<--- Received SIP request (1397 bytes) from UDP:212.210.2.2:5060 --->
INVITE sip:963760960355@media.syploop.net;user=phone SIP/2.0
Content-Length:316
From:"+37167366321" <sip:+37167366321@SIPMSS1-SIP.MGWS.LV;user=phone>;tag=icg2DbXh33XW.Z05
To:<sip:963760960355@media.syploop.net;user=phone>
Via:SIP/2.0/UDP SIPMSS1-SIP.MGWS.LV:5060;branch=z9hG4bKBhaVd3DgBXi_afY6;yop=00.00.AD31DA87.0000.700B
Call-ID:4B60557E14C869581FA4C517@0b70ffffffff
CSeq:1 INVITE
Max-Forwards:70
Record-Route:<sip:AAQACCzlwAAAnAAAAMgAAfGkK@212.210.2.2:5060;yop=00.00.AD31DA87.0000.700B;transport=UDP;lr>
Request-Disposition:no-fork
P-Asserted-Identity:<sip:+37167366321@SIPMSS1-SIP.MGWS.LV;user=phone>,<tel:+37167366321>
Session-Expires:1800;refresher=uac
Contact:<sip:212.210.2.2:5060;transport=UDP;yop=00.00.AD31DA87.0000.700B>
Supported:100rel,timer,norefersub
Allow:ACK,BYE,CANCEL,INFO,INVITE,NOTIFY,OPTIONS,PRACK,REFER,UPDATE
P-Charging-Vector:icid-value=c2BImQLw~~~~~~~~~~~~~wSR6uo2;icid-generated-at=212.210.2.2;orig-ioi=SIPMSS1-SIP.MGWS.LV
P-Early-Media:supported
Content-Type:application/sdp
Content-Disposition:session;handling=required

v=0
o=- 0 0 IN IP4 10.20.193.213
s=-
c=IN IP4 213.102.44.244
t=0 0
m=audio 45030 RTP/AVP 96 97 3 8 98
b=AS:80
a=rtpmap:96 AMR/8000
a=fmtp:96 mode-change-capability=2; max-red=0
a=rtpmap:97 GSM-EFR/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:98 telephone-event/8000
a=ptime:20
a=maxptime:20

<--- Transmitting SIP response (533 bytes) to UDP:212.210.2.2:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP SIPMSS1-SIP.MGWS.LV:5060;rport=5060;received=212.210.2.2;branch=z9hG4bKBhaVd3DgBXi_afY6;yop=00.00.AD31DA87.0000.700B
Record-Route: <sip:AAQACCzlwAAAnAAAAMgAAfGkK@212.210.2.2:5060;transport=UDP;lr;yop=00.00.AD31DA87.0000.700B>
Call-ID: 4B60557E14C869581FA4C517@0b70ffffffff
From: "+37167366321" <sip:+37167366321@SIPMSS1-SIP.MGWS.LV;user=phone>;tag=icg2DbXh33XW.Z05
To: <sip:963760960355@media.syploop.net;user=phone>
CSeq: 1 INVITE
Server: Asterisk PBX 13.11.2
Content-Length:  0


    -- Executing [963760960355@t2-1:1] Dial("PJSIP/smex-1-00000000", "PJSIP/963860960355@smex-1") in new stack
    -- Called PJSIP/963860960355@smex-1
<--- Transmitting SIP request (1032 bytes) to UDP:212.210.2.2:5060 --->
INVITE sip:963860960355@212.210.2.2:5060 SIP/2.0
Via: SIP/2.0/UDP 10.40.6.100:5060;rport;branch=z9hG4bKPj8aacefaa-2d50-4a7f-9020-e4121c9581dc
From: "+37167366321" <sip:+37167366321@10.40.6.100>;tag=b72e397b-f7ba-4876-9104-996000eac148
To: <sip:963860960355@212.210.2.2>
Contact: <sip:asterisk@10.40.6.100:5060>
Call-ID: a9d8d018-becd-4665-86b1-47b9c6018380
CSeq: 8682 INVITE
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
Require: 100rel
Session-Expires: 1800
Min-SE: 90
P-Asserted-Identity: "+37167366321" <sip:+37167366321@10.40.6.100>
Max-Forwards: 70
User-Agent: Asterisk PBX 13.11.2
Content-Type: application/sdp
Content-Length:   259

v=0
o=- 1724300690 1724300690 IN IP4 10.40.6.100
s=Asterisk
c=IN IP4 10.40.6.100
t=0 0
m=audio 12232 RTP/AVP 0 8 101
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=maxptime:150
a=sendrecv

<--- Received SIP response (309 bytes) from UDP:212.210.2.2:5060 --->
SIP/2.0 100 Trying
From:"+37167366321" <sip:+37167366321@10.40.6.100>;tag=b72e397b-f7ba-4876-9104-996000eac148
To:<sip:963860960355@212.210.2.2>
Via:SIP/2.0/UDP 10.40.6.100:5060;rport;branch=z9hG4bKPj8aacefaa-2d50-4a7f-9020-e4121c9581dc
Call-ID:a9d8d018-becd-4665-86b1-47b9c6018380
CSeq:8682 INVITE


<--- Received SIP response (1106 bytes) from UDP:212.210.2.2:5060 --->
SIP/2.0 183 Session Progress
Content-Length:194
From:"+37167366321" <sip:+37167366321@10.40.6.100>;tag=b72e397b-f7ba-4876-9104-996000eac148
To:<sip:963860960355@212.210.2.2>;tag=AjBV5D41ZXYc8h65
Via:SIP/2.0/UDP 10.40.6.100:5060;rport;branch=z9hG4bKPj8aacefaa-2d50-4a7f-9020-e4121c9581dc
Call-ID:a9d8d018-becd-4665-86b1-47b9c6018380
CSeq:8682 INVITE
Record-Route:<sip:AAQACEYpwAAAaAAAA6wAAfDwK@212.210.2.2:5060;yop=00.00.AD31DA87.0000.7011;transport=UDP;lr>
RSeq:1
Contact:<sip:212.210.2.2:5060;transport=UDP;yop=00.00.AD31DA87.0000.7011>
Require:100rel
Supported:100rel,precondition,timer,norefersub
Allow:ACK,BYE,CANCEL,INFO,INVITE,NOTIFY,OPTIONS,PRACK,REFER,UPDATE
P-Charging-Vector:icid-value=c2BImQIv2LpUe5IRcMwF7gk-;icid-generated-at=212.210.2.2;orig-ioi=SIPMSS1-SIP.MGWS.LV;term-ioi=SIPMSS1-SIP.MGWS.LV
Content-Type:application/sdp
Content-Disposition:session;handling=required

v=0
o=- 0 0 IN IP4 10.20.193.196
s=-
c=IN IP4 213.102.44.245
t=0 0
m=audio 62468 RTP/AVP 8 101
b=AS:80
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=ptime:20
a=maxptime:20

<--- Transmitting SIP request (593 bytes) to UDP:212.210.2.2:5060 --->
PRACK sip:212.210.2.2:5060;transport=UDP;yop=00.00.AD31DA87.0000.7011 SIP/2.0
Via: SIP/2.0/UDP 10.40.6.100:5060;rport;branch=z9hG4bKPj0ff08983-bacd-4de8-8d5d-2f6a72a28fb1
From: "+37167366321" <sip:+37167366321@10.40.6.100>;tag=b72e397b-f7ba-4876-9104-996000eac148
To: <sip:963860960355@212.210.2.2>;tag=AjBV5D41ZXYc8h65
Call-ID: a9d8d018-becd-4665-86b1-47b9c6018380
CSeq: 8683 PRACK
Route: <sip:AAQACEYpwAAAaAAAA6wAAfDwK@212.210.2.2:5060;transport=UDP;lr;yop=00.00.AD31DA87.0000.7011>
RAck: 1 8682 INVITE
Max-Forwards: 70
User-Agent: Asterisk PBX 13.11.2
Content-Length:  0


    -- PJSIP/smex-1-00000001 is making progress passing it to PJSIP/smex-1-00000000
<--- Transmitting SIP response (1105 bytes) to UDP:212.210.2.2:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP SIPMSS1-SIP.MGWS.LV:5060;rport=5060;received=212.210.2.2;branch=z9hG4bKBhaVd3DgBXi_afY6;yop=00.00.AD31DA87.0000.700B
Record-Route: <sip:AAQACCzlwAAAnAAAAMgAAfGkK@212.210.2.2:5060;transport=UDP;lr;yop=00.00.AD31DA87.0000.700B>
Call-ID: 4B60557E14C869581FA4C517@0b70ffffffff
From: "+37167366321" <sip:+37167366321@SIPMSS1-SIP.MGWS.LV;user=phone>;tag=icg2DbXh33XW.Z05
To: <sip:963760960355@media.syploop.net;user=phone>;tag=8f8a1573-d8e5-4f58-8adb-5475716cadba
CSeq: 1 INVITE
Server: Asterisk PBX 13.11.2
Contact: <sip:10.40.6.100:5060>
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, MESSAGE, REFER
P-Asserted-Identity: <sip:963760960355@media.syploop.net;user=phone>
Require: 100rel
RSeq: 5511
Server: Asterisk PBX 13.11.2
Content-Type: application/sdp
Content-Length:   214

v=0
o=- 0 2 IN IP4 10.40.6.100
s=Asterisk
c=IN IP4 10.40.6.100
t=0 0
m=audio 15072 RTP/AVP 8 98
a=rtpmap:8 PCMA/8000
a=rtpmap:98 telephone-event/8000
a=fmtp:98 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<--- Received SIP response (459 bytes) from UDP:212.210.2.2:5060 --->
SIP/2.0 200 OK
Content-Length:0
From:"+37167366321" <sip:+37167366321@10.40.6.100>;tag=b72e397b-f7ba-4876-9104-996000eac148
To:<sip:963860960355@212.210.2.2>;tag=AjBV5D41ZXYc8h65
Via:SIP/2.0/UDP 10.40.6.100:5060;rport;branch=z9hG4bKPj0ff08983-bacd-4de8-8d5d-2f6a72a28fb1
Call-ID:a9d8d018-becd-4665-86b1-47b9c6018380
CSeq:8683 PRACK
Supported:100rel,precondition,timer,norefersub
Allow:ACK,BYE,CANCEL,INFO,INVITE,NOTIFY,OPTIONS,PRACK,REFER,UPDATE


       > 0x7f3ff400f690 -- Probation passed - setting RTP source address to 213.102.44.244:45030
<--- Received SIP request (489 bytes) from UDP:212.210.2.2:5060 --->
PRACK sip:10.40.6.100:5060 SIP/2.0
Content-Length:0
From:"+37167366321" <sip:+37167366321@SIPMSS1-SIP.MGWS.LV;user=phone>;tag=icg2DbXh33XW.Z05
To:<sip:963760960355@media.syploop.net;user=phone>;tag=8f8a1573-d8e5-4f58-8adb-5475716cadba
Via:SIP/2.0/UDP SIPMSS1-SIP.MGWS.LV:5060;branch=z9hG4bKd8dY736bYfjBi2Y6;yop=00.00.AD31DA87.0000.700B
Call-ID:4B60557E14C869581FA4C517@0b70ffffffff
CSeq:2 PRACK
Max-Forwards:70
Request-Disposition:no-fork
RAck:5511 1 INVITE
Supported:timer


<--- Transmitting SIP response (457 bytes) to UDP:212.210.2.2:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP SIPMSS1-SIP.MGWS.LV:5060;rport=5060;received=212.210.2.2;branch=z9hG4bKd8dY736bYfjBi2Y6;yop=00.00.AD31DA87.0000.700B
Call-ID: 4B60557E14C869581FA4C517@0b70ffffffff
From: "+37167366321" <sip:+37167366321@SIPMSS1-SIP.MGWS.LV;user=phone>;tag=icg2DbXh33XW.Z05
To: <sip:963760960355@media.syploop.net;user=phone>;tag=8f8a1573-d8e5-4f58-8adb-5475716cadba
CSeq: 2 PRACK
Server: Asterisk PBX 13.11.2
Content-Length:  0


<--- Received SIP response (824 bytes) from UDP:212.210.2.2:5060 --->
SIP/2.0 180 Ringing
Content-Length:0
From:"+37167366321" <sip:+37167366321@10.40.6.100>;tag=b72e397b-f7ba-4876-9104-996000eac148
To:<sip:963860960355@212.210.2.2>;tag=AjBV5D41ZXYc8h65
Via:SIP/2.0/UDP 10.40.6.100:5060;rport;branch=z9hG4bKPj8aacefaa-2d50-4a7f-9020-e4121c9581dc
Call-ID:a9d8d018-becd-4665-86b1-47b9c6018380
CSeq:8682 INVITE
Record-Route:<sip:AAQACEYpwAAAaAAAA6wAAfDwK@212.210.2.2:5060;yop=00.00.AD31DA87.0000.7011;transport=UDP;lr>
RSeq:2
Contact:<sip:212.210.2.2:5060;transport=UDP;yop=00.00.AD31DA87.0000.7011>
Require:100rel
Supported:100rel,precondition,timer,norefersub
Allow:ACK,BYE,CANCEL,INFO,INVITE,NOTIFY,OPTIONS,PRACK,REFER,UPDATE
P-Charging-Vector:icid-value=c2BImQIv2LpUe5IRcMwF7gk-;icid-generated-at=212.210.2.2;orig-ioi=SIPMSS1-SIP.MGWS.LV;term-ioi=SIPMSS1-SIP.MGWS.LV


