10] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), pee                                                                                        r - 0x1 (telephone-event), combined - 0x1 (telephone-event)
[Sep 22 15:13:10] Peer audio RTP is at port 114.143.71.27:16384
[Sep 22 15:13:10]
<--- Transmitting (NAT) to 192.168.1.21:5060 --->
IP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.21:5060;branch=z9hG4bK-524287-1---94ffeecff90d333c;re                                                                                        ceived=192.168.1.21
From: <sip:8001@114.143.71.27:5060;rinstance=be557f71884d5872;transport=UDP;cpd=                                                                                        on>;tag=da573746
To: "S1609221429458600057" <sip:442036032149@192.168.1.6>;tag=as70598a1a
Call-ID: 19da0e7b62dd1e0d4889afe502ffae7d@192.168.1.6
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: <sip:442036032149@192.168.1.6>
Content-Length: 0

<------------>
[Sep 22 15:13:10] Audio is at 192.168.1.6 port 16002
[Sep 22 15:13:10] Adding codec 0x4 (ulaw) to SDP
[Sep 22 15:13:10] Adding codec 0x2 (gsm) to SDP
[Sep 22 15:13:10] Adding non-codec 0x1 (telephone-event) to SDP
[Sep 22 15:13:10]
<--- Reliably Transmitting (NAT) to 192.168.1.21:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.21:5060;branch=z9hG4bK-524287-1---94ffeecff90d333c;re                                                                                        ceived=192.168.1.21
From: <sip:8001@114.143.71.27:5060;rinstance=be557f71884d5872;transport=UDP;cpd=                                                                                        on>;tag=da573746
To: "S1609221429458600057" <sip:442036032149@192.168.1.6>;tag=as70598a1a
Call-ID: 19da0e7b62dd1e0d4889afe502ffae7d@192.168.1.6
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: <sip:442036032149@192.168.1.6>
Content-Type: application/sdp
Content-Length: 232

v=0
o=root 2938 2939 IN IP4 192.168.1.6
s=session
c=IN IP4 192.168.1.6
t=0 0
m=audio 16002 RTP/AVP 0 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=inactive

<------------>
[Sep 22 15:13:10]
<--- SIP read from 192.168.1.21:5060 --->
PUBLISH sip:8001@192.168.1.6;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 114.143.71.27:5060;branch=z9hG4bK-524287-1---5f0c0b10dffbdc3f
Max-Forwards: 70
Contact: <sip:8001@114.143.71.27:5060;transport=UDP>
To: <sip:8001@192.168.1.6;transport=UDP>
From: <sip:8001@192.168.1.6;transport=UDP>;tag=d24cb42e
Call-ID: ROv8wW9xsURnUTc0t4r0tg..
CSeq: 1 PUBLISH
Expires: 600
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIB                                                                                        E
Content-Type: application/pidf+xml
Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-                                                                                        serviceuri
User-Agent: Z 3.7.30891 r30851
Event: presence
Allow-Events: presence, kpml
Content-Length: 265

<?xml version="1.0" encoding="UTF-8"?>
<presence xmlns="urn:ietf:params:xml:ns:pidf"
          entity="sip:8001@192.168.1.6;transport=UDP">
  <tuple id="8001" >
     <status><basic>open</basic></status>
     <note>On the phone</note>
  </tuple>
</presence>

<------------->
[Sep 22 15:13:10] --- (16 headers 8 lines) ---
[Sep 22 15:13:10] Sending to 192.168.1.21 : 5060 (NAT)
[Sep 22 15:13:10]
<--- Transmitting (NAT) to 192.168.1.21:5060 --->
SIP/2.0 501 Method Not Implemented
Via: SIP/2.0/UDP 114.143.71.27:5060;branch=z9hG4bK-524287-1---5f0c0b10dffbdc3f;r                                                                                        eceived=192.168.1.21
From: <sip:8001@192.168.1.6;transport=UDP>;tag=d24cb42e
To: <sip:8001@192.168.1.6;transport=UDP>;tag=as2618cfdf
Call-ID: ROv8wW9xsURnUTc0t4r0tg..
CSeq: 1 PUBLISH
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0

