hello folks .
this is a continued to the post
now i have new issue .
i was able to have call working great from webrtc to sip phone .
but the problem now is when the call from webrtc—> sip phone
or from webrtc---->webrtc .
the call ring ok , and other side answer it and the two ends hear others .
but the problem is the call keep rising on the peer who received the call
so his call is mixed of ringing tone and other peer .
i will post the call when from sip phone —> webrtc
below is call from sip 100 ----> webrtc 300 … then 100 hanged
_> mudshare6CLI> _
_> mudshare6CLI> _
> <— SIP read from UDP:176.58.74.132:13240 —>
> INVITE sip:300@64.37.115.36 SIP/2.0
> Via: SIP/2.0/UDP 176.58.74.132:13240;branch=z9hG4bK-524287-1—df122163a7977477;rport
> Max-Forwards: 70
> Contact: sip:100@176.58.74.132:13240
> To: sip:300@64.37.115.36
> From: sip:100@64.37.115.36;tag=78ad7914
> Call-ID: 81140MGZlNTZiNDI1NDgxZjFhMmU4Yzk2OTdkMTQ2NzkzNjM
> CSeq: 1 INVITE
> Allow: SUBSCRIBE, NOTIFY, INVITE, ACK, CANCEL, BYE, REFER, INFO, OPTIONS, MESSAGE
> Content-Type: application/sdp
> Supported: replaces
> User-Agent: X-Lite release 4.9.5 stamp 81140
> Content-Length: 771
> v=0
> o=- 1483124364612465 1 IN IP4 176.58.74.132
> s=X-Lite release 4.9.5 stamp 81140
> c=IN IP4 176.58.74.132
> t=0 0
> a=ice-ufrag:ca424a
> a=ice-pwd:2cba9c10a7909b67f35b1e52ebf64593
> m=audio 14180 RTP/AVP 9 8 85 120 0 3 101
> a=rtpmap:85 speex/8000
> a=rtpmap:120 opus/48000/2
> a=fmtp:120 useinbandfec=1; usedtx=1; maxaveragebitrate=64000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
> a=sendrecv
> a=rtcp:14181 IN IP4 176.58.74.132
> a=candidate:1 1 UDP 659136 192.168.1.103 54764 typ host
> a=candidate:2 1 UDP 659084 176.58.74.132 14178 typ srflx raddr 192.168.1.103 rport 54764
> a=candidate:1 2 UDP 659134 192.168.1.103 54765 typ host
> a=candidate:2 2 UDP 659082 176.58.74.132 14179 typ srflx raddr 192.168.1.103 rport 54765
> a=ssrc:2961148063 cname:d4Y+u7za/p1se6Sr
> <------------->
> — (13 headers 20 lines) —
> Sending to 176.58.74.132:13240 (NAT)
> Sending to 176.58.74.132:13240 (NAT)
> Using INVITE request as basis request - 81140MGZlNTZiNDI1NDgxZjFhMmU4Yzk2OTdkMTQ2NzkzNjM
> Found peer ‘100’ for ‘100’ from 176.58.74.132:13240
> <— Reliably Transmitting (NAT) to 176.58.74.132:13240 —>
> SIP/2.0 401 Unauthorized
> Via: SIP/2.0/UDP 176.58.74.132:13240;branch=z9hG4bK-524287-1—df122163a7977477;received=176.58.74.132;rport=13240
> From: sip:100@64.37.115.36;tag=78ad7914
> To: sip:300@64.37.115.36;tag=as238664f6
> Call-ID: 81140MGZlNTZiNDI1NDgxZjFhMmU4Yzk2OTdkMTQ2NzkzNjM
> CSeq: 1 INVITE
> Server: FPBX-12.0.76.4(13.13.1)
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
> Supported: replaces, timer
> WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce=“38b52919”
> Content-Length: 0
> <------------>
> Scheduling destruction of SIP dialog ‘81140MGZlNTZiNDI1NDgxZjFhMmU4Yzk2OTdkMTQ2NzkzNjM’ in 12672 ms (Method: INVITE)
> <— SIP read from UDP:176.58.74.132:13240 —>
> ACK sip:300@64.37.115.36 SIP/2.0
> Via: SIP/2.0/UDP 176.58.74.132:13240;branch=z9hG4bK-524287-1—df122163a7977477;rport
> Max-Forwards: 70
> To: sip:300@64.37.115.36;tag=as238664f6
> From: sip:100@64.37.115.36;tag=78ad7914
> Call-ID: 81140MGZlNTZiNDI1NDgxZjFhMmU4Yzk2OTdkMTQ2NzkzNjM
> CSeq: 1 ACK
> Content-Length: 0
> <------------->
> — (8 headers 0 lines) —
> <— SIP read from UDP:176.58.74.132:13240 —>
> INVITE sip:300@64.37.115.36 SIP/2.0
> Via: SIP/2.0/UDP 176.58.74.132:13240;branch=z9hG4bK-524287-1—12f927446f5d485a;rport
> Max-Forwards: 70
> Contact: sip:100@176.58.74.132:13240
> To: sip:300@64.37.115.36
> From: sip:100@64.37.115.36;tag=78ad7914
> Call-ID: 81140MGZlNTZiNDI1NDgxZjFhMmU4Yzk2OTdkMTQ2NzkzNjM
> CSeq: 2 INVITE
> Allow: SUBSCRIBE, NOTIFY, INVITE, ACK, CANCEL, BYE, REFER, INFO, OPTIONS, MESSAGE
