<------------>
    -- Executing [1471@default:1] Wait("SIP/1471-00000000", "0.5") in new stack
    -- Executing [1471@default:2] Answer("SIP/1471-00000000", "") in new stack
Audio is at 17444
Adding codec ulaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (NAT) to 192.168.36.250:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.36.250:5060;branch=z9hG4bKdfb2723398fb4b647cac371c932100f0;received=192.168.36.250;rport=5060
Via: SIP/2.0/UDP 64.94.196.46:5060;branch=z9hG4bK1015b410e5b27a21cdf1c4df400b19c8;received=64.94.196.46
Record-Route: <sip:192.168.36.250>
From: "MCKEMIE CHRISTI" <sip:9199467059@192.168.36.250;user=phone>;tag=11f45953fef99def7
To: 1471 <sip:1471@192.168.36.250>;tag=as01f3bbfc
Call-ID: 8000016c15953fef99df03@192.168.36.250
CSeq: 9 INVITE
Server: Asterisk PBX 14.3.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:1471@192.168.36.249:5060>
Content-Type: application/sdp
Content-Length: 255

v=0
o=root 1158827916 1158827916 IN IP4 192.168.36.249
s=Asterisk PBX 14.3.0
c=IN IP4 192.168.36.249
t=0 0
m=audio 17444 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

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

<--- SIP read from UDP:192.168.36.250:5060 --->
ACK sip:1471@192.168.36.249:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.36.250:5060;branch=z9hG4bKaaf44184e2c868eec3409fc3a13fecc2
Via: SIP/2.0/UDP 64.94.196.46:5060;branch=z9hG4bK6fafd39c1e726799d68e7cfb1692aaff;received=64.94.196.46
Max-Forwards: 70
Record-Route: <sip:192.168.36.250>
From: "MCKEMIE CHRISTI" <sip:9199467059@192.168.36.250;user=phone>;tag=11f45953fef99def7
To: 1471 <sip:1471@192.168.36.250>;tag=as01f3bbfc
Call-ID: 8000016c15953fef99df03@192.168.36.250
CSeq: 9 ACK
Content-Length: 0

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

<--- SIP read from UDP:192.168.36.250:5060 --->
INVITE sip:1471@192.168.36.249:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.36.250:5060;branch=z9hG4bKeece4d6b1e03ed99a4fd8cb63f78b3eb
Via: SIP/2.0/UDP 64.94.196.46:5060;branch=z9hG4bK6ec2fa8f2c0cd9a7d97cd57c0c80cfd8;received=64.94.196.46
Max-Forwards: 70
Record-Route: <sip:192.168.36.250>
To: 1471 <sip:1471@192.168.36.250>;tag=as01f3bbfc
From: "MCKEMIE CHRISTI" <sip:9199467059@192.168.36.250;user=phone>;tag=11f45953fef99def7
Call-ID: 8000016c15953fef99df03@192.168.36.250
CSeq: 10 INVITE
Contact: 9199467059 <sip:9199467059@192.168.36.250;user=phone>
Content-Type: application/sdp
Content-Length: 259
Supported: replaces

v=0
o=9199467059 1310458221 1310458221 IN IP4 64.94.196.46
s=-
c=IN IP4 64.94.196.46
t=0 0
m=audio 22546 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:8 PCMA/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->
--- (13 headers 13 lines) ---
Sending to 192.168.36.250:5060 (NAT)
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 64.94.196.46:22546

<--- Transmitting (NAT) to 192.168.36.250:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.36.250:5060;branch=z9hG4bKeece4d6b1e03ed99a4fd8cb63f78b3eb;received=192.168.36.250;rport=5060
Via: SIP/2.0/UDP 64.94.196.46:5060;branch=z9hG4bK6ec2fa8f2c0cd9a7d97cd57c0c80cfd8;received=64.94.196.46
Record-Route: <sip:192.168.36.250>
From: "MCKEMIE CHRISTI" <sip:9199467059@192.168.36.250;user=phone>;tag=11f45953fef99def7
To: 1471 <sip:1471@192.168.36.250>;tag=as01f3bbfc
Call-ID: 8000016c15953fef99df03@192.168.36.250
CSeq: 10 INVITE
Server: Asterisk PBX 14.3.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:1471@192.168.36.249:5060>
Content-Length: 0


<------------>
Audio is at 17444
Adding codec ulaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (NAT) to 192.168.36.250:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.36.250:5060;branch=z9hG4bKeece4d6b1e03ed99a4fd8cb63f78b3eb;received=192.168.36.250;rport=5060
Via: SIP/2.0/UDP 64.94.196.46:5060;branch=z9hG4bK6ec2fa8f2c0cd9a7d97cd57c0c80cfd8;received=64.94.196.46
Record-Route: <sip:192.168.36.250>
From: "MCKEMIE CHRISTI" <sip:9199467059@192.168.36.250;user=phone>;tag=11f45953fef99def7
To: 1471 <sip:1471@192.168.36.250>;tag=as01f3bbfc
Call-ID: 8000016c15953fef99df03@192.168.36.250
CSeq: 10 INVITE
Server: Asterisk PBX 14.3.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:1471@192.168.36.249:5060>
Content-Type: application/sdp
Content-Length: 255

v=0
o=root 1158827916 1158827917 IN IP4 192.168.36.249
s=Asterisk PBX 14.3.0
c=IN IP4 192.168.36.249
t=0 0
m=audio 17444 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

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

<--- SIP read from UDP:192.168.36.250:5060 --->
OPTIONS sip:1483@192.168.36.249:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.36.250:5060;branch=z9hG4bK4ddeee72e31871abd80c43fd400da51d
Max-Forwards: 70
Record-Route: <sip:192.168.36.250>
To: 1483 <sip:1483@192.168.36.250>
From: 1483 <sip:1483@192.168.36.250>;tag=8b35953fefa3638c
Call-ID: 800000fd65953fefa36394@192.168.36.250
CSeq: 21809 OPTIONS
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Sending to 192.168.36.250:5060 (no NAT)
Looking for 1483 in default (domain 192.168.36.249)

<--- Transmitting (no NAT) to 192.168.36.250:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.36.250:5060;branch=z9hG4bK4ddeee72e31871abd80c43fd400da51d;received=192.168.36.250
Record-Route: <sip:192.168.36.250>
From: 1483 <sip:1483@192.168.36.250>;tag=8b35953fefa3638c
To: 1483 <sip:1483@192.168.36.250>;tag=as09068595
Call-ID: 800000fd65953fefa36394@192.168.36.250
CSeq: 21809 OPTIONS
Server: Asterisk PBX 14.3.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:192.168.36.249:5060>
Accept: application/sdp
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '800000fd65953fefa36394@192.168.36.250' in 32000 ms (Method: OPTIONS)

