My CLI shows after making changes in sip.conf as you said directmedia=no:
> <--- SIP read from UDP:103.44.119.75:5060 --->
> REGISTER sip:13.126.130.235:5060 SIP/2.0
> Via: SIP/2.0/UDP 103.44.119.75:5060;branch=z9hG4bK-d8754z-d26ab527b7037a1a-1---d8754z-;rport
> Max-Forwards: 70
> Contact: <sip:1010@103.44.119.75:5060;transport=UDP;rinstance=5ec7e5ff634400ff>
> To: "1010"<sip:1010@13.126.130.235:5060>
> From: "1010"<sip:1010@13.126.130.235:5060>;tag=020ea304
> Call-ID: YjhmMjY3YjFjZmE2ZTUyYmRmZWEzZDg4ZDhmNWJhZmM.
> CSeq: 179 REGISTER
> Expires: 120
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE
> Supported: replaces
> User-Agent: 3CXPhone 6.0.26523.0
> Content-Length: 0
>
> <------------->
> --- (13 headers 0 lines) ---
> Sending to 103.44.119.75:5060 (NAT)
> Sending to 103.44.119.75:5060 (NAT)
> Reliably Transmitting (NAT) to 103.44.119.75:5060:
> OPTIONS sip:1010@103.44.119.75:5060;transport=UDP;rinstance=5ec7e5ff634400ff SIP/2.0
> Via: SIP/2.0/UDP 172.26.5.241:5060;branch=z9hG4bK2f619f9d;rport
> Max-Forwards: 70
> From: "asterisk" <sip:asterisk@172.26.5.241>;tag=as50f2e67d
> To: <sip:1010@103.44.119.75:5060;transport=UDP;rinstance=5ec7e5ff634400ff>
> Contact: <sip:asterisk@172.26.5.241:5060>
> Call-ID: 21165b2f6e698915405accec5d441685@172.26.5.241:5060
> CSeq: 102 OPTIONS
> User-Agent: Asterisk PBX 13.22.0
> Date: Thu, 16 Aug 2018 10:37:31 GMT
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
> Supported: replaces, timer
> Content-Length: 0
>
>
> ---
>
> <--- Transmitting (NAT) to 103.44.119.75:5060 --->
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 103.44.119.75:5060;branch=z9hG4bK-d8754z-d26ab527b7037a1a-1---d8754z-;received=103.44.119.75;rport=5060
> From: "1010"<sip:1010@13.126.130.235:5060>;tag=020ea304
> To: "1010"<sip:1010@13.126.130.235:5060>;tag=as1845d879
> Call-ID: YjhmMjY3YjFjZmE2ZTUyYmRmZWEzZDg4ZDhmNWJhZmM.
> CSeq: 179 REGISTER
> Server: Asterisk PBX 13.22.0
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
> Supported: replaces, timer
> Expires: 120
> Contact: <sip:1010@103.44.119.75:5060;transport=UDP;rinstance=5ec7e5ff634400ff>;expires=120
> Date: Thu, 16 Aug 2018 10:37:31 GMT
> Content-Length: 0
>
>
> <------------>
> Scheduling destruction of SIP dialog 'YjhmMjY3YjFjZmE2ZTUyYmRmZWEzZDg4ZDhmNWJhZmM.' in 32000 ms (Method: REGISTER)
>
> <--- SIP read from UDP:103.44.119.75:5060 --->
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 172.26.5.241:5060;branch=z9hG4bK2f619f9d;rport=5060;received=13.126.130.235
> Contact: <sip:103.44.119.75:5060>
> To: <sip:1010@103.44.119.75:5060;transport=UDP;rinstance=5ec7e5ff634400ff>;tag=c8776629
> From: "asterisk"<sip:asterisk@172.26.5.241>;tag=as50f2e67d
> Call-ID: 21165b2f6e698915405accec5d441685@172.26.5.241:5060
> CSeq: 102 OPTIONS
> Accept: application/sdp
> Accept-Language: en
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE
> Supported: replaces
> Allow-Events: presence, message-summary, tunnel-info
> Content-Length: 0
>
> <------------->
> --- (13 headers 0 lines) ---
> Really destroying SIP dialog '21165b2f6e698915405accec5d441685@172.26.5.241:5060' Method: OPTIONS
>
> <--- SIP read from UDP:103.44.119.75:5060 --->
>
>
> <------------->
>
> <--- SIP read from UDP:103.44.119.75:35417 --->
> REGISTER sip:13.126.130.235;transport=UDP SIP/2.0
> Via: SIP/2.0/UDP 103.44.119.75:35417;branch=z9hG4bK-524287-1---35bbe818db3ed0ea;rport
> Max-Forwards: 70
> Contact: <sip:7000@103.44.119.75:35417;rinstance=4e47afd124f34ea0;transport=UDP>
> To: <sip:7000@13.126.130.235;transport=UDP>
> From: <sip:7000@13.126.130.235;transport=UDP>;tag=6e33a44a
> Call-ID: ms4D5NTtdGELRcPzGt4wWA..
