WEBRTC can not make a call


(Jack) #1

hello everyone,i got some problem with using webrtc.
i use sipML5 client ,i can login the accout ,but can not make a call.
the asterisk cli does not receive call request.
here is the debug info,

asterisk debug info
CLI>== Contact 8888/sip:8888@171.214.204.118:57704;rinstance=a90782ad2973ab75 has been deleted
  == Endpoint 8888 is now Reachable
    -- Contact 8888/sips:8888@171.214.204.118:52314;transport=ws;rtcweb-breaker=no is now Reachable.  RTT: 25.856 msec
chrome debug info

SIPml-api.js?svn=252:1 s_websocket_server_url=wss://webphone.qinweigroup.net:8089/ws
SIPml-api.js?svn=252:1 s_sip_outboundproxy_url=(null)
SIPml-api.js?svn=252:1 b_rtcweb_breaker_enabled=no
SIPml-api.js?svn=252:1 b_click2call_enabled=no
SIPml-api.js?svn=252:1 b_early_ims=yes
SIPml-api.js?svn=252:1 b_enable_media_stream_cache=no
SIPml-api.js?svn=252:1 o_bandwidth={}
SIPml-api.js?svn=252:1 o_video_size={}
SIPml-api.js?svn=252:1 SIP stack start: proxy='ns313841.ovh.net:12062', realm='<sip:webphone.qinweigroup.net>', impi='8888', impu='<sip:8888@webphone.qinweigroup.net:6871>'
SIPml-api.js?svn=252:1 Connecting to 'wss://webphone.qinweigroup.net:8089/ws'
SIPml-api.js?svn=252:1 ==stack event = starting
SIPml-api.js?svn=252:1 __tsip_transport_ws_onopen
SIPml-api.js?svn=252:1 ==stack event = started
SIPml-api.js?svn=252:1 State machine: tsip_dialog_register_Started_2_InProgress_X_oRegister
SIPml-api.js?svn=252:1 SEND: REGISTER sip:webphone.qinweigroup.net SIP/2.0
Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=z9hG4bKTkMyP5qJhDpXXsOviVnNngA258HOTGOv;rport
From: <sip:8888@webphone.qinweigroup.net:6871>;tag=F56iVTodpEi4Grw6s0i3
To: <sip:8888@webphone.qinweigroup.net:6871>
Contact: <sips:8888@df7jal23ls0d.invalid;rtcweb-breaker=no;transport=wss>;expires=200;click2call=no;+g.oma.sip-im;+audio;language="en,fr"
Call-ID: b4498468-5d91-e743-636f-f0698466e973
CSeq: 31976 REGISTER
Content-Length: 0
Max-Forwards: 70
User-Agent: IM-client/OMA1.0 sipML5-v1.2016.03.04
Organization: Doubango Telecom
Supported: path


SIPml-api.js?svn=252:1 ==session event = connecting
SIPml-api.js?svn=252:1 ==session event = sent_request
SIPml-api.js?svn=252:1 __tsip_transport_ws_onmessage
SIPml-api.js?svn=252:1 recv=SIP/2.0 401 Unauthorized
Via: SIP/2.0/WSS df7jal23ls0d.invalid;rport=53326;received=171.214.204.118;branch=z9hG4bKTkMyP5qJhDpXXsOviVnNngA258HOTGOv
From: <sip:8888@webphone.qinweigroup.net>;tag=F56iVTodpEi4Grw6s0i3
To: <sip:8888@webphone.qinweigroup.net>;tag=z9hG4bKTkMyP5qJhDpXXsOviVnNngA258HOTGOv
Call-ID: b4498468-5d91-e743-636f-f0698466e973
CSeq: 31976 REGISTER
Content-Length: 0
WWW-Authenticate: Digest realm="asterisk",qop="auth",nonce="1567862686/367e52b8394fbbba3cbe11246d8153c6",opaque="2c95fe7d750ebfcc",stale=FALSE,algorithm=md5
Server: FPBX-15.0.16(16.5.0)


SIPml-api.js?svn=252:1 State machine: tsip_dialog_register_InProgress_2_InProgress_X_401_407_421_494
SIPml-api.js?svn=252:1 SEND: REGISTER sip:webphone.qinweigroup.net SIP/2.0
Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=z9hG4bKs8Ci9qIO3ig6BVkzN9BzMX9aUQdR8b8X;rport
From: <sip:8888@webphone.qinweigroup.net:6871>;tag=F56iVTodpEi4Grw6s0i3
To: <sip:8888@webphone.qinweigroup.net:6871>
Contact: <sips:8888@df7jal23ls0d.invalid;rtcweb-breaker=no;transport=wss>;expires=200;click2call=no;+g.oma.sip-im;+audio;language="en,fr"
Call-ID: b4498468-5d91-e743-636f-f0698466e973
CSeq: 31977 REGISTER
Content-Length: 0
Max-Forwards: 70
Authorization: Digest username="8888",realm="asterisk",nonce="1567862686/367e52b8394fbbba3cbe11246d8153c6",uri="sip:webphone.qinweigroup.net",response="59a2e35a4ef4be5b29383ca478c1eccc",algorithm=md5,cnonce="6d2b1eec4b3d1f0636a0d0359c2b6b07",opaque="2c95fe7d750ebfcc",qop=auth,nc=00000001
User-Agent: IM-client/OMA1.0 sipML5-v1.2016.03.04
Organization: Doubango Telecom
Supported: path


SIPml-api.js?svn=252:1 ==session event = sent_request
SIPml-api.js?svn=252:1 __tsip_transport_ws_onmessage
SIPml-api.js?svn=252:1 recv=SIP/2.0 200 OK
Via: SIP/2.0/WSS df7jal23ls0d.invalid;rport=53326;received=171.214.204.118;branch=z9hG4bKs8Ci9qIO3ig6BVkzN9BzMX9aUQdR8b8X
From: <sip:8888@webphone.qinweigroup.net>;tag=F56iVTodpEi4Grw6s0i3
To: <sip:8888@webphone.qinweigroup.net>;tag=z9hG4bKs8Ci9qIO3ig6BVkzN9BzMX9aUQdR8b8X
Contact: <sips:8888@171.214.204.118:53326;transport=ws;rtcweb-breaker=no>;expires=199
Call-ID: b4498468-5d91-e743-636f-f0698466e973
CSeq: 31977 REGISTER
Content-Length: 0
Date: 07 Sep 2019 13:24:46 GMT;07
Server: FPBX-15.0.16(16.5.0)


SIPml-api.js?svn=252:1 State machine: tsip_dialog_register_InProgress_2_Connected_X_2xx
SIPml-api.js?svn=252:1 __tsip_transport_ws_onmessage
SIPml-api.js?svn=252:1 recv=OPTIONS sips:8888@171.214.204.118:53326;transport=ws;rtcweb-breaker=no SIP/2.0
Via: SIP/2.0/WSS 192.168.1.158:8089;rport;branch=z9hG4bKPj894c39de-a593-4e96-b2d5-8c11e964ab64;alias
From: <sip:8888@ecs-3a46>;tag=bbd36bad-e620-4909-8514-4e53a8d62f87
To: <sips:8888@171.214.204.118;rtcweb-breaker=no>
Contact: <sips:8888@ecs-3a46:5060;transport=ws>
Call-ID: 009b28b4-14ad-41f7-ab4a-b6cf81e58643
CSeq: 12187 OPTIONS
Content-Length: 0
Max-Forwards: 70
User-Agent: FPBX-15.0.16(16.5.0)