<--- Transmitting SIP request (593 bytes) to UDP:212.210.2.2:5060 --->
PRACK sip:212.210.2.2:5060;transport=UDP;yop=00.00.AD31DA87.0000.7011 SIP/2.0
Via: SIP/2.0/UDP 10.40.6.100:5060;rport;branch=z9hG4bKPj6c72ffaa-9af1-40bf-99df-fe6535803e4c
From: "+37167366321" <sip:+37167366321@10.40.6.100>;tag=b72e397b-f7ba-4876-9104-996000eac148
To: <sip:963860960355@212.210.2.2>;tag=AjBV5D41ZXYc8h65
Call-ID: a9d8d018-becd-4665-86b1-47b9c6018380
CSeq: 8684 PRACK
Route: <sip:AAQACEYpwAAAaAAAA6wAAfDwK@212.210.2.2:5060;transport=UDP;lr;yop=00.00.AD31DA87.0000.7011>
RAck: 2 8682 INVITE
Max-Forwards: 70
User-Agent: Asterisk PBX 13.11.2
Content-Length:  0


    -- PJSIP/smex-1-00000001 is ringing
<--- Transmitting SIP response (848 bytes) to UDP:212.210.2.2:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP SIPMSS1-SIP.MGWS.LV:5060;rport=5060;received=212.210.2.2;branch=z9hG4bKBhaVd3DgBXi_afY6;yop=00.00.AD31DA87.0000.700B
Record-Route: <sip:AAQACCzlwAAAnAAAAMgAAfGkK@212.210.2.2:5060;transport=UDP;lr;yop=00.00.AD31DA87.0000.700B>
Call-ID: 4B60557E14C869581FA4C517@0b70ffffffff
From: "+37167366321" <sip:+37167366321@SIPMSS1-SIP.MGWS.LV;user=phone>;tag=icg2DbXh33XW.Z05
To: <sip:963760960355@media.syploop.net;user=phone>;tag=8f8a1573-d8e5-4f58-8adb-5475716cadba
CSeq: 1 INVITE
Server: Asterisk PBX 13.11.2
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, MESSAGE, REFER
Contact: <sip:10.40.6.100:5060>
P-Asserted-Identity: <sip:963760960355@media.syploop.net;user=phone>
Require: 100rel
RSeq: 5512
Server: Asterisk PBX 13.11.2
Content-Length:  0


<--- Received SIP response (459 bytes) from UDP:212.210.2.2:5060 --->
SIP/2.0 200 OK
Content-Length:0
From:"+37167366321" <sip:+37167366321@10.40.6.100>;tag=b72e397b-f7ba-4876-9104-996000eac148
To:<sip:963860960355@212.210.2.2>;tag=AjBV5D41ZXYc8h65
Via:SIP/2.0/UDP 10.40.6.100:5060;rport;branch=z9hG4bKPj6c72ffaa-9af1-40bf-99df-fe6535803e4c
Call-ID:a9d8d018-becd-4665-86b1-47b9c6018380
CSeq:8684 PRACK
Supported:100rel,precondition,timer,norefersub
Allow:ACK,BYE,CANCEL,INFO,INVITE,NOTIFY,OPTIONS,PRACK,REFER,UPDATE


<--- Received SIP request (489 bytes) from UDP:212.210.2.2:5060 --->
PRACK sip:10.40.6.100:5060 SIP/2.0
Content-Length:0
From:"+37167366321" <sip:+37167366321@SIPMSS1-SIP.MGWS.LV;user=phone>;tag=icg2DbXh33XW.Z05
To:<sip:963760960355@media.syploop.net;user=phone>;tag=8f8a1573-d8e5-4f58-8adb-5475716cadba
Via:SIP/2.0/UDP SIPMSS1-SIP.MGWS.LV:5060;branch=z9hG4bKjCW4d4c.A1j8bB26;yop=00.00.AD31DA87.0000.700B
Call-ID:4B60557E14C869581FA4C517@0b70ffffffff
CSeq:3 PRACK
Max-Forwards:70
Request-Disposition:no-fork
RAck:5512 1 INVITE
Supported:timer


<--- Transmitting SIP response (457 bytes) to UDP:212.210.2.2:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP SIPMSS1-SIP.MGWS.LV:5060;rport=5060;received=212.210.2.2;branch=z9hG4bKjCW4d4c.A1j8bB26;yop=00.00.AD31DA87.0000.700B
Call-ID: 4B60557E14C869581FA4C517@0b70ffffffff
From: "+37167366321" <sip:+37167366321@SIPMSS1-SIP.MGWS.LV;user=phone>;tag=icg2DbXh33XW.Z05
To: <sip:963760960355@media.syploop.net;user=phone>;tag=8f8a1573-d8e5-4f58-8adb-5475716cadba
CSeq: 3 PRACK
Server: Asterisk PBX 13.11.2
Content-Length:  0


       > 0x7f3ff401c1a0 -- Probation passed - setting RTP source address to 213.102.44.245:62468


<--- Received SIP request (499 bytes) from UDP:212.210.2.2:5060 --->
CANCEL sip:963760960355@media.syploop.net;user=phone SIP/2.0
Content-Length:0
From:"+37167366321" <sip:+37167366321@SIPMSS1-SIP.MGWS.LV;user=phone>;tag=icg2DbXh33XW.Z05
To:<sip:963760960355@media.syploop.net;user=phone>
Via:SIP/2.0/UDP SIPMSS1-SIP.MGWS.LV:5060;branch=z9hG4bKBhaVd3DgBXi_afY6;yop=00.00.AD31DA87.0000.700B
Call-ID:4B60557E14C869581FA4C517@0b70ffffffff
CSeq:1 CANCEL
Max-Forwards:70
Request-Disposition:no-fork
Reason:X.int ;reasoncode=0x0000030B;add-info=0109.0001.0004


<--- Transmitting SIP response (463 bytes) to UDP:212.210.2.2:5060 --->
SIP/2.0 481 Call/Transaction Does Not Exist
Via: SIP/2.0/UDP SIPMSS1-SIP.MGWS.LV:5060;received=212.210.2.2;branch=z9hG4bKBhaVd3DgBXi_afY6;yop=00.00.AD31DA87.0000.700B
Call-ID: 4B60557E14C869581FA4C517@0b70ffffffff
From: "+37167366321" <sip:+37167366321@SIPMSS1-SIP.MGWS.LV;user=phone>;tag=icg2DbXh33XW.Z05
To: <sip:963760960355@media.syploop.net;user=phone>;tag=z9hG4bKBhaVd3DgBXi_afY6
CSeq: 1 CANCEL
Server: Asterisk PBX 13.11.2
Content-Length:  0


SMEX-MEDIA-A1*CLI>
SMEX-MEDIA-A1*CLI>
SMEX-MEDIA-A1*CLI>
SMEX-MEDIA-A1*CLI>
SMEX-MEDIA-A1*CLI>
SMEX-MEDIA-A1*CLI>
SMEX-MEDIA-A1*CLI>
SMEX-MEDIA-A1*CLI>
SMEX-MEDIA-A1*CLI>
SMEX-MEDIA-A1*CLI>
SMEX-MEDIA-A1*CLI>


<--- Received SIP response (412 bytes) from UDP:212.210.2.2:5060 --->
SIP/2.0 486 Busy Here
Content-Length:0
From:"+37167366321" <sip:+37167366321@10.40.6.100>;tag=b72e397b-f7ba-4876-9104-996000eac148
To:<sip:963860960355@212.210.2.2>;tag=AjBV5D41ZXYc8h65
Via:SIP/2.0/UDP 10.40.6.100:5060;rport;branch=z9hG4bKPj8aacefaa-2d50-4a7f-9020-e4121c9581dc
Call-ID:a9d8d018-becd-4665-86b1-47b9c6018380
CSeq:8682 INVITE
Reason:X.int ;reasoncode=0x00000005;add-info=01CA.FFFE.0000


<--- Transmitting SIP request (433 bytes) to UDP:212.210.2.2:5060 --->
ACK sip:963860960355@212.210.2.2:5060 SIP/2.0
Via: SIP/2.0/UDP 10.40.6.100:5060;rport;branch=z9hG4bKPj8aacefaa-2d50-4a7f-9020-e4121c9581dc
From: "+37167366321" <sip:+37167366321@10.40.6.100>;tag=b72e397b-f7ba-4876-9104-996000eac148
To: <sip:963860960355@212.210.2.2>;tag=AjBV5D41ZXYc8h65
Call-ID: a9d8d018-becd-4665-86b1-47b9c6018380
CSeq: 8682 ACK
Max-Forwards: 70
User-Agent: Asterisk PBX 13.11.2
Content-Length:  0


  == Everyone is busy/congested at this time (1:1/0/0)
    -- Auto fallthrough, channel 'PJSIP/smex-1-00000000' status is 'BUSY'
<--- Transmitting SIP response (812 bytes) to UDP:212.210.2.2:5060 --->
SIP/2.0 486 Busy Here
Via: SIP/2.0/UDP SIPMSS1-SIP.MGWS.LV:5060;rport=5060;received=212.210.2.2;branch=z9hG4bKBhaVd3DgBXi_afY6;yop=00.00.AD31DA87.0000.700B
Record-Route: <sip:AAQACCzlwAAAnAAAAMgAAfGkK@212.210.2.2:5060;transport=UDP;lr;yop=00.00.AD31DA87.0000.700B>
Call-ID: 4B60557E14C869581FA4C517@0b70ffffffff
From: "+37167366321" <sip:+37167366321@SIPMSS1-SIP.MGWS.LV;user=phone>;tag=icg2DbXh33XW.Z05
To: <sip:963760960355@media.syploop.net;user=phone>;tag=8f8a1573-d8e5-4f58-8adb-5475716cadba
CSeq: 1 INVITE
Server: Asterisk PBX 13.11.2
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, MESSAGE, REFER
Reason: Q.850;cause=17
P-Asserted-Identity: <sip:963760960355@media.syploop.net;user=phone>
Server: Asterisk PBX 13.11.2
Content-Length:  0


<--- Received SIP request (444 bytes) from UDP:212.210.2.2:5060 --->
ACK sip:963760960355@media.syploop.net;user=phone SIP/2.0
Content-Length:0
From:"+37167366321" <sip:+37167366321@SIPMSS1-SIP.MGWS.LV;user=phone>;tag=icg2DbXh33XW.Z05
To:<sip:963760960355@media.syploop.net;user=phone>;tag=8f8a1573-d8e5-4f58-8adb-5475716cadba
Via:SIP/2.0/UDP SIPMSS1-SIP.MGWS.LV:5060;branch=z9hG4bKBhaVd3DgBXi_afY6;yop=00.00.AD31DA87.0000.700B
Call-ID:4B60557E14C869581FA4C517@0b70ffffffff
CSeq:1 ACK
Max-Forwards:70


<--- Received SIP request (1397 bytes) from UDP:212.210.2.2:5060 --->
INVITE sip:963760960355@media.syploop.net;user=phone SIP/2.0
Content-Length:316
From:"+37167366321" <sip:+37167366321@SIPMSS1-SIP.MGWS.LV;user=phone>;tag=CfY_DgaaV3Vgg985
To:<sip:963760960355@media.syploop.net;user=phone>
Via:SIP/2.0/UDP SIPMSS1-SIP.MGWS.LV:5060;branch=z9hG4bKiCDVdADY7XjY11Y6;yop=00.00.AD31DA87.0000.700B
Call-ID:EE98417E70B9D9581E9C2B17@0b70ffffffff
CSeq:1 INVITE
Max-Forwards:70
Record-Route:<sip:AAQACCzlwAAAnAAAAMgAAfGkK@212.210.2.2:5060;yop=00.00.AD31DA87.0000.700B;transport=UDP;lr>
Request-Disposition:no-fork
P-Asserted-Identity:<sip:+37167366321@SIPMSS1-SIP.MGWS.LV;user=phone>,<tel:+37167366321>
Session-Expires:1800;refresher=uac
Contact:<sip:212.210.2.2:5060;transport=UDP;yop=00.00.AD31DA87.0000.700B>
Supported:100rel,timer,norefersub
Allow:ACK,BYE,CANCEL,INFO,INVITE,NOTIFY,OPTIONS,PRACK,REFER,UPDATE
P-Charging-Vector:icid-value=c2BImQLw~~~~~~~~~~~~~wQ97Oo2;icid-generated-at=212.210.2.2;orig-ioi=SIPMSS1-SIP.MGWS.LV
P-Early-Media:supported
Content-Type:application/sdp
Content-Disposition:session;handling=required

v=0
o=- 0 0 IN IP4 10.20.193.213
s=-
c=IN IP4 213.102.44.244
t=0 0
m=audio 45462 RTP/AVP 96 97 3 8 98
b=AS:80
a=rtpmap:96 AMR/8000
a=fmtp:96 mode-change-capability=2; max-red=0
a=rtpmap:97 GSM-EFR/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:98 telephone-event/8000
a=ptime:20
a=maxptime:20