> Content-Type: application/sdp
> Supported: replaces
> User-Agent: X-Lite release 4.9.5 stamp 81140
> Authorization: Digest username=“100”,realm=“asterisk”,nonce=“38b52919”,uri="sip:300@64.37.115.36",response=“bd37ec16d1ba1c41e1d3abfff5cb3537”,algorithm=MD5
> Content-Length: 771
> v=0
> o=- 1483124364612465 1 IN IP4 176.58.74.132
> s=X-Lite release 4.9.5 stamp 81140
> c=IN IP4 176.58.74.132
> t=0 0
> a=ice-ufrag:ca424a
> a=ice-pwd:2cba9c10a7909b67f35b1e52ebf64593
> m=audio 14180 RTP/AVP 9 8 85 120 0 3 101
> a=rtpmap:85 speex/8000
> a=rtpmap:120 opus/48000/2
> a=fmtp:120 useinbandfec=1; usedtx=1; maxaveragebitrate=64000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
> a=sendrecv
> a=rtcp:14181 IN IP4 176.58.74.132
> a=candidate:1 1 UDP 659136 192.168.1.103 54764 typ host
> a=candidate:2 1 UDP 659084 176.58.74.132 14178 typ srflx raddr 192.168.1.103 rport 54764
> a=candidate:1 2 UDP 659134 192.168.1.103 54765 typ host
> a=candidate:2 2 UDP 659082 176.58.74.132 14179 typ srflx raddr 192.168.1.103 rport 54765
> a=ssrc:2961148063 cname:d4Y+u7za/p1se6Sr
> <------------->
> — (14 headers 20 lines) —
> Sending to 176.58.74.132:13240 (NAT)
> Using INVITE request as basis request - 81140MGZlNTZiNDI1NDgxZjFhMmU4Yzk2OTdkMTQ2NzkzNjM
> Found peer ‘100’ for ‘100’ from 176.58.74.132:13240
> == Using SIP RTP TOS bits 184
> == Using SIP RTP CoS mark 5
> Found RTP audio format 9
> Found RTP audio format 8
> Found RTP audio format 85
> Found RTP audio format 120
> Found RTP audio format 0
> Found RTP audio format 3
> Found RTP audio format 101
> Found audio description format speex for ID 85
> Found audio description format opus for ID 120
> Found audio description format telephone-event for ID 101
> Capabilities: us - (ulaw|alaw|gsm|g726), peer - audio=(ulaw|gsm|alaw|g722|speex|opus)/video=(nothing)/text=(nothing), combined - (ulaw|alaw|gsm)
> Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
> Peer audio RTP is at port 176.58.74.132:14180
> Looking for 300 in from-internal (domain 64.37.115.36)
> sip_route_dump: route/path hop: sip:100@176.58.74.132:13240
> <— Transmitting (NAT) to 176.58.74.132:13240 —>
> SIP/2.0 100 Trying
> Via: SIP/2.0/UDP 176.58.74.132:13240;branch=z9hG4bK-524287-1—12f927446f5d485a;received=176.58.74.132;rport=13240
> From: sip:100@64.37.115.36;tag=78ad7914
> To: sip:300@64.37.115.36
> Call-ID: 81140MGZlNTZiNDI1NDgxZjFhMmU4Yzk2OTdkMTQ2NzkzNjM
> CSeq: 2 INVITE
> Server: FPBX-12.0.76.4(13.13.1)
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
> Supported: replaces, timer
> Contact: sip:300@64.37.115.36:5061
> Content-Length: 0
> <------------>
> – Executing [300@from-internal:1] Set(“SIP/100-00000124”, “RINGTIMER=15") in new stack
> – Executing [300@from-internal:2] Macro(“SIP/100-00000124”, “exten-vm,novm,300,0,0,0”) in new stack
> – Executing [s@macro-exten-vm:1] Macro(“SIP/100-00000124”, “user-callerid,”) in new stack
> – Executing [s@macro-user-callerid:1] Set(“SIP/100-00000124”, “TOUCH_MONITOR=1483124364.2060”) in new stack
> – Executing [s@macro-user-callerid:2] Set(“SIP/100-00000124”, “AMPUSER=100”) in new stack
> – Executing [s@macro-user-callerid:3] GotoIf(“SIP/100-00000124”, “0?report”) in new stack
> – Executing [s@macro-user-callerid:4] ExecIf(“SIP/100-00000124”, “1?Set(REALCALLERIDNUM=100)”) in new stack
> – Executing [s@macro-user-callerid:5] Set(“SIP/100-00000124”, “AMPUSER=100”) in new stack
> – Executing [s@macro-user-callerid:6] GotoIf(“SIP/100-00000124”, “0?limit”) in new stack
> – Executing [s@macro-user-callerid:7] Set(“SIP/100-00000124”, “AMPUSERCIDNAME=ahmad”) in new stack"CALLCOMPLETION(cc_callback_macro)=ccss-default") in new stack
> [2016-12-30 13:59:24] WARNING[18333][C-0000010b]: ccss.c:1012 ast_set_cc_callback_macro: Usage of cc_callback_macro is deprecated. Please use cc_callback_sub instead.