<--- SIP read from UDP:192.168.36.250:5060 --->
ACK sip:1471@192.168.36.249:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.36.250:5060;branch=z9hG4bK37c7cd8dd35ead7dfb6437391138a960
Via: SIP/2.0/UDP 64.94.196.46:5060;branch=z9hG4bKee2430d75af1ad8cb770b9f386c6d76e;received=64.94.196.46
Max-Forwards: 70
Record-Route: <sip:192.168.36.250>
From: "MCKEMIE CHRISTI" <sip:9199467059@192.168.36.250;user=phone>;tag=11f45953fef99def7
To: 1471 <sip:1471@192.168.36.250>;tag=as01f3bbfc
Call-ID: 8000016c15953fef99df03@192.168.36.250
CSeq: 10 ACK
Content-Length: 0

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

<--- SIP read from UDP:192.168.36.250:5060 --->
INVITE sip:1471@192.168.36.249:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.36.250:5060;branch=z9hG4bK7fbfd12665f31729ed9214ea4431ae67
Via: SIP/2.0/UDP 64.94.196.46:5060;branch=z9hG4bK3cd70e1cd8f1863f37c0f4540609044c;received=64.94.196.46
Max-Forwards: 70
Record-Route: <sip:192.168.36.250>
To: 1471 <sip:1471@192.168.36.250>;tag=as01f3bbfc
From: "MCKEMIE CHRISTI" <sip:9199467059@192.168.36.250;user=phone>;tag=11f45953fef99def7
Call-ID: 8000016c15953fef99df03@192.168.36.250
CSeq: 11 INVITE
Contact: 9199467059 <sip:9199467059@192.168.36.250;user=phone>
Content-Type: application/sdp
Content-Length: 259
Supported: replaces

v=0
o=9199467059 1310458254 1310458254 IN IP4 64.94.196.46
s=-
c=IN IP4 64.94.196.46
t=0 0
m=audio 22546 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:8 PCMA/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->
--- (13 headers 13 lines) ---
Sending to 192.168.36.250:5060 (NAT)
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 64.94.196.46:22546

<--- Transmitting (NAT) to 192.168.36.250:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.36.250:5060;branch=z9hG4bK7fbfd12665f31729ed9214ea4431ae67;received=192.168.36.250;rport=5060
Via: SIP/2.0/UDP 64.94.196.46:5060;branch=z9hG4bK3cd70e1cd8f1863f37c0f4540609044c;received=64.94.196.46
Record-Route: <sip:192.168.36.250>
From: "MCKEMIE CHRISTI" <sip:9199467059@192.168.36.250;user=phone>;tag=11f45953fef99def7
To: 1471 <sip:1471@192.168.36.250>;tag=as01f3bbfc
Call-ID: 8000016c15953fef99df03@192.168.36.250
CSeq: 11 INVITE
Server: Asterisk PBX 14.3.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:1471@192.168.36.249:5060>
Content-Length: 0


<------------>
Audio is at 17444
Adding codec ulaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (NAT) to 192.168.36.250:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.36.250:5060;branch=z9hG4bK7fbfd12665f31729ed9214ea4431ae67;received=192.168.36.250;rport=5060
Via: SIP/2.0/UDP 64.94.196.46:5060;branch=z9hG4bK3cd70e1cd8f1863f37c0f4540609044c;received=64.94.196.46
Record-Route: <sip:192.168.36.250>
From: "MCKEMIE CHRISTI" <sip:9199467059@192.168.36.250;user=phone>;tag=11f45953fef99def7
To: 1471 <sip:1471@192.168.36.250>;tag=as01f3bbfc
Call-ID: 8000016c15953fef99df03@192.168.36.250
CSeq: 11 INVITE
Server: Asterisk PBX 14.3.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:1471@192.168.36.249:5060>
Content-Type: application/sdp
Content-Length: 255

v=0
o=root 1158827916 1158827918 IN IP4 192.168.36.249
s=Asterisk PBX 14.3.0
c=IN IP4 192.168.36.249
t=0 0
m=audio 17444 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

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

<--- SIP read from UDP:192.168.36.250:5060 --->
ACK sip:1471@192.168.36.249:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.36.250:5060;branch=z9hG4bK50457fd1736a463311e2198d66f2e837
Via: SIP/2.0/UDP 64.94.196.46:5060;branch=z9hG4bK87d0a9b62ed18cafa05556e26dac96f7;received=64.94.196.46
Max-Forwards: 70
Record-Route: <sip:192.168.36.250>
From: "MCKEMIE CHRISTI" <sip:9199467059@192.168.36.250;user=phone>;tag=11f45953fef99def7
To: 1471 <sip:1471@192.168.36.250>;tag=as01f3bbfc
Call-ID: 8000016c15953fef99df03@192.168.36.250
CSeq: 11 ACK
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
    -- Executing [1471@default:3] AGI("SIP/1471-00000000", "callComlinktest.agi,1") in new stack
    -- Launched AGI Script /var/lib/asterisk/agi-bin/callComlinktest.agi
    -- AGI Script Executing Application: (NoOp) Options: (1471)
    -- AGI Script Executing Application: (NoOp) Options: (1471)

<--- SIP read from UDP:192.168.36.250:5060 --->
OPTIONS sip:1476@192.168.36.249:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.36.250:5060;branch=z9hG4bKcf751733146cf2a9baff99888daa8ef1
Max-Forwards: 70
Record-Route: <sip:192.168.36.250>
To: 1476 <sip:1476@192.168.36.250>
From: 1476 <sip:1476@192.168.36.250>;tag=8b75953fefb35e0c
Call-ID: 800000f665953fefb35e18@192.168.36.250
CSeq: 21823 OPTIONS
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Sending to 192.168.36.250:5060 (no NAT)
Looking for 1476 in default (domain 192.168.36.249)