SIPml-api.js?svn=252:1 Not implemented
tsk_utils_log_error @ SIPml-api.js?svn=252:1
tsip_dialog_layer.handle_incoming_message @ SIPml-api.js?svn=252:1
tsip_transport_layer.handle_incoming_message @ SIPml-api.js?svn=252:1
__tsip_transport_ws_onmessage @ SIPml-api.js?svn=252:1
SIPml-api.js?svn=252:1 SEND: SIP/2.0 405 Method Not Allowed
Via: SIP/2.0/WSS 192.168.1.158:8089;rport=8089;branch=z9hG4bKPj894c39de-a593-4e96-b2d5-8c11e964ab64;alias
From: <sip:8888@ecs-3a46>;tag=bbd36bad-e620-4909-8514-4e53a8d62f87
To: <sips:8888@171.214.204.118;rtcweb-breaker=no>
Call-ID: 009b28b4-14ad-41f7-ab4a-b6cf81e58643
CSeq: 12187 OPTIONS
Content-Length: 0


SIPml-api.js?svn=252:1 ==session event = connected
SIPml-api.js?svn=252:1 State machine: c0000_Started_2_Outgoing_X_oINVITE
SIPml-api.js?svn=252:1 ICE servers:[{"url":"stun:stun.l.google.com:19302"},{"url":"stun:stun.counterpath.net:3478"},{"url":"stun:numb.viagenie.ca:3478"}]
SIPml-api.js?svn=252:1 ==stack event = m_permission_requested
SIPml-api.js?svn=252:1 ==session event = connecting
SIPml-api.js?svn=252:1 onGetUserMediaSuccess
SIPml-api.js?svn=252:1 createOffer
SIPml-api.js?svn=252:1 onNegotiationNeeded
SIPml-api.js?svn=252:1 onCreateSdpSuccess
SIPml-api.js?svn=252:1 ==stack event = m_permission_accepted
SIPml-api.js?svn=252:1 onSignalingstateChange:have-local-offer
SIPml-api.js?svn=252:1 onSetLocalDescriptionSuccess
10SIPml-api.js?svn=252:1 onIceCandidate = gathering
SIPml-api.js?svn=252:1 __tsip_transport_ws_onmessage
SIPml-api.js?svn=252:1 recv=OPTIONS sips:6001@171.214.204.118:51073;transport=ws;rtcweb-breaker=no SIP/2.0
Via: SIP/2.0/WSS 192.168.1.158:8089;rport;branch=z9hG4bKPj84c61c76-2aee-4086-9fd5-6586c5ba2468;alias
From: <sip:6001@ecs-3a46>;tag=eaac7c40-3341-4d78-aa5f-d85564111c69
To: <sips:6001@171.214.204.118;rtcweb-breaker=no>
Contact: <sips:6001@ecs-3a46:5060;transport=ws>
Call-ID: 21c7e9db-18b8-4381-adf8-d06eade6494f
CSeq: 55275 OPTIONS
Content-Length: 0
Max-Forwards: 70
User-Agent: FPBX-15.0.16(16.5.0)


SIPml-api.js?svn=252:1 Not implemented
tsk_utils_log_error @ SIPml-api.js?svn=252:1
tsip_dialog_layer.handle_incoming_message @ SIPml-api.js?svn=252:1
tsip_transport_layer.handle_incoming_message @ SIPml-api.js?svn=252:1
__tsip_transport_ws_onmessage @ SIPml-api.js?svn=252:1
SIPml-api.js?svn=252:1 SEND: SIP/2.0 405 Method Not Allowed
Via: SIP/2.0/WSS 192.168.1.158:8089;rport=8089;branch=z9hG4bKPj84c61c76-2aee-4086-9fd5-6586c5ba2468;alias
From: <sip:6001@ecs-3a46>;tag=eaac7c40-3341-4d78-aa5f-d85564111c69
To: <sips:6001@171.214.204.118;rtcweb-breaker=no>
Call-ID: 21c7e9db-18b8-4381-adf8-d06eade6494f
CSeq: 55275 OPTIONS
Content-Length: 0


SIPml-api.js?svn=252:1 onIceCandidate = complete
SIPml-api.js?svn=252:1 ICE GATHERING COMPLETED!
SIPml-api.js?svn=252:1 onIceGatheringCompleted
SIPml-api.js?svn=252:1 SEND: INVITE sip:*69@webphone.qinweigroup.net SIP/2.0
Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=z9hG4bKXNJv6UvPQzpOrTeVItMuP4lc8TlWL74h;rport
From: <sip:8888@webphone.qinweigroup.net:6871>;tag=cTVZTf0OFuhiwaEL3epA
To: <sip:*69@webphone.qinweigroup.net>
Contact: <sips:8888@df7jal23ls0d.invalid;rtcweb-breaker=no;click2call=no;transport=wss>;+g.oma.sip-im;language="en,fr"
Call-ID: 80bc07a3-c951-b710-734c-ffad987c3cf1
CSeq: 5427 INVITE
Content-Type: application/sdp
Content-Length: 2692
Max-Forwards: 70
User-Agent: IM-client/OMA1.0 sipML5-v1.2016.03.04
Organization: Doubango Telecom

v=0
o=- 5248616704460199000 2 IN IP4 127.0.0.1
s=Doubango Telecom - chrome
t=0 0
a=group:BUNDLE 0
a=msid-semantic: WMS 1IuIFTUck7T9OZSaiZKmyXn9O8QlxOKUygF6
m=audio 52985 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 110 112 113 126
c=IN IP4 171.214.204.118
a=rtcp:52987 IN IP4 171.214.204.118
a=candidate:1302483413 1 udp 2122260223 192.168.2.70 52985 typ host generation 0 network-id 1
a=candidate:2999745851 1 udp 2122194687 192.168.56.1 52986 typ host generation 0 network-id 2
a=candidate:1302483413 2 udp 2122260222 192.168.2.70 52987 typ host generation 0 network-id 1
a=candidate:2999745851 2 udp 2122194686 192.168.56.1 52988 typ host generation 0 network-id 2
a=candidate:52538661 1 tcp 1518280447 192.168.2.70 9 typ host tcptype active generation 0 network-id 1
a=candidate:4233069003 1 tcp 1518214911 192.168.56.1 9 typ host tcptype active generation 0 network-id 2
a=candidate:52538661 2 tcp 1518280446 192.168.2.70 9 typ host tcptype active generation 0 network-id 1
a=candidate:4233069003 2 tcp 1518214910 192.168.56.1 9 typ host tcptype active generation 0 network-id 2
a=candidate:3149015041 1 udp 1686052607 171.214.204.118 52985 typ srflx raddr 192.168.2.70 rport 52985 generation 0 network-id 1
a=candidate:3149015041 2 udp 1686052606 171.214.204.118 52987 typ srflx raddr 192.168.2.70 rport 52987 generation 0 network-id 1
a=ice-ufrag:njEb
a=ice-pwd:sgS07lkCk+L7Mpt/4o7DkNn3
a=ice-options:trickle
a=fingerprint:sha-256 F1:C6:62:D6:00:E7:93:AD:E2:A5:F6:07:6E:04:D7:15:9F:37:E0:D8:39:10:EE:0D:9D:48:66:14:2E:6D:15:0F
a=setup:actpass
a=mid:0
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=extmap:2 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01
a=extmap:3 urn:ietf:params:rtp-hdrext:sdes:mid
a=extmap:4 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id
a=extmap:5 urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id
a=sendrecv
a=msid:1IuIFTUck7T9OZSaiZKmyXn9O8QlxOKUygF6 d2ae0b13-0281-4a11-be28-1bbd8828786e
a=rtcp-mux
a=rtpmap:111 opus/48000/2
a=rtcp-fb:111 transport-cc
a=fmtp:111 minptime=10;useinbandfec=1
a=rtpmap:103 ISAC/16000
a=rtpmap:104 ISAC/32000
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:106 CN/32000
a=rtpmap:105 CN/16000
a=rtpmap:13 CN/8000
a=rtpmap:110 telephone-event/48000
a=rtpmap:112 telephone-event/32000
a=rtpmap:113 telephone-event/16000
a=rtpmap:126 telephone-event/8000
a=ssrc:1023339144 cname:vMOnepzLpY47mIGp
a=ssrc:1023339144 msid:1IuIFTUck7T9OZSaiZKmyXn9O8QlxOKUygF6 d2ae0b13-0281-4a11-be28-1bbd8828786e
a=ssrc:1023339144 mslabel:1IuIFTUck7T9OZSaiZKmyXn9O8QlxOKUygF6
a=ssrc:1023339144 label:d2ae0b13-0281-4a11-be28-1bbd8828786e