> – Executing [agent_config@sub-ccss:6] ExecIf(“SIP/100-00000124”, “1?Set(CALLCOMPLETION(cc_recall_timer)=)”) in new stack
> – Executing [agent_config@sub-ccss:7] ExecIf(“SIP/100-00000124”, “1?Set(CALLCOMPLETION(cc_max_agents)=)”) in new stack
> – Executing [agent_config@sub-ccss:8] ExecIf(“SIP/100-00000124”, “0?Set(CALLCOMPLETION(cc_agent_dialstring)=Local/100_300@from-ccss-)”) in new stack
> – Executing [agent_config@sub-ccss:9] Set(“SIP/100-00000124”, “CALLCOMPLETION(cc_callback_macro)=ccss-default”) in new stack
> [2016-12-30 13:59:24] WARNING[18333][C-0000010b]: ccss.c:1012 ast_set_cc_callback_macro: Usage of cc_callback_macro is deprecated. Please use cc_callback_sub instead.
0000124”, “0?MacroExit()”) in new stack
_> – Executing [s@macro-exten-vm:8] Gosub(“SIP/100-00000124”, “sub-record-check,s,1(exten,300,dontcare)”)
> – Executing [s@macro-dial-one:44] ExecIf(“SIP/100-00000124”, “0?Set(D_OPTIONS=trII)”) in new stack
> – Executing [s@macro-dial-one:45] Dial(“SIP/100-00000124”, “SIP/99300&SIP/300,TtrI”) in new stack
> == DTLS ECDH initialized (secp256r1), faster PFS enabled
> == Using SIP RTP TOS bits 184
> == Using SIP RTP CoS mark 5
> [2016-12-30 13:59:25] WARNING[18333][C-0000010b]: app_dial.c:2525 dial_exec_full: Unable to create channel of type ‘SIP’ (cause 20 - Subscriber absent)
> Audio is at 19564
> Adding codec ulaw to SDP
> Adding codec alaw to SDP
> Adding codec gsm to SDP
> Adding codec g726 to SDP
> Adding non-codec 0x1 (telephone-event) to SDP
> Reliably Transmitting (no NAT) to 176.58.74.132:13179:
> INVITE sip:gjfia7uo@cp3kurllbgti.invalid;transport=ws SIP/2.0
> Via: SIP/2.0/WS 64.37.115.36:5061;branch=z9hG4bK50d05f53
> Max-Forwards: 70
> From: “ahmad” sip:100@64.37.115.36:5061;tag=as708085eb
> To: sip:gjfia7uo@cp3kurllbgti.invalid;transport=ws
> Contact: sip:100@64.37.115.36:5061;transport=WS
> Call-ID: 0c21caec323d0096787fa56a4c815838@64.37.115.36:5061
> CSeq: 102 INVITE
> User-Agent: FPBX-12.0.76.4(13.13.1)
> Date: Fri, 30 Dec 2016 18:59:25 GMT
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
> Supported: replaces, timer
> Content-Type: application/sdp
> Content-Length: 710
> v=0
> o=root 930509647 930509647 IN IP4 64.37.115.36
> s=Asterisk PBX 13.13.1
> c=IN IP4 64.37.115.36
> t=0 0
> m=audio 19564 RTP/SAVPF 0 8 3 111 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:3 GSM/8000
> a=rtpmap:111 G726-32/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=ptime:20
> a=maxptime:150
> a=ice-ufrag:480615cb32dd9aaf6b7566385ed9afaf
> a=ice-pwd:31c025e22b8df4443cf04615218f5ae0
> a=candidate:H40257324 1 UDP 2130706431 64.37.115.36 19564 typ host
> a=candidate:H40257324 2 UDP 2130706430 64.37.115.36 19565 typ host
> a=connection:new
> a=setup:actpass
> a=fingerprint:SHA-256 1A:49:FD:C2:53:76:8A:97:5C:2E:B3:A2:36:1A:CF:73:DF:A9:16:B0:E3:EC:D8:C9:82:6E:94:15:B1:4F:56:9C
> a=sendrecv
> —
> Really destroying SIP dialog ‘50f96ffe2dcd971215124f173f17d931@64.37.115.36:5061’ Method: INVITE
> – Called SIP/99300
> <— Transmitting (NAT) to 176.58.74.132:13240 —>
> SIP/2.0 180 Ringing
> Via: SIP/2.0/UDP 176.58.74.132:13240;branch=z9hG4bK-524287-1—12f927446f5d485a;received=176.58.74.132;rport=13240
> From: sip:100@64.37.115.36;tag=78ad7914
> To: sip:300@64.37.115.36;tag=as7e26555d
> Call-ID: 81140MGZlNTZiNDI1NDgxZjFhMmU4Yzk2OTdkMTQ2NzkzNjM
> CSeq: 2 INVITE
> Server: FPBX-12.0.76.4(13.13.1)
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
> Supported: replaces, timer
> Contact: sip:300@64.37.115.36:5061
> Content-Length: 0
> <------------>
> – Connected line update to SIP/100-00000124 prevented.