> CSeq: 357 REGISTER
> Expires: 30
> Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
> User-Agent: Z 5.2.16 rv2.8.95
> Allow-Events: presence, kpml, talk
> Content-Length: 0
>
> <------------->
> --- (13 headers 0 lines) ---
> Sending to 103.44.119.75:35417 (NAT)
> Sending to 103.44.119.75:35417 (NAT)
> Reliably Transmitting (NAT) to 103.44.119.75:35417:
> OPTIONS sip:7000@103.44.119.75:35417;rinstance=4e47afd124f34ea0;transport=UDP SIP/2.0
> Via: SIP/2.0/UDP 172.26.5.241:5060;branch=z9hG4bK1b2c9130;rport
> Max-Forwards: 70
> From: "asterisk" <sip:asterisk@172.26.5.241>;tag=as0ed0fa26
> To: <sip:7000@103.44.119.75:35417;rinstance=4e47afd124f34ea0;transport=UDP>
> Contact: <sip:asterisk@172.26.5.241:5060>
> Call-ID: 3dd9ea6767fec4b5185aba495651946d@172.26.5.241:5060
> CSeq: 102 OPTIONS
> User-Agent: Asterisk PBX 13.22.0
> Date: Thu, 16 Aug 2018 10:37:38 GMT
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
> Supported: replaces, timer
> Content-Length: 0
>
>
> ---
>
> <--- Transmitting (NAT) to 103.44.119.75:35417 --->
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 103.44.119.75:35417;branch=z9hG4bK-524287-1---35bbe818db3ed0ea;received=103.44.119.75;rport=35417
> From: <sip:7000@13.126.130.235;transport=UDP>;tag=6e33a44a
> To: <sip:7000@13.126.130.235;transport=UDP>;tag=as58f01a01
> Call-ID: ms4D5NTtdGELRcPzGt4wWA..
> CSeq: 357 REGISTER
> Server: Asterisk PBX 13.22.0
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
> Supported: replaces, timer
> Expires: 60
> Contact: <sip:7000@103.44.119.75:35417;rinstance=4e47afd124f34ea0;transport=UDP>;expires=60
> Date: Thu, 16 Aug 2018 10:37:38 GMT
> Content-Length: 0
>
>
> <------------>
> Scheduling destruction of SIP dialog 'ms4D5NTtdGELRcPzGt4wWA..' in 32000 ms (Method: REGISTER)
>
> <--- SIP read from UDP:103.44.119.75:35417 --->
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 172.26.5.241:5060;branch=z9hG4bK1b2c9130;rport=5060;received=13.126.130.235
> Contact: <sip:103.44.119.75:35417>
> To: <sip:7000@103.44.119.75:35417;rinstance=4e47afd124f34ea0;transport=UDP>;tag=dd198775
> From: "asterisk" <sip:asterisk@172.26.5.241>;tag=as0ed0fa26
> Call-ID: 3dd9ea6767fec4b5185aba495651946d@172.26.5.241:5060
> CSeq: 102 OPTIONS
> Accept: application/sdp, application/sdp
> Accept-Language: en
> Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
> Supported: replaces, norefersub, extended-refer, timer, outbound, path, X-cisco-serviceuri
> User-Agent: Z 5.2.16 rv2.8.95
> Allow-Events: presence, kpml, talk
> Content-Length: 0
>
> <------------->
> --- (14 headers 0 lines) ---
> Really destroying SIP dialog '3dd9ea6767fec4b5185aba495651946d@172.26.5.241:5060' Method: OPTIONS
>
> <--- SIP read from UDP:103.44.119.75:5060 --->
> INVITE sip:1000@13.126.130.235:5060 SIP/2.0
> Via: SIP/2.0/UDP 103.44.119.75:5060;branch=z9hG4bK-d8754z-da79a9700f3bb461-1---d8754z-;rport
> Max-Forwards: 70
> Contact: <sip:1010@103.44.119.75:5060;transport=UDP>
> To: <sip:1000@13.126.130.235:5060>
> From: "1010"<sip:1010@13.126.130.235:5060>;tag=f075701e
> Call-ID: YThmMmU1MGQ5NDFhZGQzNzUwYjRlZTg0MjhhOGYzMjU.