<--- Transmitting SIP response (533 bytes) to UDP:212.210.2.2:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP SIPMSS1-SIP.MGWS.LV:5060;rport=5060;received=212.210.2.2;branch=z9hG4bKiCDVdADY7XjY11Y6;yop=00.00.AD31DA87.0000.700B
Record-Route: <sip:AAQACCzlwAAAnAAAAMgAAfGkK@212.210.2.2:5060;transport=UDP;lr;yop=00.00.AD31DA87.0000.700B>
Call-ID: EE98417E70B9D9581E9C2B17@0b70ffffffff
From: "+37167366321" <sip:+37167366321@SIPMSS1-SIP.MGWS.LV;user=phone>;tag=CfY_DgaaV3Vgg985
To: <sip:963760960355@media.syploop.net;user=phone>
CSeq: 1 INVITE
Server: Asterisk PBX 13.11.2
Content-Length:  0


    -- Executing [963760960355@t2-1:1] Dial("PJSIP/smex-1-00000002", "PJSIP/963860960355@smex-1") in new stack
    -- Called PJSIP/963860960355@smex-1
<--- Transmitting SIP request (1033 bytes) to UDP:212.210.2.2:5060 --->
INVITE sip:963860960355@212.210.2.2:5060 SIP/2.0
Via: SIP/2.0/UDP 10.40.6.100:5060;rport;branch=z9hG4bKPj1c051983-4623-4574-9ed5-7fd14ac43474
From: "+37167366321" <sip:+37167366321@10.40.6.100>;tag=449a88fe-223e-4a48-a12d-5a590fabf154
To: <sip:963860960355@212.210.2.2>
Contact: <sip:asterisk@10.40.6.100:5060>
Call-ID: 82eb2d4d-30e9-4780-9e1d-52ae11750e38
CSeq: 26624 INVITE
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
Require: 100rel
Session-Expires: 1800
Min-SE: 90
P-Asserted-Identity: "+37167366321" <sip:+37167366321@10.40.6.100>
Max-Forwards: 70
User-Agent: Asterisk PBX 13.11.2
Content-Type: application/sdp
Content-Length:   259

v=0
o=- 1969172756 1969172756 IN IP4 10.40.6.100
s=Asterisk
c=IN IP4 10.40.6.100
t=0 0
m=audio 12930 RTP/AVP 0 8 101
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=maxptime:150
a=sendrecv

<--- Received SIP response (310 bytes) from UDP:212.210.2.2:5060 --->
SIP/2.0 100 Trying
From:"+37167366321" <sip:+37167366321@10.40.6.100>;tag=449a88fe-223e-4a48-a12d-5a590fabf154
To:<sip:963860960355@212.210.2.2>
Via:SIP/2.0/UDP 10.40.6.100:5060;rport;branch=z9hG4bKPj1c051983-4623-4574-9ed5-7fd14ac43474
Call-ID:82eb2d4d-30e9-4780-9e1d-52ae11750e38
CSeq:26624 INVITE


<--- Received SIP response (1107 bytes) from UDP:212.210.2.2:5060 --->
SIP/2.0 183 Session Progress
Content-Length:194
From:"+37167366321" <sip:+37167366321@10.40.6.100>;tag=449a88fe-223e-4a48-a12d-5a590fabf154
To:<sip:963860960355@212.210.2.2>;tag=g3.B3ZcB3ahY0Ui5
Via:SIP/2.0/UDP 10.40.6.100:5060;rport;branch=z9hG4bKPj1c051983-4623-4574-9ed5-7fd14ac43474
Call-ID:82eb2d4d-30e9-4780-9e1d-52ae11750e38
CSeq:26624 INVITE
Record-Route:<sip:AAQACCklwAADbAAAA7QAAfDYK@212.210.2.2:5060;yop=00.00.AD31DA87.0000.700A;transport=UDP;lr>
RSeq:1
Contact:<sip:212.210.2.2:5060;transport=UDP;yop=00.00.AD31DA87.0000.700A>
Require:100rel
Supported:100rel,precondition,timer,norefersub
Allow:ACK,BYE,CANCEL,INFO,INVITE,NOTIFY,OPTIONS,PRACK,REFER,UPDATE
P-Charging-Vector:icid-value=c2BImQKl7rpUPY4KcMwFXAo-;icid-generated-at=212.210.2.2;orig-ioi=SIPMSS1-SIP.MGWS.LV;term-ioi=SIPMSS1-SIP.MGWS.LV
Content-Type:application/sdp
Content-Disposition:session;handling=required

v=0
o=- 0 0 IN IP4 10.20.193.213
s=-
c=IN IP4 213.102.45.245
t=0 0
m=audio 32336 RTP/AVP 8 101
b=AS:80
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=ptime:20
a=maxptime:20

<--- Transmitting SIP request (595 bytes) to UDP:212.210.2.2:5060 --->
PRACK sip:212.210.2.2:5060;transport=UDP;yop=00.00.AD31DA87.0000.700A SIP/2.0
Via: SIP/2.0/UDP 10.40.6.100:5060;rport;branch=z9hG4bKPj312350ca-8c4a-422e-89a7-acffdfed3ef1
From: "+37167366321" <sip:+37167366321@10.40.6.100>;tag=449a88fe-223e-4a48-a12d-5a590fabf154
To: <sip:963860960355@212.210.2.2>;tag=g3.B3ZcB3ahY0Ui5
Call-ID: 82eb2d4d-30e9-4780-9e1d-52ae11750e38
CSeq: 26625 PRACK
Route: <sip:AAQACCklwAADbAAAA7QAAfDYK@212.210.2.2:5060;transport=UDP;lr;yop=00.00.AD31DA87.0000.700A>
RAck: 1 26624 INVITE
Max-Forwards: 70
User-Agent: Asterisk PBX 13.11.2
Content-Length:  0


    -- PJSIP/smex-1-00000003 is making progress passing it to PJSIP/smex-1-00000002
<--- Transmitting SIP response (1106 bytes) to UDP:212.210.2.2:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP SIPMSS1-SIP.MGWS.LV:5060;rport=5060;received=212.210.2.2;branch=z9hG4bKiCDVdADY7XjY11Y6;yop=00.00.AD31DA87.0000.700B
Record-Route: <sip:AAQACCzlwAAAnAAAAMgAAfGkK@212.210.2.2:5060;transport=UDP;lr;yop=00.00.AD31DA87.0000.700B>
Call-ID: EE98417E70B9D9581E9C2B17@0b70ffffffff
From: "+37167366321" <sip:+37167366321@SIPMSS1-SIP.MGWS.LV;user=phone>;tag=CfY_DgaaV3Vgg985
To: <sip:963760960355@media.syploop.net;user=phone>;tag=3a802c98-75ee-4afb-81bc-e5a8f1b0d676
CSeq: 1 INVITE
Server: Asterisk PBX 13.11.2
Contact: <sip:10.40.6.100:5060>
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, MESSAGE, REFER
P-Asserted-Identity: <sip:963760960355@media.syploop.net;user=phone>
Require: 100rel
RSeq: 27382
Server: Asterisk PBX 13.11.2
Content-Type: application/sdp
Content-Length:   214

v=0
o=- 0 2 IN IP4 10.40.6.100
s=Asterisk
c=IN IP4 10.40.6.100
t=0 0
m=audio 13832 RTP/AVP 8 98
a=rtpmap:8 PCMA/8000
a=rtpmap:98 telephone-event/8000
a=fmtp:98 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<--- Received SIP response (460 bytes) from UDP:212.210.2.2:5060 --->
SIP/2.0 200 OK
Content-Length:0
From:"+37167366321" <sip:+37167366321@10.40.6.100>;tag=449a88fe-223e-4a48-a12d-5a590fabf154
To:<sip:963860960355@212.210.2.2>;tag=g3.B3ZcB3ahY0Ui5
Via:SIP/2.0/UDP 10.40.6.100:5060;rport;branch=z9hG4bKPj312350ca-8c4a-422e-89a7-acffdfed3ef1
Call-ID:82eb2d4d-30e9-4780-9e1d-52ae11750e38
CSeq:26625 PRACK
Supported:100rel,precondition,timer,norefersub
Allow:ACK,BYE,CANCEL,INFO,INVITE,NOTIFY,OPTIONS,PRACK,REFER,UPDATE


       > 0x7f3ff400f560 -- Probation passed - setting RTP source address to 213.102.44.244:45462
<--- Received SIP request (490 bytes) from UDP:212.210.2.2:5060 --->
PRACK sip:10.40.6.100:5060 SIP/2.0
Content-Length:0
From:"+37167366321" <sip:+37167366321@SIPMSS1-SIP.MGWS.LV;user=phone>;tag=CfY_DgaaV3Vgg985
To:<sip:963760960355@media.syploop.net;user=phone>;tag=3a802c98-75ee-4afb-81bc-e5a8f1b0d676
Via:SIP/2.0/UDP SIPMSS1-SIP.MGWS.LV:5060;branch=z9hG4bK8hX279c5_fje8gY6;yop=00.00.AD31DA87.0000.700B
Call-ID:EE98417E70B9D9581E9C2B17@0b70ffffffff
CSeq:2 PRACK
Max-Forwards:70
Request-Disposition:no-fork
RAck:27382 1 INVITE
Supported:timer


<--- Transmitting SIP response (457 bytes) to UDP:212.210.2.2:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP SIPMSS1-SIP.MGWS.LV:5060;rport=5060;received=212.210.2.2;branch=z9hG4bK8hX279c5_fje8gY6;yop=00.00.AD31DA87.0000.700B
Call-ID: EE98417E70B9D9581E9C2B17@0b70ffffffff
From: "+37167366321" <sip:+37167366321@SIPMSS1-SIP.MGWS.LV;user=phone>;tag=CfY_DgaaV3Vgg985
To: <sip:963760960355@media.syploop.net;user=phone>;tag=3a802c98-75ee-4afb-81bc-e5a8f1b0d676
CSeq: 2 PRACK
Server: Asterisk PBX 13.11.2
Content-Length:  0


<--- Received SIP response (825 bytes) from UDP:212.210.2.2:5060 --->
SIP/2.0 180 Ringing
Content-Length:0
From:"+37167366321" <sip:+37167366321@10.40.6.100>;tag=449a88fe-223e-4a48-a12d-5a590fabf154
To:<sip:963860960355@212.210.2.2>;tag=g3.B3ZcB3ahY0Ui5
Via:SIP/2.0/UDP 10.40.6.100:5060;rport;branch=z9hG4bKPj1c051983-4623-4574-9ed5-7fd14ac43474
Call-ID:82eb2d4d-30e9-4780-9e1d-52ae11750e38
CSeq:26624 INVITE
Record-Route:<sip:AAQACCklwAADbAAAA7QAAfDYK@212.210.2.2:5060;yop=00.00.AD31DA87.0000.700A;transport=UDP;lr>
RSeq:2
Contact:<sip:212.210.2.2:5060;transport=UDP;yop=00.00.AD31DA87.0000.700A>
Require:100rel
Supported:100rel,precondition,timer,norefersub
Allow:ACK,BYE,CANCEL,INFO,INVITE,NOTIFY,OPTIONS,PRACK,REFER,UPDATE
P-Charging-Vector:icid-value=c2BImQKl7rpUPY4KcMwFXAo-;icid-generated-at=212.210.2.2;orig-ioi=SIPMSS1-SIP.MGWS.LV;term-ioi=SIPMSS1-SIP.MGWS.LV


<--- Transmitting SIP request (595 bytes) to UDP:212.210.2.2:5060 --->
PRACK sip:212.210.2.2:5060;transport=UDP;yop=00.00.AD31DA87.0000.700A SIP/2.0
Via: SIP/2.0/UDP 10.40.6.100:5060;rport;branch=z9hG4bKPj8c45ec80-624f-456d-9388-af0ec5ed2ef7
From: "+37167366321" <sip:+37167366321@10.40.6.100>;tag=449a88fe-223e-4a48-a12d-5a590fabf154
To: <sip:963860960355@212.210.2.2>;tag=g3.B3ZcB3ahY0Ui5
Call-ID: 82eb2d4d-30e9-4780-9e1d-52ae11750e38
CSeq: 26626 PRACK
Route: <sip:AAQACCklwAADbAAAA7QAAfDYK@212.210.2.2:5060;transport=UDP;lr;yop=00.00.AD31DA87.0000.700A>
RAck: 2 26624 INVITE
Max-Forwards: 70
User-Agent: Asterisk PBX 13.11.2
Content-Length:  0