> <— SIP read from WS:176.58.74.132:13179 —>
> SIP/2.0 100 Trying
> Via: SIP/2.0/WS 64.37.115.36:5061;branch=z9hG4bK50d05f53
> To: sip:gjfia7uo@cp3kurllbgti.invalid;transport=ws
> From: “ahmad” sip:100@64.37.115.36:5061;tag=as708085eb
> Call-ID: 0c21caec323d0096787fa56a4c815838@64.37.115.36:5061
> CSeq: 102 INVITE
> Supported: timer,ice,outbound
> Content-Length: 0
> <------------->
> — (8 headers 0 lines) —
> <— SIP read from WS:176.58.74.132:13179 —>
> SIP/2.0 180 Ringing
> Via: SIP/2.0/WS 64.37.115.36:5061;branch=z9hG4bK50d05f53
> To: sip:gjfia7uo@cp3kurllbgti.invalid;transport=ws;tag=7glr0mkf31
> From: “ahmad” sip:100@64.37.115.36:5061;tag=as708085eb
> Call-ID: 0c21caec323d0096787fa56a4c815838@64.37.115.36:5061
> CSeq: 102 INVITE
> Contact: sip:gjfia7uo@cp3kurllbgti.invalid;transport=ws
> Supported: timer,ice,outbound
> Content-Length: 0
> <------------->
> — (9 headers 0 lines) —
> sip_route_dump: route/path hop: sip:gjfia7uo@cp3kurllbgti.invalid;transport=ws
> – SIP/99300-00000125 is ringing
> <— Transmitting (NAT) to 176.58.74.132:13240 —>
> SIP/2.0 180 Ringing
> Via: SIP/2.0/UDP 176.58.74.132:13240;branch=z9hG4bK-524287-1—12f927446f5d485a;received=176.58.74.132;rport=13240
> From: sip:100@64.37.115.36;tag=78ad7914
> To: sip:300@64.37.115.36;tag=as7e26555d
> Call-ID: 81140MGZlNTZiNDI1NDgxZjFhMmU4Yzk2OTdkMTQ2NzkzNjM
> CSeq: 2 INVITE
> Server: FPBX-12.0.76.4(13.13.1)
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
> Supported: replaces, timer
> Contact: sip:300@64.37.115.36:5061
> Content-Length: 0
> <------------>
> <— SIP read from WS:176.58.74.132:13179 —>
> SIP/2.0 200 OK
> Via: SIP/2.0/WS 64.37.115.36:5061;branch=z9hG4bK50d05f53
> To: sip:gjfia7uo@cp3kurllbgti.invalid;transport=ws;tag=7glr0mkf31
> From: “ahmad” sip:100@64.37.115.36:5061;tag=as708085eb
> Call-ID: 0c21caec323d0096787fa56a4c815838@64.37.115.36:5061
> CSeq: 102 INVITE
> Contact: sip:gjfia7uo@cp3kurllbgti.invalid;transport=ws
> Session-Expires: 90;refresher=uas
> Supported: timer,ice,outbound
> Content-Type: application/sdp
> Content-Length: 1479
> v=0
> o=mozilla…THIS_IS_SDPARTA-50.1.0 1506672437852858885 0 IN IP4 0.0.0.0
> s=-
> t=0 0
> a=sendrecv
> a=fingerprint:sha-256 74:96:48:49:FB:04:64:97:D2:C1:D9:A9:4D:85:5B:91:47:4F:01:DA:E8:2F:39:F3:8D:BB:33:A2:01:8A:7F:2A
> a=ice-options:trickle
> a=msid-semantic:WMS *
> m=audio 52205 RTP/SAVPF 0
> c=IN IP4 192.168.1.103
> a=candidate:0 1 UDP 2122252543 192.168.1.103 52205 typ host
> a=candidate:1 1 UDP 2122187007 172.16.168.1 55724 typ host
> a=candidate:2 1 UDP 2122121471 172.16.95.1 54277 typ host
> a=candidate:3 1 UDP 2122055935 192.168.46.1 52346 typ host
> a=candidate:4 1 UDP 2121990399 192.168.98.1 62215 typ host
> a=candidate:5 1 UDP 2121924863 172.16.119.1 61265 typ host
> a=candidate:6 1 UDP 2121859327 172.16.160.1 62140 typ host
> a=candidate:0 2 UDP 2122252542 192.168.1.103 56057 typ host
> a=candidate:1 2 UDP 2122187006 172.16.168.1 54350 typ host
> a=candidate:2 2 UDP 2122121470 172.16.95.1 62821 typ host
> a=candidate:3 2 UDP 2122055934 192.168.46.1 60870 typ host
> a=candidate:4 2 UDP 2121990398 192.168.98.1 60803 typ host
> a=candidate:5 2 UDP 2121924862 172.16.119.1 55711 typ host
> a=candidate:6 2 UDP 2121859326 172.16.160.1 65390 typ host
> a=sendrecv
> a=end-of-candidates
> a=ice-pwd:dc99cccb1a1ab5a67ea336881c550f1a
> a=ice-ufrag:d195d883