> CSeq: 1 INVITE
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REGISTER, SUBSCRIBE, NOTIFY, REFER, INFO, MESSAGE
> Content-Type: application/sdp
> Supported: replaces
> User-Agent: 3CXPhone 6.0.26523.0
> Content-Length: 407
>
> v=0
> o=3cxVCE 169583265 232518075 IN IP4 103.44.119.75
> s=3cxVCE Audio Call
> c=IN IP4 103.44.119.75
> t=0 0
> m=audio 40030 RTP/AVP 0 8 3 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:3 GSM/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
> a=ptime:20
> a=sendrecv
> m=video 40028 RTP/AVP 34
> c=IN IP4 103.44.119.75
> a=rtpmap:34 H263/90000
> a=fmtp:34 QCIF=1;CIF=1;SQCIF=1;CIF4=1
> a=sendrecv
> <------------->
> --- (13 headers 18 lines) ---
> Sending to 103.44.119.75:5060 (NAT)
> Sending to 103.44.119.75:5060 (NAT)
> Using INVITE request as basis request - YThmMmU1MGQ5NDFhZGQzNzUwYjRlZTg0MjhhOGYzMjU.
> Found peer '1010' for '1010' from 103.44.119.75:5060
> == Using SIP RTP CoS mark 5
> Found RTP audio format 0
> Found RTP audio format 8
> Found RTP audio format 3
> 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 GSM for ID 3
> Found audio description format telephone-event for ID 101
> Found RTP video format 34
> Found video description format H263 for ID 34
> Capabilities: us - (ulaw|alaw), peer - audio=(ulaw|gsm|alaw)/video=(h263)/text=(nothing), combined - (ulaw|alaw)
> Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
> > 0x7fac3801ef20 -- Strict RTP learning after remote address set to: 103.44.119.75:40030
> Peer audio RTP is at port 103.44.119.75:40030
> Looking for 1000 in lasttry (domain 13.126.130.235)
> sip_route_dump: route/path hop: <sip:1010@103.44.119.75:5060;transport=UDP>
>
> <--- Transmitting (NAT) to 103.44.119.75:5060 --->
> SIP/2.0 100 Trying
> Via: SIP/2.0/UDP 103.44.119.75:5060;branch=z9hG4bK-d8754z-da79a9700f3bb461-1---d8754z-;received=103.44.119.75;rport=5060
> From: "1010"<sip:1010@13.126.130.235:5060>;tag=f075701e
> To: <sip:1000@13.126.130.235:5060>
> Call-ID: YThmMmU1MGQ5NDFhZGQzNzUwYjRlZTg0MjhhOGYzMjU.
