Re: [Doubango] sipml5 + asterisk + ip phone(or x-lite) chrome 25 no audio
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Mamadou
Feb 28, 2013, 11:11:25 AM2/28/13
to doub...@googlegroups.com, Max Builin
- please don't duplicate threads for
the same question
- I already asked not to cut your logs but it is
On 2/28/2013 4:57 PM, Max Builin wrote:
- I already asked not to cut your logs but it is
On 2/28/2013 4:57 PM, Max Builin wrote:
sipml5 settings:Display Name:1060
Private Identity*:1060
Disable Video:true
Enable RTCWeb Breaker:true
sip.conf:
[general]context = public ; Default context for incoming calls. Defaults to 'default'allowoverlap = no ; Disable overlap dialing support. (Default is yes)realm = mydomain.comudpbindaddr = 0.0.0.0:5060tcpenable = no ; Enable server for incoming TCP connections (default is no)tcpbindaddr = 0.0.0.0 ; IP address for TCP server to bind to (0.0.0.0 binds to all interfaces)transport = udp,ws,wsssrvlookup = yes ; Enable DNS SRV lookups on outbound callsavpf = yes ; Enable inter-operability with media streams using the AVPF RTP profile.subscribecontext = default[authentication]
[1060]username=1060type=friendhost=dynamicsecret=1060context=defaulthasiax = nohassip = yesencryption = yesavpf = yesicesupport = yesvideosupport=nodirectmedia=notransport=ws
[1002]username=1002type=friendhost=dynamicsecret=1111context=default
1002 ip phone, 1060 web phonechrome log:
State machine: c0000_Started_2_Outgoing_X_oINVITE tsk_utils.js:110onGetUserMediaSuccess tsk_utils.js:110createOffer tsk_utils.js:110==session event = m_stream_audio_local_added 192.168.0.5:720onCreateSdpSuccess tsk_utils.js:110==session event = connecting 192.168.0.5:720__on_state_change tsk_utils.js:110onSetLocalDescriptionSuccess tsk_utils.js:110onIceCandidate = starting tsk_utils.js:110onIceCandidate = starting tsk_utils.js:110onIceCandidate = starting tsk_utils.js:110onIceCandidate = starting tsk_utils.js:110onIceCandidate = starting tsk_utils.js:110onIceCandidate = starting tsk_utils.js:110onIceCandidate = starting tsk_utils.js:110ICE GATHERING COMPLETED! tsk_utils.js:110onIceGatheringCompleted tsk_utils.js:110SEND: INVITE sip:1002@mydomain.com SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bK019A0DOl3CGPqvy8VHtap6aVo7fmd2bB;rport From: <sip:1060@mydomain.com>;tag=tv7KgLdIMdHSkiTcRQko
To: <sip:1002@mydomain.com>
Contact: "1060"<sip:10...@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>;impi=1060;ha1=1d7964fe50fa20a57e75c4b0e338c078;+g.oma.sip-im;+sip.ice;language="en,fr" Call-ID: 6311eeaf-7270-2d75-d42d-f8eeda01f09a CSeq: 25325 INVITE Content-Type: application/sdp Content-Length: 1623 Route: <sip:192.168.0.5:5060;lr;sipml5-outbound;transport=udp> Max-Forwards: 70 User-Agent: IM-client/OMA1.0 sipML5-v1.2013.02.24 Organization: Doubango Telecom v=0 o=- 1258379063 2 IN IP4 127.0.0.1 s=Doubango Telecom - chrome t=0 0 a=group:BUNDLE audio a=msid-semantic: WMS 0adrCSMllwJxVSEOY5PJpYKiJadNG2rtJWMK m=audio 63046 RTP/SAVPF 103 104 111 0 8 107 106 105 13 126 c=IN IP4 188.162.221.228 a=rtcp:63046 IN IP4 188.162.221.228 a=candidate:2458163306 1 udp 2113937151 192.168.0.156 60771 typ host generation 0 a=candidate:2458163306 2 udp 2113937151 192.168.0.156 60771 typ host generation 0 a=candidate:332177118 1 udp 1845501695 188.162.221.228 63046 typ srflx raddr 192.168.0.156 rport 60771 generation 0 a=candidate:332177118 2 udp 1845501695 188.162.221.228 63046 typ srflx raddr 192.168.0.156 rport 60771 generation 0 a=candidate:3691472026 1 tcp 1509957375 192.168.0.156 64499 typ host generation 0 a=candidate:3691472026 2 tcp 1509957375 192.168.0.156 64499 typ host generation 0 a=ice-ufrag:jC0HVVbngjZgsUiK a=ice-pwd:WWAgn4YrlQgBoSGzmlzNG5Fk a=ice-options:google-ice a=sendrecv a=mid:audio a=rtcp-mux a=crypto:0 AES_CM_128_HMAC_SHA1_32 inline:MvkjDXMhcTT9P/ga83ed6/+7IzL7ogFghnzPvfjg a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:srRpyia+d3Bd9zAIDzdC9qP4BPWnQnQo7VTYhlU/ a=rtpmap:103 ISAC/16000 a=rtpmap:104 ISAC/32000 a=rtpmap:111 opus/48000/2 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:107 CN/48000 a=rtpmap:106 CN/32000 a=rtpmap:105 CN/16000 a=rtpmap:13 CN/8000 a=rtpmap:126 telephone-event/8000 a=ssrc:3935994346 cname:KtMPOwI1/6m27lBB a=ssrc:3935994346 msid:0adrCSMllwJxVSEOY5PJpYKiJadNG2rtJWMK a0 a=ssrc:3935994346 mslabel:0adrCSMllwJxVSEOY5PJpYKiJadNG2rtJWMK a=ssrc:3935994346 label:0adrCSMllwJxVSEOY5PJpYKiJadNG2rtJWMKa0 tsk_utils.js:110__tsip_transport_ws_onmessage tsk_utils.js:110recv=SIP/2.0 401 Unauthorized Via: SIP/2.0/WS df7jal23ls0d.invalid;rport;received=192.168.0.156;branch=z9hG4bK019A0DOl3CGPqvy8VHtap6aVo7fmd2bB From: <sip:1060@mydomain.com>;tag=tv7KgLdIMdHSkiTcRQko
To: <sip:1002@mydomain.com>;tag=as0a6b715d
Call-ID: 6311eeaf-7270-2d75-d42d-f8eeda01f09a CSeq: 25325 INVITE Content-Length: 0 Server: Asterisk PBX 11.2.1 Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH Supported: replaces,timer WWW-Authenticate: Digest realm="mydomain.com",nonce="7382e9e4",stale=FALSE,algorithm=MD5
tsk_utils.js:110SEND: ACK sip:1002@mydomain.com SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bK019A0DOl3CGPqvy8VHtap6aVo7fmd2bB;rport From: <sip:1060@mydomain.com>;tag=tv7KgLdIMdHSkiTcRQko
To: <sip:1002@mydomain.com>;tag=as0a6b715d
Call-ID: 6311eeaf-7270-2d75-d42d-f8eeda01f09a CSeq: 25325 ACK Content-Length: 0 Route: <sip:192.168.0.5:5060;lr;sipml5-outbound;transport=udp> Max-Forwards: 70 tsk_utils.js:110State machine: x0000_Any_2_Any_X_i401_407_INVITE tsk_utils.js:110SEND: INVITE sip:1002@mydomain.com SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKChc50mlKWHFbcI51HaNwKXaHganMiGMS;rport From: <sip:1060@mydomain.com>;tag=tv7KgLdIMdHSkiTcRQko
To: <sip:1002@mydomain.com>