> a=msid:{91707436-784e-c342-8905-b3b073c3459d} {7df47b88-afa0-eb49-93e4-572f3aa8891e}
> a=rtcp:56057 IN IP4 192.168.1.103
> a=rtpmap:0 PCMU/8000
> a=setup:active
> a=ssrc:690537073 cname:{d76fe920-4144-1a43-a57f-a08acff46948}
> <------------->
> — (11 headers 33 lines) —
> Found RTP audio format 0
> Found audio description format PCMU for ID 0
> Capabilities: us - (ulaw|alaw|gsm|g726), peer - audio=(ulaw)/video=(nothing)/text=(nothing), combined - (ulaw)
> Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x0 (nothing), combined - 0x0 (nothing)
> Peer audio RTP is at port 192.168.1.103:52205
> sip_route_dump: route/path hop: sip:gjfia7uo@cp3kurllbgti.invalid;transport=ws
> set_destination: Parsing sip:gjfia7uo@cp3kurllbgti.invalid;transport=ws for address/port to send to
> set_destination: URI is for WebSocket, we can’t set destination
> Transmitting (no NAT) to 176.58.74.132:5060:
> ACK sip:gjfia7uo@cp3kurllbgti.invalid;transport=ws SIP/2.0
> Via: SIP/2.0/WS 64.37.115.36:5061;branch=z9hG4bK1f4b995f
> Max-Forwards: 70
> From: “ahmad” sip:100@64.37.115.36:5061;tag=as708085eb
> To: sip:gjfia7uo@cp3kurllbgti.invalid;transport=ws;tag=7glr0mkf31
> Contact: sip:100@64.37.115.36:5061;transport=WS
> Call-ID: 0c21caec323d0096787fa56a4c815838@64.37.115.36:5061
> CSeq: 102 ACK
> User-Agent: FPBX-12.0.76.4(13.13.1)
> Content-Length: 0
> —
> – Connected line update to SIP/100-00000124 prevented.
> – SIP/99300-00000125 answered SIP/100-00000124
> Audio is at 10536
> Adding codec ulaw to SDP
> Adding codec alaw to SDP
> Adding codec gsm to SDP
> Adding non-codec 0x1 (telephone-event) to SDP
> <— Reliably Transmitting (NAT) to 176.58.74.132:13240 —>
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 176.58.74.132:13240;branch=z9hG4bK-524287-1—12f927446f5d485a;received=176.58.74.132;rport=13240
> From: sip:100@64.37.115.36;tag=78ad7914
> To: sip:300@64.37.115.36;tag=as7e26555d
> Call-ID: 81140MGZlNTZiNDI1NDgxZjFhMmU4Yzk2OTdkMTQ2NzkzNjM
> CSeq: 2 INVITE
> Server: FPBX-12.0.76.4(13.13.1)
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
> Supported: replaces, timer
> Contact: sip:300@64.37.115.36:5061
> Content-Type: application/sdp
> Content-Length: 297
> v=0
> o=root 671133073 671133073 IN IP4 64.37.115.36
> s=Asterisk PBX 13.13.1
> c=IN IP4 64.37.115.36
> t=0 0
> m=audio 10536 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 0-16
> a=ptime:20
> a=maxptime:150
> a=sendrecv
> <------------>
> – Channel SIP/99300-00000125 joined ‘simple_bridge’ basic-bridge <05943e0c-7344-45a5-a3a8-ad333cecc5f6>
> – Channel SIP/100-00000124 joined ‘simple_bridge’ basic-bridge <05943e0c-7344-45a5-a3a8-ad333cecc5f6>
> Retransmitting #1 (NAT) to 176.58.74.132:13240:
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 176.58.74.132:13240;branch=z9hG4bK-524287-1—12f927446f5d485a;received=176.58.74.132;rport=13240
> From: sip:100@64.37.115.36;tag=78ad7914
> To: sip:300@64.37.115.36;tag=as7e26555d
> Call-ID: 81140MGZlNTZiNDI1NDgxZjFhMmU4Yzk2OTdkMTQ2NzkzNjM
> CSeq: 2 INVITE
> Server: FPBX-12.0.76.4(13.13.1)
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
> Supported: replaces, timer
> Contact: sip:300@64.37.115.36:5061
> Content-Type: application/sdp
> Content-Length: 297
> v=0
> o=root 671133073 671133073 IN IP4 64.37.115.36
> s=Asterisk PBX 13.13.1
> c=IN IP4 64.37.115.36
> t=0 0