> CSeq: 1 INVITE
> Server: Asterisk PBX 13.22.0
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
> Supported: replaces, timer
> Contact: <sip:1000@172.26.5.241:5060>
> Content-Length: 0
>
>
> <------------>
> -- Executing [1000@lasttry:1] NoOp("SIP/1010-00000033", "First LIne") in new stack
> -- Executing [1000@lasttry:2] NoOp("SIP/1010-00000033", "Second Line") in new stack
> -- Executing [1000@lasttry:3] Dial("SIP/1010-00000033", "SIP/7000") in new stack
> == Using SIP RTP CoS mark 5
> Audio is at 13228
> Adding codec ulaw to SDP
> Adding codec alaw to SDP
> Adding non-codec 0x1 (telephone-event) to SDP
> Reliably Transmitting (NAT) to 103.44.119.75:35417:
> INVITE sip:7000@103.44.119.75:35417;rinstance=4e47afd124f34ea0;transport=UDP SIP/2.0
> Via: SIP/2.0/UDP 172.26.5.241:5060;branch=z9hG4bK177f5a6c;rport
> Max-Forwards: 70
> From: "1010" <sip:1010@172.26.5.241>;tag=as01637e19
> To: <sip:7000@103.44.119.75:35417;rinstance=4e47afd124f34ea0;transport=UDP>
> Contact: <sip:1010@172.26.5.241:5060>
> Call-ID: 4f6357ac236db63879213ab332564250@172.26.5.241:5060
> CSeq: 102 INVITE
> User-Agent: Asterisk PBX 13.22.0
> Date: Thu, 16 Aug 2018 10:37:45 GMT
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
> Supported: replaces, timer
> Content-Type: application/sdp
> Content-Length: 262
>
> v=0
> o=root 595161832 595161832 IN IP4 172.26.5.241
> s=Asterisk PBX 13.22.0
> c=IN IP4 172.26.5.241
> t=0 0
> m=audio 13228 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=maxptime:150
> a=sendrecv
>
> ---
> -- Called SIP/7000
>
> <--- SIP read from UDP:103.44.119.75:35417 --->
> SIP/2.0 100 Trying
> Via: SIP/2.0/UDP 172.26.5.241:5060;branch=z9hG4bK177f5a6c;rport=5060;received=13.126.130.235
> To: <sip:7000@103.44.119.75:35417;rinstance=4e47afd124f34ea0;transport=UDP>
> From: "1010" <sip:1010@172.26.5.241>;tag=as01637e19
> Call-ID: 4f6357ac236db63879213ab332564250@172.26.5.241:5060
> CSeq: 102 INVITE
> Content-Length: 0
>
> <------------->
> --- (7 headers 0 lines) ---
>
> <--- SIP read from UDP:103.44.119.75:35417 --->
> SIP/2.0 180 Ringing
> Via: SIP/2.0/UDP 172.26.5.241:5060;branch=z9hG4bK177f5a6c;rport=5060;received=13.126.130.235
> Contact: <sip:7000@103.44.119.75:35417>
> To: <sip:7000@103.44.119.75:35417;rinstance=4e47afd124f34ea0;transport=UDP>;tag=a7472331
> From: "1010" <sip:1010@172.26.5.241>;tag=as01637e19
> Call-ID: 4f6357ac236db63879213ab332564250@172.26.5.241:5060
> CSeq: 102 INVITE
> Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
> User-Agent: Z 5.2.16 rv2.8.95
> Allow-Events: presence, kpml, talk
> Content-Length: 0
>
> <------------->
> --- (11 headers 0 lines) ---
> sip_route_dump: route/path hop: <sip:7000@103.44.119.75:35417>
> -- SIP/7000-00000034 is ringing
>
> <--- Transmitting (NAT) to 103.44.119.75:5060 --->
> SIP/2.0 180 Ringing
> Via: SIP/2.0/UDP 103.44.119.75:5060;branch=z9hG4bK-d8754z-da79a9700f3bb461-1---d8754z-;received=103.44.119.75;rport=5060
> From: "1010"<sip:1010@13.126.130.235:5060>;tag=f075701e
> To: <sip:1000@13.126.130.235:5060>;tag=as4ec0d955
> Call-ID: YThmMmU1MGQ5NDFhZGQzNzUwYjRlZTg0MjhhOGYzMjU.