Contact: "1060"<sip:10...@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>;impi=1060;ha1=1d7964fe50fa20a57e75c4b0e338c078;+g.oma.sip-im;+sip.ice;language="en,fr" Call-ID: 6311eeaf-7270-2d75-d42d-f8eeda01f09a CSeq: 25326 INVITE Content-Type: application/sdp Content-Length: 1623 Route: <sip:192.168.0.5:5060;lr;sipml5-outbound;transport=udp> Max-Forwards: 70 Authorization: Digest username="1060",realm="mydomain.com",nonce="7382e9e4",uri="sip:1002@mydomain.com",response="074b7bfea32741e64a22794b6f71311a",algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2013.02.24 Organization: Doubango Telecom v=0 o=- 1258379063 2 IN IP4 127.0.0.1 s=Doubango Telecom - chrome t=0 0 a=group:BUNDLE audio a=msid-semantic: WMS 0adrCSMllwJxVSEOY5PJpYKiJadNG2rtJWMK m=audio 63046 RTP/SAVPF 103 104 111 0 8 107 106 105 13 126 c=IN IP4 188.162.221.228 a=rtcp:63046 IN IP4 188.162.221.228 a=candidate:2458163306 1 udp 2113937151 192.168.0.156 60771 typ host generation 0 a=candidate:2458163306 2 udp 2113937151 192.168.0.156 60771 typ host generation 0 a=candidate:332177118 1 udp 1845501695 188.162.221.228 63046 typ srflx raddr 192.168.0.156 rport 60771 generation 0 a=candidate:332177118 2 udp 1845501695 188.162.221.228 63046 typ srflx raddr 192.168.0.156 rport 60771 generation 0 a=candidate:3691472026 1 tcp 1509957375 192.168.0.156 64499 typ host generation 0 a=candidate:3691472026 2 tcp 1509957375 192.168.0.156 64499 typ host generation 0 a=ice-ufrag:jC0HVVbngjZgsUiK a=ice-pwd:WWAgn4YrlQgBoSGzmlzNG5Fk a=ice-options:google-ice a=sendrecv a=mid:audio a=rtcp-mux a=crypto:0 AES_CM_128_HMAC_SHA1_32 inline:MvkjDXMhcTT9P/ga83ed6/+7IzL7ogFghnzPvfjg a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:srRpyia+d3Bd9zAIDzdC9qP4BPWnQnQo7VTYhlU/ a=rtpmap:103 ISAC/16000 a=rtpmap:104 ISAC/32000 a=rtpmap:111 opus/48000/2 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:107 CN/48000 a=rtpmap:106 CN/32000 a=rtpmap:105 CN/16000 a=rtpmap:13 CN/8000 a=rtpmap:126 telephone-event/8000 a=ssrc:3935994346 cname:KtMPOwI1/6m27lBB a=ssrc:3935994346 msid:0adrCSMllwJxVSEOY5PJpYKiJadNG2rtJWMK a0 a=ssrc:3935994346 mslabel:0adrCSMllwJxVSEOY5PJpYKiJadNG2rtJWMK a=ssrc:3935994346 label:0adrCSMllwJxVSEOY5PJpYKiJadNG2rtJWMKa0 tsk_utils.js:110__tsip_transport_ws_onmessage tsk_utils.js:110recv=SIP/2.0 100 Trying Via: SIP/2.0/WS df7jal23ls0d.invalid;rport;received=192.168.0.156;branch=z9hG4bKChc50mlKWHFbcI51HaNwKXaHganMiGMS From: <sip:1060@mydomain.com>;tag=tv7KgLdIMdHSkiTcRQko
To: <sip:1002@mydomain.com>
Contact: <sip:10...@192.168.0.5:5060;transport=WS> Call-ID: 6311eeaf-7270-2d75-d42d-f8eeda01f09a CSeq: 25326 INVITE Content-Length: 0 Server: Asterisk PBX 11.2.1 Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH Supported: replaces,timer tsk_utils.js:110State machine: x0000_Any_2_Any_X_i1xx tsk_utils.js:110==session event = i_ao_request 192.168.0.5:720__tsip_transport_ws_onmessage tsk_utils.js:110recv=SIP/2.0 180 Ringing Via: SIP/2.0/WS df7jal23ls0d.invalid;rport;received=192.168.0.156;branch=z9hG4bKChc50mlKWHFbcI51HaNwKXaHganMiGMS From: <sip:1060@mydomain.com>;tag=tv7KgLdIMdHSkiTcRQko
To: <sip:1002@mydomain.com>;tag=as39383462
Contact: <sip:10...@192.168.0.5:5060;transport=WS> Call-ID: 6311eeaf-7270-2d75-d42d-f8eeda01f09a CSeq: 25326 INVITE Content-Length: 0 Server: Asterisk PBX 11.2.1 Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH Supported: replaces,timer tsk_utils.js:110State machine: x0000_Any_2_Any_X_i1xx tsk_utils.js:110==session event = i_ao_request 192.168.0.5:720__tsip_transport_ws_onmessage tsk_utils.js:110recv=SIP/2.0 200 OK Via: SIP/2.0/WS df7jal23ls0d.invalid;rport;received=192.168.0.156;branch=z9hG4bKChc50mlKWHFbcI51HaNwKXaHganMiGMS From: <sip:1060@mydomain.com>;tag=tv7KgLdIMdHSkiTcRQko
To: <sip:1002@mydomain.com>;tag=as39383462
Contact: <sip:10...@192.168.0.5:5060;transport=WS> Call-ID: 6311eeaf-7270-2d75-d42d-f8eeda01f09a CSeq: 25326 INVITE Content-Type: application/sdp Content-Length: 567 Server: Asterisk PBX 11.2.1 Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH Supported: replaces,timer v=0 o=root 1714100763 1714100763 IN IP4 192.168.0.5 s=Asterisk PBX 11.2.1 c=IN IP4 192.168.0.5 t=0 0 m=audio 17412 RTP/SAVPF 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=ptime:20 a=ice-ufrag:2cfa185c0e7a1add1fda78ea3aa8716d a=ice-pwd:65dd99774a1f179d3cc639472cc5dfd7 a=candidate:Hc0a80005 1 UDP 2130706431 192.168.0.5 17412 typ host a=candidate:Hc0a80005 2 UDP 2130706430 192.168.0.5 17413 typ host a=sendrecv a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:K0Qjiym904XbFZLGi+dmSJHtqK9p57RSAuPzMIfN tsk_utils.js:110State machine: c0000_Outgoing_2_Connected_X_i2xxINVITE tsk_utils.js:110setRemoteDescription(answer) v=0 o=root 1714100763 1714100763 IN IP4 192.168.0.5 s=Asterisk PBX 11.2.1 c=IN IP4 192.168.0.5 t=0 0 m=audio 17412 RTP/SAVPF 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=ptime:20 a=ice-ufrag:2cfa185c0e7a1add1fda78ea3aa8716d a=ice-pwd:65dd99774a1f179d3cc639472cc5dfd7 a=candidate:Hc0a80005 1 UDP 2130706431 192.168.0.5 17412 typ host a=candidate:Hc0a80005 2 UDP 2130706430 192.168.0.5 17413 typ host a=sendrecv a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:K0Qjiym904XbFZLGi+dmSJHtqK9p57RSAuPzMIfN tsk_utils.js:110SEND: ACK sip:10...@192.168.0.5:5060;transport=WS SIP/2.0 Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKB2tIVCMbvNpbpFus5yKO;rport From: <sip:1060@mydomain.com>;tag=tv7KgLdIMdHSkiTcRQko
To: <sip:1002@mydomain.com>;tag=as39383462
Contact: "1060"<sip:10...@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>;+g.oma.sip-im;+sip.ice;language="en,fr" Call-ID: 6311eeaf-7270-2d75-d42d-f8eeda01f09a CSeq: 25326 ACK Content-Length: 0 Route: <sip:192.168.0.5:5060;lr;sipml5-outbound;transport=udp> Max-Forwards: 70 Authorization: Digest username="1060",realm="mydomain.com",nonce="7382e9e4",uri="sip:10...@192.168.0.5:5060;transport=WS",response="3ce3d60175388a297dce1e7187ed065e",algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2013.02.24 Organization: Doubango Telecom tsk_utils.js:110__on_open tsk_utils.js:110__on_state_change tsk_utils.js:110onSetRemoteDescriptionError tsk_utils.js:110
- SetRemoteDescription failed. tsk_utils.js:122
==session event = m_early_media 192.168.0.5:720==session event = connected--
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Mamadou
Feb 28, 2013, 11:23:52 AM2/28/13
to doub...@googlegroups.com, Max Builin
never mind:
- your logs are cut because it's start by the INVITE request. It must start by API version, User-Agent, REGISTER request...