> m=audio 10536 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 0-16
> a=ptime:20
> a=maxptime:150
> a=sendrecv
> —
> <— SIP read from UDP:176.58.74.132:1025 —>
> ACK sip:300@64.37.115.36:5061 SIP/2.0
> Via: SIP/2.0/UDP 176.58.74.132:13240;branch=z9hG4bK-524287-1—17063c503fa47677;rport
> Max-Forwards: 70
> Contact: sip:100@176.58.74.132:13240
> To: sip:300@64.37.115.36;tag=as7e26555d
> From: sip:100@64.37.115.36;tag=78ad7914
> Call-ID: 81140MGZlNTZiNDI1NDgxZjFhMmU4Yzk2OTdkMTQ2NzkzNjM
> CSeq: 2 ACK
> User-Agent: X-Lite release 4.9.5 stamp 81140
> Content-Length: 0
> <------------->
> — (10 headers 0 lines) —
> > 0x7ff88802f940 – Probation passed - setting RTP source address to 176.58.74.132:14178
> > 0x7ff88802f940 – Probation passed - setting RTP source address to 176.58.74.132:14178
> <— SIP read from UDP:176.58.74.132:1025 —>
> ACK sip:300@64.37.115.36:5061 SIP/2.0
> Via: SIP/2.0/UDP 176.58.74.132:13240;branch=z9hG4bK-524287-1—17063c503fa47677;rport
> Max-Forwards: 70
> Contact: sip:100@176.58.74.132:13240
> To: sip:300@64.37.115.36;tag=as7e26555d
> From: sip:100@64.37.115.36;tag=78ad7914
> Call-ID: 81140MGZlNTZiNDI1NDgxZjFhMmU4Yzk2OTdkMTQ2NzkzNjM
> CSeq: 2 ACK
> User-Agent: X-Lite release 4.9.5 stamp 81140
> Content-Length: 0
> <------------->
> — (10 headers 0 lines) —
> <— SIP read from UDP:176.58.74.132:1025 —>
> INVITE sip:300@64.37.115.36:5061 SIP/2.0
> Via: SIP/2.0/UDP 176.58.74.132:13240;branch=z9hG4bK-524287-1—5255c1162cd30015;rport
> Max-Forwards: 70
> Contact: sip:100@176.58.74.132:13240
> To: sip:300@64.37.115.36;tag=as7e26555d
> From: sip:100@64.37.115.36;tag=78ad7914
> Call-ID: 81140MGZlNTZiNDI1NDgxZjFhMmU4Yzk2OTdkMTQ2NzkzNjM
> CSeq: 3 INVITE
> Allow: SUBSCRIBE, NOTIFY, INVITE, ACK, CANCEL, BYE, REFER, INFO, OPTIONS, MESSAGE
> Content-Type: application/sdp
> Supported: replaces
> User-Agent: X-Lite release 4.9.5 stamp 81140
> Authorization: Digest username=“100”,realm=“asterisk”,nonce=“38b52919”,uri=“sip:300@64.37.115.36:5061”,response=“da9f6d317f0ad4e7ff1d0b300b355064”,algorithm=MD5
> Content-Length: 378
> v=0
> o=- 1483124364612465 2 IN IP4 176.58.74.132
> s=X-Lite release 4.9.5 stamp 81140
> c=IN IP4 176.58.74.132
> t=0 0
> m=audio 14180 RTP/AVP 9 8 85 120 0 3 101
> a=rtpmap:85 speex/8000
> a=rtpmap:120 opus/48000/2
> a=fmtp:120 useinbandfec=1; usedtx=1; maxaveragebitrate=64000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
> a=ssrc:2961148063 cname:d4Y+u7za/p1se6Sr
> a=sendrecv
> <------------->
> — (14 headers 13 lines) —
> Sending to 176.58.74.132:1025 (NAT)
> Found RTP audio format 9
> Found RTP audio format 8
> Found RTP audio format 85
> Found RTP audio format 120
> Found RTP audio format 0
> Found RTP audio format 3
> Found RTP audio format 101
> Found audio description format speex for ID 85
> Found audio description format opus for ID 120
> Found audio description format telephone-event for ID 101
> Capabilities: us - (ulaw|alaw|gsm|g726), peer - audio=(ulaw|gsm|alaw|g722|speex|opus)/video=(nothing)/text=(nothing), combined - (ulaw|alaw|gsm)
> Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
> Peer audio RTP is at port 176.58.74.132:14180
> <— Transmitting (NAT) to 176.58.74.132:1025 —>
> SIP/2.0 100 Trying
> Via: SIP/2.0/UDP 176.58.74.132:13240;branch=z9hG4bK-524287-1—5255c1162cd30015;received=176.58.74.132;rport=1025
> From: sip:100@64.37.115.36;tag=78ad7914