> CSeq: 1 INVITE
> Server: Asterisk PBX 13.22.0
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
> Supported: replaces, timer
> Contact: <sip:1000@172.26.5.241:5060>
> Content-Length: 0
>
>
> <------------>
>
> <--- SIP read from UDP:103.44.119.75:35417 --->
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 172.26.5.241:5060;branch=z9hG4bK177f5a6c;rport=5060;received=13.126.130.235
> Contact: <sip:7000@103.44.119.75:35417>
> To: <sip:7000@103.44.119.75:35417;rinstance=4e47afd124f34ea0;transport=UDP>;tag=a7472331
> From: "1010" <sip:1010@172.26.5.241>;tag=as01637e19
> Call-ID: 4f6357ac236db63879213ab332564250@172.26.5.241:5060
> CSeq: 102 INVITE
> Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
> Content-Type: application/sdp
> User-Agent: Z 5.2.16 rv2.8.95
> Allow-Events: presence, kpml, talk
> Content-Length: 602
>
> v=0
> o=Z 0 1 IN IP4 103.44.119.75
> s=Z
> c=IN IP4 103.44.119.75
> t=0 0
> m=audio 8000 RTP/AVP 0 106 9 3 111 8 97 110 112 102 101 98 100 99
> a=rtpmap:106 opus/48000/2
> a=fmtp:106 minptime=20; cbr=1; maxaveragebitrate=40000; useinbandfec=1
> a=rtpmap:111 speex/16000
> a=rtpmap:97 iLBC/8000
> a=fmtp:97 mode=20
> a=rtpmap:110 speex/8000
> a=rtpmap:112 speex/32000
> a=rtpmap:102 G726-32/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=rtpmap:98 telephone-event/48000
> a=fmtp:98 0-16
> a=rtpmap:100 telephone-event/16000
> a=fmtp:100 0-16
> a=rtpmap:99 telephone-event/32000
> a=fmtp:99 0-16
> a=sendrecv
> <------------->
> --- (12 headers 23 lines) ---
> Found RTP audio format 0
> Found RTP audio format 106
> Found RTP audio format 9
> Found RTP audio format 3
> Found RTP audio format 111
> Found RTP audio format 8
> Found RTP audio format 97
> Found RTP audio format 110
> Found RTP audio format 112
> Found RTP audio format 102
> Found RTP audio format 101
> Found RTP audio format 98
> Found RTP audio format 100
> Found RTP audio format 99
> Found audio description format opus for ID 106
> Found audio description format speex for ID 111
> Found audio description format iLBC for ID 97
> Found audio description format speex for ID 110
> Found audio description format speex for ID 112
> Found audio description format G726-32 for ID 102
> Found audio description format telephone-event for ID 101
> Found unknown media description format telephone-event for ID 98
> Found unknown media description format telephone-event for ID 100
> Found unknown media description format telephone-event for ID 99
> Capabilities: us - (ulaw|alaw), peer - audio=(ulaw|gsm|alaw|g722|ilbc|g726|opus|speex|speex16|speex32)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
> Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
> > 0x7fac5400af80 -- Strict RTP learning after remote address set to: 103.44.119.75:8000
> Peer audio RTP is at port 103.44.119.75:8000
> sip_route_dump: route/path hop: <sip:7000@103.44.119.75:35417>
> Transmitting (NAT) to 103.44.119.75:35417:
> ACK sip:7000@103.44.119.75:35417 SIP/2.0
> Via: SIP/2.0/UDP 172.26.5.241:5060;branch=z9hG4bK05553d4c;rport
> Max-Forwards: 70
> From: "1010" <sip:1010@172.26.5.241>;tag=as01637e19
> To: <sip:7000@103.44.119.75:35417;rinstance=4e47afd124f34ea0;transport=UDP>;tag=a7472331
> Contact: <sip:1010@172.26.5.241:5060>
> Call-ID: 4f6357ac236db63879213ab332564250@172.26.5.241:5060
> CSeq: 102 ACK
> User-Agent: Asterisk PBX 13.22.0
> Content-Length: 0
>
>
> ---
> -- SIP/7000-00000034 answered SIP/1010-00000033
> Audio is at 19216
> Adding codec ulaw to SDP
> Adding codec alaw to SDP
> Adding non-codec 0x1 (telephone-event) to SDP
>
> <--- Reliably Transmitting (NAT) to 103.44.119.75:5060 --->
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 103.44.119.75:5060;branch=z9hG4bK-d8754z-da79a9700f3bb461-1---d8754z-;received=103.44.119.75;rport=5060
> From: "1010"<sip:1010@13.126.130.235:5060>;tag=f075701e
> To: <sip:1000@13.126.130.235:5060>;tag=as4ec0d955
> Call-ID: YThmMmU1MGQ5NDFhZGQzNzUwYjRlZTg0MjhhOGYzMjU.