<------------>
[Sep 22 15:13:10]
<--- SIP read from 192.168.1.21:5060 --->
SUBSCRIBE sip:8001@192.168.1.6;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 114.143.71.27:5060;branch=z9hG4bK-524287-1---e2832f8bb11612c4
Max-Forwards: 70
Contact: <sip:8001@114.143.71.27:5060;transport=UDP>
To: <sip:8001@192.168.1.6;transport=UDP>
From: <sip:8001@192.168.1.6;transport=UDP>;tag=d9079662
Call-ID: oK075qwmQxNYebwAD2Eo7Q..
CSeq: 1 SUBSCRIBE
Expires: 600
Accept: application/watcherinfo+xml
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIB                                                                                        E
Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-                                                                                        serviceuri
User-Agent: Z 3.7.30891 r30851
Event: presence.winfo
Allow-Events: presence, kpml
Content-Length: 0

<------------->
[Sep 22 15:13:10] --- (16 headers 0 lines) ---
[Sep 22 15:13:10] Creating new subscription
[Sep 22 15:13:10] Sending to 192.168.1.21 : 5060 (NAT)
[Sep 22 15:13:10] list_route: hop: <sip:8001@114.143.71.27:5060;transport=UDP>
[Sep 22 15:13:10] Found peer '8001'
[Sep 22 15:13:10]
<--- Transmitting (NAT) to 192.168.1.21:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 114.143.71.27:5060;branch=z9hG4bK-524287-1---e2832f8bb11612c4;r                                                                                        eceived=192.168.1.21
From: <sip:8001@192.168.1.6;transport=UDP>;tag=d9079662
To: <sip:8001@192.168.1.6;transport=UDP>;tag=as51b8ad06
Call-ID: oK075qwmQxNYebwAD2Eo7Q..
CSeq: 1 SUBSCRIBE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="7be29d4d"
Content-Length: 0

<------------>
[Sep 22 15:13:10] Scheduling destruction of SIP dialog 'oK075qwmQxNYebwAD2Eo7Q..                                                                                        ' in 6400 ms (Method: SUBSCRIBE)
[Sep 22 15:13:10]
<--- SIP read from 192.168.1.21:5060 --->
ACK sip:442036032149@192.168.1.6 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.21:5060;branch=z9hG4bK-524287-1---2771b950ea06185d
Max-Forwards: 70
Contact: <sip:8001@114.143.71.27:5060>
To: "S1609221429458600057" <sip:442036032149@192.168.1.6>;tag=as70598a1a
From: <sip:8001@114.143.71.27:5060;rinstance=be557f71884d5872;transport=UDP;cpd=                                                                                        on>;tag=da573746
Call-ID: 19da0e7b62dd1e0d4889afe502ffae7d@192.168.1.6
CSeq: 2 ACK
User-Agent: Z 3.7.30891 r30851
Content-Length: 0

<------------->
[Sep 22 15:13:10] --- (10 headers 0 lines) ---
[Sep 22 15:13:10]
<--- SIP read from 192.168.1.21:5060 --->
SUBSCRIBE sip:8001@192.168.1.6;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 114.143.71.27:5060;branch=z9hG4bK-524287-1---e6db151c456a1cef
Max-Forwards: 70
Contact: <sip:8001@114.143.71.27:5060;transport=UDP>
To: <sip:8001@192.168.1.6;transport=UDP>
From: <sip:8001@192.168.1.6;transport=UDP>;tag=d9079662
Call-ID: oK075qwmQxNYebwAD2Eo7Q..
CSeq: 2 SUBSCRIBE
Expires: 600
Accept: application/watcherinfo+xml
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIB                                                                                        E
Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-                                                                                        serviceuri
User-Agent: Z 3.7.30891 r30851
Authorization: Digest username="8001",realm="asterisk",nonce="7be29d4d",uri="sip                                                                                        :8001@192.168.1.6;transport=UDP",response="eea1d5d02efdb4798e6376374ff1e374",alg                                                                                        orithm=MD5
Event: presence.winfo
Allow-Events: presence, kpml
Content-Length: 0