- the 200 OK shows that the crypto tag is not correct which means you haven't patched your Asterisk server as explained at https://code.google.com/p/sipml5/wiki/Asterisk
- your logs are cut because it's start by the INVITE request. It must start by API version, User-Agent, REGISTER request...
- the 200 OK shows that the crypto tag is not correct which means you haven't patched your Asterisk server as explained at https://code.google.com/p/sipml5/wiki/Asterisk
Mamadou
Feb 28, 2013, 11:39:19 AM2/28/13
to Максим Буйлин, doub...@googlegroups.com
What is important is the information
about the browser, API, capabilities... displayed when the browser
starts.
Another important notice: when you reply you have to use "reply to all" otherwise "doub...@googlegroups.com" will not be CCed. For example, your latest mail will not be received by the group because you're sending it to me only and will not also be listed on the web (https://groups.google.com/group/doubango/browse_thread/thread/a64bb48bcf45c1b9). Even better, just add the group mail and not mine (I'll receive it).
On 2/28/2013 5:32 PM, Максим Буйлин wrote:
Another important notice: when you reply you have to use "reply to all" otherwise "doub...@googlegroups.com" will not be CCed. For example, your latest mail will not be received by the group because you're sending it to me only and will not also be listed on the web (https://groups.google.com/group/doubango/browse_thread/thread/a64bb48bcf45c1b9). Even better, just add the group mail and not mine (I'll receive it).
On 2/28/2013 5:32 PM, Максим Буйлин wrote:
Thank you to explain! Now I understand you.I read the link, and patched Asterisk.
Try it again.
If not, can I create a theme with this problem (this time will show the full log)?Log cut because I thought registration information will be redundant, I apologize.
Максим Буйлин
Feb 28, 2013, 12:03:30 PM2/28/13
to doub...@googlegroups.com
Thanks, Mamadou.
There is full log:
YOUR ARE USING DEBUG CODE. PLEASE USE CODE UNDER 'release' FOLDER. SIPml-api.js:19
User-Agent=Mozilla/5.0 (Windows NT 6.1; WOW64) AppleWebKit/537.22 (KHTML, like Gecko) Chrome/25.0.1364.97 Safari/537.22 tsk_utils.js:110
Navigator friendly name = chrome tsk_utils.js:110
OS friendly name = windows tsk_utils.js:110
Engine initialized tsk_utils.js:110
s_websocket_server_url=ws://192.168.0.5:8088/ws tsk_utils.js:110
s_sip_outboundproxy_url=udp://192.168.0.5:5060 tsk_utils.js:110
b_rtcweb_breaker_enabled=yes tsk_utils.js:110
SIP stack start: proxy='sipml5.org:12062', realm='<sip:mydomain.com>', impi='1060', impu='<sip:1060@mydomain.com>' tsk_utils.js:110
Connecting to 'ws://192.168.0.5:8088/ws' tsk_utils.js:110
==stack event = starting tsk_utils.js:110
__tsip_transport_ws_onopen tsk_utils.js:110
==stack event = started tsk_utils.js:110
State machine: tsip_dialog_register_Started_2_InProgress_X_oRegister tsk_utils.js:110
SEND: REGISTER sip:mydomain.com SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKPI3oSqmcnDpBm4czdKgTlf1DgLEZNNpG;rport
From: <sip:1060@mydomain.com>;tag=UiypRMoE0TLITQKaai34
To: <sip:1060@mydomain.com>
Contact: "1060"<sip:10...@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>;expires=200;+g.oma.sip-im;+audio;language="en,fr"
Call-ID: 2662b811-e4e8-32ed-f741-668737d93448
CSeq: 13223 REGISTER
Content-Length: 0
Route: <sip:192.168.0.5:5060;lr;sipml5-outbound;transport=udp>
Max-Forwards: 70
User-Agent: IM-client/OMA1.0 sipML5-v1.2013.02.24
Organization: Doubango Telecom
Supported: path
tsk_utils.js:110
==session event = connecting 192.168.0.5:720
==session event = sent_request 192.168.0.5:720
__tsip_transport_ws_onmessage tsk_utils.js:110
recv=SIP/2.0 401 Unauthorized
Via: SIP/2.0/WS df7jal23ls0d.invalid;rport;received=192.168.0.156;branch=z9hG4bKPI3oSqmcnDpBm4czdKgTlf1DgLEZNNpG
From: <sip:1060@mydomain.com>;tag=UiypRMoE0TLITQKaai34
To: <sip:1060@mydomain.com>;tag=as5286dc7a
Call-ID: 2662b811-e4e8-32ed-f741-668737d93448
CSeq: 13223 REGISTER
Content-Length: 0
Server: Asterisk PBX 11.2.1
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH
Supported: replaces,timer
WWW-Authenticate: Digest realm="mydomain.com",nonce="67ed3842",stale=FALSE,algorithm=MD5
tsk_utils.js:110
State machine: tsip_dialog_register_InProgress_2_InProgress_X_401_407_421_494 tsk_utils.js:110
SEND: REGISTER sip:mydomain.com SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKb2rb3pAhfTE4rThZ87kS8s62sk8dCd8w;rport
From: <sip:1060@mydomain.com>;tag=UiypRMoE0TLITQKaai34
To: <sip:1060@mydomain.com>
Contact: "1060"<sip:10...@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>;expires=200;+g.oma.sip-im;+audio;language="en,fr"
Call-ID: 2662b811-e4e8-32ed-f741-668737d93448
CSeq: 13224 REGISTER
Content-Length: 0
Route: <sip:192.168.0.5:5060;lr;sipml5-outbound;transport=udp>
Max-Forwards: 70
Authorization: Digest username="1060",realm="mydomain.com",nonce="67ed3842",uri="sip:mydomain.com",response="5ee3b786a1895fb1d7c7f889129bb057",algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2013.02.24
Organization: Doubango Telecom
Supported: path
tsk_utils.js:110
==session event = sent_request 192.168.0.5:720
__tsip_transport_ws_onmessage tsk_utils.js:110
recv=SIP/2.0 200 OK
Via: SIP/2.0/WS df7jal23ls0d.invalid;rport;received=192.168.0.156;branch=z9hG4bKb2rb3pAhfTE4rThZ87kS8s62sk8dCd8w
From: <sip:1060@mydomain.com>;tag=UiypRMoE0TLITQKaai34
To: <sip:1060@mydomain.com>;tag=as5286dc7a
Contact: <sip:10...@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>;expires=200
Call-ID: 2662b811-e4e8-32ed-f741-668737d93448
CSeq: 13224 REGISTER
Expires: 200
Content-Length: 0
Server: Asterisk PBX 11.2.1
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH
Supported: replaces,timer
Date: 28 Feb 2013 16:57:21 GMT;28
tsk_utils.js:110
State machine: tsip_dialog_register_InProgress_2_Connected_X_2xx tsk_utils.js:110
==session event = connected
State machine: c0000_Started_2_Outgoing_X_oINVITE tsk_utils.js:110
PeerConnectionClass = function RTCPeerConnection() { [native code] } SessionDescriptionClass = function RTCSessionDescription() { [native code] } IceCandidateClass = function RTCIceCandidate() { [native code] } tsk_utils.js:110
==stack event = m_permission_requested tsk_utils.js:110
==session event = connecting 192.168.0.5:720
onGetUserMediaSuccess tsk_utils.js:110
createOffer tsk_utils.js:110
==stack event = m_permission_accepted tsk_utils.js:110
onCreateSdpSuccess tsk_utils.js:110
==session event = m_stream_audio_local_added 192.168.0.5:720
__on_state_change tsk_utils.js:110
onSetLocalDescriptionSuccess tsk_utils.js:110
onIceCandidate = starting tsk_utils.js:110
onIceCandidate = starting tsk_utils.js:110
onIceCandidate = starting tsk_utils.js:110
onIceCandidate = starting tsk_utils.js:110
onIceCandidate = starting tsk_utils.js:110
onIceCandidate = starting tsk_utils.js:110
onIceCandidate = starting tsk_utils.js:110
ICE GATHERING COMPLETED! tsk_utils.js:110
onIceGatheringCompleted tsk_utils.js:110
SEND: INVITE sip:1002@mydomain.com SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKfxLlKUhTE8NPbMORi9MsGVgd9FgwcnKl;rport
From: <sip:1060@mydomain.com>;tag=Hg5xdv0I9rZoMZTT52wV
To: <sip:1002@mydomain.com>
Contact: "1060"<sip:10...@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>;impi=1060;ha1=1d7964fe50fa20a57e75c4b0e338c078;+g.oma.sip-im;+sip.ice;language="en,fr"
Call-ID: dfdf3f4d-e504-4fdf-d2f3-ac7ad09e8d47
CSeq: 7774 INVITE
Content-Type: application/sdp
Content-Length: 1617
Route: <sip:192.168.0.5:5060;lr;sipml5-outbound;transport=udp>
Max-Forwards: 70
User-Agent: IM-client/OMA1.0 sipML5-v1.2013.02.24
Organization: Doubango Telecom
v=0
o=- 35107539 2 IN IP4 127.0.0.1
s=Doubango Telecom - chrome
t=0 0
a=group:BUNDLE audio
a=msid-semantic: WMS BN8d6T0Eop5xJNG1VLw2gG0TSqzzG0DUmN4I
m=audio 61204 RTP/SAVPF 103 104 111 0 8 107 106 105 13 126
c=IN IP4 188.162.221.228
a=rtcp:61204 IN IP4 188.162.221.228
a=candidate:2458163306 1 udp 2113937151 192.168.0.156 63214 typ host generation 0
a=candidate:2458163306 2 udp 2113937151 192.168.0.156 63214 typ host generation 0
a=candidate:332177118 1 udp 1845501695 188.162.221.228 61204 typ srflx raddr 192.168.0.156 rport 63214 generation 0
a=candidate:332177118 2 udp 1845501695 188.162.221.228 61204 typ srflx raddr 192.168.0.156 rport 63214 generation 0
a=candidate:3691472026 1 tcp 1509957375 192.168.0.156 50236 typ host generation 0
a=candidate:3691472026 2 tcp 1509957375 192.168.0.156 50236 typ host generation 0
a=ice-ufrag:IusuaKdtPzIkuHjU
a=ice-pwd:HOu2lqfo81RMCw3YWM6NSDil
a=ice-options:google-ice
a=sendrecv
a=mid:audio
a=rtcp-mux
a=crypto:0 AES_CM_128_HMAC_SHA1_32 inline:gRROdLQD+acGQn9D4DHIr+cbH6XbTP6IhyY2Yz/d
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:tTBzngs2Qbz/TVC7Bda7P1XI3IkVsb4F75jez8Vg
a=rtpmap:103 ISAC/16000
a=rtpmap:104 ISAC/32000
a=rtpmap:111 opus/48000/2
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:107 CN/48000
a=rtpmap:106 CN/32000
a=rtpmap:105 CN/16000
a=rtpmap:13 CN/8000
a=rtpmap:126 telephone-event/8000
a=ssrc:876278491 cname:/fEFRazkAkDYYLQ6
a=ssrc:876278491 msid:BN8d6T0Eop5xJNG1VLw2gG0TSqzzG0DUmN4I a0
a=ssrc:876278491 mslabel:BN8d6T0Eop5xJNG1VLw2gG0TSqzzG0DUmN4I
a=ssrc:876278491 label:BN8d6T0Eop5xJNG1VLw2gG0TSqzzG0DUmN4Ia0
tsk_utils.js:110
__tsip_transport_ws_onmessage tsk_utils.js:110
recv=SIP/2.0 401 Unauthorized
Via: SIP/2.0/WS df7jal23ls0d.invalid;rport;received=192.168.0.156;branch=z9hG4bKfxLlKUhTE8NPbMORi9MsGVgd9FgwcnKl
From: <sip:1060@mydomain.com>;tag=Hg5xdv0I9rZoMZTT52wV
To: <sip:1002@mydomain.com>;tag=as2c33d73e
Call-ID: dfdf3f4d-e504-4fdf-d2f3-ac7ad09e8d47
CSeq: 7774 INVITE
Content-Length: 0
Server: Asterisk PBX 11.2.1
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH
Supported: replaces,timer
WWW-Authenticate: Digest realm="mydomain.com",nonce="6ff5367f",stale=FALSE,algorithm=MD5
tsk_utils.js:110
SEND: ACK sip:1002@mydomain.com SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKfxLlKUhTE8NPbMORi9MsGVgd9FgwcnKl;rport
From: <sip:1060@mydomain.com>;tag=Hg5xdv0I9rZoMZTT52wV
To: <sip:1002@mydomain.com>;tag=as2c33d73e
Call-ID: dfdf3f4d-e504-4fdf-d2f3-ac7ad09e8d47
CSeq: 7774 ACK
Content-Length: 0
Route: <sip:192.168.0.5:5060;lr;sipml5-outbound;transport=udp>
Max-Forwards: 70
tsk_utils.js:110
State machine: x0000_Any_2_Any_X_i401_407_INVITE tsk_utils.js:110
SEND: INVITE sip:1002@mydomain.com SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKfPs0WTTurKIqCxCyB8kTOGByxD4aJTna;rport
From: <sip:1060@mydomain.com>;tag=Hg5xdv0I9rZoMZTT52wV
To: <sip:1002@mydomain.com>
Contact: "1060"<sip:10...@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>;impi=1060;ha1=1d7964fe50fa20a57e75c4b0e338c078;+g.oma.sip-im;+sip.ice;language="en,fr"
Call-ID: dfdf3f4d-e504-4fdf-d2f3-ac7ad09e8d47
CSeq: 7775 INVITE
Content-Type: application/sdp
Content-Length: 1617
Route: <sip:192.168.0.5:5060;lr;sipml5-outbound;transport=udp>
Max-Forwards: 70
Authorization: Digest username="1060",realm="mydomain.com",nonce="6ff5367f",uri="sip:1002@mydomain.com",response="9eda82a44639856db9948d9cfb393b70",algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2013.02.24
Organization: Doubango Telecom
v=0
o=- 35107539 2 IN IP4 127.0.0.1
s=Doubango Telecom - chrome
t=0 0
a=group:BUNDLE audio
a=msid-semantic: WMS BN8d6T0Eop5xJNG1VLw2gG0TSqzzG0DUmN4I
m=audio 61204 RTP/SAVPF 103 104 111 0 8 107 106 105 13 126