> CSeq: 1 INVITE
> Server: Asterisk PBX 13.22.0
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
> Supported: replaces, timer
> Contact: <sip:1000@172.26.5.241:5060>
> Content-Type: application/sdp
> Content-Length: 286
>
> v=0
> o=root 1504670142 1504670142 IN IP4 172.26.5.241
> s=Asterisk PBX 13.22.0
> c=IN IP4 172.26.5.241
> t=0 0
> m=audio 19216 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=maxptime:150
> a=sendrecv
> m=video 0 RTP/AVP 34
>
> <------------>
> -- Channel SIP/7000-00000034 joined 'simple_bridge' basic-bridge <618e2d17-a2a8-43d3-9f92-93a4ed2bf723>
> -- Channel SIP/1010-00000033 joined 'simple_bridge' basic-bridge <618e2d17-a2a8-43d3-9f92-93a4ed2bf723>
> > Bridge 618e2d17-a2a8-43d3-9f92-93a4ed2bf723: switching from simple_bridge technology to native_rtp
> > Locally RTP bridged 'SIP/1010-00000033' and 'SIP/7000-00000034' in stack
> Retransmitting #1 (NAT) to 103.44.119.75:5060:
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 103.44.119.75:5060;branch=z9hG4bK-d8754z-da79a9700f3bb461-1---d8754z-;received=103.44.119.75;rport=5060
> From: "1010"<sip:1010@13.126.130.235:5060>;tag=f075701e
> To: <sip:1000@13.126.130.235:5060>;tag=as4ec0d955
> Call-ID: YThmMmU1MGQ5NDFhZGQzNzUwYjRlZTg0MjhhOGYzMjU.
> CSeq: 1 INVITE
> Server: Asterisk PBX 13.22.0
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
> Supported: replaces, timer
> Contact: <sip:1000@172.26.5.241:5060>
> Content-Type: application/sdp
> Content-Length: 286
>
> v=0
> o=root 1504670142 1504670142 IN IP4 172.26.5.241
> s=Asterisk PBX 13.22.0
> c=IN IP4 172.26.5.241
> t=0 0
> m=audio 19216 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=maxptime:150
> a=sendrecv
> m=video 0 RTP/AVP 34
>
> ---
> Retransmitting #2 (NAT) to 103.44.119.75:5060:
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 103.44.119.75:5060;branch=z9hG4bK-d8754z-da79a9700f3bb461-1---d8754z-;received=103.44.119.75;rport=5060
> From: "1010"<sip:1010@13.126.130.235:5060>;tag=f075701e
> To: <sip:1000@13.126.130.235:5060>;tag=as4ec0d955
> Call-ID: YThmMmU1MGQ5NDFhZGQzNzUwYjRlZTg0MjhhOGYzMjU.
> CSeq: 1 INVITE
> Server: Asterisk PBX 13.22.0
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
> Supported: replaces, timer
> Contact: <sip:1000@172.26.5.241:5060>
> Content-Type: application/sdp
> Content-Length: 286
>
> v=0
> o=root 1504670142 1504670142 IN IP4 172.26.5.241
> s=Asterisk PBX 13.22.0
> c=IN IP4 172.26.5.241
> t=0 0
> m=audio 19216 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=maxptime:150
> a=sendrecv
> m=video 0 RTP/AVP 34
>
> ---
> Retransmitting #3 (NAT) to 103.44.119.75:5060:
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 103.44.119.75:5060;branch=z9hG4bK-d8754z-da79a9700f3bb461-1---d8754z-;received=103.44.119.75;rport=5060
> From: "1010"<sip:1010@13.126.130.235:5060>;tag=f075701e
> To: <sip:1000@13.126.130.235:5060>;tag=as4ec0d955
> Call-ID: YThmMmU1MGQ5NDFhZGQzNzUwYjRlZTg0MjhhOGYzMjU.
> CSeq: 1 INVITE
> Server: Asterisk PBX 13.22.0
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
> Supported: replaces, timer
> Contact: <sip:1000@172.26.5.241:5060>
> Content-Type: application/sdp
> Content-Length: 286
>
> v=0
> o=root 1504670142 1504670142 IN IP4 172.26.5.241
> s=Asterisk PBX 13.22.0
> c=IN IP4 172.26.5.241
> t=0 0
> m=audio 19216 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=maxptime:150
> a=sendrecv
> m=video 0 RTP/AVP 34
>
> ---
> Retransmitting #4 (NAT) to 103.44.119.75:5060:
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 103.44.119.75:5060;branch=z9hG4bK-d8754z-da79a9700f3bb461-1---d8754z-;received=103.44.119.75;rport=5060
> From: "1010"<sip:1010@13.126.130.235:5060>;tag=f075701e
> To: <sip:1000@13.126.130.235:5060>;tag=as4ec0d955
> Call-ID: YThmMmU1MGQ5NDFhZGQzNzUwYjRlZTg0MjhhOGYzMjU.