    -- PJSIP/smex-1-00000003 is ringing
<--- Transmitting SIP response (849 bytes) to UDP:212.210.2.2:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP SIPMSS1-SIP.MGWS.LV:5060;rport=5060;received=212.210.2.2;branch=z9hG4bKiCDVdADY7XjY11Y6;yop=00.00.AD31DA87.0000.700B
Record-Route: <sip:AAQACCzlwAAAnAAAAMgAAfGkK@212.210.2.2:5060;transport=UDP;lr;yop=00.00.AD31DA87.0000.700B>
Call-ID: EE98417E70B9D9581E9C2B17@0b70ffffffff
From: "+37167366321" <sip:+37167366321@SIPMSS1-SIP.MGWS.LV;user=phone>;tag=CfY_DgaaV3Vgg985
To: <sip:963760960355@media.syploop.net;user=phone>;tag=3a802c98-75ee-4afb-81bc-e5a8f1b0d676
CSeq: 1 INVITE
Server: Asterisk PBX 13.11.2
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, MESSAGE, REFER
Contact: <sip:10.40.6.100:5060>
P-Asserted-Identity: <sip:963760960355@media.syploop.net;user=phone>
Require: 100rel
RSeq: 27383
Server: Asterisk PBX 13.11.2
Content-Length:  0


<--- Received SIP response (460 bytes) from UDP:212.210.2.2:5060 --->
SIP/2.0 200 OK
Content-Length:0
From:"+37167366321" <sip:+37167366321@10.40.6.100>;tag=449a88fe-223e-4a48-a12d-5a590fabf154
To:<sip:963860960355@212.210.2.2>;tag=g3.B3ZcB3ahY0Ui5
Via:SIP/2.0/UDP 10.40.6.100:5060;rport;branch=z9hG4bKPj8c45ec80-624f-456d-9388-af0ec5ed2ef7
Call-ID:82eb2d4d-30e9-4780-9e1d-52ae11750e38
CSeq:26626 PRACK
Supported:100rel,precondition,timer,norefersub
Allow:ACK,BYE,CANCEL,INFO,INVITE,NOTIFY,OPTIONS,PRACK,REFER,UPDATE


<--- Received SIP request (490 bytes) from UDP:212.210.2.2:5060 --->
PRACK sip:10.40.6.100:5060 SIP/2.0
Content-Length:0
From:"+37167366321" <sip:+37167366321@SIPMSS1-SIP.MGWS.LV;user=phone>;tag=CfY_DgaaV3Vgg985
To:<sip:963760960355@media.syploop.net;user=phone>;tag=3a802c98-75ee-4afb-81bc-e5a8f1b0d676
Via:SIP/2.0/UDP SIPMSS1-SIP.MGWS.LV:5060;branch=z9hG4bKiaCUd.B23Xgdb626;yop=00.00.AD31DA87.0000.700B
Call-ID:EE98417E70B9D9581E9C2B17@0b70ffffffff
CSeq:3 PRACK
Max-Forwards:70
Request-Disposition:no-fork
RAck:27383 1 INVITE
Supported:timer


<--- Transmitting SIP response (457 bytes) to UDP:212.210.2.2:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP SIPMSS1-SIP.MGWS.LV:5060;rport=5060;received=212.210.2.2;branch=z9hG4bKiaCUd.B23Xgdb626;yop=00.00.AD31DA87.0000.700B
Call-ID: EE98417E70B9D9581E9C2B17@0b70ffffffff
From: "+37167366321" <sip:+37167366321@SIPMSS1-SIP.MGWS.LV;user=phone>;tag=CfY_DgaaV3Vgg985
To: <sip:963760960355@media.syploop.net;user=phone>;tag=3a802c98-75ee-4afb-81bc-e5a8f1b0d676
CSeq: 3 PRACK
Server: Asterisk PBX 13.11.2
Content-Length:  0


       > 0x7f3ff402d010 -- Probation passed - setting RTP source address to 213.102.45.245:32336
<--- Received SIP request (499 bytes) from UDP:212.210.2.2:5060 --->
CANCEL sip:963760960355@media.syploop.net;user=phone SIP/2.0
Content-Length:0
From:"+37167366321" <sip:+37167366321@SIPMSS1-SIP.MGWS.LV;user=phone>;tag=CfY_DgaaV3Vgg985
To:<sip:963760960355@media.syploop.net;user=phone>
Via:SIP/2.0/UDP SIPMSS1-SIP.MGWS.LV:5060;branch=z9hG4bKiCDVdADY7XjY11Y6;yop=00.00.AD31DA87.0000.700B
Call-ID:EE98417E70B9D9581E9C2B17@0b70ffffffff
CSeq:1 CANCEL
Max-Forwards:70
Request-Disposition:no-fork
Reason:X.int ;reasoncode=0x0000030B;add-info=0109.0001.0004


<--- Transmitting SIP response (458 bytes) to UDP:212.210.2.2:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP SIPMSS1-SIP.MGWS.LV:5060;rport=5060;received=212.210.2.2;branch=z9hG4bKiCDVdADY7XjY11Y6;yop=00.00.AD31DA87.0000.700B
Call-ID: EE98417E70B9D9581E9C2B17@0b70ffffffff
From: "+37167366321" <sip:+37167366321@SIPMSS1-SIP.MGWS.LV;user=phone>;tag=CfY_DgaaV3Vgg985
To: <sip:963760960355@media.syploop.net;user=phone>;tag=3a802c98-75ee-4afb-81bc-e5a8f1b0d676
CSeq: 1 CANCEL
Server: Asterisk PBX 13.11.2
Content-Length:  0


<--- Transmitting SIP response (827 bytes) to UDP:212.210.2.2:5060 --->
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP SIPMSS1-SIP.MGWS.LV:5060;rport=5060;received=212.210.2.2;branch=z9hG4bKiCDVdADY7XjY11Y6;yop=00.00.AD31DA87.0000.700B
Record-Route: <sip:AAQACCzlwAAAnAAAAMgAAfGkK@212.210.2.2:5060;transport=UDP;lr;yop=00.00.AD31DA87.0000.700B>
Call-ID: EE98417E70B9D9581E9C2B17@0b70ffffffff
From: "+37167366321" <sip:+37167366321@SIPMSS1-SIP.MGWS.LV;user=phone>;tag=CfY_DgaaV3Vgg985
To: <sip:963760960355@media.syploop.net;user=phone>;tag=3a802c98-75ee-4afb-81bc-e5a8f1b0d676
CSeq: 1 INVITE
Server: Asterisk PBX 13.11.2
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, MESSAGE, REFER
Server: Asterisk PBX 13.11.2
P-Asserted-Identity: <sip:963760960355@media.syploop.net;user=phone>
Server: Asterisk PBX 13.11.2
Content-Length:  0


  == Spawn extension (t2-1, 963760960355, 1) exited non-zero on 'PJSIP/smex-1-00000002'
<--- Transmitting SIP request (442 bytes) to UDP:212.210.2.2:5060 --->
CANCEL sip:963860960355@212.210.2.2:5060 SIP/2.0
Via: SIP/2.0/UDP 10.40.6.100:5060;rport;branch=z9hG4bKPj1c051983-4623-4574-9ed5-7fd14ac43474
From: "+37167366321" <sip:+37167366321@10.40.6.100>;tag=449a88fe-223e-4a48-a12d-5a590fabf154
To: <sip:963860960355@212.210.2.2>
Call-ID: 82eb2d4d-30e9-4780-9e1d-52ae11750e38
CSeq: 26624 CANCEL
Reason: Q.850;cause=0
Max-Forwards: 70
User-Agent: Asterisk PBX 13.11.2
Content-Length:  0


<--- Received SIP request (444 bytes) from UDP:212.210.2.2:5060 --->
ACK sip:963760960355@media.syploop.net;user=phone SIP/2.0
Content-Length:0
From:"+37167366321" <sip:+37167366321@SIPMSS1-SIP.MGWS.LV;user=phone>;tag=CfY_DgaaV3Vgg985
To:<sip:963760960355@media.syploop.net;user=phone>;tag=3a802c98-75ee-4afb-81bc-e5a8f1b0d676
Via:SIP/2.0/UDP SIPMSS1-SIP.MGWS.LV:5060;branch=z9hG4bKiCDVdADY7XjY11Y6;yop=00.00.AD31DA87.0000.700B
Call-ID:EE98417E70B9D9581E9C2B17@0b70ffffffff
CSeq:1 ACK
Max-Forwards:70


<--- Received SIP response (461 bytes) from UDP:212.210.2.2:5060 --->
SIP/2.0 200 OK
Content-Length:0
From:"+37167366321" <sip:+37167366321@10.40.6.100>;tag=449a88fe-223e-4a48-a12d-5a590fabf154
To:<sip:963860960355@212.210.2.2>;tag=g3.B3ZcB3ahY0Ui5
Via:SIP/2.0/UDP 10.40.6.100:5060;rport;branch=z9hG4bKPj1c051983-4623-4574-9ed5-7fd14ac43474
Call-ID:82eb2d4d-30e9-4780-9e1d-52ae11750e38
CSeq:26624 CANCEL
Supported:100rel,precondition,timer,norefersub
Allow:ACK,BYE,CANCEL,INFO,INVITE,NOTIFY,OPTIONS,PRACK,REFER,UPDATE


<--- Received SIP response (421 bytes) from UDP:212.210.2.2:5060 --->
SIP/2.0 487 Request Cancelled
Content-Length:0
From:"+37167366321" <sip:+37167366321@10.40.6.100>;tag=449a88fe-223e-4a48-a12d-5a590fabf154
To:<sip:963860960355@212.210.2.2>;tag=g3.B3ZcB3ahY0Ui5
Via:SIP/2.0/UDP 10.40.6.100:5060;rport;branch=z9hG4bKPj1c051983-4623-4574-9ed5-7fd14ac43474
Call-ID:82eb2d4d-30e9-4780-9e1d-52ae11750e38
CSeq:26624 INVITE
Reason:X.int ;reasoncode=0x0000030B;add-info=05CC.0001.0002


<--- Transmitting SIP request (434 bytes) to UDP:212.210.2.2:5060 --->
ACK sip:963860960355@212.210.2.2:5060 SIP/2.0
Via: SIP/2.0/UDP 10.40.6.100:5060;rport;branch=z9hG4bKPj1c051983-4623-4574-9ed5-7fd14ac43474
From: "+37167366321" <sip:+37167366321@10.40.6.100>;tag=449a88fe-223e-4a48-a12d-5a590fabf154
To: <sip:963860960355@212.210.2.2>;tag=g3.B3ZcB3ahY0Ui5
Call-ID: 82eb2d4d-30e9-4780-9e1d-52ae11750e38
CSeq: 26624 ACK
Max-Forwards: 70
User-Agent: Asterisk PBX 13.11.2
Content-Length:  0


<--- Received SIP request (1397 bytes) from UDP:212.210.2.2:5060 --->
INVITE sip:963760960355@media.syploop.net;user=phone SIP/2.0
Content-Length:316
From:"+37167366321" <sip:+37167366321@SIPMSS1-SIP.MGWS.LV;user=phone>;tag=Ud1iDW0gb3UC9A45
To:<sip:963760960355@media.syploop.net;user=phone>
Via:SIP/2.0/UDP SIPMSS1-SIP.MGWS.LV:5060;branch=z9hG4bKW9f6d2fbfXhCZY26;yop=00.00.AD31DA87.0000.700B
Call-ID:C5F92F7EC2C3F158F64FBA17@0b70ffffffff
CSeq:1 INVITE
Max-Forwards:70
Record-Route:<sip:AAQACCzlwAAAnAAAAMgAAfGkK@212.210.2.2:5060;yop=00.00.AD31DA87.0000.700B;transport=UDP;lr>
Request-Disposition:no-fork
P-Asserted-Identity:<sip:+37167366321@SIPMSS1-SIP.MGWS.LV;user=phone>,<tel:+37167366321>
Session-Expires:1800;refresher=uac
Contact:<sip:212.210.2.2:5060;transport=UDP;yop=00.00.AD31DA87.0000.700B>
Supported:100rel,timer,norefersub
Allow:ACK,BYE,CANCEL,INFO,INVITE,NOTIFY,OPTIONS,PRACK,REFER,UPDATE
P-Charging-Vector:icid-value=c2BImQLw~~~~~~~~~~~~~wQA7eo2;icid-generated-at=212.210.2.2;orig-ioi=SIPMSS1-SIP.MGWS.LV
P-Early-Media:supported
Content-Type:application/sdp
Content-Disposition:session;handling=required

v=0
o=- 0 0 IN IP4 10.20.193.213
s=-
c=IN IP4 213.102.44.245
t=0 0
m=audio 63072 RTP/AVP 96 97 3 8 98
b=AS:80
a=rtpmap:96 AMR/8000
a=fmtp:96 mode-change-capability=2; max-red=0
a=rtpmap:97 GSM-EFR/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:98 telephone-event/8000
a=ptime:20
a=maxptime:20