c=IN IP4 188.162.221.228
a=rtcp:61204 IN IP4 188.162.221.228
a=candidate:2458163306 1 udp 2113937151 192.168.0.156 63214 typ host generation 0
a=candidate:2458163306 2 udp 2113937151 192.168.0.156 63214 typ host generation 0
a=candidate:332177118 1 udp 1845501695 188.162.221.228 61204 typ srflx raddr 192.168.0.156 rport 63214 generation 0
a=candidate:332177118 2 udp 1845501695 188.162.221.228 61204 typ srflx raddr 192.168.0.156 rport 63214 generation 0
a=candidate:3691472026 1 tcp 1509957375 192.168.0.156 50236 typ host generation 0
a=candidate:3691472026 2 tcp 1509957375 192.168.0.156 50236 typ host generation 0
a=ice-ufrag:IusuaKdtPzIkuHjU
a=ice-pwd:HOu2lqfo81RMCw3YWM6NSDil
a=ice-options:google-ice
a=sendrecv
a=mid:audio
a=rtcp-mux
a=crypto:0 AES_CM_128_HMAC_SHA1_32 inline:gRROdLQD+acGQn9D4DHIr+cbH6XbTP6IhyY2Yz/d
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:tTBzngs2Qbz/TVC7Bda7P1XI3IkVsb4F75jez8Vg
a=rtpmap:103 ISAC/16000
a=rtpmap:104 ISAC/32000
a=rtpmap:111 opus/48000/2
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:107 CN/48000
a=rtpmap:106 CN/32000
a=rtpmap:105 CN/16000
a=rtpmap:13 CN/8000
a=rtpmap:126 telephone-event/8000
a=ssrc:876278491 cname:/fEFRazkAkDYYLQ6
a=ssrc:876278491 msid:BN8d6T0Eop5xJNG1VLw2gG0TSqzzG0DUmN4I a0
a=ssrc:876278491 mslabel:BN8d6T0Eop5xJNG1VLw2gG0TSqzzG0DUmN4I
a=ssrc:876278491 label:BN8d6T0Eop5xJNG1VLw2gG0TSqzzG0DUmN4Ia0
tsk_utils.js:110
__tsip_transport_ws_onmessage tsk_utils.js:110
recv=SIP/2.0 100 Trying
Via: SIP/2.0/WS df7jal23ls0d.invalid;rport;received=192.168.0.156;branch=z9hG4bKfPs0WTTurKIqCxCyB8kTOGByxD4aJTna
From: <sip:1060@mydomain.com>;tag=Hg5xdv0I9rZoMZTT52wV
To: <sip:1002@mydomain.com>
Contact: <sip:10...@192.168.0.5:5060;transport=WS>
Call-ID: dfdf3f4d-e504-4fdf-d2f3-ac7ad09e8d47
CSeq: 7775 INVITE
Content-Length: 0
Server: Asterisk PBX 11.2.1
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH
Supported: replaces,timer
tsk_utils.js:110
State machine: x0000_Any_2_Any_X_i1xx tsk_utils.js:110
__tsip_transport_ws_onmessage tsk_utils.js:110
recv=SIP/2.0 180 Ringing
Via: SIP/2.0/WS df7jal23ls0d.invalid;rport;received=192.168.0.156;branch=z9hG4bKfPs0WTTurKIqCxCyB8kTOGByxD4aJTna
From: <sip:1060@mydomain.com>;tag=Hg5xdv0I9rZoMZTT52wV
To: <sip:1002@mydomain.com>;tag=as09d55b02
Contact: <sip:10...@192.168.0.5:5060;transport=WS>
Call-ID: dfdf3f4d-e504-4fdf-d2f3-ac7ad09e8d47
CSeq: 7775 INVITE
Content-Length: 0
Server: Asterisk PBX 11.2.1
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH
Supported: replaces,timer
tsk_utils.js:110
State machine: x0000_Any_2_Any_X_i1xx tsk_utils.js:110
==session event = i_ao_request 192.168.0.5:720
==session event = i_ao_request 192.168.0.5:720
__tsip_transport_ws_onmessage tsk_utils.js:110
recv=SIP/2.0 200 OK
Via: SIP/2.0/WS df7jal23ls0d.invalid;rport;received=192.168.0.156;branch=z9hG4bKfPs0WTTurKIqCxCyB8kTOGByxD4aJTna
From: <sip:1060@mydomain.com>;tag=Hg5xdv0I9rZoMZTT52wV
To: <sip:1002@mydomain.com>;tag=as09d55b02
Contact: <sip:10...@192.168.0.5:5060;transport=WS>
Call-ID: dfdf3f4d-e504-4fdf-d2f3-ac7ad09e8d47
CSeq: 7775 INVITE
Content-Type: application/sdp
Content-Length: 709
Server: Asterisk PBX 11.2.1
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH
Supported: replaces,timer
v=0
o=root 960348188 960348188 IN IP4 192.168.0.5
s=Asterisk PBX 11.2.1
c=IN IP4 192.168.0.5
t=0 0
m=audio 19982 RTP/SAVPF 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=ptime:20
a=ice-ufrag:41b93ba86a6ca3aa76acd3dd5b24ce45
a=ice-pwd:6751616109b8f0fd6028fb9505bdfae2
a=candidate:Hc0a80005 1 UDP 2130706431 192.168.0.5 19982 typ host
a=candidate:Sbca2dde4 1 UDP 1694498815 188.162.221.228 61206 typ srflx
a=candidate:Hc0a80005 2 UDP 2130706430 192.168.0.5 19983 typ host
a=candidate:Sbca2dde4 2 UDP 1694498814 188.162.221.228 61206 typ srflx
a=sendrecv
a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:ioAx7Jdr6OHwW+/wzew1Z175H5FddvqhFhsMs3VA
tsk_utils.js:110
State machine: c0000_Outgoing_2_Connected_X_i2xxINVITE tsk_utils.js:110
setRemoteDescription(answer)
v=0
o=root 960348188 960348188 IN IP4 192.168.0.5
s=Asterisk PBX 11.2.1
c=IN IP4 192.168.0.5
t=0 0
m=audio 19982 RTP/SAVPF 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=ptime:20
a=ice-ufrag:41b93ba86a6ca3aa76acd3dd5b24ce45
a=ice-pwd:6751616109b8f0fd6028fb9505bdfae2
a=candidate:Hc0a80005 1 UDP 2130706431 192.168.0.5 19982 typ host
a=candidate:Sbca2dde4 1 UDP 1694498815 188.162.221.228 61206 typ srflx
a=candidate:Hc0a80005 2 UDP 2130706430 192.168.0.5 19983 typ host
a=candidate:Sbca2dde4 2 UDP 1694498814 188.162.221.228 61206 typ srflx
a=sendrecv
a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:ioAx7Jdr6OHwW+/wzew1Z175H5FddvqhFhsMs3VA
tsk_utils.js:110
SEND: ACK sip:10...@192.168.0.5:5060;transport=WS SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKksNu3fqGQFAaCI3M6sxd;rport
From: <sip:1060@mydomain.com>;tag=Hg5xdv0I9rZoMZTT52wV
To: <sip:1002@mydomain.com>;tag=as09d55b02
Contact: "1060"<sip:10...@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>;+g.oma.sip-im;+sip.ice;language="en,fr"
Call-ID: dfdf3f4d-e504-4fdf-d2f3-ac7ad09e8d47
CSeq: 7775 ACK
Content-Length: 0
Route: <sip:192.168.0.5:5060;lr;sipml5-outbound;transport=udp>
Max-Forwards: 70
Authorization: Digest username="1060",realm="mydomain.com",nonce="6ff5367f",uri="sip:10...@192.168.0.5:5060;transport=WS",response="28c6fd039b33d2d55a64acd6ebd78946",algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2013.02.24
Organization: Doubango Telecom
tsk_utils.js:110
__on_open tsk_utils.js:110
__on_state_change tsk_utils.js:110
onSetRemoteDescriptionError tsk_utils.js:110
- SetRemoteDescription failed. tsk_utils.js:122
==session event = m_early_media 192.168.0.5:720