<------------->
[Sep 22 15:13:10] --- (17 headers 0 lines) ---
[Sep 22 15:13:10] Creating new subscription
[Sep 22 15:13:10] Sending to 192.168.1.21 : 5060 (NAT)
[Sep 22 15:13:10] Found peer '8001'
[Sep 22 15:13:10] Looking for 8001 in default (domain 192.168.1.6)
[Sep 22 15:13:10]
<--- Transmitting (NAT) to 192.168.1.21:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 114.143.71.27:5060;branch=z9hG4bK-524287-1---e6db151c456a1cef;r                                                                                        eceived=192.168.1.21
From: <sip:8001@192.168.1.6;transport=UDP>;tag=d9079662
To: <sip:8001@192.168.1.6;transport=UDP>;tag=as51b8ad06
Call-ID: oK075qwmQxNYebwAD2Eo7Q..
CSeq: 2 SUBSCRIBE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0

<------------>
[Sep 22 15:13:10]   == Manager 'sendcron' logged off from 127.0.0.1
[Sep 22 15:13:10] Really destroying SIP dialog 'oK075qwmQxNYebwAD2Eo7Q..' Method                                                                                        : SUBSCRIBE
[Sep 22 15:13:10]   == Manager 'sendcron' logged off from 127.0.0.1
[Sep 22 15:13:11]   == Manager 'sendcron' logged off from 127.0.0.1
[Sep 22 15:13:11]   == Manager 'sendcron' logged off from 127.0.0.1
[Sep 22 15:13:16] Reliably Transmitting (NAT) to 192.168.1.30:5060:
OPTIONS sip:8010@114.143.71.27:28132;rinstance=f2d19d2714928398;transport=UDP;cp                                                                                        d=on SIP/2.0
Via: SIP/2.0/UDP 192.168.1.6:5060;branch=z9hG4bK0ae40565;rport
From: "asterisk" <sip:asterisk@192.168.1.6>;tag=as4a1a16cb
To: <sip:8010@114.143.71.27:28132;rinstance=f2d19d2714928398;transport=UDP;cpd=o                                                                                        n>
Contact: <sip:asterisk@192.168.1.6>
Call-ID: 08a8e3771666e096500c4596240438fd@192.168.1.6
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Thu, 22 Sep 2016 09:43:16 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0

---
[Sep 22 15:13:16]
<--- SIP read from 192.168.1.30:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.6:5060;branch=z9hG4bK0ae40565;rport=5060
Contact: <sip:192.168.1.30:5060>
To: <sip:8010@114.143.71.27:28132;rinstance=f2d19d2714928398;transport=UDP;cpd=o                                                                                        n>;tag=7a356913
From: "asterisk"<sip:asterisk@192.168.1.6>;tag=as4a1a16cb
Call-ID: 08a8e3771666e096500c4596240438fd@192.168.1.6
CSeq: 102 OPTIONS
Accept: application/sdp, application/sdp
Accept-Language: en
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIB                                                                                        E
Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri
User-Agent: Z 3.6.25251 r25476
Allow-Events: presence, kpml
Content-Length: 0