SIPml-api.js?svn=252:1 __tsip_transport_ws_onmessage
SIPml-api.js?svn=252:1 recv=SIP/2.0 401 Unauthorized
Via: SIP/2.0/WSS df7jal23ls0d.invalid;rport=53326;received=171.214.204.118;branch=z9hG4bKXNJv6UvPQzpOrTeVItMuP4lc8TlWL74h
From: <sip:8888@webphone.qinweigroup.net>;tag=cTVZTf0OFuhiwaEL3epA
To: <sip:*69@webphone.qinweigroup.net>;tag=z9hG4bKXNJv6UvPQzpOrTeVItMuP4lc8TlWL74h
Call-ID: 80bc07a3-c951-b710-734c-ffad987c3cf1
CSeq: 5427 INVITE
Content-Length: 0
WWW-Authenticate: Digest realm="asterisk",qop="auth",nonce="1567862736/d140d4f5a784df87f6f657c642098d3c",opaque="6bebb1a730b11b48",stale=FALSE,algorithm=md5
Server: FPBX-15.0.16(16.5.0)


SIPml-api.js?svn=252:1 SEND: ACK sip:*69@webphone.qinweigroup.net SIP/2.0
Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=z9hG4bKXNJv6UvPQzpOrTeVItMuP4lc8TlWL74h;rport
From: <sip:8888@webphone.qinweigroup.net:6871>;tag=cTVZTf0OFuhiwaEL3epA
To: <sip:*69@webphone.qinweigroup.net>;tag=z9hG4bKXNJv6UvPQzpOrTeVItMuP4lc8TlWL74h
Call-ID: 80bc07a3-c951-b710-734c-ffad987c3cf1
CSeq: 5427 ACK
Content-Length: 0
Max-Forwards: 70


SIPml-api.js?svn=252:1 State machine: x0000_Any_2_Any_X_i401_407_INVITE
SIPml-api.js?svn=252:1 SEND: INVITE sip:*69@webphone.qinweigroup.net SIP/2.0
Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=z9hG4bK4awxH4d0g6b7w95Jg3o6nac2AHm4r6vq;rport
From: <sip:8888@webphone.qinweigroup.net:6871>;tag=cTVZTf0OFuhiwaEL3epA
To: <sip:*69@webphone.qinweigroup.net>
Contact: <sips:8888@df7jal23ls0d.invalid;rtcweb-breaker=no;click2call=no;transport=wss>;+g.oma.sip-im;language="en,fr"
Call-ID: 80bc07a3-c951-b710-734c-ffad987c3cf1
CSeq: 5428 INVITE
Content-Type: application/sdp
Content-Length: 2692
Max-Forwards: 70
Authorization: Digest username="8888",realm="asterisk",nonce="1567862736/d140d4f5a784df87f6f657c642098d3c",uri="sip:*69@webphone.qinweigroup.net",response="20270395d0a5bed5a1757e50eea4377b",algorithm=md5,cnonce="b3e6cb0ee6bb427a6edc186655476f81",opaque="6bebb1a730b11b48",qop=auth,nc=00000001
User-Agent: IM-client/OMA1.0 sipML5-v1.2016.03.04
Organization: Doubango Telecom

v=0
o=- 5248616704460199000 2 IN IP4 127.0.0.1
s=Doubango Telecom - chrome
t=0 0
a=group:BUNDLE 0
a=msid-semantic: WMS 1IuIFTUck7T9OZSaiZKmyXn9O8QlxOKUygF6
m=audio 52985 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 110 112 113 126
c=IN IP4 171.214.204.118
a=rtcp:52987 IN IP4 171.214.204.118
a=candidate:1302483413 1 udp 2122260223 192.168.2.70 52985 typ host generation 0 network-id 1
a=candidate:2999745851 1 udp 2122194687 192.168.56.1 52986 typ host generation 0 network-id 2
a=candidate:1302483413 2 udp 2122260222 192.168.2.70 52987 typ host generation 0 network-id 1
a=candidate:2999745851 2 udp 2122194686 192.168.56.1 52988 typ host generation 0 network-id 2
a=candidate:52538661 1 tcp 1518280447 192.168.2.70 9 typ host tcptype active generation 0 network-id 1
a=candidate:4233069003 1 tcp 1518214911 192.168.56.1 9 typ host tcptype active generation 0 network-id 2
a=candidate:52538661 2 tcp 1518280446 192.168.2.70 9 typ host tcptype active generation 0 network-id 1
a=candidate:4233069003 2 tcp 1518214910 192.168.56.1 9 typ host tcptype active generation 0 network-id 2
a=candidate:3149015041 1 udp 1686052607 171.214.204.118 52985 typ srflx raddr 192.168.2.70 rport 52985 generation 0 network-id 1
a=candidate:3149015041 2 udp 1686052606 171.214.204.118 52987 typ srflx raddr 192.168.2.70 rport 52987 generation 0 network-id 1
a=ice-ufrag:njEb
a=ice-pwd:sgS07lkCk+L7Mpt/4o7DkNn3
a=ice-options:trickle
a=fingerprint:sha-256 F1:C6:62:D6:00:E7:93:AD:E2:A5:F6:07:6E:04:D7:15:9F:37:E0:D8:39:10:EE:0D:9D:48:66:14:2E:6D:15:0F
a=setup:actpass
a=mid:0
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=extmap:2 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01
a=extmap:3 urn:ietf:params:rtp-hdrext:sdes:mid
a=extmap:4 urn:ietf:params:rtp-hdrext:sdes:rtp-stream-id
a=extmap:5 urn:ietf:params:rtp-hdrext:sdes:repaired-rtp-stream-id
a=sendrecv
a=msid:1IuIFTUck7T9OZSaiZKmyXn9O8QlxOKUygF6 d2ae0b13-0281-4a11-be28-1bbd8828786e
a=rtcp-mux
a=rtpmap:111 opus/48000/2
a=rtcp-fb:111 transport-cc
a=fmtp:111 minptime=10;useinbandfec=1
a=rtpmap:103 ISAC/16000
a=rtpmap:104 ISAC/32000
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:106 CN/32000
a=rtpmap:105 CN/16000
a=rtpmap:13 CN/8000
a=rtpmap:110 telephone-event/48000
a=rtpmap:112 telephone-event/32000
a=rtpmap:113 telephone-event/16000
a=rtpmap:126 telephone-event/8000
a=ssrc:1023339144 cname:vMOnepzLpY47mIGp
a=ssrc:1023339144 msid:1IuIFTUck7T9OZSaiZKmyXn9O8QlxOKUygF6 d2ae0b13-0281-4a11-be28-1bbd8828786e
a=ssrc:1023339144 mslabel:1IuIFTUck7T9OZSaiZKmyXn9O8QlxOKUygF6
a=ssrc:1023339144 label:d2ae0b13-0281-4a11-be28-1bbd8828786e