==session event = connected 192.168.0.5:720
State machine: tsip_dialog_register_Connected_2_InProgress_X_oRegister tsk_utils.js:110
SEND: REGISTER sip:mydomain.com SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKBiH0R5f0ivBhan1vFuQTos91NR7gR3yf;rport
From: <sip:1060@mydomain.com>;tag=UiypRMoE0TLITQKaai34
To: <sip:1060@mydomain.com>
Contact: "1060"<sip:10...@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>;expires=200;+g.oma.sip-im;+audio;language="en,fr"
Call-ID: 2662b811-e4e8-32ed-f741-668737d93448
CSeq: 13225 REGISTER
Content-Length: 0
Route: <sip:192.168.0.5:5060;lr;sipml5-outbound;transport=udp>
Max-Forwards: 70
Authorization: Digest username="1060",realm="mydomain.com",nonce="67ed3842",uri="sip:mydomain.com",response="5ee3b786a1895fb1d7c7f889129bb057",algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2013.02.24
Organization: Doubango Telecom
tsk_utils.js:110
==session event = sent_request 192.168.0.5:720
__tsip_transport_ws_onmessage tsk_utils.js:110
recv=SIP/2.0 401 Unauthorized
Via: SIP/2.0/WS df7jal23ls0d.invalid;rport;received=192.168.0.156;branch=z9hG4bKBiH0R5f0ivBhan1vFuQTos91NR7gR3yf
From: <sip:1060@mydomain.com>;tag=UiypRMoE0TLITQKaai34
To: <sip:1060@mydomain.com>;tag=as4947b657
Call-ID: 2662b811-e4e8-32ed-f741-668737d93448
CSeq: 13225 REGISTER
Content-Length: 0
Server: Asterisk PBX 11.2.1
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH
Supported: replaces,timer
WWW-Authenticate: Digest realm="mydomain.com",nonce="0664f8f3",stale=FALSE,algorithm=MD5
tsk_utils.js:110
State machine: tsip_dialog_register_InProgress_2_InProgress_X_401_407_421_494 tsk_utils.js:110
SEND: REGISTER sip:mydomain.com SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKvs3VsEmTKrxnxTDolAUF6WtsH8BrxIKo;rport
From: <sip:1060@mydomain.com>;tag=UiypRMoE0TLITQKaai34
To: <sip:1060@mydomain.com>
Contact: "1060"<sip:10...@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>;expires=200;+g.oma.sip-im;+audio;language="en,fr"
Call-ID: 2662b811-e4e8-32ed-f741-668737d93448
CSeq: 13226 REGISTER
Content-Length: 0
Route: <sip:192.168.0.5:5060;lr;sipml5-outbound;transport=udp>
Max-Forwards: 70
Authorization: Digest username="1060",realm="mydomain.com",nonce="0664f8f3",uri="sip:mydomain.com",response="74e6497d99ef1febcc31af4c56402149",algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2013.02.24
Organization: Doubango Telecom
tsk_utils.js:110
==session event = sent_request 192.168.0.5:720
__tsip_transport_ws_onmessage tsk_utils.js:110
recv=SIP/2.0 200 OK
Via: SIP/2.0/WS df7jal23ls0d.invalid;rport;received=192.168.0.156;branch=z9hG4bKvs3VsEmTKrxnxTDolAUF6WtsH8BrxIKo
From: <sip:1060@mydomain.com>;tag=UiypRMoE0TLITQKaai34
To: <sip:1060@mydomain.com>;tag=as4947b657
Contact: <sip:10...@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>;expires=200
Call-ID: 2662b811-e4e8-32ed-f741-668737d93448
CSeq: 13226 REGISTER
Expires: 200
Content-Length: 0
Server: Asterisk PBX 11.2.1
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH
Supported: replaces,timer
Date: 28 Feb 2013 16:59:01 GMT;28
tsk_utils.js:110
State machine: tsip_dialog_register_InProgress_2_Connected_X_2xx tsk_utils.js:110
2013/2/28 Mamadou <diopm...@doubango.org>
Mamadou
Feb 28, 2013, 12:05:32 PM2/28/13
to doub...@googlegroups.com, Максим Буйлин
I confirm the crypto tag issue.
Максим Буйлин
Mar 1, 2013, 8:48:51 AM3/1/13
to doub...@googlegroups.com
i uninstall, and install asterisk
root@IASRPI:/usr/src/asterisk-11.2.1# patch -p0 -i ./asterisk_379070.patch
patching file channels/chan_sip.c
Reversed (or previously applied) patch detected! Assume -R? [n] y
Hunk #1 succeeded at 12517 (offset -82 lines).
Hunk #2 succeeded at 12533 (offset -82 lines).
Hunk #3 succeeded at 13190 (offset -83 lines).
patching file channels/sip/sdp_crypto.c
Reversed (or previously applied) patch detected! Assume -R? [n] y
patch has been successfully installed, but still no sound
where i call from chrome to x-lite, have error in chrome console.
chrome log:
s_websocket_server_url=ws://192.168.0.5:8088/ws SIPml-api.js:1
s_sip_outboundproxy_url=udp://192.168.0.5:5060 SIPml-api.js:1
b_rtcweb_breaker_enabled=yes SIPml-api.js:1
SIP stack start: proxy='sipml5.org:10062', realm='<sip:telecor.ru>', impi='1060', impu='<sip:10...@telecor.ru>' SIPml-api.js:1
Connecting to 'ws://192.168.0.5:8088/ws' SIPml-api.js:1
==stack event = starting SIPml-api.js:1
__tsip_transport_ws_onopen SIPml-api.js:1
==stack event = started SIPml-api.js:1
State machine: tsip_dialog_register_Started_2_InProgress_X_oRegister SIPml-api.js:1
SEND: REGISTER sip:telecor.ru SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKyA3u6kFdtP5TewgXFWoUTouhoUSBZTd0;rport
From: <sip:10...@telecor.ru>;tag=09NXJqgiN7U6NbtqnwUu
To: <sip:10...@telecor.ru>
Contact: "1060"<sip:10...@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>;expires=200;+g.oma.sip-im;+audio;language="en,fr"
Call-ID: 0f33c773-8370-52bd-1ad4-a480c70b1a42
CSeq: 40689 REGISTER
Content-Length: 0
Route: <sip:192.168.0.5:5060;lr;sipml5-outbound;transport=udp>
Max-Forwards: 70
User-Agent: IM-client/OMA1.0 sipML5-v1.2013.02.26
Organization: Doubango Telecom
Supported: path
SIPml-api.js:1
==session event = connecting call.htm:720
==session event = sent_request call.htm:720
__tsip_transport_ws_onmessage SIPml-api.js:1
recv=SIP/2.0 401 Unauthorized
Via: SIP/2.0/WS df7jal23ls0d.invalid;rport;received=192.168.0.156;branch=z9hG4bKyA3u6kFdtP5TewgXFWoUTouhoUSBZTd0
From: <sip:10...@telecor.ru>;tag=09NXJqgiN7U6NbtqnwUu
To: <sip:10...@telecor.ru>;tag=as451043f3
Call-ID: 0f33c773-8370-52bd-1ad4-a480c70b1a42
CSeq: 40689 REGISTER
Content-Length: 0
Server: Asterisk PBX 11.2.1
