```
<--- Received SIP request (1273 bytes) from UDP:192.168.1.243:54711 --->
INVITE sip:Zone_2@192.168.1.18 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.243:54711;branch=z9hG4bK66788284;rport
From: <sip:103@192.168.1.18>;tag=1105452050
To: <sip:Zone_2@192.168.1.18>
Call-ID: 963469123-54711-4@BJC.BGI.B.CED
CSeq: 30 INVITE
Contact: <sip:103@192.168.1.243:54711>
Max-Forwards: 70
User-Agent: Grandstream Wave 1.0.3.27
Privacy: none
P-Preferred-Identity: <sip:103@192.168.1.18>
Supported: replaces, path, timer, eventlist
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Type: application/sdp
Accept: application/sdp, application/dtmf-relay
Content-Length:   632

v=0
o=103 8000 8000 IN IP4 192.168.1.243
s=SIP Call
c=IN IP4 192.168.1.243
t=0 0
m=audio 14078 RTP/AVP 0 8 9 123 2 97 3 101
a=sendrecv
a=rtcp:14079 IN IP4 192.168.1.243
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:123 opus/48000�<--- Transmitting SIP response (475 bytes) to UDP:192.168.1.243:54711 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.243:54711;rport=54711;received=192.168.1.243;branch=z9hG4bK66788284
Call-ID: 963469123-54711-4@BJC.BGI.B.CED
From: <sip:103@192.168.1.18>;tag=1105452050
To: <sip:Zone_2@192.168.1.18>;tag=z9hG4bK66788284
CSeq: 30 INVITE
WWW-Authenticate: Digest  realm="asterisk",nonce="1554804972/82cd9a20101613aa10c08179bccd0eb7",opaque="5906c406244e6169",algorithm=md5,qop="auth"
Server: Asterisk PBX 15.2.2
Content-Length:  0


<--- Received SIP request (277 bytes) from UDP:192.168.1.243:54711 --->
ACK sip:Zone_2@192.168.1.18 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.243:54711;branch=z9hG4bK66788284;rport
From: <sip:103@192.168.1.18>;tag=1105452050
To: <sip:Zone_2@192.168.1.18>;tag=z9hG4bK66788284
Call-ID: 963469123-54711-4@BJC.BGI.B.CED
CSeq: 30 ACK
Content-Length: 0


<--- Received SIP request (1544 bytes) from UDP:192.168.1.243:54711 --->
INVITE sip:Zone_2@192.168.1.18 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.243:54711;branch=z9hG4bK1145485573;rport
From: <sip:103@192.168.1.18>;tag=1105452050
To: <sip:Zone_2@192.168.1.18>
Call-ID: 963469123-54711-4@BJC.BGI.B.CED
CSeq: 31 INVITE
Contact: <sip:103@192.168.1.243:54711>
Authorization: Digest username="103", realm="asterisk", nonce="1554804972/82cd9a20101613aa10c08179bccd0eb7", uri="sip:Zone_2@192.168.1.18", response="6d3a353788f6ec9561541cf55799d8c3", algorithm=md5, cnonce="04775183", opaque="5906c406244e6169", qop=auth, nc=00000005
Max-Forwards: 70
User-Agent: Grandstream Wave 1.0.3.27
Privacy: none
P-Preferred-Identity: <sip:103@192.168.1.18>
Supported: replaces, path, timer, eventlist
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Type: application/sdp
Accept: application/sdp, application/dtmf-relay
Content-Length:   632

v=0
o=103 8000 8[Apr  9 10:16:12] NOTICE[2932]: res_pjsip_session.c:2941 new_invite: Call from '103' (UDP:192.168.1.243:54711) to extension 'Zone_2' rejected because extension not found in context 'default'.