> To: sip:300@64.37.115.36;tag=as7e26555d
> Call-ID: 81140MGZlNTZiNDI1NDgxZjFhMmU4Yzk2OTdkMTQ2NzkzNjM
> CSeq: 3 INVITE
> Server: FPBX-12.0.76.4(13.13.1)
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
> Supported: replaces, timer
> Contact: sip:300@64.37.115.36:5061
> Content-Length: 0
> <------------>
> Audio is at 10536
> Adding codec ulaw to SDP
> Adding codec alaw to SDP
> Adding codec gsm to SDP
> Adding non-codec 0x1 (telephone-event) to SDP
> <— Reliably Transmitting (NAT) to 176.58.74.132:1025 —>
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 176.58.74.132:13240;branch=z9hG4bK-524287-1—5255c1162cd30015;received=176.58.74.132;rport=1025
> From: sip:100@64.37.115.36;tag=78ad7914
> To: sip:300@64.37.115.36;tag=as7e26555d
> Call-ID: 81140MGZlNTZiNDI1NDgxZjFhMmU4Yzk2OTdkMTQ2NzkzNjM
> CSeq: 3 INVITE
> Server: FPBX-12.0.76.4(13.13.1)
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
> Supported: replaces, timer
> Contact: sip:300@64.37.115.36:5061
> Content-Type: application/sdp
> Content-Length: 297
> v=0
> o=root 671133073 671133074 IN IP4 64.37.115.36
> s=Asterisk PBX 13.13.1
> c=IN IP4 64.37.115.36
> t=0 0
> m=audio 10536 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 0-16
> a=ptime:20
> a=maxptime:150
> a=sendrecv
> <------------>
> > 0x7ff89c29b1c0 – Probation passed - setting RTP source address to 176.58.74.132:14184
> > 0x7ff88802f940 – Probation passed - setting RTP source address to 176.58.74.132:14178
> > 0x7ff88802f940 – Probation passed - setting RTP source address to 176.58.74.132:14178
> Retransmitting #1 (NAT) to 176.58.74.132:1025:
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 176.58.74.132:13240;branch=z9hG4bK-524287-1—5255c1162cd30015;received=176.58.74.132;rport=1025
> From: sip:100@64.37.115.36;tag=78ad7914
> To: sip:300@64.37.115.36;tag=as7e26555d
> Call-ID: 81140MGZlNTZiNDI1NDgxZjFhMmU4Yzk2OTdkMTQ2NzkzNjM
> CSeq: 3 INVITE
> Server: FPBX-12.0.76.4(13.13.1)
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
> Supported: replaces, timer
> Contact: sip:300@64.37.115.36:5061
> Content-Type: application/sdp
> Content-Length: 297
> v=0
> o=root 671133073 671133074 IN IP4 64.37.115.36
> s=Asterisk PBX 13.13.1
> c=IN IP4 64.37.115.36
> t=0 0
> m=audio 10536 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 0-16
> a=ptime:20
> a=maxptime:150
> a=sendrecv
> —
> <— SIP read from UDP:176.58.74.132:1025 —>
> ACK sip:300@64.37.115.36:5061 SIP/2.0
> Via: SIP/2.0/UDP 176.58.74.132:13240;branch=z9hG4bK-524287-1—4b68b93e99044f2f;rport
> Max-Forwards: 70
> Contact: sip:100@176.58.74.132:13240
> To: sip:300@64.37.115.36;tag=as7e26555d
> From: sip:100@64.37.115.36;tag=78ad7914
> Call-ID: 81140MGZlNTZiNDI1NDgxZjFhMmU4Yzk2OTdkMTQ2NzkzNjM
> CSeq: 3 ACK
> User-Agent: X-Lite release 4.9.5 stamp 81140
> Content-Length: 0
> <------------->
> — (10 headers 0 lines) —
> <— SIP read from UDP:176.58.74.132:1025 —>
> ACK sip:300@64.37.115.36:5061 SIP/2.0
> Via: SIP/2.0/UDP 176.58.74.132:13240;branch=z9hG4bK-524287-1—4b68b93e99044f2f;rport
> Max-Forwards: 70
> Contact: sip:100@176.58.74.132:13240
> To: sip:300@64.37.115.36;tag=as7e26555d
> From: sip:100@64.37.115.36;tag=78ad7914
> Call-ID: 81140MGZlNTZiNDI1NDgxZjFhMmU4Yzk2OTdkMTQ2NzkzNjM
> CSeq: 3 ACK
> User-Agent: X-Lite release 4.9.5 stamp 81140
> Content-Length: 0
> <------------->