SIPml-api.js?svn=252:1 __tsip_transport_ws_onmessage
SIPml-api.js?svn=252:1 recv=SIP/2.0 100 Trying
Via: SIP/2.0/WSS df7jal23ls0d.invalid;rport=53326;received=171.214.204.118;branch=z9hG4bK4awxH4d0g6b7w95Jg3o6nac2AHm4r6vq
From: <sip:8888@webphone.qinweigroup.net>;tag=cTVZTf0OFuhiwaEL3epA
To: <sip:*69@webphone.qinweigroup.net>
Call-ID: 80bc07a3-c951-b710-734c-ffad987c3cf1
CSeq: 5428 INVITE
Content-Length: 0
Server: FPBX-15.0.16(16.5.0)


SIPml-api.js?svn=252:1 State machine: x0000_Any_2_Any_X_i1xx
SIPml-api.js?svn=252:1 __tsip_transport_ws_onmessage
SIPml-api.js?svn=252:1 recv=SIP/2.0 488 Not Acceptable Here
Via: SIP/2.0/WSS df7jal23ls0d.invalid;rport=53326;received=171.214.204.118;branch=z9hG4bK4awxH4d0g6b7w95Jg3o6nac2AHm4r6vq
From: <sip:8888@webphone.qinweigroup.net>;tag=cTVZTf0OFuhiwaEL3epA
To: <sip:*69@webphone.qinweigroup.net>;tag=bc984f23-7939-43c2-91fb-edd7f4633fb8
Call-ID: 80bc07a3-c951-b710-734c-ffad987c3cf1
CSeq: 5428 INVITE
Content-Length: 0
Server: FPBX-15.0.16(16.5.0)


SIPml-api.js?svn=252:1 SEND: ACK sip:*69@webphone.qinweigroup.net SIP/2.0
Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=z9hG4bK4awxH4d0g6b7w95Jg3o6nac2AHm4r6vq;rport
From: <sip:8888@webphone.qinweigroup.net:6871>;tag=cTVZTf0OFuhiwaEL3epA
To: <sip:*69@webphone.qinweigroup.net>;tag=bc984f23-7939-43c2-91fb-edd7f4633fb8
Call-ID: 80bc07a3-c951-b710-734c-ffad987c3cf1
CSeq: 5428 ACK
Content-Length: 0
Max-Forwards: 70


SIPml-api.js?svn=252:1 State machine: c0000_Outgoing_2_Terminated_X_i300_to_i699INVITE
SIPml-api.js?svn=252:1 === INVITE Dialog terminated ===
SIPml-api.js?svn=252:1 PeerConnection::stop()
2SIPml-api.js?svn=252:1 ==session event = i_ao_request
SIPml-api.js?svn=252:1 ==session event = terminated
SIPml-api.js?svn=252:1 The FSM is in the final state
tsk_utils_log_warn @ SIPml-api.js?svn=252:1
tsk_fsm.act @ SIPml-api.js?svn=252:1
tsip_transac.fsm_act @ SIPml-api.js?svn=252:1
(anonymous) @ SIPml-api.js?svn=252:1
setTimeout (async)
tsip_transac_layer.cancel_by_dialog @ SIPml-api.js?svn=252:1
tsip_dialog.deinit @ SIPml-api.js?svn=252:1
__tsip_dialog_invite_onterm @ SIPml-api.js?svn=252:1
tsk_fsm.act @ SIPml-api.js?svn=252:1
tsip_dialog.fsm_act @ SIPml-api.js?svn=252:1
__tsip_dialog_invite_event_callback @ SIPml-api.js?svn=252:1
tsip_dialog.callback @ SIPml-api.js?svn=252:1
__tsip_transac_ict_Proceeding_2_Completed_X_300_to_699 @ SIPml-api.js?svn=252:1
tsk_fsm.act @ SIPml-api.js?svn=252:1
tsip_transac.fsm_act @ SIPml-api.js?svn=252:1
__tsip_transac_ict_event_callback @ SIPml-api.js?svn=252:1
tsip_transac.callback @ SIPml-api.js?svn=252:1
tsip_transac_layer.handle_incoming_message @ SIPml-api.js?svn=252:1
tsip_transport_layer.handle_incoming_message @ SIPml-api.js?svn=252:1
__tsip_transport_ws_onmessage @ SIPml-api.js?svn=252:1
SIPml-api.js?svn=252:1 __tsip_transport_ws_onmessage
SIPml-api.js?svn=252:1 recv=OPTIONS sips:8888@171.214.204.118:53326;transport=ws;rtcweb-breaker=no SIP/2.0
Via: SIP/2.0/WSS 192.168.1.158:8089;rport;branch=z9hG4bKPjab895d3c-9a73-4471-aa40-bdf5ed7205f0;alias
From: <sip:8888@ecs-3a46>;tag=bbb3efa4-0e14-4e3b-9d26-f284108a1094
To: <sips:8888@171.214.204.118;rtcweb-breaker=no>
Contact: <sips:8888@ecs-3a46:5060;transport=ws>
Call-ID: 30a53df1-ff64-416b-bb20-04796e7d840f
CSeq: 24739 OPTIONS
Content-Length: 0
Max-Forwards: 70
User-Agent: FPBX-15.0.16(16.5.0)