<--- Transmitting SIP response (348 bytes) to UDP:192.168.1.243:54711 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.1.243:54711;rport=54711;received=192.168.1.243;branch=z9hG4bK1145485573
Call-ID: 963469123-54711-4@BJC.BGI.B.CED
From: <sip:103@192.168.1.18>;tag=1105452050
To: <sip:Zone_2@192.168.1.18>;tag=a45a106e-9dce-45ba-814a-386c9feedefc
CSeq: 31 INVITE
Server: Asterisk PBX 15.2.2
Content-Length:  0


<--- Received SIP request (300 bytes) from UDP:192.168.1.243:54711 --->
ACK sip:Zone_2@192.168.1.18 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.243:54711;branch=z9hG4bK1145485573;rport
From: <sip:103@192.168.1.18>;tag=1105452050
To: <sip:Zone_2@192.168.1.18>;tag=a45a106e-9dce-45ba-814a-386c9feedefc
Call-ID: 963469123-54711-4@BJC.BGI.B.CED
CSeq: 31 ACK
Content-Length: 0


<--- Transmitting SIP request (419 bytes) to UDP:192.168.1.243:54711 --->
OPTIONS sip:103@192.168.1.243:54711 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.18:5060;rport;branch=z9hG4bKPj17134138-6348-4abf-9d24-d2551649d513
From: <sip:103@192.168.1.18>;tag=5cee2d33-c566-44cf-8c27-8141e3660294
To: <sip:103@192.168.1.243>
Contact: <sip:103@192.168.1.18:5060>
Call-ID: cdb90efc-b230-4814-9957-0287bbcd3464
CSeq: 31175 OPTIONS
Max-Forwards: 70
User-Agent: Asterisk PBX 15.2.2
Content-Length:  0


<--- Received SIP response (488 bytes) from UDP:192.168.1.243:54711 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.18:5060;rport=5060;branch=z9hG4bKPj17134138-6348-4abf-9d24-d2551649d513
From: <sip:103@192.168.1.18>;tag=5cee2d33-c566-44cf-8c27-8141e3660294
To: <sip:103@192.168.1.243>;tag=1786750529
Call-ID: cdb90efc-b230-4814-9957-0287bbcd3464
CSeq: 31175 OPTIONS
Supported: replaces, path, eventlist
User-Agent: Grandstream Wave 1.0.3.27
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0


<--- Received SIP request (1269 bytes) from UDP:192.168.1.243:54711 --->
INVITE sip:102@192.168.1.18 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.243:54711;branch=z9hG4bK138347666;rport
From: <sip:103@192.168.1.18>;tag=1184169627
To: <sip:102@192.168.1.18>
Call-ID: 1900865602-54711-5@BJC.BGI.B.CED
CSeq: 40 INVITE
Contact: <sip:103@192.168.1.243:54711>
Max-Forwards: 70
User-Agent: Grandstream Wave 1.0.3.27
Privacy: none
P-Preferred-Identity: <sip:103@192.168.1.18>
Supported: replaces, path, timer, eventlist
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Type: application/sdp
Accept: application/sdp, application/dtmf-relay
Content-Length:   632

v=0
o=103 8000 8000 IN IP4 192.168.1.243
s=SIP Call
c=IN IP4 192.168.1.243
t=0 0
m=audio 17016 RTP/AVP 0 8 9 123 2 97 3 101
a=sendrecv
a=rtcp:17017 IN IP4 192.168.1.243
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:123 opus/48000/2
<--- Transmitting SIP response (475 bytes) to UDP:192.168.1.243:54711 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.243:54711;rport=54711;received=192.168.1.243;branch=z9hG4bK138347666
Call-ID: 1900865602-54711-5@BJC.BGI.B.CED
From: <sip:103@192.168.1.18>;tag=1184169627
To: <sip:102@192.168.1.18>;tag=z9hG4bK138347666
CSeq: 40 INVITE
WWW-Authenticate: Digest  realm="asterisk",nonce="1554804983/93536db7c0b3352cf9d4cd30fa6e9451",opaque="2a4a9c57657228b4",algorithm=md5,qop="auth"
Server: Asterisk PBX 15.2.2
Content-Length:  0


<--- Received SIP request (274 bytes) from UDP:192.168.1.243:54711 --->
ACK sip:102@192.168.1.18 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.243:54711;branch=z9hG4bK138347666;rport
From: <sip:103@192.168.1.18>;tag=1184169627
To: <sip:102@192.168.1.18>;tag=z9hG4bK138347666