<--- Transmitting SIP response (533 bytes) to UDP:212.210.2.2:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP SIPMSS1-SIP.MGWS.LV:5060;rport=5060;received=212.210.2.2;branch=z9hG4bKW9f6d2fbfXhCZY26;yop=00.00.AD31DA87.0000.700B
Record-Route: <sip:AAQACCzlwAAAnAAAAMgAAfGkK@212.210.2.2:5060;transport=UDP;lr;yop=00.00.AD31DA87.0000.700B>
Call-ID: C5F92F7EC2C3F158F64FBA17@0b70ffffffff
From: "+37167366321" <sip:+37167366321@SIPMSS1-SIP.MGWS.LV;user=phone>;tag=Ud1iDW0gb3UC9A45
To: <sip:963760960355@media.syploop.net;user=phone>
CSeq: 1 INVITE
Server: Asterisk PBX 13.11.2
Content-Length:  0


    -- Executing [963760960355@t2-1:1] Dial("PJSIP/smex-1-00000004", "PJSIP/963860960355@smex-1") in new stack
    -- Called PJSIP/963860960355@smex-1
<--- Transmitting SIP request (1031 bytes) to UDP:212.210.2.2:5060 --->
INVITE sip:963860960355@212.210.2.2:5060 SIP/2.0
Via: SIP/2.0/UDP 10.40.6.100:5060;rport;branch=z9hG4bKPj79f09bce-5e8f-475f-b556-6a7002a0f586
From: "+37167366321" <sip:+37167366321@10.40.6.100>;tag=68c95d15-ea04-4031-8a21-e720c6f0da19
To: <sip:963860960355@212.210.2.2>
Contact: <sip:asterisk@10.40.6.100:5060>
Call-ID: 07ae7746-3520-499f-b206-119381ec55ff
CSeq: 10312 INVITE
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
Require: 100rel
Session-Expires: 1800
Min-SE: 90
P-Asserted-Identity: "+37167366321" <sip:+37167366321@10.40.6.100>
Max-Forwards: 70
User-Agent: Asterisk PBX 13.11.2
Content-Type: application/sdp
Content-Length:   257

v=0
o=- 521358322 521358322 IN IP4 10.40.6.100
s=Asterisk
c=IN IP4 10.40.6.100
t=0 0
m=audio 11370 RTP/AVP 0 8 101
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=maxptime:150
a=sendrecv

<--- Received SIP response (310 bytes) from UDP:212.210.2.2:5060 --->
SIP/2.0 100 Trying
From:"+37167366321" <sip:+37167366321@10.40.6.100>;tag=68c95d15-ea04-4031-8a21-e720c6f0da19
To:<sip:963860960355@212.210.2.2>
Via:SIP/2.0/UDP 10.40.6.100:5060;rport;branch=z9hG4bKPj79f09bce-5e8f-475f-b556-6a7002a0f586
Call-ID:07ae7746-3520-499f-b206-119381ec55ff
CSeq:10312 INVITE


<--- Received SIP response (1107 bytes) from UDP:212.210.2.2:5060 --->
SIP/2.0 183 Session Progress
Content-Length:194
From:"+37167366321" <sip:+37167366321@10.40.6.100>;tag=68c95d15-ea04-4031-8a21-e720c6f0da19
To:<sip:963860960355@212.210.2.2>;tag=cah034X1AhcA7i25
Via:SIP/2.0/UDP 10.40.6.100:5060;rport;branch=z9hG4bKPj79f09bce-5e8f-475f-b556-6a7002a0f586
Call-ID:07ae7746-3520-499f-b206-119381ec55ff
CSeq:10312 INVITE
Record-Route:<sip:AAQACBZJwAADHAAAACgAAfPkK@212.210.2.2:5060;yop=00.00.AD31DA87.0000.7005;transport=UDP;lr>
RSeq:1
Contact:<sip:212.210.2.2:5060;transport=UDP;yop=00.00.AD31DA87.0000.7005>
Require:100rel
Supported:100rel,precondition,timer,norefersub
Allow:ACK,BYE,CANCEL,INFO,INVITE,NOTIFY,OPTIONS,PRACK,REFER,UPDATE
P-Charging-Vector:icid-value=c2BImQLi9rpUC5UFcMwFDAs-;icid-generated-at=212.210.2.2;orig-ioi=SIPMSS1-SIP.MGWS.LV;term-ioi=SIPMSS1-SIP.MGWS.LV
Content-Type:application/sdp
Content-Disposition:session;handling=required

v=0
o=- 0 0 IN IP4 10.20.193.213
s=-
c=IN IP4 213.102.45.245
t=0 0
m=audio 32538 RTP/AVP 8 101
b=AS:80
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=ptime:20
a=maxptime:20

<--- Transmitting SIP request (595 bytes) to UDP:212.210.2.2:5060 --->
PRACK sip:212.210.2.2:5060;transport=UDP;yop=00.00.AD31DA87.0000.7005 SIP/2.0
Via: SIP/2.0/UDP 10.40.6.100:5060;rport;branch=z9hG4bKPj0b216daf-7802-4563-b5b9-bccda869fc5c
From: "+37167366321" <sip:+37167366321@10.40.6.100>;tag=68c95d15-ea04-4031-8a21-e720c6f0da19
To: <sip:963860960355@212.210.2.2>;tag=cah034X1AhcA7i25
Call-ID: 07ae7746-3520-499f-b206-119381ec55ff
CSeq: 10313 PRACK
Route: <sip:AAQACBZJwAADHAAAACgAAfPkK@212.210.2.2:5060;transport=UDP;lr;yop=00.00.AD31DA87.0000.7005>
RAck: 1 10312 INVITE
Max-Forwards: 70
User-Agent: Asterisk PBX 13.11.2
Content-Length:  0


    -- PJSIP/smex-1-00000005 is making progress passing it to PJSIP/smex-1-00000004
<--- Transmitting SIP response (1106 bytes) to UDP:212.210.2.2:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP SIPMSS1-SIP.MGWS.LV:5060;rport=5060;received=212.210.2.2;branch=z9hG4bKW9f6d2fbfXhCZY26;yop=00.00.AD31DA87.0000.700B
Record-Route: <sip:AAQACCzlwAAAnAAAAMgAAfGkK@212.210.2.2:5060;transport=UDP;lr;yop=00.00.AD31DA87.0000.700B>
Call-ID: C5F92F7EC2C3F158F64FBA17@0b70ffffffff
From: "+37167366321" <sip:+37167366321@SIPMSS1-SIP.MGWS.LV;user=phone>;tag=Ud1iDW0gb3UC9A45
To: <sip:963760960355@media.syploop.net;user=phone>;tag=1d9b58e8-0467-4778-8a11-b4d5ee18122f
CSeq: 1 INVITE
Server: Asterisk PBX 13.11.2
Contact: <sip:10.40.6.100:5060>
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, MESSAGE, REFER
P-Asserted-Identity: <sip:963760960355@media.syploop.net;user=phone>
Require: 100rel
RSeq: 28398
Server: Asterisk PBX 13.11.2
Content-Type: application/sdp
Content-Length:   214

v=0
o=- 0 2 IN IP4 10.40.6.100
s=Asterisk
c=IN IP4 10.40.6.100
t=0 0
m=audio 18584 RTP/AVP 8 98
a=rtpmap:8 PCMA/8000
a=rtpmap:98 telephone-event/8000
a=fmtp:98 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<--- Received SIP response (460 bytes) from UDP:212.210.2.2:5060 --->
SIP/2.0 200 OK
Content-Length:0
From:"+37167366321" <sip:+37167366321@10.40.6.100>;tag=68c95d15-ea04-4031-8a21-e720c6f0da19
To:<sip:963860960355@212.210.2.2>;tag=cah034X1AhcA7i25
Via:SIP/2.0/UDP 10.40.6.100:5060;rport;branch=z9hG4bKPj0b216daf-7802-4563-b5b9-bccda869fc5c
Call-ID:07ae7746-3520-499f-b206-119381ec55ff
CSeq:10313 PRACK
Supported:100rel,precondition,timer,norefersub
Allow:ACK,BYE,CANCEL,INFO,INVITE,NOTIFY,OPTIONS,PRACK,REFER,UPDATE


       > 0x7f3ff402d010 -- Probation passed - setting RTP source address to 213.102.44.245:63072
<--- Received SIP request (490 bytes) from UDP:212.210.2.2:5060 --->
PRACK sip:10.40.6.100:5060 SIP/2.0
Content-Length:0
From:"+37167366321" <sip:+37167366321@SIPMSS1-SIP.MGWS.LV;user=phone>;tag=Ud1iDW0gb3UC9A45
To:<sip:963760960355@media.syploop.net;user=phone>;tag=1d9b58e8-0467-4778-8a11-b4d5ee18122f
Via:SIP/2.0/UDP SIPMSS1-SIP.MGWS.LV:5060;branch=z9hG4bKe0A97A9DX9idiB26;yop=00.00.AD31DA87.0000.700B
Call-ID:C5F92F7EC2C3F158F64FBA17@0b70ffffffff
CSeq:2 PRACK
Max-Forwards:70
Request-Disposition:no-fork
RAck:28398 1 INVITE
Supported:timer


<--- Transmitting SIP response (457 bytes) to UDP:212.210.2.2:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP SIPMSS1-SIP.MGWS.LV:5060;rport=5060;received=212.210.2.2;branch=z9hG4bKe0A97A9DX9idiB26;yop=00.00.AD31DA87.0000.700B
Call-ID: C5F92F7EC2C3F158F64FBA17@0b70ffffffff
From: "+37167366321" <sip:+37167366321@SIPMSS1-SIP.MGWS.LV;user=phone>;tag=Ud1iDW0gb3UC9A45
To: <sip:963760960355@media.syploop.net;user=phone>;tag=1d9b58e8-0467-4778-8a11-b4d5ee18122f
CSeq: 2 PRACK
Server: Asterisk PBX 13.11.2
Content-Length:  0


<--- Received SIP response (825 bytes) from UDP:212.210.2.2:5060 --->
SIP/2.0 180 Ringing
Content-Length:0
From:"+37167366321" <sip:+37167366321@10.40.6.100>;tag=68c95d15-ea04-4031-8a21-e720c6f0da19
To:<sip:963860960355@212.210.2.2>;tag=cah034X1AhcA7i25
Via:SIP/2.0/UDP 10.40.6.100:5060;rport;branch=z9hG4bKPj79f09bce-5e8f-475f-b556-6a7002a0f586
Call-ID:07ae7746-3520-499f-b206-119381ec55ff
CSeq:10312 INVITE
Record-Route:<sip:AAQACBZJwAADHAAAACgAAfPkK@212.210.2.2:5060;yop=00.00.AD31DA87.0000.7005;transport=UDP;lr>
RSeq:2
Contact:<sip:212.210.2.2:5060;transport=UDP;yop=00.00.AD31DA87.0000.7005>
Require:100rel
Supported:100rel,precondition,timer,norefersub
Allow:ACK,BYE,CANCEL,INFO,INVITE,NOTIFY,OPTIONS,PRACK,REFER,UPDATE
P-Charging-Vector:icid-value=c2BImQLi9rpUC5UFcMwFDAs-;icid-generated-at=212.210.2.2;orig-ioi=SIPMSS1-SIP.MGWS.LV;term-ioi=SIPMSS1-SIP.MGWS.LV


<--- Transmitting SIP request (595 bytes) to UDP:212.210.2.2:5060 --->
PRACK sip:212.210.2.2:5060;transport=UDP;yop=00.00.AD31DA87.0000.7005 SIP/2.0
Via: SIP/2.0/UDP 10.40.6.100:5060;rport;branch=z9hG4bKPj626d510a-6be3-47c8-9812-f52e03b45044
From: "+37167366321" <sip:+37167366321@10.40.6.100>;tag=68c95d15-ea04-4031-8a21-e720c6f0da19
To: <sip:963860960355@212.210.2.2>;tag=cah034X1AhcA7i25
Call-ID: 07ae7746-3520-499f-b206-119381ec55ff
CSeq: 10314 PRACK
Route: <sip:AAQACBZJwAADHAAAACgAAfPkK@212.210.2.2:5060;transport=UDP;lr;yop=00.00.AD31DA87.0000.7005>
RAck: 2 10312 INVITE
Max-Forwards: 70
User-Agent: Asterisk PBX 13.11.2
Content-Length:  0