<--- Transmitting (no NAT) to 192.168.36.250:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.36.250:5060;branch=z9hG4bKcf751733146cf2a9baff99888daa8ef1;received=192.168.36.250
Record-Route: <sip:192.168.36.250>
From: 1476 <sip:1476@192.168.36.250>;tag=8b75953fefb35e0c
To: 1476 <sip:1476@192.168.36.250>;tag=as6ff25c74
Call-ID: 800000f665953fefb35e18@192.168.36.250
CSeq: 21823 OPTIONS
Server: Asterisk PBX 14.3.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:192.168.36.249:5060>
Accept: application/sdp
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '800000f665953fefb35e18@192.168.36.250' in 32000 ms (Method: OPTIONS)
    -- AGI Script Executing Application: (NoOp) Options: (Start)
    -- AGI Script Executing Application: (NoOp) Options: (T/F=TRUE)
    -- AGI Script Executing Application: (NoOp) Options: (*)
    -- AGI Script Executing Application: (NoOp) Options: (5)
    -- AGI Script Executing Application: (NoOp) Options: (T/F=TRUE)
    -- AGI Script Executing Application: (NoOp) Options: (*)
    -- AGI Script Executing Application: (NoOp) Options: (T/F=TRUE)
    -- AGI Script Executing Application: (NoOp) Options: (*)
    -- AGI Script Executing Application: (NoOp) Options: (T/F=TRUE)
    -- AGI Script Executing Application: (NoOp) Options: (*)
    -- AGI Script Executing Application: (NoOp) Options: (T/F=TRUE)
    -- AGI Script Executing Application: (NoOp) Options: (*)
    -- AGI Script Executing Application: (NoOp) Options: (T/F=TRUE)
    -- AGI Script Executing Application: (NoOp) Options: (*)
    -- AGI Script Executing Application: (NoOp) Options: (T/F=TRUE)
    -- AGI Script Executing Application: (NoOp) Options: (*)
    -- AGI Script Executing Application: (NoOp) Options: (T/F=TRUE)
    -- AGI Script Executing Application: (NoOp) Options: (*)
    -- AGI Script Executing Application: (NoOp) Options: (T/F=TRUE)
    -- AGI Script Executing Application: (NoOp) Options: (*)
    -- AGI Script Executing Application: (NoOp) Options: (1)
    -- AGI Script Executing Application: (NoOp) Options: (1332)
    -- AGI Script Executing Application: (NoOp) Options: (1340)
    -- AGI Script Executing Application: (NoOp) Options: (1344)
    -- AGI Script Executing Application: (NoOp) Options: (1348)
    -- AGI Script Executing Application: (NoOp) Options: (1357)
    -- AGI Script Executing Application: (read) Options: (storedAnswer,ipiwelcome&membprov,1,n,1,5)
    -- Accepting a maximum of 1 digits.
    -- <SIP/1471-00000000> Playing 'ipiwelcome.ulaw' (language 'en')
       > 0x7fc87805e9a0 -- Probation passed - setting RTP source address to 64.94.196.46:22546

<--- SIP read from UDP:192.168.36.250:5060 --->
OPTIONS sip:1475@192.168.36.249:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.36.250:5060;branch=z9hG4bK36364f170e0e4e5df66675a86d248b80
Max-Forwards: 70
Record-Route: <sip:192.168.36.250>
To: 1475 <sip:1475@192.168.36.250>
From: 1475 <sip:1475@192.168.36.250>;tag=8b55953fefc3536c
Call-ID: 800000f565953fefc35379@192.168.36.250
CSeq: 21793 OPTIONS
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Sending to 192.168.36.250:5060 (no NAT)
Looking for 1475 in default (domain 192.168.36.249)

<--- Transmitting (no NAT) to 192.168.36.250:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.36.250:5060;branch=z9hG4bK36364f170e0e4e5df66675a86d248b80;received=192.168.36.250
Record-Route: <sip:192.168.36.250>
From: 1475 <sip:1475@192.168.36.250>;tag=8b55953fefc3536c
To: 1475 <sip:1475@192.168.36.250>;tag=as19c5c805
Call-ID: 800000f565953fefc35379@192.168.36.250
CSeq: 21793 OPTIONS
Server: Asterisk PBX 14.3.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:192.168.36.249:5060>
Accept: application/sdp
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '800000f565953fefc35379@192.168.36.250' in 32000 ms (Method: OPTIONS)

<--- SIP read from UDP:192.168.36.250:5060 --->
OPTIONS sip:1486@192.168.36.249:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.36.250:5060;branch=z9hG4bKe47fb0cf6dbd8407eb86b32e3430f71c
Max-Forwards: 70
Record-Route: <sip:192.168.36.250>
To: 1486 <sip:1486@192.168.36.250>
From: 1486 <sip:1486@192.168.36.250>;tag=8b95953fefd37303
Call-ID: 8000010065953fefd3730f@192.168.36.250
CSeq: 21807 OPTIONS
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Sending to 192.168.36.250:5060 (no NAT)
Looking for 1486 in default (domain 192.168.36.249)

<--- Transmitting (no NAT) to 192.168.36.250:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.36.250:5060;branch=z9hG4bKe47fb0cf6dbd8407eb86b32e3430f71c;received=192.168.36.250
Record-Route: <sip:192.168.36.250>
From: 1486 <sip:1486@192.168.36.250>;tag=8b95953fefd37303
To: 1486 <sip:1486@192.168.36.250>;tag=as3b693495
Call-ID: 8000010065953fefd3730f@192.168.36.250
CSeq: 21807 OPTIONS
Server: Asterisk PBX 14.3.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:192.168.36.249:5060>
Accept: application/sdp
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '8000010065953fefd3730f@192.168.36.250' in 32000 ms (Method: OPTIONS)

<--- SIP read from UDP:192.168.36.250:5060 --->
OPTIONS sip:1473@192.168.36.249:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.36.250:5060;branch=z9hG4bKd8562b973d994bdf46aa3f3678bbde55
Max-Forwards: 70
Record-Route: <sip:192.168.36.250>
To: 1473 <sip:1473@192.168.36.250>
From: 1473 <sip:1473@192.168.36.250>;tag=8b15953fefe368a8
Call-ID: 800000f365953fefe368b5@192.168.36.250
CSeq: 21827 OPTIONS
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Sending to 192.168.36.250:5060 (no NAT)
Looking for 1473 in default (domain 192.168.36.249)

<--- Transmitting (no NAT) to 192.168.36.250:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.36.250:5060;branch=z9hG4bKd8562b973d994bdf46aa3f3678bbde55;received=192.168.36.250
Record-Route: <sip:192.168.36.250>
From: 1473 <sip:1473@192.168.36.250>;tag=8b15953fefe368a8
To: 1473 <sip:1473@192.168.36.250>;tag=as68451648
Call-ID: 800000f365953fefe368b5@192.168.36.250
CSeq: 21827 OPTIONS
Server: Asterisk PBX 14.3.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:192.168.36.249:5060>
Accept: application/sdp
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '800000f365953fefe368b5@192.168.36.250' in 32000 ms (Method: OPTIONS)