SIPml-api.js?svn=252:1 Not implemented
tsk_utils_log_error @ SIPml-api.js?svn=252:1
tsip_dialog_layer.handle_incoming_message @ SIPml-api.js?svn=252:1
tsip_transport_layer.handle_incoming_message @ SIPml-api.js?svn=252:1
__tsip_transport_ws_onmessage @ SIPml-api.js?svn=252:1
SIPml-api.js?svn=252:1 SEND: SIP/2.0 405 Method Not Allowed
Via: SIP/2.0/WSS 192.168.1.158:8089;rport=8089;branch=z9hG4bKPjab895d3c-9a73-4471-aa40-bdf5ed7205f0;alias
From: <sip:8888@ecs-3a46>;tag=bbb3efa4-0e14-4e3b-9d26-f284108a1094
To: <sips:8888@171.214.204.118;rtcweb-breaker=no>
Call-ID: 30a53df1-ff64-416b-bb20-04796e7d840f
CSeq: 24739 OPTIONS
Content-Length: 0


```
could someone can help me?
i realy need you help .thx.

#2

Asterisk is sending you a reply of 488 Not Acceptable Here. This is because of the media you are offering.

You are offering bundled media, ICE, DTLS, rtcp-mux - are these all turned on in your extension config?



Missing from the freepbx extension interface is the option to set webrtc=yes which enables all the necessary webrtc stuff without going through and picking the options individually.


(Jack) #3

thank you billsimon. it’s work for me.
asterisk can receive the call request ,but i can’t hear the playback sound .
i can see the message in the CLI

Using SIP RTP Audio TOS bits 184
  == Using SIP RTP Audio TOS bits 184 in TCLASS field.
  == Using SIP RTP Audio CoS mark 5
    -- Executing [*69@from-internal:1] Goto("PJSIP/6001-0000001f", "app-calltrace-perform,s,1") in new stack
    -- Goto (app-calltrace-perform,s,1)
    -- Executing [s@app-calltrace-perform:1] Set("PJSIP/6001-0000001f", "CONNECTEDLINE(name-charset,i)=utf8") in new stack
    -- Executing [s@app-calltrace-perform:2] Set("PJSIP/6001-0000001f", "CONNECTEDLINE(name,i)=呼叫追踪") in new stack
    -- Executing [s@app-calltrace-perform:3] Set("PJSIP/6001-0000001f", "CONNECTEDLINE(num,i)=s") in new stack
    -- Executing [s@app-calltrace-perform:4] Answer("PJSIP/6001-0000001f", "") in new stack
    -- Executing [s@app-calltrace-perform:5] Wait("PJSIP/6001-0000001f", "1") in new stack
    -- Executing [s@app-calltrace-perform:6] Macro("PJSIP/6001-0000001f", "user-callerid,") in new stack
    -- Executing [s@macro-user-callerid:1] Set("PJSIP/6001-0000001f", "TOUCH_MONITOR=1567866966.31") in new stack
    -- Executing [s@macro-user-callerid:2] Set("PJSIP/6001-0000001f", "AMPUSER=6001") in new stack
    -- Executing [s@macro-user-callerid:3] GotoIf("PJSIP/6001-0000001f", "0?report") in new stack
    -- Executing [s@macro-user-callerid:4] ExecIf("PJSIP/6001-0000001f", "1?Set(REALCALLERIDNUM=6001)") in new stack
    -- Executing [s@macro-user-callerid:5] Set("PJSIP/6001-0000001f", "AMPUSER=6001") in new stack
    -- Executing [s@macro-user-callerid:6] GotoIf("PJSIP/6001-0000001f", "0?limit") in new stack
    -- Executing [s@macro-user-callerid:7] Set("PJSIP/6001-0000001f", "AMPUSERCIDNAME=web888") in new stack
    -- Executing [s@macro-user-callerid:8] ExecIf("PJSIP/6001-0000001f", "0?Set(__CIDMASQUERADING=TRUE)") in new stack
    -- Executing [s@macro-user-callerid:9] GotoIf("PJSIP/6001-0000001f", "0?report") in new stack
    -- Executing [s@macro-user-callerid:10] Set("PJSIP/6001-0000001f", "AMPUSERCID=6001") in new stack
    -- Executing [s@macro-user-callerid:11] Set("PJSIP/6001-0000001f", "__DIAL_OPTIONS=HhTtr") in new stack
    -- Executing [s@macro-user-callerid:12] Set("PJSIP/6001-0000001f", "CALLERID(all)="web888" <6001>") in new stack
    -- Executing [s@macro-user-callerid:13] ExecIf("PJSIP/6001-0000001f", "0?Set(CALLERID(all)=)") in new stack
    -- Executing [s@macro-user-callerid:14] GotoIf("PJSIP/6001-0000001f", "0?limit") in new stack
    -- Executing [s@macro-user-callerid:15] ExecIf("PJSIP/6001-0000001f", "0?Set(GROUP(concurrency_limit)=6001)") in new stack
    -- Executing [s@macro-user-callerid:16] NoOp("PJSIP/6001-0000001f", "Macro Depth is 1") in new stack
    -- Executing [s@macro-user-callerid:17] GotoIf("PJSIP/6001-0000001f", "1?report2:macroerror") in new stack
    -- Goto (macro-user-callerid,s,18)
    -- Executing [s@macro-user-callerid:18] GotoIf("PJSIP/6001-0000001f", "0?continue") in new stack
    -- Executing [s@macro-user-callerid:19] Set("PJSIP/6001-0000001f", "__TTL=64") in new stack
    -- Executing [s@macro-user-callerid:20] GotoIf("PJSIP/6001-0000001f", "1?continue") in new stack
    -- Goto (macro-user-callerid,s,36)
    -- Executing [s@macro-user-callerid:36] Set("PJSIP/6001-0000001f", "CALLERID(number)=6001") in new stack
    -- Executing [s@macro-user-callerid:37] Set("PJSIP/6001-0000001f", "CALLERID(name)=web888") in new stack
    -- Executing [s@macro-user-callerid:38] GotoIf("PJSIP/6001-0000001f", "0?cnum") in new stack
    -- Executing [s@macro-user-callerid:39] Set("PJSIP/6001-0000001f", "CDR(cnam)=web888") in new stack
    -- Executing [s@macro-user-callerid:40] Set("PJSIP/6001-0000001f", "CDR(cnum)=6001") in new stack
    -- Executing [s@macro-user-callerid:41] Set("PJSIP/6001-0000001f", "CHANNEL(language)=en") in new stack
    -- Executing [s@app-calltrace-perform:7] Set("PJSIP/6001-0000001f", "INVALID_LOOPCOUNT=0") in new stack
    -- Executing [s@app-calltrace-perform:8] Playback("PJSIP/6001-0000001f", "info-about-last-call&telephone-number") in new stack
    -- <PJSIP/6001-0000001f> Playing 'info-about-last-call.ulaw' (language 'en')
    -- <PJSIP/6001-0000001f> Playing 'telephone-number.ulaw' (language 'en')
    -- Executing [s@app-calltrace-perform:9] Set("PJSIP/6001-0000001f", "lastcaller=8888") in new stack
    -- Executing [s@app-calltrace-perform:10] GotoIf("PJSIP/6001-0000001f", "0?noinfo") in new stack
    -- Executing [s@app-calltrace-perform:11] SayDigits("PJSIP/6001-0000001f", "8888") in new stack
    -- <PJSIP/6001-0000001f> Playing 'digits/8.ulaw' (language 'en')
    -- <PJSIP/6001-0000001f> Playing 'digits/8.ulaw' (language 'en')