> CSeq: 1 INVITE
> Server: Asterisk PBX 13.22.0
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
> Supported: replaces, timer
> Contact: <sip:1000@172.26.5.241:5060>
> Content-Type: application/sdp
> Content-Length: 286
>
> v=0
> o=root 1504670142 1504670142 IN IP4 172.26.5.241
> s=Asterisk PBX 13.22.0
> c=IN IP4 172.26.5.241
> t=0 0
> m=audio 19216 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=maxptime:150
> a=sendrecv
> m=video 0 RTP/AVP 34
>
> ---
> Retransmitting #5 (NAT) to 103.44.119.75:5060:
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 103.44.119.75:5060;branch=z9hG4bK-d8754z-da79a9700f3bb461-1---d8754z-;received=103.44.119.75;rport=5060
> From: "1010"<sip:1010@13.126.130.235:5060>;tag=f075701e
> To: <sip:1000@13.126.130.235:5060>;tag=as4ec0d955
> Call-ID: YThmMmU1MGQ5NDFhZGQzNzUwYjRlZTg0MjhhOGYzMjU.
> CSeq: 1 INVITE
> Server: Asterisk PBX 13.22.0
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
> Supported: replaces, timer
> Contact: <sip:1000@172.26.5.241:5060>
> Content-Type: application/sdp
> Content-Length: 286
>
> v=0
> o=root 1504670142 1504670142 IN IP4 172.26.5.241
> s=Asterisk PBX 13.22.0
> c=IN IP4 172.26.5.241
> t=0 0
> m=audio 19216 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=maxptime:150
> a=sendrecv
> m=video 0 RTP/AVP 34
>
> ---
> Retransmitting #6 (NAT) to 103.44.119.75:5060:
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 103.44.119.75:5060;branch=z9hG4bK-d8754z-da79a9700f3bb461-1---d8754z-;received=103.44.119.75;rport=5060
> From: "1010"<sip:1010@13.126.130.235:5060>;tag=f075701e
> To: <sip:1000@13.126.130.235:5060>;tag=as4ec0d955
> Call-ID: YThmMmU1MGQ5NDFhZGQzNzUwYjRlZTg0MjhhOGYzMjU.
> CSeq: 1 INVITE
> Server: Asterisk PBX 13.22.0
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
> Supported: replaces, timer
> Contact: <sip:1000@172.26.5.241:5060>
> Content-Type: application/sdp
> Content-Length: 286
>
> v=0
> o=root 1504670142 1504670142 IN IP4 172.26.5.241
> s=Asterisk PBX 13.22.0
> c=IN IP4 172.26.5.241
> t=0 0
> m=audio 19216 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=maxptime:150
> a=sendrecv
> m=video 0 RTP/AVP 34
>
> ---
> [Aug 16 10:38:02] WARNING[1305]: chan_sip.c:4068 retrans_pkt: Retransmission timeout reached on transmission YThmMmU1MGQ5NDFhZGQzNzUwYjRlZTg0MjhhOGYzMjU. for seqno 1 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
> Packet timed out after 6976ms with no response
> [Aug 16 10:38:02] WARNING[1305]: chan_sip.c:4092 retrans_pkt: Hanging up call YThmMmU1MGQ5NDFhZGQzNzUwYjRlZTg0MjhhOGYzMjU. - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).