<--- SIP read from UDP:192.168.36.250:5060 --->
OPTIONS sip:1477@192.168.36.249:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.36.250:5060;branch=z9hG4bK0cc6b62a63efacd465eb04edf8e75269
Max-Forwards: 70
Record-Route: <sip:192.168.36.250>
To: 1477 <sip:1477@192.168.36.250>
From: 1477 <sip:1477@192.168.36.250>;tag=8b95953feff3841a
Call-ID: 800000f765953feff38426@192.168.36.250
CSeq: 6995 OPTIONS
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Sending to 192.168.36.250:5060 (no NAT)
Looking for 1477 in default (domain 192.168.36.249)

<--- Transmitting (no NAT) to 192.168.36.250:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.36.250:5060;branch=z9hG4bK0cc6b62a63efacd465eb04edf8e75269;received=192.168.36.250
Record-Route: <sip:192.168.36.250>
From: 1477 <sip:1477@192.168.36.250>;tag=8b95953feff3841a
To: 1477 <sip:1477@192.168.36.250>;tag=as6a941f05
Call-ID: 800000f765953feff38426@192.168.36.250
CSeq: 6995 OPTIONS
Server: Asterisk PBX 14.3.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:192.168.36.249:5060>
Accept: application/sdp
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '800000f765953feff38426@192.168.36.250' in 32000 ms (Method: OPTIONS)

<--- SIP read from UDP:192.168.36.250:5060 --->
OPTIONS sip:1474@192.168.36.249:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.36.250:5060;branch=z9hG4bK3d58b464d0f7bc8bce90852a758c210a
Max-Forwards: 70
Record-Route: <sip:192.168.36.250>
To: 1474 <sip:1474@192.168.36.250>
From: 1474 <sip:1474@192.168.36.250>;tag=8b35953ff0037924
Call-ID: 800000f465953ff0037930@192.168.36.250
CSeq: 21813 OPTIONS
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Sending to 192.168.36.250:5060 (no NAT)
Looking for 1474 in default (domain 192.168.36.249)

<--- Transmitting (no NAT) to 192.168.36.250:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.36.250:5060;branch=z9hG4bK3d58b464d0f7bc8bce90852a758c210a;received=192.168.36.250
Record-Route: <sip:192.168.36.250>
From: 1474 <sip:1474@192.168.36.250>;tag=8b35953ff0037924
To: 1474 <sip:1474@192.168.36.250>;tag=as0d76010b
Call-ID: 800000f465953ff0037930@192.168.36.250
CSeq: 21813 OPTIONS
Server: Asterisk PBX 14.3.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:192.168.36.249:5060>
Accept: application/sdp
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '800000f465953ff0037930@192.168.36.250' in 32000 ms (Method: OPTIONS)

<--- SIP read from UDP:192.168.36.250:5060 --->
OPTIONS sip:1472@192.168.36.249:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.36.250:5060;branch=z9hG4bKde8c97c99d0f813d9574b38c0621dab9
Max-Forwards: 70
Record-Route: <sip:192.168.36.250>
To: 1472 <sip:1472@192.168.36.250>
From: 1472 <sip:1472@192.168.36.250>;tag=8af5953ff0136f39
Call-ID: 800000f265953ff0136f45@192.168.36.250
CSeq: 21793 OPTIONS
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Sending to 192.168.36.250:5060 (no NAT)
Looking for 1472 in default (domain 192.168.36.249)

<--- Transmitting (no NAT) to 192.168.36.250:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.36.250:5060;branch=z9hG4bKde8c97c99d0f813d9574b38c0621dab9;received=192.168.36.250
Record-Route: <sip:192.168.36.250>
From: 1472 <sip:1472@192.168.36.250>;tag=8af5953ff0136f39
To: 1472 <sip:1472@192.168.36.250>;tag=as69cbc6a6
Call-ID: 800000f265953ff0136f45@192.168.36.250
CSeq: 21793 OPTIONS
Server: Asterisk PBX 14.3.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:192.168.36.249:5060>
Accept: application/sdp
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '800000f265953ff0136f45@192.168.36.250' in 32000 ms (Method: OPTIONS)
    -- User entered '8'
    -- AGI Script Executing Application: (NoOp) Options: (break)
    -- AGI Script Executing Application: (NoOp) Options: (read)
    -- AGI Script Executing Application: (NoOp) Options: (8)
    -- AGI Script Executing Application: (NoOp) Options: (QUEF)
    -- AGI Script Executing Application: (NoOp) Options: (QUEF)
    -- AGI Script Executing Application: (NoOp) Options: (T/F=TRUE)
    -- AGI Script Executing Application: (NoOp) Options: (*)
    -- AGI Script Executing Application: (NoOp) Options: (LOOK4)
    -- AGI Script Executing Application: (NoOp) Options: (T/F=TRUE)
    -- AGI Script Executing Application: (NoOp) Options: (*)
    -- AGI Script Executing Application: (NoOp) Options: (conclusion:)
    -- AGI Script Executing Application: (NoOp) Options: (-JUMP2JEXT)
    -- AGI Script Executing Application: (NoOp) Options: (T/F=TRUE)
    -- AGI Script Executing Application: (NoOp) Options: (*)
    -- AGI Script Executing Application: (NoOp) Options: (T/F=TRUE)
    -- AGI Script Executing Application: (NoOp) Options: (*)
    -- AGI Script Executing Application: (NoOp) Options: (T/F=TRUE)
    -- AGI Script Executing Application: (NoOp) Options: (*)
    -- AGI Script Executing Application: (NoOp) Options: (T/F=TRUE)
    -- AGI Script Executing Application: (NoOp) Options: (*)
    -- AGI Script Executing Application: (NoOp) Options: (T/F=TRUE)
    -- AGI Script Executing Application: (NoOp) Options: (*)
    -- AGI Script Executing Application: (NoOp) Options: (T/F=TRUE)
    -- AGI Script Executing Application: (NoOp) Options: (*)
    -- AGI Script Executing Application: (NoOp) Options: (1)
    -- AGI Script Executing Application: (NoOp) Options: (1332)
    -- AGI Script Executing Application: (NoOp) Options: (1340)
    -- AGI Script Executing Application: (NoOp) Options: (1344)
    -- AGI Script Executing Application: (NoOp) Options: (1348)
    -- AGI Script Executing Application: (NoOp) Options: (1357)
    -- AGI Script Executing Application: (read) Options: (storedAnswer,enterextn,4,n,1,5)
    -- Accepting a maximum of 4 digits.
    -- <SIP/1471-00000000> Playing 'enterextn.ulaw' (language 'en')