    -- <PJSIP/6001-0000001f> Playing 'digits/8.ulaw' (language 'en')
    -- <PJSIP/6001-0000001f> Playing 'digits/8.ulaw' (language 'en')
    -- Executing [s@app-calltrace-perform:12] Set("PJSIP/6001-0000001f", "TIMEOUT(digit)=3") in new stack
    -- Digit timeout set to 3.000
    -- Executing [s@app-calltrace-perform:13] Set("PJSIP/6001-0000001f", "TIMEOUT(response)=7") in new stack
    -- Response timeout set to 7.000
    -- Executing [s@app-calltrace-perform:14] Set("PJSIP/6001-0000001f", "INVALID_LOOPCOUNT=1") in new stack
    -- Executing [s@app-calltrace-perform:15] Read("PJSIP/6001-0000001f", "EXT,to-call-this-number&vm-press&digits/1,1,,0,10") in new stack
    -- Accepting a maximum of 1 digits.
    -- <PJSIP/6001-0000001f> Playing 'to-call-this-number.ulaw' (language 'en')
    -- <PJSIP/6001-0000001f> Playing 'vm-press.ulaw' (language 'en')
    -- <PJSIP/6001-0000001f> Playing 'digits/1.ulaw' (language 'en')
    -- User entered nothing.
    -- Executing [s@app-calltrace-perform:16] GotoIf("PJSIP/6001-0000001f", "1?i,invalid") in new stack
    -- Goto (app-calltrace-perform,i,1)
    -- Executing [i@app-calltrace-perform:1] Playback("PJSIP/6001-0000001f", "no-valid-responce-pls-try-again") in new stack
    -- <PJSIP/6001-0000001f> Playing 'no-valid-responce-pls-try-again.slin' (language 'en')
    -- Executing [i@app-calltrace-perform:2] GotoIf("PJSIP/6001-0000001f", "1?s,repeatoption") in new stack
    -- Goto (app-calltrace-perform,s,14)
    -- Executing [s@app-calltrace-perform:14] Set("PJSIP/6001-0000001f", "INVALID_LOOPCOUNT=2") in new stack
    -- Executing [s@app-calltrace-perform:15] Read("PJSIP/6001-0000001f", "EXT,to-call-this-number&vm-press&digits/1,1,,0,10") in new stack
    -- Accepting a maximum of 1 digits.
    -- <PJSIP/6001-0000001f> Playing 'to-call-this-number.ulaw' (language 'en')
    -- <PJSIP/6001-0000001f> Playing 'vm-press.ulaw' (language 'en')
    -- <PJSIP/6001-0000001f> Playing 'digits/1.ulaw' (language 'en')
    -- User entered nothing.
    -- Executing [s@app-calltrace-perform:16] GotoIf("PJSIP/6001-0000001f", "1?i,invalid") in new stack
    -- Goto (app-calltrace-perform,i,1)
    -- Executing [i@app-calltrace-perform:1] Playback("PJSIP/6001-0000001f", "no-valid-responce-pls-try-again") in new stack
    -- <PJSIP/6001-0000001f> Playing 'no-valid-responce-pls-try-again.slin' (language 'en')
    -- Executing [i@app-calltrace-perform:2] GotoIf("PJSIP/6001-0000001f", "1?s,repeatoption") in new stack
    -- Goto (app-calltrace-perform,s,14)
    -- Executing [s@app-calltrace-perform:14] Set("PJSIP/6001-0000001f", "INVALID_LOOPCOUNT=3") in new stack
    -- Executing [s@app-calltrace-perform:15] Read("PJSIP/6001-0000001f", "EXT,to-call-this-number&vm-press&digits/1,1,,0,10") in new stack
    -- Accepting a maximum of 1 digits.
    -- <PJSIP/6001-0000001f> Playing 'to-call-this-number.ulaw' (language 'en')
    -- <PJSIP/6001-0000001f> Playing 'vm-press.ulaw' (language 'en')
    -- <PJSIP/6001-0000001f> Playing 'digits/1.ulaw' (language 'en')
    -- Added contact 'sips:6001@171.214.204.118:55871;transport=ws;rtcweb-breaker=no' to AOR '6001' with expiration of 200 seconds
    -- Removed contact 'sips:6001@171.214.204.118:51073;transport=ws;rtcweb-breaker=no' from AOR '6001' due to remove existing
  == Contact 6001/sips:6001@171.214.204.118:51073;transport=ws;rtcweb-breaker=no has been deleted
  == Endpoint 6001 is now Unreachable
  == Endpoint 6001 is now Reachable
    -- Contact 6001/sips:6001@171.214.204.118:55871;transport=ws;rtcweb-breaker=no is now Reachable.  RTT: 26.160 msec
    -- Added contact 'sips:6001@171.214.204.118:51073;transport=ws;rtcweb-breaker=no' to AOR '6001' with expiration of 200 seconds
    -- Removed contact 'sips:6001@171.214.204.118:55871;transport=ws;rtcweb-breaker=no' from AOR '6001' due to remove existing
  == Contact 6001/sips:6001@171.214.204.118:55871;transport=ws;rtcweb-breaker=no has been deleted
  == Endpoint 6001 is now Unreachable
  == Endpoint 6001 is now Reachable
    -- Contact 6001/sips:6001@171.214.204.118:51073;transport=ws;rtcweb-breaker=no is now Reachable.  RTT: 32.201 msec
    -- User entered nothing.
    -- Executing [s@app-calltrace-perform:16] GotoIf("PJSIP/6001-0000001f", "1?i,invalid") in new stack
    -- Goto (app-calltrace-perform,i,1)
    -- Executing [i@app-calltrace-perform:1] Playback("PJSIP/6001-0000001f", "no-valid-responce-pls-try-again") in new stack
    -- <PJSIP/6001-0000001f> Playing 'no-valid-responce-pls-try-again.slin' (language 'en')
    -- Executing [i@app-calltrace-perform:2] GotoIf("PJSIP/6001-0000001f", "0?s,repeatoption") in new stack
    -- Executing [i@app-calltrace-perform:3] Playback("PJSIP/6001-0000001f", "vm-goodbye") in new stack
    -- <PJSIP/6001-0000001f> Playing 'vm-goodbye.ulaw' (language 'en')
    -- Executing [i@app-calltrace-perform:4] Macro("PJSIP/6001-0000001f", "hangupcall,") in new stack
    -- Executing [s@macro-hangupcall:1] GotoIf("PJSIP/6001-0000001f", "1?theend") in new stack
    -- Goto (macro-hangupcall,s,3)
    -- Executing [s@macro-hangupcall:3] ExecIf("PJSIP/6001-0000001f", "0?Set(CDR(recordingfile)=)") in new stack
    -- Executing [s@macro-hangupcall:4] NoOp("PJSIP/6001-0000001f", " montior file= ") in new stack
    -- Executing [s@macro-hangupcall:5] GotoIf("PJSIP/6001-0000001f", "1?skipagi") in new stack
    -- Goto (macro-hangupcall,s,7)
    -- Executing [s@macro-hangupcall:7] Hangup("PJSIP/6001-0000001f", "") in new stack
  == Spawn extension (macro-hangupcall, s, 7) exited non-zero on 'PJSIP/6001-0000001f' in macro 'hangupcall'
  == Spawn extension (app-calltrace-perform, i, 4) exited non-zero on 'PJSIP/6001-0000001f'
    -- Added contact 'sips:6001@171.214.204.118:55871;transport=ws;rtcweb-breaker=no' to AOR '6001' with expiration of 200 seconds
    -- Removed contact 'sips:6001@171.214.204.118:51073;transport=ws;rtcweb-breaker=no' from AOR '6001' due to remove existing
