Asterisk 11.13.1~dfsg-2+deb8u1, Copyright (C) 1999 - 2013 Digium, Inc. and others.
Created by Mark Spencer <markster@digium.com>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'core show license' for details.
=========================================================================
Connected to Asterisk 11.13.1~dfsg-2+deb8u1 currently running on david (pid = 21747)
david*CLI> sip set debug on
SIP Debugging re-enabled

<--- SIP read from UDP:192.168.0.13:5060 --->
jaK
<------------->

<--- SIP read from UDP:192.168.0.6:5060 --->
INVITE sip:0981722187@192.168.0.2 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.6:5060;branch=z9hG4bK2069829577;rport
From: <sip:FXS@192.168.0.2>;tag=184258689
To: <sip:0981722187@192.168.0.2>
Call-ID: 1633570378-5060-10@BJC.BGI.A.G
CSeq: 90 INVITE
Contact: <sip:FXS@192.168.0.6:5060>
Max-Forwards: 70
User-Agent: Grandstream HT-503 V1.4A 1.0.8.4 chip V2.2
Privacy: none
P-Preferred-Identity: <sip:FXS@192.168.0.2>
Supported: replaces, path, timer, eventlist
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE
Content-Type: application/sdp
Accept: application/sdp, application/dtmf-relay
Content-Length: 384

v=0
o=FXS 8000 8000 IN IP4 192.168.0.6
s=SIP Call
c=IN IP4 192.168.0.6
t=0 0
m=audio 5004 RTP/AVP 0 8 4 18 2 97 102 100
a=sendrecv
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:8 PCMA/8000
a=rtpmap:4 G723/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:2 G726-32/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=20
a=rtpmap:102 G729E/8000
a=rtpmap:100 AAL2-G726-16/8000
<------------->
--- (16 headers 18 lines) ---
Sending to 192.168.0.6:5060 (no NAT)
Sending to 192.168.0.6:5060 (no NAT)
Using INVITE request as basis request - 1633570378-5060-10@BJC.BGI.A.G
Found peer 'FXS' for 'FXS' from 192.168.0.6:5060

<--- Reliably Transmitting (no NAT) to 192.168.0.6:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.0.6:5060;branch=z9hG4bK2069829577;received=192.168.0.6;rport=5060
From: <sip:FXS@192.168.0.2>;tag=184258689
To: <sip:0981722187@192.168.0.2>;tag=as14383432
Call-ID: 1633570378-5060-10@BJC.BGI.A.G
CSeq: 90 INVITE
Server: Asterisk PBX 11.13.1~dfsg-2+deb8u1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="753329ab"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '1633570378-5060-10@BJC.BGI.A.G' in 6400 ms (Method: INVITE)

<--- SIP read from UDP:192.168.0.6:5060 --->
ACK sip:0981722187@192.168.0.2 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.6:5060;branch=z9hG4bK2069829577;rport
From: <sip:FXS@192.168.0.2>;tag=184258689
To: <sip:0981722187@192.168.0.2>;tag=as14383432
Call-ID: 1633570378-5060-10@BJC.BGI.A.G
CSeq: 90 ACK
Content-Length: 0

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

<--- SIP read from UDP:192.168.0.6:5060 --->
INVITE sip:0981722187@192.168.0.2 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.6:5060;branch=z9hG4bK1494493287;rport
From: <sip:FXS@192.168.0.2>;tag=184258689
To: <sip:0981722187@192.168.0.2>
Call-ID: 1633570378-5060-10@BJC.BGI.A.G
CSeq: 91 INVITE
Contact: <sip:FXS@192.168.0.6:5060>
Authorization: Digest username="FXS", realm="asterisk", nonce="753329ab", uri="sip:0981722187@192.168.0.2", response="42cc3d9c8de4e9b8cae98cb704969f90", algorithm=MD5
Max-Forwards: 70
User-Agent: Grandstream HT-503 V1.4A 1.0.8.4 chip V2.2
Privacy: none
P-Preferred-Identity: <sip:FXS@192.168.0.2>
Supported: replaces, path, timer, eventlist
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE
Content-Type: application/sdp
Accept: application/sdp, application/dtmf-relay
Content-Length: 384