<--- SIP read from UDP:192.168.36.250:5060 --->
OPTIONS sip:1471@192.168.36.249:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.36.250:5060;branch=z9hG4bKeab44d5b1bba65efb1ac1ed4ab85965a
Max-Forwards: 70
Record-Route: <sip:192.168.36.250>
To: 1471 <sip:1471@192.168.36.250>
From: 1471 <sip:1471@192.168.36.250>;tag=8ad5953ff0238cd7
Call-ID: 8000016c65953ff0238cde@192.168.36.250
CSeq: 12 OPTIONS
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Sending to 192.168.36.250:5060 (no NAT)
Looking for 1471 in default (domain 192.168.36.249)

<--- Transmitting (no NAT) to 192.168.36.250:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.36.250:5060;branch=z9hG4bKeab44d5b1bba65efb1ac1ed4ab85965a;received=192.168.36.250
Record-Route: <sip:192.168.36.250>
From: 1471 <sip:1471@192.168.36.250>;tag=8ad5953ff0238cd7
To: 1471 <sip:1471@192.168.36.250>;tag=as600e243d
Call-ID: 8000016c65953ff0238cde@192.168.36.250
CSeq: 12 OPTIONS
Server: Asterisk PBX 14.3.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:192.168.36.249:5060>
Accept: application/sdp
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '8000016c65953ff0238cde@192.168.36.250' in 32000 ms (Method: OPTIONS)
Really destroying SIP dialog '800000f665953fee32dfc4@192.168.36.250' Method: OPTIONS
Really destroying SIP dialog '8000016cc5953fee32eaf4@192.168.36.250' Method: NOTIFY
Really destroying SIP dialog '800000f2c5953fee32f08e@192.168.36.250' Method: NOTIFY
Really destroying SIP dialog '800000f4c5953fee32f5dc@192.168.36.250' Method: NOTIFY
Really destroying SIP dialog '800000f7c5953fee32fb4c@192.168.36.250' Method: NOTIFY
Really destroying SIP dialog '800000f3c5953fee3300b6@192.168.36.250' Method: NOTIFY
Really destroying SIP dialog '80000100c5953fee3305f4@192.168.36.250' Method: NOTIFY
Really destroying SIP dialog '800000f5c5953fee330b85@192.168.36.250' Method: NOTIFY
Really destroying SIP dialog '800000f6c5953fee3310c4@192.168.36.250' Method: NOTIFY
Really destroying SIP dialog '800000fdc5953fee331609@192.168.36.250' Method: NOTIFY
Really destroying SIP dialog '800000fec5953fee331b7b@192.168.36.250' Method: NOTIFY
Really destroying SIP dialog '800000f8c5953fee3320c3@192.168.36.250' Method: NOTIFY
Really destroying SIP dialog '800000f9c5953fee3325ff@192.168.36.250' Method: NOTIFY
Really destroying SIP dialog '800000fbc5953fee332b43@192.168.36.250' Method: NOTIFY
Really destroying SIP dialog '800000fac5953fee333150@192.168.36.250' Method: NOTIFY
Really destroying SIP dialog '800000ffc5953fee333683@192.168.36.250' Method: NOTIFY
Really destroying SIP dialog '800000fcc5953fee333bd4@192.168.36.250' Method: NOTIFY
Really destroying SIP dialog '800000f865953fee42e208@192.168.36.250' Method: OPTIONS
    -- User entered '312'
    -- AGI Script Executing Application: (NoOp) Options: (break)
    -- AGI Script Executing Application: (NoOp) Options: (read)
    -- AGI Script Executing Application: (NoOp) Options: (312)
    -- AGI Script Executing Application: (NoOp) Options: (QUEF)
    -- AGI Script Executing Application: (NoOp) Options: (QUEF)
    -- AGI Script Executing Application: (NoOp) Options: (T/F=TRUE)
    -- AGI Script Executing Application: (NoOp) Options: (*)
    -- AGI Script Executing Application: (NoOp) Options: (T/F=TRUE)
    -- AGI Script Executing Application: (NoOp) Options: (*)
    -- AGI Script Executing Application: (Dial) Options: (SIP/1471/312)
  == Using SIP RTP CoS mark 5
Audio is at 19372
Adding codec ulaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 192.168.36.250:5060:
INVITE sip:312@192.168.36.250:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.36.249:5060;branch=z9hG4bK6d7af3b0;rport
Max-Forwards: 70
From: "Asterix" <sip:1471@192.168.36.250>;tag=as6f80eda9
To: <sip:312@192.168.36.250:5060>
Contact: <sip:1471@192.168.36.249:5060>
Call-ID: 7520117f5dbf5ca72d7ee7a865f18423@192.168.36.250
CSeq: 102 INVITE
User-Agent: Asterisk PBX 14.3.0
Date: Wed, 28 Jun 2017 19:09:57 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 255

v=0
o=root 1067227258 1067227258 IN IP4 192.168.36.249
s=Asterisk PBX 14.3.0
c=IN IP4 192.168.36.249
t=0 0
m=audio 19372 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

---
    -- Called SIP/1471/312
Really destroying SIP dialog '800000fe65953fee52fd09@192.168.36.250' Method: OPTIONS

<--- SIP read from UDP:192.168.36.250:5060 --->
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.36.249:5060;branch=z9hG4bK6d7af3b0;rport;received=192.168.36.249
Proxy-Authenticate: Digest algorithm=MD5,realm="192.168.36.250",domain="192.168.36.250",qop="auth",nonce="e6dc119c40dd780e57c5e90fabf2ac52",opaque="",stale=FALSE
From: "Asterix" <sip:1471@192.168.36.250>;tag=as6f80eda9
To: <sip:312@192.168.36.250:5060>;tag=7bd5953ff052f37e
Call-ID: 7520117f5dbf5ca72d7ee7a865f18423@192.168.36.250
CSeq: 102 INVITE
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Transmitting (NAT) to 192.168.36.250:5060:
ACK sip:312@192.168.36.250:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.36.249:5060;branch=z9hG4bK6d7af3b0;rport
Max-Forwards: 70
From: "Asterix" <sip:1471@192.168.36.250>;tag=as6f80eda9
To: <sip:312@192.168.36.250:5060>;tag=7bd5953ff052f37e
Contact: <sip:1471@192.168.36.249:5060>
Call-ID: 7520117f5dbf5ca72d7ee7a865f18423@192.168.36.250
CSeq: 102 ACK
User-Agent: Asterisk PBX 14.3.0
Content-Length: 0