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH
Supported: replaces,timer
WWW-Authenticate: Digest realm="telecor.ru",nonce="3ae06236",stale=FALSE,algorithm=MD5
SIPml-api.js:1
State machine: tsip_dialog_register_InProgress_2_InProgress_X_401_407_421_494 SIPml-api.js:1
SEND: REGISTER sip:telecor.ru SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKOozaL53IvZ9TuWiwUIfay0PP8fQ0kGa2;rport
From: <sip:10...@telecor.ru>;tag=09NXJqgiN7U6NbtqnwUu
To: <sip:10...@telecor.ru>
Contact: "1060"<sip:10...@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>;expires=200;+g.oma.sip-im;+audio;language="en,fr"
Call-ID: 0f33c773-8370-52bd-1ad4-a480c70b1a42
CSeq: 40690 REGISTER
Content-Length: 0
Route: <sip:192.168.0.5:5060;lr;sipml5-outbound;transport=udp>
Max-Forwards: 70
Authorization: Digest username="1060",realm="telecor.ru",nonce="3ae06236",uri="sip:telecor.ru",response="dd06b9e4318a3944c675b7551de68326",algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2013.02.26
Organization: Doubango Telecom
Supported: path
SIPml-api.js:1
==session event = sent_request call.htm:720
__tsip_transport_ws_onmessage SIPml-api.js:1
recv=SIP/2.0 200 OK
Via: SIP/2.0/WS df7jal23ls0d.invalid;rport;received=192.168.0.156;branch=z9hG4bKOozaL53IvZ9TuWiwUIfay0PP8fQ0kGa2
From: <sip:10...@telecor.ru>;tag=09NXJqgiN7U6NbtqnwUu
To: <sip:10...@telecor.ru>;tag=as451043f3
Contact: <sip:10...@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>;expires=200
Call-ID: 0f33c773-8370-52bd-1ad4-a480c70b1a42
CSeq: 40690 REGISTER
Expires: 200
Content-Length: 0
Server: Asterisk PBX 11.2.1
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH
Supported: replaces,timer
Date: 01 Mar 2013 13:39:49 GMT;01
SIPml-api.js:1
State machine: tsip_dialog_register_InProgress_2_Connected_X_2xx SIPml-api.js:1
==session event = connected call.htm:720
State machine: c0000_Started_2_Outgoing_X_oINVITE SIPml-api.js:1
onGetUserMediaSuccess SIPml-api.js:1
createOffer SIPml-api.js:1
==session event = m_stream_audio_local_added call.htm:720
==session event = connecting call.htm:720
onCreateSdpSuccess SIPml-api.js:1
__on_state_change SIPml-api.js:1
onSetLocalDescriptionSuccess SIPml-api.js:1
onIceCandidate = starting SIPml-api.js:1
onIceCandidate = starting SIPml-api.js:1
onIceCandidate = starting SIPml-api.js:1
onIceCandidate = starting SIPml-api.js:1
onIceCandidate = starting SIPml-api.js:1
onIceCandidate = starting SIPml-api.js:1
onIceCandidate = starting SIPml-api.js:1
ICE GATHERING COMPLETED! SIPml-api.js:1
onIceGatheringCompleted SIPml-api.js:1
SEND: INVITE sip:10...@telecor.ru SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKqhBX7Oal1X3vof26me0nkrJXwTH2qPNT;rport
From: <sip:10...@telecor.ru>;tag=ilM7AJCt5andtC2JXLyG
To: <sip:10...@telecor.ru>
Contact: "1060"<sip:10...@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>;impi=1060;ha1=1d7964fe50fa20a57e75c4b0e338c078;+g.oma.sip-im;+sip.ice;language="en,fr"
Call-ID: 2a2777d0-17d9-ad5e-c3c5-b941b54ff22f
CSeq: 27484 INVITE
Content-Type: application/sdp
Content-Length: 1623
Route: <sip:192.168.0.5:5060;lr;sipml5-outbound;transport=udp>
Max-Forwards: 70
User-Agent: IM-client/OMA1.0 sipML5-v1.2013.02.26
Organization: Doubango Telecom
v=0
o=- 3763232849 2 IN IP4 127.0.0.1
s=Doubango Telecom - chrome
t=0 0
a=group:BUNDLE audio
a=msid-semantic: WMS qttfRzNkEeEF2u2nYDUX5iuazO3GHPwhViE8
m=audio 63540 RTP/SAVPF 103 104 111 0 8 107 106 105 13 126
c=IN IP4 188.162.221.228
a=rtcp:63540 IN IP4 188.162.221.228
a=candidate:2458163306 1 udp 2113937151 192.168.0.156 61241 typ host generation 0
a=candidate:2458163306 2 udp 2113937151 192.168.0.156 61241 typ host generation 0
a=candidate:332177118 1 udp 1845501695 188.162.221.228 63540 typ srflx raddr 192.168.0.156 rport 61241 generation 0
a=candidate:332177118 2 udp 1845501695 188.162.221.228 63540 typ srflx raddr 192.168.0.156 rport 61241 generation 0
a=candidate:3691472026 1 tcp 1509957375 192.168.0.156 51004 typ host generation 0
a=candidate:3691472026 2 tcp 1509957375 192.168.0.156 51004 typ host generation 0
a=ice-ufrag:pHRGQ8AvUMvZlEDS
a=ice-pwd:SGnEBzb6Iq1U/BcyRCbucJ4D
a=ice-options:google-ice
a=sendrecv
a=mid:audio
a=rtcp-mux
a=crypto:0 AES_CM_128_HMAC_SHA1_32 inline:bcBpTZwqm2/OeOnD6vUFKttNG+e8YgDT3B7qIYhV
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:63g3ZWTcF2G5ZocWD/IO82nUZ0ngmmjpZTIbR4xr
a=rtpmap:103 ISAC/16000
a=rtpmap:104 ISAC/32000
a=rtpmap:111 opus/48000/2
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:107 CN/48000
a=rtpmap:106 CN/32000
a=rtpmap:105 CN/16000
a=rtpmap:13 CN/8000
a=rtpmap:126 telephone-event/8000
a=ssrc:1606234715 cname:flk4tln4eoju2Wgz
a=ssrc:1606234715 msid:qttfRzNkEeEF2u2nYDUX5iuazO3GHPwhViE8 a0
a=ssrc:1606234715 mslabel:qttfRzNkEeEF2u2nYDUX5iuazO3GHPwhViE8
a=ssrc:1606234715 label:qttfRzNkEeEF2u2nYDUX5iuazO3GHPwhViE8a0
SIPml-api.js:1
__tsip_transport_ws_onmessage SIPml-api.js:1
recv=SIP/2.0 401 Unauthorized
Via: SIP/2.0/WS df7jal23ls0d.invalid;rport;received=192.168.0.156;branch=z9hG4bKqhBX7Oal1X3vof26me0nkrJXwTH2qPNT
From: <sip:10...@telecor.ru>;tag=ilM7AJCt5andtC2JXLyG
To: <sip:10...@telecor.ru>;tag=as7af0254a
Call-ID: 2a2777d0-17d9-ad5e-c3c5-b941b54ff22f
CSeq: 27484 INVITE
Content-Length: 0
Server: Asterisk PBX 11.2.1
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH
Supported: replaces,timer
WWW-Authenticate: Digest realm="telecor.ru",nonce="7baccc98",stale=FALSE,algorithm=MD5
SIPml-api.js:1
SEND: ACK sip:10...@telecor.ru SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKqhBX7Oal1X3vof26me0nkrJXwTH2qPNT;rport
From: <sip:10...@telecor.ru>;tag=ilM7AJCt5andtC2JXLyG
To: <sip:10...@telecor.ru>;tag=as7af0254a
Call-ID: 2a2777d0-17d9-ad5e-c3c5-b941b54ff22f