> -- Channel SIP/1010-00000033 left 'native_rtp' basic-bridge <618e2d17-a2a8-43d3-9f92-93a4ed2bf723>
> == Spawn extension (lasttry, 1000, 3) exited non-zero on 'SIP/1010-00000033'
> Scheduling destruction of SIP dialog 'YThmMmU1MGQ5NDFhZGQzNzUwYjRlZTg0MjhhOGYzMjU.' in 6976 ms (Method: INVITE)
> Reliably Transmitting (NAT) to 103.44.119.75:5060:
> BYE sip:1010@103.44.119.75:5060;transport=UDP SIP/2.0
> Via: SIP/2.0/UDP 172.26.5.241:5060;branch=z9hG4bK5a3572ae;rport
> Max-Forwards: 70
> From: <sip:1000@13.126.130.235:5060>;tag=as4ec0d955
> To: "1010"<sip:1010@13.126.130.235:5060>;tag=f075701e
> Call-ID: YThmMmU1MGQ5NDFhZGQzNzUwYjRlZTg0MjhhOGYzMjU.
> CSeq: 102 BYE
> User-Agent: Asterisk PBX 13.22.0
> X-Asterisk-HangupCause: No user responding
> X-Asterisk-HangupCauseCode: 18
> Content-Length: 0
>
>
> ---
> -- Channel SIP/7000-00000034 left 'native_rtp' basic-bridge <618e2d17-a2a8-43d3-9f92-93a4ed2bf723>
> Scheduling destruction of SIP dialog '4f6357ac236db63879213ab332564250@172.26.5.241:5060' in 6400 ms (Method: INVITE)
> Reliably Transmitting (NAT) to 103.44.119.75:35417:
> BYE sip:7000@103.44.119.75:35417 SIP/2.0
> Via: SIP/2.0/UDP 172.26.5.241:5060;branch=z9hG4bK66122094;rport
> Max-Forwards: 70
> From: "1010" <sip:1010@172.26.5.241>;tag=as01637e19
> To: <sip:7000@103.44.119.75:35417;rinstance=4e47afd124f34ea0;transport=UDP>;tag=a7472331
> Call-ID: 4f6357ac236db63879213ab332564250@172.26.5.241:5060
> CSeq: 103 BYE
> User-Agent: Asterisk PBX 13.22.0
> X-Asterisk-HangupCause: No user responding
> X-Asterisk-HangupCauseCode: 18
> Content-Length: 0
>
>
> ---
>
> <--- SIP read from UDP:103.44.119.75:35417 --->
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 172.26.5.241:5060;branch=z9hG4bK66122094;rport=5060;received=13.126.130.235
> Contact: <sip:7000@103.44.119.75:35417>
> To: <sip:7000@103.44.119.75:35417;rinstance=4e47afd124f34ea0;transport=UDP>;tag=a7472331
> From: "1010" <sip:1010@172.26.5.241>;tag=as01637e19
> Call-ID: 4f6357ac236db63879213ab332564250@172.26.5.241:5060
> CSeq: 103 BYE
> User-Agent: Z 5.2.16 rv2.8.95
> Content-Length: 0
>
> <------------->
> --- (9 headers 0 lines) ---
> Really destroying SIP dialog '4f6357ac236db63879213ab332564250@172.26.5.241:5060' Method: INVITE
>
> <--- SIP read from UDP:103.44.119.75:5060 --->
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 172.26.5.241:5060;branch=z9hG4bK5a3572ae;rport=5060;received=13.126.130.235
> Contact: <sip:1010@103.44.119.75:5060;transport=UDP>
> To: "1010"<sip:1010@13.126.130.235:5060>;tag=f075701e
> From: <sip:1000@13.126.130.235:5060>;tag=as4ec0d955
> Call-ID: YThmMmU1MGQ5NDFhZGQzNzUwYjRlZTg0MjhhOGYzMjU.
> CSeq: 102 BYE
> User-Agent: 3CXPhone 6.0.26523.0
> Content-Length: 0
>
> <------------->
> --- (9 headers 0 lines) ---
> SIP Response message for INCOMING dialog BYE arrived
> Really destroying SIP dialog 'YThmMmU1MGQ5NDFhZGQzNzUwYjRlZTg0MjhhOGYzMjU.' Method: INVITE
> Really destroying SIP dialog 'YjhmMjY3YjFjZmE2ZTUyYmRmZWEzZDg4ZDhmNWJhZmM.' Method: REGISTER
>
> <--- SIP read from UDP:103.44.119.75:5060 --->
>
>
> <------------->
>
> <--- SIP read from UDP:103.44.119.75:35417 --->
>
>
> <------------->
> Really destroying SIP dialog 'ms4D5NTtdGELRcPzGt4wWA..' Method: REGISTER