---
Audio is at 19372
Adding codec ulaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 192.168.36.250:5060:
INVITE sip:312@192.168.36.250:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.36.249:5060;branch=z9hG4bK14f6d7d0;rport
Max-Forwards: 70
From: "Asterix" <sip:1471@192.168.36.250>;tag=as6f80eda9
To: <sip:312@192.168.36.250:5060>
Contact: <sip:1471@192.168.36.249:5060>
Call-ID: 7520117f5dbf5ca72d7ee7a865f18423@192.168.36.250
CSeq: 103 INVITE
User-Agent: Asterisk PBX 14.3.0
Proxy-Authorization: Digest username="1471", realm="192.168.36.250", algorithm=MD5, uri="sip:192.168.36.250", nonce="e6dc119c40dd780e57c5e90fabf2ac52", response="a0744543f70b560e2aab0e81fe80a272", qop=auth, cnonce="13481fa1", nc=00000001
Date: Wed, 28 Jun 2017 19:09:57 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 255

v=0
o=root 1067227258 1067227259 IN IP4 192.168.36.249
s=Asterisk PBX 14.3.0
c=IN IP4 192.168.36.249
t=0 0
m=audio 19372 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

---

<--- SIP read from UDP:192.168.36.250:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.36.249:5060;branch=z9hG4bK14f6d7d0;rport;received=192.168.36.249
Record-Route: <sip:192.168.36.250>
From: "Asterix" <sip:1471@192.168.36.250>;tag=as6f80eda9
To: <sip:312@192.168.36.250:5060>;tag=7bd5953ff0531cce
Call-ID: 7520117f5dbf5ca72d7ee7a865f18423@192.168.36.250
CSeq: 103 INVITE
Content-Length: 0

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

<--- SIP read from UDP:192.168.36.250:5060 --->
SIP/2.0 480 Temporarily not available
Via: SIP/2.0/UDP 192.168.36.249:5060;branch=z9hG4bK14f6d7d0;rport;received=192.168.36.249
From: "Asterix" <sip:1471@192.168.36.250>;tag=as6f80eda9
To: <sip:312@192.168.36.250:5060>;tag=7bd5953ff0531cce
Call-ID: 7520117f5dbf5ca72d7ee7a865f18423@192.168.36.250
CSeq: 103 INVITE
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---
Transmitting (NAT) to 192.168.36.250:5060:
ACK sip:312@192.168.36.250:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.36.249:5060;branch=z9hG4bK14f6d7d0;rport
Max-Forwards: 70
From: "Asterix" <sip:1471@192.168.36.250>;tag=as6f80eda9
To: <sip:312@192.168.36.250:5060>;tag=7bd5953ff0531cce
Contact: <sip:1471@192.168.36.249:5060>
Call-ID: 7520117f5dbf5ca72d7ee7a865f18423@192.168.36.250
CSeq: 103 ACK
User-Agent: Asterisk PBX 14.3.0
Content-Length: 0


---
    -- SIP/1471-00000001 redirecting info has changed, passing it to SIP/1471-00000000
    -- SIP/1471-00000001 is busy
Scheduling destruction of SIP dialog '7520117f5dbf5ca72d7ee7a865f18423@192.168.36.250' in 32000 ms (Method: INVITE)
  == Everyone is busy/congested at this time (1:1/0/0)
    -- AGI Script Executing Application: (NoOp) Options: (T/F=TRUE)
    -- AGI Script Executing Application: (NoOp) Options: (*)
    -- AGI Script Executing Application: (NoOp) Options: (T/F=TRUE)
    -- AGI Script Executing Application: (NoOp) Options: (*)
    -- AGI Script Executing Application: (NoOp) Options: (conclusion:)
    -- AGI Script Executing Application: (NoOp) Options: (-JUMP2JEXT)
    -- AGI Script Executing Application: (NoOp) Options: (T/F=TRUE)
    -- AGI Script Executing Application: (NoOp) Options: (*)
    -- AGI Script Executing Application: (NoOp) Options: (T/F=TRUE)
    -- AGI Script Executing Application: (NoOp) Options: (*)
    -- AGI Script Executing Application: (NoOp) Options: (T/F=TRUE)
    -- AGI Script Executing Application: (NoOp) Options: (*)
    -- AGI Script Executing Application: (NoOp) Options: (T/F=TRUE)
    -- AGI Script Executing Application: (NoOp) Options: (*)
    -- AGI Script Executing Application: (NoOp) Options: (T/F=TRUE)
    -- AGI Script Executing Application: (NoOp) Options: (*)
    -- AGI Script Executing Application: (NoOp) Options: (T/F=TRUE)
    -- AGI Script Executing Application: (NoOp) Options: (*)
    -- AGI Script Executing Application: (NoOp) Options: (1)
    -- AGI Script Executing Application: (NoOp) Options: (1332)
    -- AGI Script Executing Application: (NoOp) Options: (1340)
    -- AGI Script Executing Application: (NoOp) Options: (1344)
    -- AGI Script Executing Application: (NoOp) Options: (1348)
    -- AGI Script Executing Application: (NoOp) Options: (1357)
    -- AGI Script Executing Application: (read) Options: (storedAnswer,holdtrans&badextn&enterextn,4,n,1,5)
    -- Accepting a maximum of 4 digits.
    -- <SIP/1471-00000000> Playing 'holdtrans.ulaw' (language 'en')
Really destroying SIP dialog '800000fd65953fee62f0e1@192.168.36.250' Method: OPTIONS
Really destroying SIP dialog '800000f665953fee730c5e@192.168.36.250' Method: OPTIONS

<--- SIP read from UDP:192.168.36.250:5060 --->
OPTIONS sip:1482@192.168.36.249:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.36.250:5060;branch=z9hG4bKd86a6350a523344033edcccb93638b21
Max-Forwards: 70
Record-Route: <sip:192.168.36.250>
To: 1482 <sip:1482@192.168.36.250>
From: 1482 <sip:1482@192.168.36.250>;tag=8b15953ff073a0be
Call-ID: 800000fc65953ff073a0cb@192.168.36.250
CSeq: 21816 OPTIONS
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Sending to 192.168.36.250:5060 (no NAT)
Looking for 1482 in default (domain 192.168.36.249)

<--- Transmitting (no NAT) to 192.168.36.250:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.36.250:5060;branch=z9hG4bKd86a6350a523344033edcccb93638b21;received=192.168.36.250
Record-Route: <sip:192.168.36.250>
From: 1482 <sip:1482@192.168.36.250>;tag=8b15953ff073a0be
To: 1482 <sip:1482@192.168.36.250>;tag=as787eb8a6
Call-ID: 800000fc65953ff073a0cb@192.168.36.250
CSeq: 21816 OPTIONS
Server: Asterisk PBX 14.3.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:192.168.36.249:5060>
Accept: application/sdp
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '800000fc65953ff073a0cb@192.168.36.250' in 32000 ms (Method: OPTIONS)
Really destroying SIP dialog '800000f565953fee830178@192.168.36.250' Method: OPTIONS