Call-ID: 1900865602-54711-5@BJC.BGI.B.CED
CSeq: 40 ACK
Content-Length: 0


<--- Received SIP request (1535 bytes) from UDP:192.168.1.243:54711 --->
INVITE sip:102@192.168.1.18 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.243:54711;branch=z9hG4bK891806338;rport
From: <sip:103@192.168.1.18>;tag=1184169627
To: <sip:102@192.168.1.18>
Call-ID: 1900865602-54711-5@BJC.BGI.B.CED
CSeq: 41 INVITE
Contact: <sip:103@192.168.1.243:54711>
Authorization: Digest username="103", realm="asterisk", nonce="1554804983/93536db7c0b3352cf9d4cd30fa6e9451", uri="sip:102@192.168.1.18", response="ed69d80fba8eb1c7e3e6764d53349f59", algorithm=md5, cnonce="00506422", opaque="2a4a9c57657228b4", qop=auth, nc=00000006
Max-Forwards: 70
User-Agent: Grandstream Wave 1.0.3.27
Privacy: none
P-Preferred-Identity: <sip:103@192.168.1.18>
Supported: replaces, path, timer, eventlist
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Type: application/sdp
Accept: application/sdp, application/dtmf-relay
Content-Length:   632

v=0
o=103 8000 8000 IN IP  == Setting global variable 'SIPDOMAIN' to '192.168.1.18'
<--- Transmitting SIP response (301 bytes) to UDP:192.168.1.243:54711 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.243:54711;rport=54711;received=192.168.1.243;branch=z9hG4bK891806338
Call-ID: 1900865602-54711-5@BJC.BGI.B.CED
From: <sip:103@192.168.1.18>;tag=1184169627
To: <sip:102@192.168.1.18>
CSeq: 41 INVITE
Server: Asterisk PBX 15.2.2
Content-Length:  0


  == Using SIP RTP Audio TOS bits 184
  == Using SIP RTP Audio TOS bits 184 in TCLASS field.
  == Using SIP RTP Video TOS bits 136
  == Using SIP RTP Video TOS bits 136 in TCLASS field.
    -- Executing [102@default:1] Dial("PJSIP/103-0000000b", "PJSIP/102/sip:102@192.168.1.236:5064") in new stack
    -- Called PJSIP/102/sip:102@192.168.1.236:5064
  == Using SIP RTP Audio TOS bits 184
  == Using SIP RTP Audio TOS bits 184 in TCLASS field.
  == Using SIP RTP Video TOS bits 136
  == Using SIP RTP Video TOS bits 136 in TCLASS field.
<--- Transmitting SIP request (1087 bytes) to UDP:192.168.1.236:5064 --->
INVITE sip:102@192.168.1.236:5064 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.18:5060;rport;branch=z9hG4bKPjd7a0fabb-77ef-4ed9-b3bc-83d3807c7749
From: "103" <sip:Zone_3@192.168.1.18>;tag=b1e57e23-7605-413c-889d-f64b1d58c1c4
To: <sip:102@192.168.1.236>
Contact: <sip:asterisk@192.168.1.18:5060>
Call-ID: 806b6eb3-90c1-4f31-abaf-e0ea5f420afd
CSeq: 18476 INVITE
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, MESSAGE, REFER
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800
Min-SE: 90
P-Asserted-Identity: "103" <sip:Zone_3@192.168.1.18>
Max-Forwards: 70
User-Agent: Asterisk PBX 15.2.2
Content-Type: application/sdp
Content-Length:   375

v=0
o=- 514112189 514112189 IN IP4 192.168.1.18
s=Asterisk
c=IN IP4 192.168.1.18
t=0 0
m=audio 33548 RTP/AVP 0 8 3 111 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:111 G7<--- Received SIP response (489 bytes) from UDP:192.168.1.236:5064 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.18:5060;rport=5060;branch=z9hG4bKPjd7a0fabb-77ef-4ed9-b3bc-83d3807c7749
From: "103" <sip:Zone_3@192.168.1.18>;tag=b1e57e23-7605-413c-889d-f64b1d58c1c4
To: <sip:102@192.168.1.236>
Call-ID: 806b6eb3-90c1-4f31-abaf-e0ea5f420afd
CSeq: 18476 INVITE
Supported: replaces, path, eventlist
User-Agent: Grandstream GXV3275 1.0.3.207
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0


<--- Received SIP response (576 bytes) from UDP:192.168.1.236:5064 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.18:5060;rport=5060;branch=z9hG4bKPjd7a0fabb-77ef-4ed9-b3bc-83d3807c7749