<------------->
[Sep 22 15:13:16] --- (14 headers 0 lines) ---
[Sep 22 15:13:16] Really destroying SIP dialog '08a8e3771666e096500c4596240438fd                                                                                        @192.168.1.6' Method: OPTIONS
[Sep 22 15:13:17] Reliably Transmitting (NAT) to 192.168.1.21:5060:
OPTIONS sip:8001@114.143.71.27:5060;rinstance=be557f71884d5872;transport=UDP;cpd                                                                                        =on SIP/2.0
Via: SIP/2.0/UDP 192.168.1.6:5060;branch=z9hG4bK29de28c3;rport
From: "asterisk" <sip:asterisk@192.168.1.6>;tag=as4f6209af
To: <sip:8001@114.143.71.27:5060;rinstance=be557f71884d5872;transport=UDP;cpd=on                                                                                        >
Contact: <sip:asterisk@192.168.1.6>
Call-ID: 78c8c27870243ec11bd8b5c255419a5f@192.168.1.6
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Thu, 22 Sep 2016 09:43:17 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0

---
[Sep 22 15:13:17]
<--- SIP read from 192.168.1.21:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.6:5060;branch=z9hG4bK29de28c3;rport=5060
Contact: <sip:192.168.1.21:5060>
To: <sip:8001@114.143.71.27:5060;rinstance=be557f71884d5872;transport=UDP;cpd=on                                                                                        >;tag=5e4bbb65
From: "asterisk" <sip:asterisk@192.168.1.6>;tag=as4f6209af
Call-ID: 78c8c27870243ec11bd8b5c255419a5f@192.168.1.6
CSeq: 102 OPTIONS
Accept: application/sdp, application/sdp
Accept-Language: en
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIB                                                                                        E
Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-                                                                                        serviceuri
User-Agent: Z 3.7.30891 r30851
Allow-Events: presence, kpml
Content-Length: 0

<------------->
[Sep 22 15:13:17] --- (14 headers 0 lines) ---
[Sep 22 15:13:17] Really destroying SIP dialog '78c8c27870243ec11bd8b5c255419a5f                                                                                        @192.168.1.6' Method: OPTIONS
[Sep 22 15:13:18] Reliably Transmitting (NAT) to 192.168.1.80:38860:
OPTIONS sip:8017@114.143.71.27:29214;rinstance=4d89e88116fe6fec;transport=UDP;cp                                                                                        d=on SIP/2.0
Via: SIP/2.0/UDP 192.168.1.6:5060;branch=z9hG4bK7d054523;rport
From: "asterisk" <sip:asterisk@192.168.1.6>;tag=as54b504ff
To: <sip:8017@114.143.71.27:29214;rinstance=4d89e88116fe6fec;transport=UDP;cpd=o                                                                                        n>
Contact: <sip:asterisk@192.168.1.6>
Call-ID: 606725ee17e9d16e3c41e89e112de1ce@192.168.1.6
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Thu, 22 Sep 2016 09:43:18 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0

---
[Sep 22 15:13:18]
<--- SIP read from 192.168.1.80:38860 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.6:5060;branch=z9hG4bK7d054523;rport=5060
Contact: <sip:192.168.1.80:38860>
To: <sip:8017@114.143.71.27:29214;rinstance=4d89e88116fe6fec;transport=UDP;cpd=o                                                                                        n>;tag=772e436c
From: "asterisk" <sip:asterisk@192.168.1.6>;tag=as54b504ff
Call-ID: 606725ee17e9d16e3c41e89e112de1ce@192.168.1.6
CSeq: 102 OPTIONS
Accept: application/sdp, application/sdp
Accept-Language: en
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIB                                                                                        E
Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-                                                                                        serviceuri
User-Agent: Z 3.7.30891 r30851
Allow-Events: presence, kpml
Content-Length: 0