<--- SIP read from UDP:192.168.36.250:5060 --->
OPTIONS sip:1485@192.168.36.249:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.36.250:5060;branch=z9hG4bK5de5893749e5b26acf531c21a70a3291
Max-Forwards: 70
Record-Route: <sip:192.168.36.250>
To: 1485 <sip:1485@192.168.36.250>
From: 1485 <sip:1485@192.168.36.250>;tag=8b75953ff0839575
Call-ID: 800000ff65953ff0839581@192.168.36.250
CSeq: 21825 OPTIONS
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Sending to 192.168.36.250:5060 (no NAT)
Looking for 1485 in default (domain 192.168.36.249)

<--- Transmitting (no NAT) to 192.168.36.250:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.36.250:5060;branch=z9hG4bK5de5893749e5b26acf531c21a70a3291;received=192.168.36.250
Record-Route: <sip:192.168.36.250>
From: 1485 <sip:1485@192.168.36.250>;tag=8b75953ff0839575
To: 1485 <sip:1485@192.168.36.250>;tag=as0d6efb28
Call-ID: 800000ff65953ff0839581@192.168.36.250
CSeq: 21825 OPTIONS
Server: Asterisk PBX 14.3.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:192.168.36.249:5060>
Accept: application/sdp
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '800000ff65953ff0839581@192.168.36.250' in 32000 ms (Method: OPTIONS)
    -- <SIP/1471-00000000> Playing 'badextn.ulaw' (language 'en')
Really destroying SIP dialog '8000010065953fee92f6f4@192.168.36.250' Method: OPTIONS

<--- SIP read from UDP:192.168.36.250:5060 --->
OPTIONS sip:1480@192.168.36.249:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.36.250:5060;branch=z9hG4bK8ba12fd0856c5356f19518138de63311
Max-Forwards: 70
Record-Route: <sip:192.168.36.250>
To: 1480 <sip:1480@192.168.36.250>
From: 1480 <sip:1480@192.168.36.250>;tag=8ad5953ff0939cec
Call-ID: 800000fa65953ff0939cf8@192.168.36.250
CSeq: 21849 OPTIONS
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Sending to 192.168.36.250:5060 (no NAT)
Looking for 1480 in default (domain 192.168.36.249)

<--- Transmitting (no NAT) to 192.168.36.250:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.36.250:5060;branch=z9hG4bK8ba12fd0856c5356f19518138de63311;received=192.168.36.250
Record-Route: <sip:192.168.36.250>
From: 1480 <sip:1480@192.168.36.250>;tag=8ad5953ff0939cec
To: 1480 <sip:1480@192.168.36.250>;tag=as6dca5372
Call-ID: 800000fa65953ff0939cf8@192.168.36.250
CSeq: 21849 OPTIONS
Server: Asterisk PBX 14.3.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:192.168.36.249:5060>
Accept: application/sdp
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '800000fa65953ff0939cf8@192.168.36.250' in 32000 ms (Method: OPTIONS)
Really destroying SIP dialog '800000f365953feea313a3@192.168.36.250' Method: OPTIONS

<--- SIP read from UDP:192.168.36.250:5060 --->
OPTIONS sip:1481@192.168.36.249:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.36.250:5060;branch=z9hG4bKa7d131f7a640f573b89095cb422b1882
Max-Forwards: 70
Record-Route: <sip:192.168.36.250>
To: 1481 <sip:1481@192.168.36.250>
From: 1481 <sip:1481@192.168.36.250>;tag=8af5953ff0a3b8c8
Call-ID: 800000fb65953ff0a3b8d5@192.168.36.250
CSeq: 21844 OPTIONS
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Sending to 192.168.36.250:5060 (no NAT)
Looking for 1481 in default (domain 192.168.36.249)

<--- Transmitting (no NAT) to 192.168.36.250:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.36.250:5060;branch=z9hG4bKa7d131f7a640f573b89095cb422b1882;received=192.168.36.250
Record-Route: <sip:192.168.36.250>
From: 1481 <sip:1481@192.168.36.250>;tag=8af5953ff0a3b8c8
To: 1481 <sip:1481@192.168.36.250>;tag=as2cd8df11
Call-ID: 800000fb65953ff0a3b8d5@192.168.36.250
CSeq: 21844 OPTIONS
Server: Asterisk PBX 14.3.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:192.168.36.249:5060>
Accept: application/sdp
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '800000fb65953ff0a3b8d5@192.168.36.250' in 32000 ms (Method: OPTIONS)

<--- SIP read from UDP:192.168.36.250:5060 --->
BYE sip:1471@192.168.36.249:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.36.250:5060;branch=z9hG4bK66bc41162a45f28a390aad666de06c18
Via: SIP/2.0/UDP 64.94.196.46:5060;branch=z9hG4bKea837f9e142e96fb02b317d5e5b24db4;received=64.94.196.46
Max-Forwards: 70
Record-Route: <sip:192.168.36.250>
To: 1471 <sip:1471@192.168.36.250>;tag=as01f3bbfc
From: "MCKEMIE CHRISTI" <sip:9199467059@192.168.36.250;user=phone>;tag=11f45953fef99def7
Call-ID: 8000016c15953fef99df03@192.168.36.250
CSeq: 13 BYE
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Sending to 192.168.36.250:5060 (NAT)
Scheduling destruction of SIP dialog '8000016c15953fef99df03@192.168.36.250' in 32000 ms (Method: BYE)

<--- Transmitting (NAT) to 192.168.36.250:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.36.250:5060;branch=z9hG4bK66bc41162a45f28a390aad666de06c18;received=192.168.36.250;rport=5060
Via: SIP/2.0/UDP 64.94.196.46:5060;branch=z9hG4bKea837f9e142e96fb02b317d5e5b24db4;received=64.94.196.46
Record-Route: <sip:192.168.36.250>
From: "MCKEMIE CHRISTI" <sip:9199467059@192.168.36.250;user=phone>;tag=11f45953fef99def7
To: 1471 <sip:1471@192.168.36.250>;tag=as01f3bbfc
Call-ID: 8000016c15953fef99df03@192.168.36.250
CSeq: 13 BYE
Server: Asterisk PBX 14.3.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