From: "103" <sip:Zone_3@192.168.1.18>;tag=b1e57e23-7605-413c-889d-f64b1d58c1c4
To: <sip:102@192.168.1.236>;tag=189391962
Call-ID: 806b6eb3-90c1-4f31-abaf-e0ea5f420afd
CSeq: 18476 INVITE
Contact: <sip:102@192.168.1.236:5064>
Supported: replaces, path, timer, eventlist
User-Agent: Grandstream GXV3275 1.0.3.207
Allow-Events: talk, hold
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0


    -- PJSIP/102-0000000c is ringing
<--- Transmitting SIP response (542 bytes) to UDP:192.168.1.243:54711 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.243:54711;rport=54711;received=192.168.1.243;branch=z9hG4bK891806338
Call-ID: 1900865602-54711-5@BJC.BGI.B.CED
From: <sip:103@192.168.1.18>;tag=1184169627
To: <sip:102@192.168.1.18>;tag=a6dd344b-f613-4d77-ad41-75bc2f118c67
CSeq: 41 INVITE
Server: Asterisk PBX 15.2.2
Contact: <sip:192.168.1.18:5060>
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, MESSAGE, REFER
P-Asserted-Identity: "102" <sip:Zone_2@192.168.1.18>
Content-Length:  0


<--- Received SIP response (1061 bytes) from UDP:192.168.1.236:5064 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.18:5060;rport=5060;branch=z9hG4bKPjd7a0fabb-77ef-4ed9-b3bc-83d3807c7749
From: "103" <sip:Zone_3@192.168.1.18>;tag=b1e57e23-7605-413c-889d-f64b1d58c1c4
To: <sip:102@192.168.1.236>;tag=189391962
Call-ID: 806b6eb3-90c1-4f31-abaf-e0ea5f420afd
CSeq: 18476 INVITE
Contact: <sip:102@192.168.1.236:5064>
Supported: replaces, path, timer, eventlist
User-Agent: Grandstream GXV3275 1.0.3.207
Session-Expires: 1800;refresher=uac
Require: timer
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Type: application/sdp
Content-Length:   428

v=0
o=102 8000 8000 IN IP4 192.168.1.236
s=SIP Call
c=IN IP4 192.168.1.236
t=0 0
m=audio 5004 RTP/AVP 0 8 101
a=sendrecv
a=rtcp:5005 IN IP4 192.168.1.236
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
m=video 5006 RTP/�0x188e1e0 -- Strict RTP learning after remote address set to: 192.168.1.236:5004
       > 0x1890c60 -- Strict RTP learning after remote address set to: 192.168.1.236:5006
<--- Transmitting SIP request (395 bytes) to UDP:192.168.1.236:5064 --->
ACK sip:102@192.168.1.236:5064 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.18:5060;rport;branch=z9hG4bKPj3ebe310b-170f-437b-a717-1184ca525aca
From: "103" <sip:Zone_3@192.168.1.18>;tag=b1e57e23-7605-413c-889d-f64b1d58c1c4
To: <sip:102@192.168.1.236>;tag=189391962
Call-ID: 806b6eb3-90c1-4f31-abaf-e0ea5f420afd
CSeq: 18476 ACK
Max-Forwards: 70
User-Agent: Asterisk PBX 15.2.2
Content-Length:  0


    -- PJSIP/102-0000000c answered PJSIP/103-0000000b
       > 0x1888ce0 -- Strict RTP learning after remote address set to: 192.168.1.243:17016
       > 0x188b760 -- Strict RTP learning after remote address set to: 192.168.1.243:28594
<--- Transmitting SIP response (982 bytes) to UDP:192.168.1.243:54711 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.243:54711;rport=54711;received=192.168.1.243;branch=z9hG4bK891806338
Call-ID: 1900865602-54711-5@BJC.BGI.B.CED
From: <sip:103@192.168.1.18>;tag=1184169627
To: <sip:102@192.168.1.18>;tag=a6dd344b-f613-4d77-ad41-75bc2f118c67
CSeq: 41 INVITE
Server: Asterisk PBX 15.2.2
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, MESSAGE, REFER
Contact: <sip:192.168.1.18:5060>
Supported: 100rel, timer, replaces, norefersub
P-Asserted-Identity: "102" <sip:Zone_2@192.168.1.18>
Content-Type: application/sdp
Content-Length:   363

v=0
o=- 8000 8002 IN IP4 192.168.1.18