  == Contact 6001/sips:6001@171.214.204.118:51073;transport=ws;rtcweb-breaker=no has been deleted
  == Endpoint 6001 is now Unreachable
  == Endpoint 6001 is now Reachable
    -- Contact 6001/sips:6001@171.214.204.118:55871;transport=ws;rtcweb-breaker=no is now Reachable.  RTT: 20.158 msec
    -- Added contact 'sips:6001@171.214.204.118:51073;transport=ws;rtcweb-breaker=no' to AOR '6001' with expiration of 200 seconds
    -- Removed contact 'sips:6001@171.214.204.118:55871;transport=ws;rtcweb-breaker=no' from AOR '6001' due to remove existing
  == Contact 6001/sips:6001@171.214.204.118:55871;transport=ws;rtcweb-breaker=no has been deleted
  == Endpoint 6001 is now Unreachable
  == Endpoint 6001 is now Reachable
    -- Contact 6001/sips:6001@171.214.204.118:51073;transport=ws;rtcweb-breaker=no is now Reachable.  RTT: 26.175 msec
  == Setting global variable 'SIPDOMAIN' to 'webphone.qinweigroup.net'
  == Using SIP RTP Audio TOS bits 184
  == Using SIP RTP Audio TOS bits 184 in TCLASS field.
  == Using SIP RTP Audio CoS mark 5
    -- Executing [*69@from-internal:1] Goto("PJSIP/6001-00000020", "app-calltrace-perform,s,1") in new stack
    -- Goto (app-calltrace-perform,s,1)
    -- Executing [s@app-calltrace-perform:1] Set("PJSIP/6001-00000020", "CONNECTEDLINE(name-charset,i)=utf8") in new stack
    -- Executing [s@app-calltrace-perform:2] Set("PJSIP/6001-00000020", "CONNECTEDLINE(name,i)=呼叫追踪") in new stack
    -- Executing [s@app-calltrace-perform:3] Set("PJSIP/6001-00000020", "CONNECTEDLINE(num,i)=s") in new stack
    -- Executing [s@app-calltrace-perform:4] Answer("PJSIP/6001-00000020", "") in new stack
    -- Executing [s@app-calltrace-perform:5] Wait("PJSIP/6001-00000020", "1") in new stack
    -- Executing [s@app-calltrace-perform:6] Macro("PJSIP/6001-00000020", "user-callerid,") in new stack
    -- Executing [s@macro-user-callerid:1] Set("PJSIP/6001-00000020", "TOUCH_MONITOR=1567867189.32") in new stack
    -- Executing [s@macro-user-callerid:2] Set("PJSIP/6001-00000020", "AMPUSER=6001") in new stack
    -- Executing [s@macro-user-callerid:3] GotoIf("PJSIP/6001-00000020", "0?report") in new stack
    -- Executing [s@macro-user-callerid:4] ExecIf("PJSIP/6001-00000020", "1?Set(REALCALLERIDNUM=6001)") in new stack
    -- Executing [s@macro-user-callerid:5] Set("PJSIP/6001-00000020", "AMPUSER=6001") in new stack
    -- Executing [s@macro-user-callerid:6] GotoIf("PJSIP/6001-00000020", "0?limit") in new stack
    -- Executing [s@macro-user-callerid:7] Set("PJSIP/6001-00000020", "AMPUSERCIDNAME=web888") in new stack
    -- Executing [s@macro-user-callerid:8] ExecIf("PJSIP/6001-00000020", "0?Set(__CIDMASQUERADING=TRUE)") in new stack
    -- Executing [s@macro-user-callerid:9] GotoIf("PJSIP/6001-00000020", "0?report") in new stack
    -- Executing [s@macro-user-callerid:10] Set("PJSIP/6001-00000020", "AMPUSERCID=6001") in new stack
    -- Executing [s@macro-user-callerid:11] Set("PJSIP/6001-00000020", "__DIAL_OPTIONS=HhTtr") in new stack
    -- Executing [s@macro-user-callerid:12] Set("PJSIP/6001-00000020", "CALLERID(all)="web888" <6001>") in new stack
    -- Executing [s@macro-user-callerid:13] ExecIf("PJSIP/6001-00000020", "0?Set(CALLERID(all)=)") in new stack
    -- Executing [s@macro-user-callerid:14] GotoIf("PJSIP/6001-00000020", "0?limit") in new stack
    -- Executing [s@macro-user-callerid:15] ExecIf("PJSIP/6001-00000020", "0?Set(GROUP(concurrency_limit)=6001)") in new stack
    -- Executing [s@macro-user-callerid:16] NoOp("PJSIP/6001-00000020", "Macro Depth is 1") in new stack
    -- Executing [s@macro-user-callerid:17] GotoIf("PJSIP/6001-00000020", "1?report2:macroerror") in new stack
    -- Goto (macro-user-callerid,s,18)
    -- Executing [s@macro-user-callerid:18] GotoIf("PJSIP/6001-00000020", "0?continue") in new stack
    -- Executing [s@macro-user-callerid:19] Set("PJSIP/6001-00000020", "__TTL=64") in new stack
    -- Executing [s@macro-user-callerid:20] GotoIf("PJSIP/6001-00000020", "1?continue") in new stack
    -- Goto (macro-user-callerid,s,36)
    -- Executing [s@macro-user-callerid:36] Set("PJSIP/6001-00000020", "CALLERID(number)=6001") in new stack
    -- Executing [s@macro-user-callerid:37] Set("PJSIP/6001-00000020", "CALLERID(name)=web888") in new stack
    -- Executing [s@macro-user-callerid:38] GotoIf("PJSIP/6001-00000020", "0?cnum") in new stack
    -- Executing [s@macro-user-callerid:39] Set("PJSIP/6001-00000020", "CDR(cnam)=web888") in new stack
    -- Executing [s@macro-user-callerid:40] Set("PJSIP/6001-00000020", "CDR(cnum)=6001") in new stack
    -- Executing [s@macro-user-callerid:41] Set("PJSIP/6001-00000020", "CHANNEL(language)=en") in new stack
    -- Executing [s@app-calltrace-perform:7] Set("PJSIP/6001-00000020", "INVALID_LOOPCOUNT=0") in new stack
    -- Executing [s@app-calltrace-perform:8] Playback("PJSIP/6001-00000020", "info-about-last-call&telephone-number") in new stack
    -- <PJSIP/6001-00000020> Playing 'info-about-last-call.ulaw' (language 'en')
    -- <PJSIP/6001-00000020> Playing 'telephone-number.ulaw' (language 'en')
    -- Executing [s@app-calltrace-perform:9] Set("PJSIP/6001-00000020", "lastcaller=8888") in new stack
    -- Executing [s@app-calltrace-perform:10] GotoIf("PJSIP/6001-00000020", "0?noinfo") in new stack
    -- Executing [s@app-calltrace-perform:11] SayDigits("PJSIP/6001-00000020", "8888") in new stack
    -- <PJSIP/6001-00000020> Playing 'digits/8.ulaw' (language 'en')
    -- <PJSIP/6001-00000020> Playing 'digits/8.ulaw' (language 'en')
    -- <PJSIP/6001-00000020> Playing 'digits/8.ulaw' (language 'en')
    -- <PJSIP/6001-00000020> Playing 'digits/8.ulaw' (language 'en')
    -- Executing [s@app-calltrace-perform:12] Set("PJSIP/6001-00000020", "TIMEOUT(digit)=3") in new stack
    -- Digit timeout set to 3.000
    -- Executing [s@app-calltrace-perform:13] Set("PJSIP/6001-00000020", "TIMEOUT(response)=7") in new stack