v=0
o=FXS 8000 8000 IN IP4 192.168.0.6
s=SIP Call
c=IN IP4 192.168.0.6
t=0 0
m=audio 5004 RTP/AVP 0 8 4 18 2 97 102 100
a=sendrecv
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:8 PCMA/8000
a=rtpmap:4 G723/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:2 G726-32/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=20
a=rtpmap:102 G729E/8000
a=rtpmap:100 AAL2-G726-16/8000
<------------->
--- (17 headers 18 lines) ---
Sending to 192.168.0.6:5060 (no NAT)
Using INVITE request as basis request - 1633570378-5060-10@BJC.BGI.A.G
Found peer 'FXS' for 'FXS' from 192.168.0.6:5060
  == Using SIP RTP CoS mark 5
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 4
Found RTP audio format 18
Found RTP audio format 2
Found RTP audio format 97
Found RTP audio format 102
Found RTP audio format 100
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format G723 for ID 4
Found audio description format G729 for ID 18
Found audio description format G726-32 for ID 2
Found audio description format iLBC for ID 97
Found unknown media description format G729E for ID 102
Found unknown media description format AAL2-G726-16 for ID 100
Capabilities: us - (gsm|ulaw|alaw|h263|testlaw), peer - audio=(g723|ulaw|alaw|g726|g729|ilbc)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port 192.168.0.6:5004
Looking for 0981722187 in phones (domain 192.168.0.2)
list_route: hop: <sip:FXS@192.168.0.6:5060>

<--- Transmitting (no NAT) to 192.168.0.6:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.6:5060;branch=z9hG4bK1494493287;received=192.168.0.6;rport=5060
From: <sip:FXS@192.168.0.2>;tag=184258689
To: <sip:0981722187@192.168.0.2>
Call-ID: 1633570378-5060-10@BJC.BGI.A.G
CSeq: 91 INVITE
Server: Asterisk PBX 11.13.1~dfsg-2+deb8u1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:0981722187@192.168.0.2:5060>
Content-Length: 0


<------------>
    -- Executing [0981722187@phones:1] Goto("SIP/FXS-00000019", "outgoing,0,1") in new stack
    -- Goto (outgoing,0,1)
    -- Executing [0@outgoing:1] Dial("SIP/FXS-00000019", "SIP/FXO") in new stack
  == Using SIP RTP CoS mark 5
Audio is at 11780
Adding codec 100003 (ulaw) to SDP
Adding codec 100004 (alaw) to SDP
Adding codec 100008 (g729) to SDP
Adding codec 100002 (gsm) to SDP
Adding codec 100017 (testlaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 192.168.0.6:5062:
INVITE sip:FXO@192.168.0.6:5062 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.2:5060;branch=z9hG4bK731bc225
Max-Forwards: 70
From: <sip:FXS@192.168.0.2>;tag=as15053e3a
To: <sip:FXO@192.168.0.6:5062>
Contact: <sip:FXS@192.168.0.2:5060>
Call-ID: 23a0eb10794a4ed31b2c817f686cb8f0@192.168.0.2:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 11.13.1~dfsg-2+deb8u1
Date: Sun, 22 Jan 2017 13:53:20 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 342

v=0
o=root 2129092439 2129092439 IN IP4 192.168.0.2
s=Asterisk PBX 11.13.1~dfsg-2+deb8u1
c=IN IP4 192.168.0.2
t=0 0
m=audio 11780 RTP/AVP 0 8 18 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
    -- Called SIP/FXO

<--- SIP read from UDP:192.168.0.6:5062 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.2:5060;branch=z9hG4bK731bc225
From: <sip:FXS@192.168.0.2>;tag=as15053e3a
To: <sip:FXO@192.168.0.6:5062>
Call-ID: 23a0eb10794a4ed31b2c817f686cb8f0@192.168.0.2:5060
CSeq: 102 INVITE
Supported: replaces, path, timer, eventlist
User-Agent: Grandstream HT-503 V1.4A 1.0.8.4 chip V2.2
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE
Content-Length: 0