    -- PJSIP/smex-1-00000005 is ringing
<--- Transmitting SIP response (849 bytes) to UDP:212.210.2.2:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP SIPMSS1-SIP.MGWS.LV:5060;rport=5060;received=212.210.2.2;branch=z9hG4bKW9f6d2fbfXhCZY26;yop=00.00.AD31DA87.0000.700B
Record-Route: <sip:AAQACCzlwAAAnAAAAMgAAfGkK@212.210.2.2:5060;transport=UDP;lr;yop=00.00.AD31DA87.0000.700B>
Call-ID: C5F92F7EC2C3F158F64FBA17@0b70ffffffff
From: "+37167366321" <sip:+37167366321@SIPMSS1-SIP.MGWS.LV;user=phone>;tag=Ud1iDW0gb3UC9A45
To: <sip:963760960355@media.syploop.net;user=phone>;tag=1d9b58e8-0467-4778-8a11-b4d5ee18122f
CSeq: 1 INVITE
Server: Asterisk PBX 13.11.2
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, MESSAGE, REFER
Contact: <sip:10.40.6.100:5060>
P-Asserted-Identity: <sip:963760960355@media.syploop.net;user=phone>
Require: 100rel
RSeq: 28399
Server: Asterisk PBX 13.11.2
Content-Length:  0


<--- Received SIP response (460 bytes) from UDP:212.210.2.2:5060 --->
SIP/2.0 200 OK
Content-Length:0
From:"+37167366321" <sip:+37167366321@10.40.6.100>;tag=68c95d15-ea04-4031-8a21-e720c6f0da19
To:<sip:963860960355@212.210.2.2>;tag=cah034X1AhcA7i25
Via:SIP/2.0/UDP 10.40.6.100:5060;rport;branch=z9hG4bKPj626d510a-6be3-47c8-9812-f52e03b45044
Call-ID:07ae7746-3520-499f-b206-119381ec55ff
CSeq:10314 PRACK
Supported:100rel,precondition,timer,norefersub
Allow:ACK,BYE,CANCEL,INFO,INVITE,NOTIFY,OPTIONS,PRACK,REFER,UPDATE


<--- Received SIP request (490 bytes) from UDP:212.210.2.2:5060 --->
PRACK sip:10.40.6.100:5060 SIP/2.0
Content-Length:0
From:"+37167366321" <sip:+37167366321@SIPMSS1-SIP.MGWS.LV;user=phone>;tag=Ud1iDW0gb3UC9A45
To:<sip:963760960355@media.syploop.net;user=phone>;tag=1d9b58e8-0467-4778-8a11-b4d5ee18122f
Via:SIP/2.0/UDP SIPMSS1-SIP.MGWS.LV:5060;branch=z9hG4bKUUBbd0h0c1i_jcY6;yop=00.00.AD31DA87.0000.700B
Call-ID:C5F92F7EC2C3F158F64FBA17@0b70ffffffff
CSeq:3 PRACK
Max-Forwards:70
Request-Disposition:no-fork
RAck:28399 1 INVITE
Supported:timer


<--- Transmitting SIP response (457 bytes) to UDP:212.210.2.2:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP SIPMSS1-SIP.MGWS.LV:5060;rport=5060;received=212.210.2.2;branch=z9hG4bKUUBbd0h0c1i_jcY6;yop=00.00.AD31DA87.0000.700B
Call-ID: C5F92F7EC2C3F158F64FBA17@0b70ffffffff
From: "+37167366321" <sip:+37167366321@SIPMSS1-SIP.MGWS.LV;user=phone>;tag=Ud1iDW0gb3UC9A45
To: <sip:963760960355@media.syploop.net;user=phone>;tag=1d9b58e8-0467-4778-8a11-b4d5ee18122f
CSeq: 3 PRACK
Server: Asterisk PBX 13.11.2
Content-Length:  0


       > 0x7f3ff4029c00 -- Probation passed - setting RTP source address to 213.102.45.245:32538
<--- Received SIP request (499 bytes) from UDP:212.210.2.2:5060 --->
CANCEL sip:963760960355@media.syploop.net;user=phone SIP/2.0
Content-Length:0
From:"+37167366321" <sip:+37167366321@SIPMSS1-SIP.MGWS.LV;user=phone>;tag=Ud1iDW0gb3UC9A45
To:<sip:963760960355@media.syploop.net;user=phone>
Via:SIP/2.0/UDP SIPMSS1-SIP.MGWS.LV:5060;branch=z9hG4bKW9f6d2fbfXhCZY26;yop=00.00.AD31DA87.0000.700B
Call-ID:C5F92F7EC2C3F158F64FBA17@0b70ffffffff
CSeq:1 CANCEL
Max-Forwards:70
Request-Disposition:no-fork
Reason:X.int ;reasoncode=0x0000030B;add-info=0109.0001.0004


<--- Transmitting SIP response (458 bytes) to UDP:212.210.2.2:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP SIPMSS1-SIP.MGWS.LV:5060;rport=5060;received=212.210.2.2;branch=z9hG4bKW9f6d2fbfXhCZY26;yop=00.00.AD31DA87.0000.700B
Call-ID: C5F92F7EC2C3F158F64FBA17@0b70ffffffff
From: "+37167366321" <sip:+37167366321@SIPMSS1-SIP.MGWS.LV;user=phone>;tag=Ud1iDW0gb3UC9A45
To: <sip:963760960355@media.syploop.net;user=phone>;tag=1d9b58e8-0467-4778-8a11-b4d5ee18122f
CSeq: 1 CANCEL
Server: Asterisk PBX 13.11.2
Content-Length:  0


<--- Transmitting SIP response (827 bytes) to UDP:212.210.2.2:5060 --->
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP SIPMSS1-SIP.MGWS.LV:5060;rport=5060;received=212.210.2.2;branch=z9hG4bKW9f6d2fbfXhCZY26;yop=00.00.AD31DA87.0000.700B
Record-Route: <sip:AAQACCzlwAAAnAAAAMgAAfGkK@212.210.2.2:5060;transport=UDP;lr;yop=00.00.AD31DA87.0000.700B>
Call-ID: C5F92F7EC2C3F158F64FBA17@0b70ffffffff
From: "+37167366321" <sip:+37167366321@SIPMSS1-SIP.MGWS.LV;user=phone>;tag=Ud1iDW0gb3UC9A45
To: <sip:963760960355@media.syploop.net;user=phone>;tag=1d9b58e8-0467-4778-8a11-b4d5ee18122f
CSeq: 1 INVITE
Server: Asterisk PBX 13.11.2
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, MESSAGE, REFER
Server: Asterisk PBX 13.11.2
P-Asserted-Identity: <sip:963760960355@media.syploop.net;user=phone>
Server: Asterisk PBX 13.11.2
Content-Length:  0


  == Spawn extension (t2-1, 963760960355, 1) exited non-zero on 'PJSIP/smex-1-00000004'
<--- Transmitting SIP request (442 bytes) to UDP:212.210.2.2:5060 --->
CANCEL sip:963860960355@212.210.2.2:5060 SIP/2.0
Via: SIP/2.0/UDP 10.40.6.100:5060;rport;branch=z9hG4bKPj79f09bce-5e8f-475f-b556-6a7002a0f586
From: "+37167366321" <sip:+37167366321@10.40.6.100>;tag=68c95d15-ea04-4031-8a21-e720c6f0da19
To: <sip:963860960355@212.210.2.2>
Call-ID: 07ae7746-3520-499f-b206-119381ec55ff
CSeq: 10312 CANCEL
Reason: Q.850;cause=0
Max-Forwards: 70
User-Agent: Asterisk PBX 13.11.2
Content-Length:  0


<--- Received SIP request (444 bytes) from UDP:212.210.2.2:5060 --->
ACK sip:963760960355@media.syploop.net;user=phone SIP/2.0
Content-Length:0
From:"+37167366321" <sip:+37167366321@SIPMSS1-SIP.MGWS.LV;user=phone>;tag=Ud1iDW0gb3UC9A45
To:<sip:963760960355@media.syploop.net;user=phone>;tag=1d9b58e8-0467-4778-8a11-b4d5ee18122f
Via:SIP/2.0/UDP SIPMSS1-SIP.MGWS.LV:5060;branch=z9hG4bKW9f6d2fbfXhCZY26;yop=00.00.AD31DA87.0000.700B
Call-ID:C5F92F7EC2C3F158F64FBA17@0b70ffffffff
CSeq:1 ACK
Max-Forwards:70


<--- Received SIP response (461 bytes) from UDP:212.210.2.2:5060 --->
SIP/2.0 200 OK
Content-Length:0
From:"+37167366321" <sip:+37167366321@10.40.6.100>;tag=68c95d15-ea04-4031-8a21-e720c6f0da19
To:<sip:963860960355@212.210.2.2>;tag=cah034X1AhcA7i25
Via:SIP/2.0/UDP 10.40.6.100:5060;rport;branch=z9hG4bKPj79f09bce-5e8f-475f-b556-6a7002a0f586
Call-ID:07ae7746-3520-499f-b206-119381ec55ff
CSeq:10312 CANCEL
Supported:100rel,precondition,timer,norefersub
Allow:ACK,BYE,CANCEL,INFO,INVITE,NOTIFY,OPTIONS,PRACK,REFER,UPDATE


<--- Received SIP response (421 bytes) from UDP:212.210.2.2:5060 --->
SIP/2.0 487 Request Cancelled
Content-Length:0
From:"+37167366321" <sip:+37167366321@10.40.6.100>;tag=68c95d15-ea04-4031-8a21-e720c6f0da19
To:<sip:963860960355@212.210.2.2>;tag=cah034X1AhcA7i25
Via:SIP/2.0/UDP 10.40.6.100:5060;rport;branch=z9hG4bKPj79f09bce-5e8f-475f-b556-6a7002a0f586
Call-ID:07ae7746-3520-499f-b206-119381ec55ff
CSeq:10312 INVITE
Reason:X.int ;reasoncode=0x0000030B;add-info=05CC.0001.0002


<--- Transmitting SIP request (434 bytes) to UDP:212.210.2.2:5060 --->
ACK sip:963860960355@212.210.2.2:5060 SIP/2.0
Via: SIP/2.0/UDP 10.40.6.100:5060;rport;branch=z9hG4bKPj79f09bce-5e8f-475f-b556-6a7002a0f586
From: "+37167366321" <sip:+37167366321@10.40.6.100>;tag=68c95d15-ea04-4031-8a21-e720c6f0da19
To: <sip:963860960355@212.210.2.2>;tag=cah034X1AhcA7i25
Call-ID: 07ae7746-3520-499f-b206-119381ec55ff
CSeq: 10312 ACK
Max-Forwards: 70
User-Agent: Asterisk PBX 13.11.2
Content-Length:  0


<--- Transmitting SIP request (423 bytes) to UDP:212.210.2.2:5060 --->
OPTIONS sip:212.210.2.2:5060 SIP/2.0
Via: SIP/2.0/UDP 10.40.6.100:5060;rport;branch=z9hG4bKPj6cc9837e-8220-4da9-8294-67dedfed5b72
From: <sip:smex-1@10.40.6.100>;tag=05bdd69c-8f25-483e-afb2-d96dba2a48b2
To: <sip:212.210.2.2>
Contact: <sip:smex-1@10.40.6.100:5060>
Call-ID: 2f71b371-a5f4-4a61-b386-e77901b58978
CSeq: 9057 OPTIONS
Max-Forwards: 70
User-Agent: Asterisk PBX 13.11.2
Content-Length:  0


<--- Received SIP response (305 bytes) from UDP:212.210.2.2:5060 --->
SIP/2.0 200 OK
Content-Length:0
From:<sip:smex-1@10.40.6.100>;tag=05bdd69c-8f25-483e-afb2-d96dba2a48b2
To:<sip:212.210.2.2>;tag=dummy
Via:SIP/2.0/UDP 10.40.6.100:5060;rport;branch=z9hG4bKPj6cc9837e-8220-4da9-8294-67dedfed5b72
Call-ID:2f71b371-a5f4-4a61-b386-e77901b58978
CSeq:9057 OPTIONS