s=Asterisk
c=IN IP4 192.168.1.18
t=0 0
m=audio 37706 RTP/AVP 0 8 3 2 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
    -- Channel PJSIP/102-0000000c joined 'simple_bridge' basic-bridge <682de07f-5906-4ce0-89df-d8089be024da>
    -- Channel PJSIP/103-0000000b joined 'simple_bridge' basic-bridge <682de07f-5906-4ce0-89df-d8089be024da>
<--- Received SIP request (529 bytes) from UDP:192.168.1.243:54711 --->
ACK sip:192.168.1.18:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.243:54711;branch=z9hG4bK1680716700;rport
From: <sip:103@192.168.1.18>;tag=1184169627
To: <sip:102@192.168.1.18>;tag=a6dd344b-f613-4d77-ad41-75bc2f118c67
Call-ID: 1900865602-54711-5@BJC.BGI.B.CED
CSeq: 41 ACK
Contact: <sip:103@192.168.1.243:54711>
Max-Forwards: 70
Supported: replaces, path, timer, eventlist
User-Agent: Grandstream Wave 1.0.3.27
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0


<--- Received SIP request (744 bytes) from UDP:192.168.1.243:54711 --->
INFO sip:192.168.1.18:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.243:54711;branch=z9hG4bK1054233038;rport
From: <sip:103@192.168.1.18>;tag=1184169627
To: <sip:102@192.168.1.18>;tag=a6dd344b-f613-4d77-ad41-75bc2f118c67
Call-ID: 1900865602-54711-5@BJC.BGI.B.CED
CSeq: 42 INFO
Contact: <sip:103@192.168.1.243:54711>
Max-Forwards: 70
Supported: replaces, path, timer, eventlist
User-Agent: Grandstream Wave 1.0.3.27
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Type: application/media_control+xml
Content-Length:   164

<?xml version="1.0" encoding="utf-8" ?><media_control>  <vc_primitive>    <to_encoder>      <picture_fast_update/>    </to_encoder>  </vc_primitive></media_control>
<--- Transmitting SIP response (337 bytes) to UDP:192.168.1.243:54711 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.243:54711;rport=54711;received=192.168.1.243;branch=z9hG4bK1054233038
Call-ID: 1900865602-54711-5@BJC.BGI.B.CED
From: <sip:103@192.168.1.18>;tag=1184169627
To: <sip:102@192.168.1.18>;tag=a6dd344b-f613-4d77-ad41-75bc2f118c67
CSeq: 42 INFO
Server: Asterisk PBX 15.2.2
Content-Length:  0


<--- Transmitting SIP request (623 bytes) to UDP:192.168.1.236:5064 --->
INFO sip:102@192.168.1.236:5064 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.18:5060;rport;branch=z9hG4bKPje6060001-aaae-433b-ac4b-383ef42b796b
From: "103" <sip:Zone_3@192.168.1.18>;tag=b1e57e23-7605-413c-889d-f64b1d58c1c4
To: <sip:102@192.168.1.236>;tag=189391962
Call-ID: 806b6eb3-90c1-4f31-abaf-e0ea5f420afd
CSeq: 18477 INFO
Max-Forwards: 70
User-Agent: Asterisk PBX 15.2.2
Content-Type: application/media_control+xml
Content-Length:   178

<?xml version="1.0" encoding="utf-8" ?>
 <media_control>
  <vc_primitive>
   <to_encoder>
    <picture_fast_update/>
   </to_encoder>
  </vc_primitive>
 </media_control>

<--- Received SIP response (543 bytes) from UDP:192.168.1.236:5064 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.18:5060;rport=5060;branch=z9hG4bKPje6060001-aaae-433b-ac4b-383ef42b796b
From: "103" <sip:Zone_3@192.168.1.18>;tag=b1e57e23-7605-413c-889d-f64b1d58c1c4
To: <sip:102@192.168.1.236>;tag=189391962
Call-ID: 806b6eb3-90c1-4f31-abaf-e0ea5f420afd
CSeq: 18477 INFO
Contact: <sip:102@192.168.1.236:5064>
Supported: replaces, path, timer, eventlist
User-Agent: Grandstream GXV3275 1.0.3.207
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0


       > 0x1888ce0 -- Strict RTP switching to RTP target address 192.168.1.243:17016 as source
       > 0x188e1e0 -- Strict RTP switching to RTP target address 192.168.1.236:5004 as source