> — (10 headers 0 lines) —
> <— SIP read from UDP:176.58.74.132:1025 —>
> BYE sip:300@64.37.115.36:5061 SIP/2.0
> Via: SIP/2.0/UDP 176.58.74.132:13240;branch=z9hG4bK-524287-1—45ed62204de1117f;rport
> Max-Forwards: 70
> Contact: sip:100@176.58.74.132:13240
> To: sip:300@64.37.115.36;tag=as7e26555d
> From: sip:100@64.37.115.36;tag=78ad7914
> Call-ID: 81140MGZlNTZiNDI1NDgxZjFhMmU4Yzk2OTdkMTQ2NzkzNjM
> CSeq: 4 BYE
> User-Agent: X-Lite release 4.9.5 stamp 81140
> Authorization: Digest username=“100”,realm=“asterisk”,nonce=“38b52919”,uri=“sip:300@64.37.115.36:5061”,response=“f8535412bfa1d45b8baa29d039721e31”,algorithm=MD5
> Content-Length: 0
> <------------->
> — (11 headers 0 lines) —
> Sending to 176.58.74.132:1025 (NAT)
> Scheduling destruction of SIP dialog ‘81140MGZlNTZiNDI1NDgxZjFhMmU4Yzk2OTdkMTQ2NzkzNjM’ in 12672 ms (Method: BYE)
> <— Transmitting (NAT) to 176.58.74.132:1025 —>
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 176.58.74.132:13240;branch=z9hG4bK-524287-1—45ed62204de1117f;received=176.58.74.132;rport=1025
> From: sip:100@64.37.115.36;tag=78ad7914
> To: sip:300@64.37.115.36;tag=as7e26555d
> Call-ID: 81140MGZlNTZiNDI1NDgxZjFhMmU4Yzk2OTdkMTQ2NzkzNjM
> CSeq: 4 BYE
> Server: FPBX-12.0.76.4(13.13.1)
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
> Supported: replaces, timer
> Content-Length: 0
> <------------>
> – Channel SIP/100-00000124 left ‘simple_bridge’ basic-bridge <05943e0c-7344-45a5-a3a8-ad333cecc5f6>
> == Spawn extension (macro-dial-one, s, 45) exited non-zero on ‘SIP/100-00000124’ in macro ‘dial-one’
> == Spawn extension (macro-exten-vm, s, 16) exited non-zero on ‘SIP/100-00000124’ in macro ‘exten-vm’
> == Spawn extension (from-internal, 300, 2) exited non-zero on ‘SIP/100-00000124’
> – Executing [h@from-internal:1] Hangup(“SIP/100-00000124”, “”) in new stack
> == Spawn extension (from-internal, h, 1) exited non-zero on ‘SIP/100-00000124’
> – Channel SIP/99300-00000125 left ‘simple_bridge’ basic-bridge <05943e0c-7344-45a5-a3a8-ad333cecc5f6>
> Scheduling destruction of SIP dialog ‘0c21caec323d0096787fa56a4c815838@64.37.115.36:5061’ in 13056 ms (Method: INVITE)
> set_destination: Parsing sip:gjfia7uo@cp3kurllbgti.invalid;transport=ws for address/port to send to
> set_destination: URI is for WebSocket, we can’t set destination
> Reliably Transmitting (no NAT) to 176.58.74.132:5060:
> BYE sip:gjfia7uo@cp3kurllbgti.invalid;transport=ws SIP/2.0
> Via: SIP/2.0/WS 64.37.115.36:5061;branch=z9hG4bK4e25d467
> Max-Forwards: 70
> From: “ahmad” sip:100@64.37.115.36:5061;tag=as708085eb
> To: sip:gjfia7uo@cp3kurllbgti.invalid;transport=ws;tag=7glr0mkf31
> Call-ID: 0c21caec323d0096787fa56a4c815838@64.37.115.36:5061
> CSeq: 103 BYE
> User-Agent: FPBX-12.0.76.4(13.13.1)
> X-Asterisk-HangupCause: Normal Clearing
> X-Asterisk-HangupCauseCode: 16
> Content-Length: 0
> <— SIP read from WS:176.58.74.132:13179 —>
> SIP/2.0 200 OK
> Via: SIP/2.0/WS 64.37.115.36:5061;branch=z9hG4bK4e25d467
> To: sip:gjfia7uo@cp3kurllbgti.invalid;transport=ws;tag=7glr0mkf31
> From: “ahmad” sip:100@64.37.115.36:5061;tag=as708085eb
> Call-ID: 0c21caec323d0096787fa56a4c815838@64.37.115.36:5061
> CSeq: 103 BYE
> Supported: outbound
> Content-Length: 0
> <------------->
> — (8 headers 0 lines) —
> Really destroying SIP dialog ‘0c21caec323d0096787fa56a4c815838@64.37.115.36:5061’ Method: INVITE