<--- Received SIP request (1397 bytes) from UDP:212.210.2.2:5060 --->
INVITE sip:963760960355@media.syploop.net;user=phone SIP/2.0
Content-Length:316
From:"+37167366321" <sip:+37167366321@SIPMSS1-SIP.MGWS.LV;user=phone>;tag=D9XaDgW7j3Xd7045
To:<sip:963760960355@media.syploop.net;user=phone>
Via:SIP/2.0/UDP SIPMSS1-SIP.MGWS.LV:5060;branch=z9hG4bK8ifC7_b4e9hUYbY6;yop=00.00.AD31DA87.0000.700B
Call-ID:A929277ED3F151588CD91617@0b70ffffffff
CSeq:1 INVITE
Max-Forwards:70
Record-Route:<sip:AAQACCzlwAAAnAAAAMgAAfGkK@212.210.2.2:5060;yop=00.00.AD31DA87.0000.700B;transport=UDP;lr>
Request-Disposition:no-fork
P-Asserted-Identity:<sip:+37167366321@SIPMSS1-SIP.MGWS.LV;user=phone>,<tel:+37167366321>
Session-Expires:1800;refresher=uac
Contact:<sip:212.210.2.2:5060;transport=UDP;yop=00.00.AD31DA87.0000.700B>
Supported:100rel,timer,norefersub
Allow:ACK,BYE,CANCEL,INFO,INVITE,NOTIFY,OPTIONS,PRACK,REFER,UPDATE
P-Charging-Vector:icid-value=c2BImQLw~~~~~~~~~~~~~wSK7eo2;icid-generated-at=212.210.2.2;orig-ioi=SIPMSS1-SIP.MGWS.LV
P-Early-Media:supported
Content-Type:application/sdp
Content-Disposition:session;handling=required

v=0
o=- 0 0 IN IP4 10.20.193.213
s=-
c=IN IP4 213.102.44.245
t=0 0
m=audio 63230 RTP/AVP 96 97 3 8 98
b=AS:80
a=rtpmap:96 AMR/8000
a=fmtp:96 mode-change-capability=2; max-red=0
a=rtpmap:97 GSM-EFR/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:98 telephone-event/8000
a=ptime:20
a=maxptime:20

<--- Transmitting SIP response (533 bytes) to UDP:212.210.2.2:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP SIPMSS1-SIP.MGWS.LV:5060;rport=5060;received=212.210.2.2;branch=z9hG4bK8ifC7_b4e9hUYbY6;yop=00.00.AD31DA87.0000.700B
Record-Route: <sip:AAQACCzlwAAAnAAAAMgAAfGkK@212.210.2.2:5060;transport=UDP;lr;yop=00.00.AD31DA87.0000.700B>
Call-ID: A929277ED3F151588CD91617@0b70ffffffff
From: "+37167366321" <sip:+37167366321@SIPMSS1-SIP.MGWS.LV;user=phone>;tag=D9XaDgW7j3Xd7045
To: <sip:963760960355@media.syploop.net;user=phone>
CSeq: 1 INVITE
Server: Asterisk PBX 13.11.2
Content-Length:  0


    -- Executing [963760960355@t2-1:1] Dial("PJSIP/smex-1-00000006", "PJSIP/963860960355@smex-1") in new stack
    -- Called PJSIP/963860960355@smex-1
<--- Transmitting SIP request (1032 bytes) to UDP:212.210.2.2:5060 --->
INVITE sip:963860960355@212.210.2.2:5060 SIP/2.0
Via: SIP/2.0/UDP 10.40.6.100:5060;rport;branch=z9hG4bKPj9c6fb510-03cb-4071-9e41-d1114451390b
From: "+37167366321" <sip:+37167366321@10.40.6.100>;tag=16cc0afc-1482-4e4b-98ec-4c5033943d9d
To: <sip:963860960355@212.210.2.2>
Contact: <sip:asterisk@10.40.6.100:5060>
Call-ID: 7e00a784-3eac-4e35-9aab-eda898d9cb46
CSeq: 8080 INVITE
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
Require: 100rel
Session-Expires: 1800
Min-SE: 90
P-Asserted-Identity: "+37167366321" <sip:+37167366321@10.40.6.100>
Max-Forwards: 70
User-Agent: Asterisk PBX 13.11.2
Content-Type: application/sdp
Content-Length:   259

v=0
o=- 1494597313 1494597313 IN IP4 10.40.6.100
s=Asterisk
c=IN IP4 10.40.6.100
t=0 0
m=audio 16792 RTP/AVP 0 8 101
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=maxptime:150
a=sendrecv

<--- Received SIP response (309 bytes) from UDP:212.210.2.2:5060 --->
SIP/2.0 100 Trying
From:"+37167366321" <sip:+37167366321@10.40.6.100>;tag=16cc0afc-1482-4e4b-98ec-4c5033943d9d
To:<sip:963860960355@212.210.2.2>
Via:SIP/2.0/UDP 10.40.6.100:5060;rport;branch=z9hG4bKPj9c6fb510-03cb-4071-9e41-d1114451390b
Call-ID:7e00a784-3eac-4e35-9aab-eda898d9cb46
CSeq:8080 INVITE


<--- Received SIP response (1106 bytes) from UDP:212.210.2.2:5060 --->
SIP/2.0 183 Session Progress
Content-Length:194
From:"+37167366321" <sip:+37167366321@10.40.6.100>;tag=16cc0afc-1482-4e4b-98ec-4c5033943d9d
To:<sip:963860960355@212.210.2.2>;tag=ZCA13_hZgDjXYe65
Via:SIP/2.0/UDP 10.40.6.100:5060;rport;branch=z9hG4bKPj9c6fb510-03cb-4071-9e41-d1114451390b
Call-ID:7e00a784-3eac-4e35-9aab-eda898d9cb46
CSeq:8080 INVITE
Record-Route:<sip:AAQACDydwAABrAAAAgAAAfL4K@212.210.2.2:5060;yop=00.00.AD31DA87.0000.700F;transport=UDP;lr>
RSeq:1
Contact:<sip:212.210.2.2:5060;transport=UDP;yop=00.00.AD31DA87.0000.700F>
Require:100rel
Supported:100rel,precondition,timer,norefersub
Allow:ACK,BYE,CANCEL,INFO,INVITE,NOTIFY,OPTIONS,PRACK,REFER,UPDATE
P-Charging-Vector:icid-value=c2BImQI8ALtUDY8PcMwFlBg-;icid-generated-at=212.210.2.2;orig-ioi=SIPMSS1-SIP.MGWS.LV;term-ioi=SIPMSS1-SIP.MGWS.LV
Content-Type:application/sdp
Content-Disposition:session;handling=required

v=0
o=- 0 0 IN IP4 10.20.193.213
s=-
c=IN IP4 213.102.45.246
t=0 0
m=audio 15974 RTP/AVP 8 101
b=AS:80
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=ptime:20
a=maxptime:20

<--- Transmitting SIP request (593 bytes) to UDP:212.210.2.2:5060 --->
PRACK sip:212.210.2.2:5060;transport=UDP;yop=00.00.AD31DA87.0000.700F SIP/2.0
Via: SIP/2.0/UDP 10.40.6.100:5060;rport;branch=z9hG4bKPj8a318be8-2d18-435b-89a4-49a5fbc5bd6c
From: "+37167366321" <sip:+37167366321@10.40.6.100>;tag=16cc0afc-1482-4e4b-98ec-4c5033943d9d
To: <sip:963860960355@212.210.2.2>;tag=ZCA13_hZgDjXYe65
Call-ID: 7e00a784-3eac-4e35-9aab-eda898d9cb46
CSeq: 8081 PRACK
Route: <sip:AAQACDydwAABrAAAAgAAAfL4K@212.210.2.2:5060;transport=UDP;lr;yop=00.00.AD31DA87.0000.700F>
RAck: 1 8080 INVITE
Max-Forwards: 70
User-Agent: Asterisk PBX 13.11.2
Content-Length:  0


    -- PJSIP/smex-1-00000007 is making progress passing it to PJSIP/smex-1-00000006
<--- Transmitting SIP response (1106 bytes) to UDP:212.210.2.2:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP SIPMSS1-SIP.MGWS.LV:5060;rport=5060;received=212.210.2.2;branch=z9hG4bK8ifC7_b4e9hUYbY6;yop=00.00.AD31DA87.0000.700B
Record-Route: <sip:AAQACCzlwAAAnAAAAMgAAfGkK@212.210.2.2:5060;transport=UDP;lr;yop=00.00.AD31DA87.0000.700B>
Call-ID: A929277ED3F151588CD91617@0b70ffffffff
From: "+37167366321" <sip:+37167366321@SIPMSS1-SIP.MGWS.LV;user=phone>;tag=D9XaDgW7j3Xd7045
To: <sip:963760960355@media.syploop.net;user=phone>;tag=1299bfb1-4d0a-422a-89c9-972f57dadfca
CSeq: 1 INVITE
Server: Asterisk PBX 13.11.2
Contact: <sip:10.40.6.100:5060>
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, MESSAGE, REFER
P-Asserted-Identity: <sip:963760960355@media.syploop.net;user=phone>
Require: 100rel
RSeq: 17902
Server: Asterisk PBX 13.11.2
Content-Type: application/sdp
Content-Length:   214

v=0
o=- 0 2 IN IP4 10.40.6.100
s=Asterisk
c=IN IP4 10.40.6.100
t=0 0
m=audio 10268 RTP/AVP 8 98
a=rtpmap:8 PCMA/8000
a=rtpmap:98 telephone-event/8000
a=fmtp:98 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<--- Received SIP response (459 bytes) from UDP:212.210.2.2:5060 --->
SIP/2.0 200 OK
Content-Length:0
From:"+37167366321" <sip:+37167366321@10.40.6.100>;tag=16cc0afc-1482-4e4b-98ec-4c5033943d9d
To:<sip:963860960355@212.210.2.2>;tag=ZCA13_hZgDjXYe65
Via:SIP/2.0/UDP 10.40.6.100:5060;rport;branch=z9hG4bKPj8a318be8-2d18-435b-89a4-49a5fbc5bd6c
Call-ID:7e00a784-3eac-4e35-9aab-eda898d9cb46
CSeq:8081 PRACK
Supported:100rel,precondition,timer,norefersub
Allow:ACK,BYE,CANCEL,INFO,INVITE,NOTIFY,OPTIONS,PRACK,REFER,UPDATE


       > 0x7f3ff402d010 -- Probation passed - setting RTP source address to 213.102.44.245:63230
<--- Received SIP request (490 bytes) from UDP:212.210.2.2:5060 --->
PRACK sip:10.40.6.100:5060 SIP/2.0
Content-Length:0
From:"+37167366321" <sip:+37167366321@SIPMSS1-SIP.MGWS.LV;user=phone>;tag=D9XaDgW7j3Xd7045
To:<sip:963760960355@media.syploop.net;user=phone>;tag=1299bfb1-4d0a-422a-89c9-972f57dadfca
Via:SIP/2.0/UDP SIPMSS1-SIP.MGWS.LV:5060;branch=z9hG4bK63bUdDZ9VXidXU26;yop=00.00.AD31DA87.0000.700B
Call-ID:A929277ED3F151588CD91617@0b70ffffffff
CSeq:2 PRACK
Max-Forwards:70
Request-Disposition:no-fork
RAck:17902 1 INVITE
Supported:timer


<--- Transmitting SIP response (457 bytes) to UDP:212.210.2.2:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP SIPMSS1-SIP.MGWS.LV:5060;rport=5060;received=212.210.2.2;branch=z9hG4bK63bUdDZ9VXidXU26;yop=00.00.AD31DA87.0000.700B
Call-ID: A929277ED3F151588CD91617@0b70ffffffff
From: "+37167366321" <sip:+37167366321@SIPMSS1-SIP.MGWS.LV;user=phone>;tag=D9XaDgW7j3Xd7045
To: <sip:963760960355@media.syploop.net;user=phone>;tag=1299bfb1-4d0a-422a-89c9-972f57dadfca
CSeq: 2 PRACK
Server: Asterisk PBX 13.11.2
Content-Length:  0


<--- Received SIP response (824 bytes) from UDP:212.210.2.2:5060 --->
SIP/2.0 180 Ringing
Content-Length:0
From:"+37167366321" <sip:+37167366321@10.40.6.100>;tag=16cc0afc-1482-4e4b-98ec-4c5033943d9d
To:<sip:963860960355@212.210.2.2>;tag=ZCA13_hZgDjXYe65
Via:SIP/2.0/UDP 10.40.6.100:5060;rport;branch=z9hG4bKPj9c6fb510-03cb-4071-9e41-d1114451390b
Call-ID:7e00a784-3eac-4e35-9aab-eda898d9cb46
CSeq:8080 INVITE
Record-Route:<sip:AAQACDydwAABrAAAAgAAAfL4K@212.210.2.2:5060;yop=00.00.AD31DA87.0000.700F;transport=UDP;lr>
RSeq:2
Contact:<sip:212.210.2.2:5060;transport=UDP;yop=00.00.AD31DA87.0000.700F>
Require:100rel
Supported:100rel,precondition,timer,norefersub
Allow:ACK,BYE,CANCEL,INFO,INVITE,NOTIFY,OPTIONS,PRACK,REFER,UPDATE
P-Charging-Vector:icid-value=c2BImQI8ALtUDY8PcMwFlBg-;icid-generated-at=212.210.2.2;orig-ioi=SIPMSS1-SIP.MGWS.LV;term-ioi=SIPMSS1-SIP.MGWS.LV


