vicidial.org

by **andrewmarba** » Mon Sep 19, 2016 9:43 am

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.......

by **ambiorixg12** » Mon Sep 19, 2016 10:16 am

Start by disallowing guest calls on your sip.conf file

[general]

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

by **Vince-0** » Mon Sep 19, 2016 2:42 pm

Show us the Asterisk console logs for this call event.

by **andrewmarba** » Tue Sep 20, 2016 3:48 am

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

by **andrewmarba** » Thu Sep 22, 2016 4:35 am

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

by **andrewmarba** » Thu Sep 22, 2016 5:08 am

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

by **ambiorixg12** » Thu Sep 22, 2016 9:42 pm

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>

by **andrewmarba** » Fri Sep 23, 2016 3:42 am

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.

by **ambiorixg12** » Fri Sep 23, 2016 10:14 am

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,.

by **andrewmarba** » Mon Sep 26, 2016 4:04 am

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???

vicidial.org

Receiving Unknown calls on extension

Receiving Unknown calls on extension

Re: Receiving Unknown calls on extension

Re: Receiving Unknown calls on extension

Re: Receiving Unknown calls on extension

Re: Receiving Unknown calls on extension

Re: Receiving Unknown calls on extension

Re: Receiving Unknown calls on extension

Re: Receiving Unknown calls on extension

Re: Receiving Unknown calls on extension

Re: Receiving Unknown calls on extension

Who is online