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

<--- SIP read from UDP:192.168.0.6:5062 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.0.2:5060;branch=z9hG4bK731bc225
From: <sip:FXS@192.168.0.2>;tag=as15053e3a
To: <sip:FXO@192.168.0.6:5062>;tag=1180830562
Call-ID: 23a0eb10794a4ed31b2c817f686cb8f0@192.168.0.2:5060
CSeq: 102 INVITE
Contact: <sip:FXO@192.168.0.6:5062>
Supported: replaces, path, timer, eventlist
User-Agent: Grandstream HT-503 V1.4A 1.0.8.4 chip V2.2
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
list_route: hop: <sip:FXO@192.168.0.6:5062>
    -- SIP/FXO-0000001a is ringing

<--- Transmitting (no NAT) to 192.168.0.6:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.0.6:5060;branch=z9hG4bK1494493287;received=192.168.0.6;rport=5060
From: <sip:FXS@192.168.0.2>;tag=184258689
To: <sip:0981722187@192.168.0.2>;tag=as5b914a26
Call-ID: 1633570378-5060-10@BJC.BGI.A.G
CSeq: 91 INVITE
Server: Asterisk PBX 11.13.1~dfsg-2+deb8u1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:0981722187@192.168.0.2:5060>
Content-Length: 0


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

<--- SIP read from UDP:192.168.0.6:5062 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.2:5060;branch=z9hG4bK731bc225
From: <sip:FXS@192.168.0.2>;tag=as15053e3a
To: <sip:FXO@192.168.0.6:5062>;tag=1180830562
Call-ID: 23a0eb10794a4ed31b2c817f686cb8f0@192.168.0.2:5060
CSeq: 102 INVITE
Contact: <sip:FXO@192.168.0.6:5062>
Supported: replaces, path, timer, eventlist
User-Agent: Grandstream HT-503 V1.4A 1.0.8.4 chip V2.2
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE
Content-Type: application/sdp
Content-Length: 223

v=0
o=FXO 8002 8000 IN IP4 192.168.0.6
s=SIP Call
c=IN IP4 192.168.0.6
t=0 0
m=audio 5012 RTP/AVP 0 8 18
a=sendrecv
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
<------------->
--- (12 headers 12 lines) ---
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 18
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format G729 for ID 18
Capabilities: us - (gsm|ulaw|alaw|g729|h263|testlaw), peer - audio=(ulaw|alaw|g729)/video=(nothing)/text=(nothing), combined - (ulaw|alaw|g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port 192.168.0.6:5012
list_route: hop: <sip:FXO@192.168.0.6:5062>
set_destination: Parsing <sip:FXO@192.168.0.6:5062> for address/port to send to
set_destination: set destination to 192.168.0.6:5062
Transmitting (no NAT) to 192.168.0.6:5062:
ACK sip:FXO@192.168.0.6:5062 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.2:5060;branch=z9hG4bK5421bf9b
Max-Forwards: 70
From: <sip:FXS@192.168.0.2>;tag=as15053e3a
To: <sip:FXO@192.168.0.6:5062>;tag=1180830562
Contact: <sip:FXS@192.168.0.2:5060>
Call-ID: 23a0eb10794a4ed31b2c817f686cb8f0@192.168.0.2:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 11.13.1~dfsg-2+deb8u1
Content-Length: 0


---
    -- SIP/FXO-0000001a answered SIP/FXS-00000019
Audio is at 13260
Adding codec 100003 (ulaw) to SDP
Adding codec 100004 (alaw) to SDP

<--- Reliably Transmitting (no NAT) to 192.168.0.6:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.6:5060;branch=z9hG4bK1494493287;received=192.168.0.6;rport=5060
From: <sip:FXS@192.168.0.2>;tag=184258689
To: <sip:0981722187@192.168.0.2>;tag=as5b914a26
Call-ID: 1633570378-5060-10@BJC.BGI.A.G
CSeq: 91 INVITE
Server: Asterisk PBX 11.13.1~dfsg-2+deb8u1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:0981722187@192.168.0.2:5060>
Content-Type: application/sdp
Require: timer
Content-Length: 216

v=0
o=root 1863387715 1863387715 IN IP4 192.168.0.2
s=Asterisk PBX 11.13.1~dfsg-2+deb8u1
c=IN IP4 192.168.0.2
t=0 0
m=audio 13260 RTP/AVP 0 8
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=ptime:20
a=sendrecv