<------------>
    -- <SIP/1471-00000000> Playing 'enterextn.ulaw' (language 'en')
    -- User disconnected
    -- <SIP/1471-00000000>AGI Script callComlinktest.agi completed, returning 4
  == Spawn extension (default, 1471, 3) exited non-zero on 'SIP/1471-00000000'
Really destroying SIP dialog '800000f765953feeb30ac0@192.168.36.250' Method: OPTIONS

<--- SIP read from UDP:192.168.36.250:5060 --->
OPTIONS sip:1478@192.168.36.249:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.36.250:5060;branch=z9hG4bKca6c9e628c5f2f52d01d3aa6b48d16c3
Max-Forwards: 70
Record-Route: <sip:192.168.36.250>
To: 1479 <sip:1479@192.168.36.250>
From: 1479 <sip:1479@192.168.36.250>;tag=8bd5953ff0b3afe1
Call-ID: 800000f965953ff0b3afed@192.168.36.250
CSeq: 21839 OPTIONS
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Sending to 192.168.36.250:5060 (no NAT)
Looking for 1478 in default (domain 192.168.36.249)

<--- Transmitting (no NAT) to 192.168.36.250:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.36.250:5060;branch=z9hG4bKca6c9e628c5f2f52d01d3aa6b48d16c3;received=192.168.36.250
Record-Route: <sip:192.168.36.250>
From: 1479 <sip:1479@192.168.36.250>;tag=8bd5953ff0b3afe1
To: 1479 <sip:1479@192.168.36.250>;tag=as3319732a
Call-ID: 800000f965953ff0b3afed@192.168.36.250
CSeq: 21839 OPTIONS
Server: Asterisk PBX 14.3.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:192.168.36.249:5060>
Accept: application/sdp
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '800000f965953ff0b3afed@192.168.36.250' in 32000 ms (Method: OPTIONS)
Really destroying SIP dialog '800000f465953feec327d2@192.168.36.250' Method: OPTIONS

<--- SIP read from UDP:192.168.36.250:5060 --->
OPTIONS sip:1478@192.168.36.249:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.36.250:5060;branch=z9hG4bK5df2a2b68aaa29cdc0b9a62e03ed8047
Max-Forwards: 70
Record-Route: <sip:192.168.36.250>
To: 1478 <sip:1478@192.168.36.250>
From: 1478 <sip:1478@192.168.36.250>;tag=8bb5953ff0c3a5a9
Call-ID: 800000f865953ff0c3a5b7@192.168.36.250
CSeq: 21837 OPTIONS
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Sending to 192.168.36.250:5060 (no NAT)
Looking for 1478 in default (domain 192.168.36.249)

<--- Transmitting (no NAT) to 192.168.36.250:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.36.250:5060;branch=z9hG4bK5df2a2b68aaa29cdc0b9a62e03ed8047;received=192.168.36.250
Record-Route: <sip:192.168.36.250>
From: 1478 <sip:1478@192.168.36.250>;tag=8bb5953ff0c3a5a9
To: 1478 <sip:1478@192.168.36.250>;tag=as09e6cbe3
Call-ID: 800000f865953ff0c3a5b7@192.168.36.250
CSeq: 21837 OPTIONS
Server: Asterisk PBX 14.3.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:192.168.36.249:5060>
Accept: application/sdp
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '800000f865953ff0c3a5b7@192.168.36.250' in 32000 ms (Method: OPTIONS)
Really destroying SIP dialog '800000f265953feed31b28@192.168.36.250' Method: OPTIONS

<--- SIP read from UDP:192.168.36.250:5060 --->
OPTIONS sip:1484@192.168.36.249:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.36.250:5060;branch=z9hG4bKb3ac6ff1c55bfc5b74f95828b07f56c8
Max-Forwards: 70
Record-Route: <sip:192.168.36.250>
To: 1484 <sip:1484@192.168.36.250>
From: 1484 <sip:1484@192.168.36.250>;tag=8b55953ff0d3c200
Call-ID: 800000fe65953ff0d3c20d@192.168.36.250
CSeq: 21835 OPTIONS
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Sending to 192.168.36.250:5060 (no NAT)
Looking for 1484 in default (domain 192.168.36.249)

<--- Transmitting (no NAT) to 192.168.36.250:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.36.250:5060;branch=z9hG4bKb3ac6ff1c55bfc5b74f95828b07f56c8;received=192.168.36.250
Record-Route: <sip:192.168.36.250>
From: 1484 <sip:1484@192.168.36.250>;tag=8b55953ff0d3c200
To: 1484 <sip:1484@192.168.36.250>;tag=as5c9b399f
Call-ID: 800000fe65953ff0d3c20d@192.168.36.250
CSeq: 21835 OPTIONS
Server: Asterisk PBX 14.3.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:192.168.36.249:5060>
Accept: application/sdp
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '800000fe65953ff0d3c20d@192.168.36.250' in 32000 ms (Method: OPTIONS)
Really destroying SIP dialog '8000016c65953feee31f38@192.168.36.250' Method: OPTIONS

<--- SIP read from UDP:192.168.36.250:5060 --->
OPTIONS sip:1483@192.168.36.249:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.36.250:5060;branch=z9hG4bKc29f873e0ccca7142f7bc08d01d8bdf1
Max-Forwards: 70
Record-Route: <sip:192.168.36.250>
To: 1483 <sip:1483@192.168.36.250>
From: 1483 <sip:1483@192.168.36.250>;tag=8b35953ff0e3b7a6
Call-ID: 800000fd65953ff0e3b7b3@192.168.36.250
CSeq: 21810 OPTIONS
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Sending to 192.168.36.250:5060 (no NAT)
Looking for 1483 in default (domain 192.168.36.249)

<--- Transmitting (no NAT) to 192.168.36.250:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.36.250:5060;branch=z9hG4bKc29f873e0ccca7142f7bc08d01d8bdf1;received=192.168.36.250
Record-Route: <sip:192.168.36.250>
From: 1483 <sip:1483@192.168.36.250>;tag=8b35953ff0e3b7a6
To: 1483 <sip:1483@192.168.36.250>;tag=as78dd572c
Call-ID: 800000fd65953ff0e3b7b3@192.168.36.250
CSeq: 21810 OPTIONS
Server: Asterisk PBX 14.3.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:192.168.36.249:5060>
Accept: application/sdp
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '800000fd65953ff0e3b7b3@192.168.36.250' in 32000 ms (Method: OPTIONS)
asterisk*CLI>
Disconnected from Asterisk server
Asterisk cleanly ending (0).
Executing last minute cleanups