       > 0x188b760 -- Strict RTP switching to RTP target address 192.168.1.243:28594 as source
       > 0x1890c60 -- Strict RTP switching to RTP target address 192.168.1.236:5006 as source
<--- Received SIP request (556 bytes) from UDP:192.168.1.236:5064 --->
BYE sip:asterisk@192.168.1.18:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.236:5064;branch=z9hG4bK2032915779;rport
From: <sip:102@192.168.1.236>;tag=189391962
To: "103" <sip:Zone_3@192.168.1.18>;tag=b1e57e23-7605-413c-889d-f64b1d58c1c4
Call-ID: 806b6eb3-90c1-4f31-abaf-e0ea5f420afd
CSeq: 18477 BYE
Contact: <sip:102@192.168.1.236:5064>
Max-Forwards: 70
Supported: replaces, path, timer, eventlist
User-Agent: Grandstream GXV3275 1.0.3.207
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0


<--- Transmitting SIP response (350 bytes) to UDP:192.168.1.236:5064 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.236:5064;rport=5064;received=192.168.1.236;branch=z9hG4bK2032915779
Call-ID: 806b6eb3-90c1-4f31-abaf-e0ea5f420afd
From: <sip:102@192.168.1.236>;tag=189391962
To: "103" <sip:Zone_3@192.168.1.18>;tag=b1e57e23-7605-413c-889d-f64b1d58c1c4
CSeq: 18477 BYE
Server: Asterisk PBX 15.2.2
Content-Length:  0


    -- Channel PJSIP/102-0000000c left 'simple_bridge' basic-bridge <682de07f-5906-4ce0-89df-d8089be024da>
    -- Channel PJSIP/103-0000000b left 'simple_bridge' basic-bridge <682de07f-5906-4ce0-89df-d8089be024da>
  == Spawn extension (default, 102, 1) exited non-zero on 'PJSIP/103-0000000b'
<--- Transmitting SIP request (406 bytes) to UDP:192.168.1.243:54711 --->
BYE sip:103@192.168.1.243:54711 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.18:5060;rport;branch=z9hG4bKPj8f14f4a7-a037-45b9-b20e-2b83c5d8ffda
From: <sip:102@192.168.1.18>;tag=a6dd344b-f613-4d77-ad41-75bc2f118c67
To: <sip:103@192.168.1.18>;tag=1184169627
Call-ID: 1900865602-54711-5@BJC.BGI.B.CED
CSeq: 2107 BYE
Reason: Q.850;cause=16
Max-Forwards: 70
User-Agent: Asterisk PBX 15.2.2
Content-Length:  0


<--- Received SIP response (525 bytes) from UDP:192.168.1.243:54711 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.18:5060;rport=5060;branch=z9hG4bKPj8f14f4a7-a037-45b9-b20e-2b83c5d8ffda
From: <sip:102@192.168.1.18>;tag=a6dd344b-f613-4d77-ad41-75bc2f118c67
To: <sip:103@192.168.1.18>;tag=1184169627
Call-ID: 1900865602-54711-5@BJC.BGI.B.CED
CSeq: 2107 BYE
Contact: <sip:103@192.168.1.243:54711>
Supported: replaces, path, timer, eventlist
User-Agent: Grandstream Wave 1.0.3.27
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0


<--- Transmitting SIP request (418 bytes) to UDP:192.168.1.236:5064 --->
OPTIONS sip:102@192.168.1.236:5064 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.18:5060;rport;branch=z9hG4bKPj81a1f665-7854-4af2-a353-80820150adfc
From: <sip:102@192.168.1.18>;tag=f430ddee-cede-48cd-8693-4fc511c0f1ce
To: <sip:102@192.168.1.236>
Contact: <sip:102@192.168.1.18:5060>
Call-ID: d86a7fbd-01d0-4eb8-a3fe-b03556d25dce
CSeq: 61615 OPTIONS
Max-Forwards: 70
User-Agent: Asterisk PBX 15.2.2
Content-Length:  0


<--- Received SIP response (492 bytes) from UDP:192.168.1.236:5064 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.18:5060;rport=5060;branch=z9hG4bKPj81a1f665-7854-4af2-a353-80820150adfc
From: <sip:102@192.168.1.18>;tag=f430ddee-cede-48cd-8693-4fc511c0f1ce
To: <sip:102@192.168.1.236>;tag=1345834777
Call-ID: d86a7fbd-01d0-4eb8-a3fe-b03556d25dce
CSeq: 61615 OPTIONS
Supported: replaces, path, eventlist
User-Agent: Grandstream GXV3275 1.0.3.207
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0