<------------>
    -- Locally bridging SIP/FXS-00000019 and SIP/FXO-0000001a

<--- SIP read from UDP:192.168.0.6:5060 --->
ACK sip:0981722187@192.168.0.2:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.6:5060;branch=z9hG4bK836299420;rport
From: <sip:FXS@192.168.0.2>;tag=184258689
To: <sip:0981722187@192.168.0.2>;tag=as5b914a26
Call-ID: 1633570378-5060-10@BJC.BGI.A.G
CSeq: 91 ACK
Contact: <sip:FXS@192.168.0.6:5060>
Max-Forwards: 70
Supported: replaces, path, timer, eventlist
User-Agent: Grandstream HT-503 V1.4A 1.0.8.4 chip V2.2
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE
Content-Length: 0

<------------->
--- (12 headers 0 lines) ---
       > 0x7ffa480118f0 -- Probation passed - setting RTP source address to 192.168.0.6:5012
       > 0x7ff9f8014b90 -- Probation passed - setting RTP source address to 192.168.0.6:5004
Reliably Transmitting (no NAT) to 192.168.0.6:5060:
OPTIONS sip:FXS@192.168.0.6:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.2:5060;branch=z9hG4bK4611d094
Max-Forwards: 70
From: "asterisk" <sip:asterisk@192.168.0.2>;tag=as7e3f4d25
To: <sip:FXS@192.168.0.6:5060>
Contact: <sip:asterisk@192.168.0.2:5060>
Call-ID: 6c54a8816ca680fd778efe223a263882@192.168.0.2:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 11.13.1~dfsg-2+deb8u1
Date: Sun, 22 Jan 2017 13:53:21 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:192.168.0.6:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.2:5060;branch=z9hG4bK4611d094
From: "asterisk" <sip:asterisk@192.168.0.2>;tag=as7e3f4d25
To: <sip:FXS@192.168.0.6:5060>;tag=669306034
Call-ID: 6c54a8816ca680fd778efe223a263882@192.168.0.2:5060
CSeq: 102 OPTIONS
Supported: replaces, path, timer, eventlist
User-Agent: Grandstream HT-503 V1.4A 1.0.8.4 chip V2.2
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '6c54a8816ca680fd778efe223a263882@192.168.0.2:5060' Method: OPTIONS

<--- SIP read from UDP:192.168.0.13:5060 --->
jaK
<------------->
Reliably Transmitting (no NAT) to 192.168.0.6:5062:
OPTIONS sip:FXO@192.168.0.6:5062 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.2:5060;branch=z9hG4bK19a6063f
Max-Forwards: 70
From: "asterisk" <sip:asterisk@192.168.0.2>;tag=as04b2ab3b
To: <sip:FXO@192.168.0.6:5062>
Contact: <sip:asterisk@192.168.0.2:5060>
Call-ID: 4bd9b47d3b09551e655f6b9c53636fd8@192.168.0.2:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 11.13.1~dfsg-2+deb8u1
Date: Sun, 22 Jan 2017 13:53:23 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:192.168.0.6:5062 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.2:5060;branch=z9hG4bK19a6063f
From: "asterisk" <sip:asterisk@192.168.0.2>;tag=as04b2ab3b
To: <sip:FXO@192.168.0.6:5062>;tag=431133086
Call-ID: 4bd9b47d3b09551e655f6b9c53636fd8@192.168.0.2:5060
CSeq: 102 OPTIONS
Supported: replaces, path, timer, eventlist
User-Agent: Grandstream HT-503 V1.4A 1.0.8.4 chip V2.2
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '4bd9b47d3b09551e655f6b9c53636fd8@192.168.0.2:5060' Method: OPTIONS
Reliably Transmitting (no NAT) to 192.168.0.101:5060:
OPTIONS sip:101@192.168.0.101:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.2:5060;branch=z9hG4bK51ea7c8b
Max-Forwards: 70
From: "asterisk" <sip:asterisk@192.168.0.2>;tag=as0c987bfd
To: <sip:101@192.168.0.101:5060>
Contact: <sip:asterisk@192.168.0.2:5060>
Call-ID: 592036f66ba3f52e21adfcee0d1d6a1b@192.168.0.2:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 11.13.1~dfsg-2+deb8u1
Date: Sun, 22 Jan 2017 13:53:29 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:192.168.0.101:5060 --->
SIP/2.0 200 OK
To: <sip:101@192.168.0.101:5060>;tag=15dab1caf0e81ff8i0
From: "asterisk" <sip:asterisk@192.168.0.2>;tag=as0c987bfd
Call-ID: 592036f66ba3f52e21adfcee0d1d6a1b@192.168.0.2:5060
CSeq: 102 OPTIONS
Via: SIP/2.0/UDP 192.168.0.2:5060;branch=z9hG4bK51ea7c8b
Server: Linksys/SPA942-6.1.5(a)
Content-Length: 0
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: replaces