<------------->
[Sep 22 15:13:18] --- (14 headers 0 lines) ---
[Sep 22 15:13:18] Really destroying SIP dialog '606725ee17e9d16e3c41e89e112de1ce                                                                                        @192.168.1.6' Method: OPTIONS
[Sep 22 15:13:20] Reliably Transmitting (NAT) to 192.168.1.38:37952:
OPTIONS sip:8018@114.143.71.27:28974;rinstance=4669cc6ef2ec4707;transport=UDP;cp                                                                                        d=on SIP/2.0
Via: SIP/2.0/UDP 192.168.1.6:5060;branch=z9hG4bK7080e96c;rport
From: "asterisk" <sip:asterisk@192.168.1.6>;tag=as227127b7
To: <sip:8018@114.143.71.27:28974;rinstance=4669cc6ef2ec4707;transport=UDP;cpd=o                                                                                        n>
Contact: <sip:asterisk@192.168.1.6>
Call-ID: 7a91ecae3ff5152b07549d2b436c8a7c@192.168.1.6
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Thu, 22 Sep 2016 09:43:20 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0

---
[Sep 22 15:13:20]
<--- SIP read from 192.168.1.38:37952 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.6:5060;branch=z9hG4bK7080e96c;rport=5060
Contact: <sip:192.168.1.38:37952>
To: <sip:8018@114.143.71.27:28974;rinstance=4669cc6ef2ec4707;transport=UDP;cpd=o                                                                                        n>;tag=b80d7160
From: "asterisk" <sip:asterisk@192.168.1.6>;tag=as227127b7
Call-ID: 7a91ecae3ff5152b07549d2b436c8a7c@192.168.1.6
CSeq: 102 OPTIONS
Accept: application/sdp, application/sdp
Accept-Language: en
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIB                                                                                        E
Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-                                                                                        serviceuri
User-Agent: Z 3.7.30891 r30851
Allow-Events: presence, kpml
Content-Length: 0

<------------->
[Sep 22 15:13:20] --- (14 headers 0 lines) ---
[Sep 22 15:13:20] Really destroying SIP dialog '7a91ecae3ff5152b07549d2b436c8a7c                                                                                        @192.168.1.6' Method: OPTIONS
[Sep 22 15:13:20]   == Refreshing DNS lookups.
[Sep 22 15:13:21] Reliably Transmitting (NAT) to 192.168.1.31:38647:
OPTIONS sip:8011@114.143.71.27:28994;rinstance=77d5d05a4a14bb81;transport=UDP;cp                                                                                        d=on SIP/2.0
Via: SIP/2.0/UDP 192.168.1.6:5060;branch=z9hG4bK53473677;rport
From: "asterisk" <sip:asterisk@192.168.1.6>;tag=as0228e257
To: <sip:8011@114.143.71.27:28994;rinstance=77d5d05a4a14bb81;transport=UDP;cpd=o                                                                                        n>
Contact: <sip:asterisk@192.168.1.6>
Call-ID: 748f2dc76e16711e4da0eb72313ed0d8@192.168.1.6
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Thu, 22 Sep 2016 09:43:21 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0

---
[Sep 22 15:13:21]
<--- SIP read from 192.168.1.31:38647 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.6:5060;branch=z9hG4bK53473677;rport=5060
Contact: <sip:192.168.1.31:38647>
To: <sip:8011@114.143.71.27:28994;rinstance=77d5d05a4a14bb81;transport=UDP;cpd=o                                                                                        n>;tag=23005218
From: "asterisk"<sip:asterisk@192.168.1.6>;tag=as0228e257
Call-ID: 748f2dc76e16711e4da0eb72313ed0d8@192.168.1.6
CSeq: 102 OPTIONS
Accept: application/sdp, application/sdp
Accept-Language: en
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIB                                                                                        E
Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri
User-Agent: Z 3.6.25251 r25476
Allow-Events: presence, kpml
Content-Length: 0