<--- Transmitting SIP request (593 bytes) to UDP:212.210.2.2:5060 --->
PRACK sip:212.210.2.2:5060;transport=UDP;yop=00.00.AD31DA87.0000.700F SIP/2.0
Via: SIP/2.0/UDP 10.40.6.100:5060;rport;branch=z9hG4bKPjbb3ac4e9-104a-463a-a0f7-bc2afcb3cbea
From: "+37167366321" <sip:+37167366321@10.40.6.100>;tag=16cc0afc-1482-4e4b-98ec-4c5033943d9d
To: <sip:963860960355@212.210.2.2>;tag=ZCA13_hZgDjXYe65
Call-ID: 7e00a784-3eac-4e35-9aab-eda898d9cb46
CSeq: 8082 PRACK
Route: <sip:AAQACDydwAABrAAAAgAAAfL4K@212.210.2.2:5060;transport=UDP;lr;yop=00.00.AD31DA87.0000.700F>
RAck: 2 8080 INVITE
Max-Forwards: 70
User-Agent: Asterisk PBX 13.11.2
Content-Length:  0


    -- PJSIP/smex-1-00000007 is ringing
<--- Transmitting SIP response (849 bytes) to UDP:212.210.2.2:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP SIPMSS1-SIP.MGWS.LV:5060;rport=5060;received=212.210.2.2;branch=z9hG4bK8ifC7_b4e9hUYbY6;yop=00.00.AD31DA87.0000.700B
Record-Route: <sip:AAQACCzlwAAAnAAAAMgAAfGkK@212.210.2.2:5060;transport=UDP;lr;yop=00.00.AD31DA87.0000.700B>
Call-ID: A929277ED3F151588CD91617@0b70ffffffff
From: "+37167366321" <sip:+37167366321@SIPMSS1-SIP.MGWS.LV;user=phone>;tag=D9XaDgW7j3Xd7045
To: <sip:963760960355@media.syploop.net;user=phone>;tag=1299bfb1-4d0a-422a-89c9-972f57dadfca
CSeq: 1 INVITE
Server: Asterisk PBX 13.11.2
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, MESSAGE, REFER
Contact: <sip:10.40.6.100:5060>
P-Asserted-Identity: <sip:963760960355@media.syploop.net;user=phone>
Require: 100rel
RSeq: 17903
Server: Asterisk PBX 13.11.2
Content-Length:  0


<--- Received SIP response (459 bytes) from UDP:212.210.2.2:5060 --->
SIP/2.0 200 OK
Content-Length:0
From:"+37167366321" <sip:+37167366321@10.40.6.100>;tag=16cc0afc-1482-4e4b-98ec-4c5033943d9d
To:<sip:963860960355@212.210.2.2>;tag=ZCA13_hZgDjXYe65
Via:SIP/2.0/UDP 10.40.6.100:5060;rport;branch=z9hG4bKPjbb3ac4e9-104a-463a-a0f7-bc2afcb3cbea
Call-ID:7e00a784-3eac-4e35-9aab-eda898d9cb46
CSeq:8082 PRACK
Supported:100rel,precondition,timer,norefersub
Allow:ACK,BYE,CANCEL,INFO,INVITE,NOTIFY,OPTIONS,PRACK,REFER,UPDATE


<--- Received SIP request (490 bytes) from UDP:212.210.2.2:5060 --->
PRACK sip:10.40.6.100:5060 SIP/2.0
Content-Length:0
From:"+37167366321" <sip:+37167366321@SIPMSS1-SIP.MGWS.LV;user=phone>;tag=D9XaDgW7j3Xd7045
To:<sip:963760960355@media.syploop.net;user=phone>;tag=1299bfb1-4d0a-422a-89c9-972f57dadfca
Via:SIP/2.0/UDP SIPMSS1-SIP.MGWS.LV:5060;branch=z9hG4bKXZ7177gj79i.2726;yop=00.00.AD31DA87.0000.700B
Call-ID:A929277ED3F151588CD91617@0b70ffffffff
CSeq:3 PRACK
Max-Forwards:70
Request-Disposition:no-fork
RAck:17903 1 INVITE
Supported:timer


<--- Transmitting SIP response (457 bytes) to UDP:212.210.2.2:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP SIPMSS1-SIP.MGWS.LV:5060;rport=5060;received=212.210.2.2;branch=z9hG4bKXZ7177gj79i.2726;yop=00.00.AD31DA87.0000.700B
Call-ID: A929277ED3F151588CD91617@0b70ffffffff
From: "+37167366321" <sip:+37167366321@SIPMSS1-SIP.MGWS.LV;user=phone>;tag=D9XaDgW7j3Xd7045
To: <sip:963760960355@media.syploop.net;user=phone>;tag=1299bfb1-4d0a-422a-89c9-972f57dadfca
CSeq: 3 PRACK
Server: Asterisk PBX 13.11.2
Content-Length:  0


       > 0x7f3ff4029b70 -- Probation passed - setting RTP source address to 213.102.45.246:15974
<--- Received SIP request (499 bytes) from UDP:212.210.2.2:5060 --->
CANCEL sip:963760960355@media.syploop.net;user=phone SIP/2.0
Content-Length:0
From:"+37167366321" <sip:+37167366321@SIPMSS1-SIP.MGWS.LV;user=phone>;tag=D9XaDgW7j3Xd7045
To:<sip:963760960355@media.syploop.net;user=phone>
Via:SIP/2.0/UDP SIPMSS1-SIP.MGWS.LV:5060;branch=z9hG4bK8ifC7_b4e9hUYbY6;yop=00.00.AD31DA87.0000.700B
Call-ID:A929277ED3F151588CD91617@0b70ffffffff
CSeq:1 CANCEL
Max-Forwards:70
Request-Disposition:no-fork
Reason:X.int ;reasoncode=0x0000030B;add-info=0109.0001.0004


<--- Transmitting SIP response (463 bytes) to UDP:212.210.2.2:5060 --->
SIP/2.0 481 Call/Transaction Does Not Exist
Via: SIP/2.0/UDP SIPMSS1-SIP.MGWS.LV:5060;received=212.210.2.2;branch=z9hG4bK8ifC7_b4e9hUYbY6;yop=00.00.AD31DA87.0000.700B
Call-ID: A929277ED3F151588CD91617@0b70ffffffff
From: "+37167366321" <sip:+37167366321@SIPMSS1-SIP.MGWS.LV;user=phone>;tag=D9XaDgW7j3Xd7045
To: <sip:963760960355@media.syploop.net;user=phone>;tag=z9hG4bK8ifC7_b4e9hUYbY6
CSeq: 1 CANCEL
Server: Asterisk PBX 13.11.2
Content-Length:  0



<--- Received SIP response (428 bytes) from UDP:212.210.2.2:5060 --->
SIP/2.0 480 Temporarily not available
Content-Length:0
From:"+37167366321" <sip:+37167366321@10.40.6.100>;tag=16cc0afc-1482-4e4b-98ec-4c5033943d9d
To:<sip:963860960355@212.210.2.2>;tag=ZCA13_hZgDjXYe65
Via:SIP/2.0/UDP 10.40.6.100:5060;rport;branch=z9hG4bKPj9c6fb510-03cb-4071-9e41-d1114451390b
Call-ID:7e00a784-3eac-4e35-9aab-eda898d9cb46
CSeq:8080 INVITE
Reason:X.int ;reasoncode=0x00000006;add-info=01CA.FFFE.0000


<--- Transmitting SIP request (433 bytes) to UDP:212.210.2.2:5060 --->
ACK sip:963860960355@212.210.2.2:5060 SIP/2.0
Via: SIP/2.0/UDP 10.40.6.100:5060;rport;branch=z9hG4bKPj9c6fb510-03cb-4071-9e41-d1114451390b
From: "+37167366321" <sip:+37167366321@10.40.6.100>;tag=16cc0afc-1482-4e4b-98ec-4c5033943d9d
To: <sip:963860960355@212.210.2.2>;tag=ZCA13_hZgDjXYe65
Call-ID: 7e00a784-3eac-4e35-9aab-eda898d9cb46
CSeq: 8080 ACK
Max-Forwards: 70
User-Agent: Asterisk PBX 13.11.2
Content-Length:  0


    -- No one is available to answer at this time (1:0/0/0)
    -- Auto fallthrough, channel 'PJSIP/smex-1-00000006' status is 'NOANSWER'
<--- Transmitting SIP response (826 bytes) to UDP:212.210.2.2:5060 --->
SIP/2.0 480 Temporarily Unavailable
Via: SIP/2.0/UDP SIPMSS1-SIP.MGWS.LV:5060;rport=5060;received=212.210.2.2;branch=z9hG4bK8ifC7_b4e9hUYbY6;yop=00.00.AD31DA87.0000.700B
Record-Route: <sip:AAQACCzlwAAAnAAAAMgAAfGkK@212.210.2.2:5060;transport=UDP;lr;yop=00.00.AD31DA87.0000.700B>
Call-ID: A929277ED3F151588CD91617@0b70ffffffff
From: "+37167366321" <sip:+37167366321@SIPMSS1-SIP.MGWS.LV;user=phone>;tag=D9XaDgW7j3Xd7045
To: <sip:963760960355@media.syploop.net;user=phone>;tag=1299bfb1-4d0a-422a-89c9-972f57dadfca
CSeq: 1 INVITE
Server: Asterisk PBX 13.11.2
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, MESSAGE, REFER
Reason: Q.850;cause=19
P-Asserted-Identity: <sip:963760960355@media.syploop.net;user=phone>
Server: Asterisk PBX 13.11.2
Content-Length:  0


<--- Received SIP request (444 bytes) from UDP:212.210.2.2:5060 --->
ACK sip:963760960355@media.syploop.net;user=phone SIP/2.0
Content-Length:0
From:"+37167366321" <sip:+37167366321@SIPMSS1-SIP.MGWS.LV;user=phone>;tag=D9XaDgW7j3Xd7045
To:<sip:963760960355@media.syploop.net;user=phone>;tag=1299bfb1-4d0a-422a-89c9-972f57dadfca
Via:SIP/2.0/UDP SIPMSS1-SIP.MGWS.LV:5060;branch=z9hG4bK8ifC7_b4e9hUYbY6;yop=00.00.AD31DA87.0000.700B
Call-ID:A929277ED3F151588CD91617@0b70ffffffff
CSeq:1 ACK
Max-Forwards:70


<--- Transmitting SIP request (424 bytes) to UDP:212.210.2.2:5060 --->
OPTIONS sip:212.210.2.2:5060 SIP/2.0
Via: SIP/2.0/UDP 10.40.6.100:5060;rport;branch=z9hG4bKPjb71a3a72-1be7-4c88-88ea-dc41bc95c682
From: <sip:smex-1@10.40.6.100>;tag=99bbbe98-ca27-4d5d-9f1e-dc67e50af2f2
To: <sip:212.210.2.2>
Contact: <sip:smex-1@10.40.6.100:5060>
Call-ID: f781dd4b-3a71-4809-a167-044bc1ccc4a8
CSeq: 26954 OPTIONS
Max-Forwards: 70
User-Agent: Asterisk PBX 13.11.2
Content-Length:  0


<--- Received SIP response (306 bytes) from UDP:212.210.2.2:5060 --->
SIP/2.0 200 OK
Content-Length:0
From:<sip:smex-1@10.40.6.100>;tag=99bbbe98-ca27-4d5d-9f1e-dc67e50af2f2
To:<sip:212.210.2.2>;tag=dummy
Via:SIP/2.0/UDP 10.40.6.100:5060;rport;branch=z9hG4bKPjb71a3a72-1be7-4c88-88ea-dc41bc95c682
Call-ID:f781dd4b-3a71-4809-a167-044bc1ccc4a8
CSeq:26954 OPTIONS


<--- Transmitting SIP request (423 bytes) to UDP:212.210.2.2:5060 --->
OPTIONS sip:212.210.2.2:5060 SIP/2.0
Via: SIP/2.0/UDP 10.40.6.100:5060;rport;branch=z9hG4bKPj0d4f379e-3cf0-4dee-8d47-18a449c0526b
From: <sip:smex-1@10.40.6.100>;tag=d1b64097-864a-4ef9-88a6-77347e674a25
To: <sip:212.210.2.2>
Contact: <sip:smex-1@10.40.6.100:5060>
Call-ID: 46e43bff-2259-455b-9190-6b7c774481bf
CSeq: 2002 OPTIONS
Max-Forwards: 70
User-Agent: Asterisk PBX 13.11.2
Content-Length:  0


<--- Received SIP response (305 bytes) from UDP:212.210.2.2:5060 --->
SIP/2.0 200 OK
Content-Length:0
From:<sip:smex-1@10.40.6.100>;tag=d1b64097-864a-4ef9-88a6-77347e674a25
To:<sip:212.210.2.2>;tag=dummy
Via:SIP/2.0/UDP 10.40.6.100:5060;rport;branch=z9hG4bKPj0d4f379e-3cf0-4dee-8d47-18a449c0526b
Call-ID:46e43bff-2259-455b-9190-6b7c774481bf
CSeq:2002 OPTIONS


SMEX-MEDIA-A1*CLI>