<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '592036f66ba3f52e21adfcee0d1d6a1b@192.168.0.2:5060' Method: OPTIONS

<--- SIP read from UDP:192.168.0.13:5060 --->
jaK
<------------->

<--- SIP read from UDP:192.168.0.6:5060 --->
BYE sip:0981722187@192.168.0.2:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.6:5060;branch=z9hG4bK1609328712;rport
From: <sip:FXS@192.168.0.2>;tag=184258689
To: <sip:0981722187@192.168.0.2>;tag=as5b914a26
Call-ID: 1633570378-5060-10@BJC.BGI.A.G
CSeq: 92 BYE
Contact: <sip:FXS@192.168.0.6:5060>
Max-Forwards: 70
Supported: replaces, path, timer, eventlist
User-Agent: Grandstream HT-503 V1.4A 1.0.8.4 chip V2.2
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE
Content-Length: 0

<------------->
--- (12 headers 0 lines) ---
Sending to 192.168.0.6:5060 (no NAT)
Scheduling destruction of SIP dialog '1633570378-5060-10@BJC.BGI.A.G' in 6400 ms (Method: BYE)

<--- Transmitting (no NAT) to 192.168.0.6:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.6:5060;branch=z9hG4bK1609328712;received=192.168.0.6;rport=5060
From: <sip:FXS@192.168.0.2>;tag=184258689
To: <sip:0981722187@192.168.0.2>;tag=as5b914a26
Call-ID: 1633570378-5060-10@BJC.BGI.A.G
CSeq: 92 BYE
Server: Asterisk PBX 11.13.1~dfsg-2+deb8u1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '23a0eb10794a4ed31b2c817f686cb8f0@192.168.0.2:5060' in 6400 ms (Method: INVITE)
set_destination: Parsing <sip:FXO@192.168.0.6:5062> for address/port to send to
set_destination: set destination to 192.168.0.6:5062
Reliably Transmitting (no NAT) to 192.168.0.6:5062:
BYE sip:FXO@192.168.0.6:5062 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.2:5060;branch=z9hG4bK0881f39f
Max-Forwards: 70
From: <sip:FXS@192.168.0.2>;tag=as15053e3a
To: <sip:FXO@192.168.0.6:5062>;tag=1180830562
Call-ID: 23a0eb10794a4ed31b2c817f686cb8f0@192.168.0.2:5060
CSeq: 103 BYE
User-Agent: Asterisk PBX 11.13.1~dfsg-2+deb8u1
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


---
  == Spawn extension (outgoing, 0, 1) exited non-zero on 'SIP/FXS-00000019'

<--- SIP read from UDP:192.168.0.6:5062 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.2:5060;branch=z9hG4bK0881f39f
From: <sip:FXS@192.168.0.2>;tag=as15053e3a
To: <sip:FXO@192.168.0.6:5062>;tag=1180830562
Call-ID: 23a0eb10794a4ed31b2c817f686cb8f0@192.168.0.2:5060
CSeq: 103 BYE
Contact: <sip:FXO@192.168.0.6:5062>
Supported: replaces, path, timer, eventlist
User-Agent: Grandstream HT-503 V1.4A 1.0.8.4 chip V2.2
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
Really destroying SIP dialog '23a0eb10794a4ed31b2c817f686cb8f0@192.168.0.2:5060' Method: INVITE
david*CLI> exit