<------------->
[Sep 22 15:13:21] --- (14 headers 0 lines) ---
[Sep 22 15:13:21] Really destroying SIP dialog '748f2dc76e16711e4da0eb72313ed0d8                                                                                        @192.168.1.6' Method: OPTIONS
[Sep 22 15:13:22]
<--- SIP read from 192.168.1.40:38523 --->
PUBLISH sip:8020@192.168.1.6;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 114.143.71.27:28789;branch=z9hG4bK-d8754z-eb5bcd971e1a1b00-1---                                                                                        d8754z-
Max-Forwards: 70
Contact: <sip:8020@114.143.71.27:28789;transport=UDP>
To: <sip:8020@192.168.1.6;transport=UDP>
From: <sip:8020@192.168.1.6;transport=UDP>;tag=7f5b2004
Call-ID: NjlkNTQwNjM5YWFkNDYxYjVjNDI3MTAyMDRkMDU1YmM.
CSeq: 1 PUBLISH
Expires: 600
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIB                                                                                        E
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: 265

<?xml version="1.0" encoding="UTF-8"?>
<presence xmlns="urn:ietf:params:xml:ns:pidf"
          entity="sip:8020@192.168.1.6;transport=UDP">
  <tuple id="8020" >
     <status><basic>open</basic></status>
     <note>On the phone</note>
  </tuple>
</presence>

<------------->
[Sep 22 15:13:22] --- (16 headers 8 lines) ---
[Sep 22 15:13:22] Sending to 192.168.1.40 : 38523 (NAT)
[Sep 22 15:13:22]
<--- Transmitting (NAT) to 192.168.1.40:38523 --->
SIP/2.0 501 Method Not Implemented
Via: SIP/2.0/UDP 114.143.71.27:28789;branch=z9hG4bK-d8754z-eb5bcd971e1a1b00-1---                                                                                        d8754z-;received=192.168.1.40
From: <sip:8020@192.168.1.6;transport=UDP>;tag=7f5b2004
To: <sip:8020@192.168.1.6;transport=UDP>;tag=as51570442
Call-ID: NjlkNTQwNjM5YWFkNDYxYjVjNDI3MTAyMDRkMDU1YmM.
CSeq: 1 PUBLISH
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0

<------------>
[Sep 22 15:13:22]
<--- SIP read from 192.168.1.40:38523 --->
SUBSCRIBE sip:8020@192.168.1.6;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 114.143.71.27:28789;branch=z9hG4bK-d8754z-8509ad6139ec3599-1---                                                                                        d8754z-
Max-Forwards: 70
Contact: <sip:8020@114.143.71.27:28789;transport=UDP>
To: <sip:8020@192.168.1.6;transport=UDP>
From: <sip:8020@192.168.1.6;transport=UDP>;tag=bc518127
Call-ID: YWQwZTIxOGQxYjljMTMxNzgzNGQ0NjIxMWE5YzQyY2E.
CSeq: 1 SUBSCRIBE
Expires: 600
Accept: application/watcherinfo+xml
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIB                                                                                        E
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

<------------->
[Sep 22 15:13:22] --- (16 headers 0 lines) ---
[Sep 22 15:13:22] Creating new subscription
[Sep 22 15:13:22] Sending to 192.168.1.40 : 38523 (NAT)
[Sep 22 15:13:22] list_route: hop: <sip:8020@114.143.71.27:28789;transport=UDP>
[Sep 22 15:13:22] Found peer '8020'
[Sep 22 15:13:22]
<--- Transmitting (NAT) to 192.168.1.40:38523 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 114.143.71.27:28789;branch=z9hG4bK-d8754z-8509ad6139ec3599-1---                                                                                        d8754z-;received=192.168.1.40
From: <sip:8020@192.168.1.6;transport=UDP>;tag=bc518127
To: <sip:8020@192.168.1.6;transport=UDP>;tag=as1ef24e96
Call-ID: YWQwZTIxOGQxYjljMTMxNzgzNGQ0NjIxMWE5YzQyY2E.
CSeq: 1 SUBSCRIBE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="173f9eba"
Content-Length: 0