    -- Response timeout set to 7.000
    -- Executing [s@app-calltrace-perform:14] Set("PJSIP/6001-00000020", "INVALID_LOOPCOUNT=1") in new stack
    -- Executing [s@app-calltrace-perform:15] Read("PJSIP/6001-00000020", "EXT,to-call-this-number&vm-press&digits/1,1,,0,10") in new stack
    -- Accepting a maximum of 1 digits.
    -- <PJSIP/6001-00000020> Playing 'to-call-this-number.ulaw' (language 'en')
    -- <PJSIP/6001-00000020> Playing 'vm-press.ulaw' (language 'en')
    -- <PJSIP/6001-00000020> Playing 'digits/1.ulaw' (language 'en')
    -- User entered nothing.
    -- Executing [s@app-calltrace-perform:16] GotoIf("PJSIP/6001-00000020", "1?i,invalid") in new stack
    -- Goto (app-calltrace-perform,i,1)
    -- Executing [i@app-calltrace-perform:1] Playback("PJSIP/6001-00000020", "no-valid-responce-pls-try-again") in new stack
    -- <PJSIP/6001-00000020> Playing 'no-valid-responce-pls-try-again.slin' (language 'en')
    -- Added contact 'sips:6001@171.214.204.118:55871;transport=ws;rtcweb-breaker=no' to AOR '6001' with expiration of 200 seconds
    -- Removed contact 'sips:6001@171.214.204.118:51073;transport=ws;rtcweb-breaker=no' from AOR '6001' due to remove existing
  == Contact 6001/sips:6001@171.214.204.118:51073;transport=ws;rtcweb-breaker=no has been deleted
  == Endpoint 6001 is now Unreachable
  == Endpoint 6001 is now Reachable
    -- Contact 6001/sips:6001@171.214.204.118:55871;transport=ws;rtcweb-breaker=no is now Reachable.  RTT: 15.963 msec
    -- Executing [i@app-calltrace-perform:2] GotoIf("PJSIP/6001-00000020", "1?s,repeatoption") in new stack
    -- Goto (app-calltrace-perform,s,14)
    -- Executing [s@app-calltrace-perform:14] Set("PJSIP/6001-00000020", "INVALID_LOOPCOUNT=2") in new stack
    -- Executing [s@app-calltrace-perform:15] Read("PJSIP/6001-00000020", "EXT,to-call-this-number&vm-press&digits/1,1,,0,10") in new stack
    -- Accepting a maximum of 1 digits.
    -- <PJSIP/6001-00000020> Playing 'to-call-this-number.ulaw' (language 'en')
    -- <PJSIP/6001-00000020> Playing 'vm-press.ulaw' (language 'en')
    -- <PJSIP/6001-00000020> Playing 'digits/1.ulaw' (language 'en')
    -- Added contact 'sips:6001@171.214.204.118:51073;transport=ws;rtcweb-breaker=no' to AOR '6001' with expiration of 200 seconds
    -- Removed contact 'sips:6001@171.214.204.118:55871;transport=ws;rtcweb-breaker=no' from AOR '6001' due to remove existing
  == Contact 6001/sips:6001@171.214.204.118:55871;transport=ws;rtcweb-breaker=no has been deleted
  == Endpoint 6001 is now Unreachable
  == Endpoint 6001 is now Reachable
    -- Contact 6001/sips:6001@171.214.204.118:51073;transport=ws;rtcweb-breaker=no is now Reachable.  RTT: 34.014 msec
    -- User entered nothing.
    -- Executing [s@app-calltrace-perform:16] GotoIf("PJSIP/6001-00000020", "1?i,invalid") in new stack
    -- Goto (app-calltrace-perform,i,1)
    -- Executing [i@app-calltrace-perform:1] Playback("PJSIP/6001-00000020", "no-valid-responce-pls-try-again") in new stack
    -- <PJSIP/6001-00000020> Playing 'no-valid-responce-pls-try-again.slin' (language 'en')
    -- Executing [i@app-calltrace-perform:2] GotoIf("PJSIP/6001-00000020", "1?s,repeatoption") in new stack
    -- Goto (app-calltrace-perform,s,14)
    -- Executing [s@app-calltrace-perform:14] Set("PJSIP/6001-00000020", "INVALID_LOOPCOUNT=3") in new stack
    -- Executing [s@app-calltrace-perform:15] Read("PJSIP/6001-00000020", "EXT,to-call-this-number&vm-press&digits/1,1,,0,10") in new stack
    -- Accepting a maximum of 1 digits.
    -- <PJSIP/6001-00000020> Playing 'to-call-this-number.ulaw' (language 'en')
    -- <PJSIP/6001-00000020> Playing 'vm-press.ulaw' (language 'en')
    -- <PJSIP/6001-00000020> Playing 'digits/1.ulaw' (language 'en')
    -- User entered nothing.
    -- Executing [s@app-calltrace-perform:16] GotoIf("PJSIP/6001-00000020", "1?i,invalid") in new stack
    -- Goto (app-calltrace-perform,i,1)
    -- Executing [i@app-calltrace-perform:1] Playback("PJSIP/6001-00000020", "no-valid-responce-pls-try-again") in new stack
    -- <PJSIP/6001-00000020> Playing 'no-valid-responce-pls-try-again.slin' (language 'en')
    -- Executing [i@app-calltrace-perform:2] GotoIf("PJSIP/6001-00000020", "0?s,repeatoption") in new stack
    -- Executing [i@app-calltrace-perform:3] Playback("PJSIP/6001-00000020", "vm-goodbye") in new stack
    -- <PJSIP/6001-00000020> Playing 'vm-goodbye.ulaw' (language 'en')
    -- Executing [i@app-calltrace-perform:4] Macro("PJSIP/6001-00000020", "hangupcall,") in new stack
    -- Executing [s@macro-hangupcall:1] GotoIf("PJSIP/6001-00000020", "1?theend") in new stack
    -- Goto (macro-hangupcall,s,3)
    -- Executing [s@macro-hangupcall:3] ExecIf("PJSIP/6001-00000020", "0?Set(CDR(recordingfile)=)") in new stack
    -- Executing [s@macro-hangupcall:4] NoOp("PJSIP/6001-00000020", " montior file= ") in new stack
    -- Executing [s@macro-hangupcall:5] GotoIf("PJSIP/6001-00000020", "1?skipagi") in new stack
    -- Goto (macro-hangupcall,s,7)
    -- Executing [s@macro-hangupcall:7] Hangup("PJSIP/6001-00000020", "") in new stack
  == Spawn extension (macro-hangupcall, s, 7) exited non-zero on 'PJSIP/6001-00000020' in macro 'hangupcall'
  == Spawn extension (app-calltrace-perform, i, 4) exited non-zero on 'PJSIP/6001-00000020'


(Jack) #4

where can i find the option or can i add it by myself?


(Tom Ray) #5

I’d be more concerned with the fact your device goes unreachable then reachable a few times during the call. Which means Asterisk isn’t getting replies back from the device for its keepalive’s. If you can’t hear the incoming audio, your device keeps dropping its registration / timing out and that the PBX isn’t accepting your DTMF entries I would say this is a NAT or issue with your WebRTC client.


(system) closed #6

This topic was automatically closed 7 days after the last reply. New replies are no longer allowed.