<------------>
[Sep 22 15:13:22] Scheduling destruction of SIP dialog 'YWQwZTIxOGQxYjljMTMxNzgz                                                                                        NGQ0NjIxMWE5YzQyY2E.' in 6400 ms (Method: SUBSCRIBE)
[Sep 22 15:13:22]
<--- SIP read from 192.168.1.40:38523 --->
SUBSCRIBE sip:8020@192.168.1.6;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 114.143.71.27:28789;branch=z9hG4bK-d8754z-fcd99ed4431ea108-1---                                                                                        d8754z-
Max-Forwards: 70
Contact: <sip:8020@114.143.71.27:28789;transport=UDP>
To: <sip:8020@192.168.1.6;transport=UDP>
From: <sip:8020@192.168.1.6;transport=UDP>;tag=bc518127
Call-ID: YWQwZTIxOGQxYjljMTMxNzgzNGQ0NjIxMWE5YzQyY2E.
CSeq: 2 SUBSCRIBE
Expires: 600
Accept: application/watcherinfo+xml
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIB                                                                                        E
Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri
User-Agent: Z 3.6.25251 r25476
Authorization: Digest username="8020",realm="asterisk",nonce="173f9eba",uri="sip                                                                                        :8020@192.168.1.6;transport=UDP",response="358b2cce7606698321db2153a9792459",alg                                                                                        orithm=MD5
Event: presence.winfo
Allow-Events: presence, kpml
Content-Length: 0

<------------->
[Sep 22 15:13:22] --- (17 headers 0 lines) ---
[Sep 22 15:13:22] Creating new subscription
[Sep 22 15:13:22] Sending to 192.168.1.40 : 38523 (NAT)
[Sep 22 15:13:22] Found peer '8020'
[Sep 22 15:13:22] Looking for 8020 in default (domain 192.168.1.6)
[Sep 22 15:13:22]
<--- Transmitting (NAT) to 192.168.1.40:38523 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 114.143.71.27:28789;branch=z9hG4bK-d8754z-fcd99ed4431ea108-1---                                                                                        d8754z-;received=192.168.1.40
From: <sip:8020@192.168.1.6;transport=UDP>;tag=bc518127
To: <sip:8020@192.168.1.6;transport=UDP>;tag=as1ef24e96
Call-ID: YWQwZTIxOGQxYjljMTMxNzgzNGQ0NjIxMWE5YzQyY2E.
CSeq: 2 SUBSCRIBE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0

<------------>
[Sep 22 15:13:22] Really destroying SIP dialog 'YWQwZTIxOGQxYjljMTMxNzgzNGQ0NjIx                                                                                        MWE5YzQyY2E.' Method: SUBSCRIBE
[Sep 22 15:13:24] Reliably Transmitting (NAT) to 192.168.1.23:5060:
OPTIONS sip:8003@114.143.71.27:28146;rinstance=e76d8c66848156ae;transport=UDP;cp                                                                                        d=on SIP/2.0
Via: SIP/2.0/UDP 192.168.1.6:5060;branch=z9hG4bK068c4963;rport
From: "asterisk" <sip:asterisk@192.168.1.6>;tag=as20b649dd
To: <sip:8003@114.143.71.27:28146;rinstance=e76d8c66848156ae;transport=UDP;cpd=o                                                                                        n>
Contact: <sip:asterisk@192.168.1.6>
Call-ID: 715838e641647e7c6568b77958120cba@192.168.1.6
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Thu, 22 Sep 2016 09:43:24 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0

<------------->
[Sep 22 15:13:24] --- (14 headers 0 lines) ---
[Sep 22 15:13:24] Really destroying SIP dialog '715838e641647e7c6568b77958120cba                                                                                        @192.168.1.6' Method: OPTIONS
[Sep 22 15:13:28]   == Parsing '/etc/asterisk/manager.conf': [Sep 22 15:13:28] F                                                                                        ound
[Sep 22 15:13:28]   == Manager 'sendcron' logged on from 127.0.0.1
[Sep 22 15:13:28]     -- Executing [8600058@default:1] MeetMe("Local/8600058@def                                                                                        ault-0694,2", "8600058|F") in new stack
[Sep 22 15:13:28]        > Channel Local/8600058@default-0694,1 was answered.
[Sep 22 15:13:28]     -- Executing [900441805625668@default:1] AGI("Local/860005                                                                                        8@default-0694,1", "agi://127.0.0.1:4577/call_log") in new stack
[Sep 22 15:13:28]     -- AGI Script agi://127.0.0.1:4577/call_log completed, ret                                                                                        urning 0
[Sep 22 15:13:28]     -- Executing [900441805625668@default:2] Dial("Local/86000                                                                                        58@default-0694,1", "SIP/00441805625668@TataVoip||tTo") in new stack
[Sep 22 15:13:28] Audio is at 192.168.1.6 port 16564
[Sep 22 15:13:28] Adding codec 0x2 (gsm) to SDP
[Sep 22 15:13:28] Adding codec 0x4 (ulaw) to SDP
[Sep 22 15:13:28] Adding codec 0x8 (alaw) to SDP
[Sep 22 15:13:28] Adding non-codec 0x1 (telephone-event) to SDP
[Sep 22 15:13:28] Reliably Transmitting (no NAT) to 202.54.112.194:5060:
INVITE sip:00441805625668@202.54.112.194;cpd=on SIP/2.0
Via: SIP/2.0/UDP 192.168.1.6:5060;branch=z9hG4bK7fb6e208;rport
From: "M9221513280000183167" <sip:442036032149@192.168.1.6>;tag=as1a2e4278
To: <sip:00441805625668@202.54.112.194;cpd=on>
Contact: <sip:442036032149@192.168.1.6>
Call-ID: 0b8df3d76042f8d76aa95c6f51edf951@192.168.1.6
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "M9221513280000183167" <sip:442036032149@192.168.1.6>;privacy=o                                                                                        ff;screen=no
Date: Thu, 22 Sep 2016 09:43:28 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Type: application/sdp
Content-Length: 256

v=0
o=root 2938 2938 IN IP4 192.168.1.6
s=session
c=IN IP4 192.168.1.6
t=0 0
m=audio 16564 RTP/AVP 3 0 8 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
[Sep 22 15:13:28]     -- Called 00441805625668@TataVoip
[Sep 22 15:13:28]
<--- SIP read from 202.54.112.194:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.6:5060;rport=5060;received=123.252.131.2;branch=z9hG4                                                                                        bK7fb6e208
From: "M9221513280000183167" <sip:442036032149@192.168.1.6:5060>;tag=as1a2e4278
To: <sip:00441805625668@202.54.112.194:5060;cpd=on>
Call-ID: 0b8df3d76042f8d76aa95c6f51edf951@192.168.1.6
CSeq: 102 INVITE
Content-Length: 0

Re: Receiving Unknown calls on extension

PostPosted: Thu Sep 22, 2016 9:42 pm
by ambiorixg12
The first sip trace it is a normal outbound call From: "M9221458130000082364" <sip:442036032149@192.168.1.6>;tag=as49eedae To: <sip:00441243771354@202.54.112.194;cpd=on>

Re: Receiving Unknown calls on extension

PostPosted: Fri Sep 23, 2016 3:42 am
by andrewmarba
Yes but my concern is that random agents get inbound from then extension that doesnot exist in the dialer like 9000 or 1000. This is happening today also.

Re: Receiving Unknown calls on extension

PostPosted: Fri Sep 23, 2016 10:14 am
by ambiorixg12
First If these calls are detected on the Asterisk CLI or log file, if not the calls are directly sent to the client IP, so changing the standard SIP port would help to eliminate these calls,.

Re: Receiving Unknown calls on extension

PostPosted: Mon Sep 26, 2016 4:04 am
by andrewmarba
ambiorixg12 wrote:First If these calls are detected on the Asterisk CLI or log file, if not the calls are directly sent to the client IP, so changing the standard SIP port would help to eliminate these calls,.


And how would i do that???
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