CSeq: 27484 ACK
Content-Length: 0
Route: <sip:192.168.0.5:5060;lr;sipml5-outbound;transport=udp>
Max-Forwards: 70
SIPml-api.js:1
State machine: x0000_Any_2_Any_X_i401_407_INVITE SIPml-api.js:1
SEND: INVITE sip:10...@telecor.ru SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKt6Gxp0oEiaGFpg8FhM3t7lln15LInmpQ;rport
From: <sip:10...@telecor.ru>;tag=ilM7AJCt5andtC2JXLyG
To: <sip:10...@telecor.ru>
Contact: "1060"<sip:10...@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>;impi=1060;ha1=1d7964fe50fa20a57e75c4b0e338c078;+g.oma.sip-im;+sip.ice;language="en,fr"
Call-ID: 2a2777d0-17d9-ad5e-c3c5-b941b54ff22f
CSeq: 27485 INVITE
Content-Type: application/sdp
Content-Length: 1623
Route: <sip:192.168.0.5:5060;lr;sipml5-outbound;transport=udp>
Max-Forwards: 70
Authorization: Digest username="1060",realm="telecor.ru",nonce="7baccc98",uri="sip:10...@telecor.ru",response="cf627ee027b94fead6b4cd714eaa1429",algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2013.02.26
Organization: Doubango Telecom
v=0
o=- 3763232849 2 IN IP4 127.0.0.1
s=Doubango Telecom - chrome
t=0 0
a=group:BUNDLE audio
a=msid-semantic: WMS qttfRzNkEeEF2u2nYDUX5iuazO3GHPwhViE8
m=audio 63540 RTP/SAVPF 103 104 111 0 8 107 106 105 13 126
c=IN IP4 188.162.221.228
a=rtcp:63540 IN IP4 188.162.221.228
a=candidate:2458163306 1 udp 2113937151 192.168.0.156 61241 typ host generation 0
a=candidate:2458163306 2 udp 2113937151 192.168.0.156 61241 typ host generation 0
a=candidate:332177118 1 udp 1845501695 188.162.221.228 63540 typ srflx raddr 192.168.0.156 rport 61241 generation 0
a=candidate:332177118 2 udp 1845501695 188.162.221.228 63540 typ srflx raddr 192.168.0.156 rport 61241 generation 0
a=candidate:3691472026 1 tcp 1509957375 192.168.0.156 51004 typ host generation 0
a=candidate:3691472026 2 tcp 1509957375 192.168.0.156 51004 typ host generation 0
a=ice-ufrag:pHRGQ8AvUMvZlEDS
a=ice-pwd:SGnEBzb6Iq1U/BcyRCbucJ4D
a=ice-options:google-ice
a=sendrecv
a=mid:audio
a=rtcp-mux
a=crypto:0 AES_CM_128_HMAC_SHA1_32 inline:bcBpTZwqm2/OeOnD6vUFKttNG+e8YgDT3B7qIYhV
a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:63g3ZWTcF2G5ZocWD/IO82nUZ0ngmmjpZTIbR4xr
a=rtpmap:103 ISAC/16000
a=rtpmap:104 ISAC/32000
a=rtpmap:111 opus/48000/2
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:107 CN/48000
a=rtpmap:106 CN/32000
a=rtpmap:105 CN/16000
a=rtpmap:13 CN/8000
a=rtpmap:126 telephone-event/8000
a=ssrc:1606234715 cname:flk4tln4eoju2Wgz
a=ssrc:1606234715 msid:qttfRzNkEeEF2u2nYDUX5iuazO3GHPwhViE8 a0
a=ssrc:1606234715 mslabel:qttfRzNkEeEF2u2nYDUX5iuazO3GHPwhViE8
a=ssrc:1606234715 label:qttfRzNkEeEF2u2nYDUX5iuazO3GHPwhViE8a0
SIPml-api.js:1
__tsip_transport_ws_onmessage SIPml-api.js:1
recv=SIP/2.0 100 Trying
Via: SIP/2.0/WS df7jal23ls0d.invalid;rport;received=192.168.0.156;branch=z9hG4bKt6Gxp0oEiaGFpg8FhM3t7lln15LInmpQ
From: <sip:10...@telecor.ru>;tag=ilM7AJCt5andtC2JXLyG
To: <sip:10...@telecor.ru>
Contact: <sip:10...@192.168.0.5:5060;transport=WS>
Call-ID: 2a2777d0-17d9-ad5e-c3c5-b941b54ff22f
CSeq: 27485 INVITE
Content-Length: 0
Server: Asterisk PBX 11.2.1
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH
Supported: replaces,timer
SIPml-api.js:1
State machine: x0000_Any_2_Any_X_i1xx SIPml-api.js:1
==session event = i_ao_request call.htm:720
__tsip_transport_ws_onmessage SIPml-api.js:1
recv=SIP/2.0 180 Ringing
Via: SIP/2.0/WS df7jal23ls0d.invalid;rport;received=192.168.0.156;branch=z9hG4bKt6Gxp0oEiaGFpg8FhM3t7lln15LInmpQ
From: <sip:10...@telecor.ru>;tag=ilM7AJCt5andtC2JXLyG
To: <sip:10...@telecor.ru>;tag=as560ec8ac
Contact: <sip:10...@192.168.0.5:5060;transport=WS>
Call-ID: 2a2777d0-17d9-ad5e-c3c5-b941b54ff22f
CSeq: 27485 INVITE
Content-Length: 0
Server: Asterisk PBX 11.2.1
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH
Supported: replaces,timer
SIPml-api.js:1
State machine: x0000_Any_2_Any_X_i1xx SIPml-api.js:1
==session event = i_ao_request call.htm:720
__tsip_transport_ws_onmessage SIPml-api.js:1
recv=SIP/2.0 200 OK
Via: SIP/2.0/WS df7jal23ls0d.invalid;rport;received=192.168.0.156;branch=z9hG4bKt6Gxp0oEiaGFpg8FhM3t7lln15LInmpQ
From: <sip:10...@telecor.ru>;tag=ilM7AJCt5andtC2JXLyG
To: <sip:10...@telecor.ru>;tag=as560ec8ac
Contact: <sip:10...@192.168.0.5:5060;transport=WS>
Call-ID: 2a2777d0-17d9-ad5e-c3c5-b941b54ff22f
CSeq: 27485 INVITE
Content-Type: application/sdp
Content-Length: 711
Server: Asterisk PBX 11.2.1
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REFER,SUBSCRIBE,NOTIFY,INFO,PUBLISH
Supported: replaces,timer
v=0
o=root 1769125430 1769125430 IN IP4 192.168.0.5
s=Asterisk PBX 11.2.1
c=IN IP4 192.168.0.5
t=0 0
m=audio 19362 RTP/SAVPF 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=ptime:20
a=ice-ufrag:4aeb19a3262cfe3c14283bd4261df468
a=ice-pwd:4018f66d4830155640734cd67431b89e
a=candidate:Hc0a80005 1 UDP 2130706431 192.168.0.5 19362 typ host
a=candidate:Sbca2dde4 1 UDP 1694498815 188.162.221.228 63544 typ srflx
a=candidate:Hc0a80005 2 UDP 2130706430 192.168.0.5 19363 typ host
a=candidate:Sbca2dde4 2 UDP 1694498814 188.162.221.228 63544 typ srflx
a=sendrecv
a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:dVQWpjH/5WjkzQA2k2oFi+TKMxRDdIg/IpQD4Jd/
SIPml-api.js:1
State machine: c0000_Outgoing_2_Connected_X_i2xxINVITE SIPml-api.js:1
setRemoteDescription(answer)
v=0
o=root 1769125430 1769125430 IN IP4 192.168.0.5
s=Asterisk PBX 11.2.1
c=IN IP4 192.168.0.5
t=0 0
m=audio 19362 RTP/SAVPF 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=ptime:20
a=ice-ufrag:4aeb19a3262cfe3c14283bd4261df468
a=ice-pwd:4018f66d4830155640734cd67431b89e
a=candidate:Hc0a80005 1 UDP 2130706431 192.168.0.5 19362 typ host
a=candidate:Sbca2dde4 1 UDP 1694498815 188.162.221.228 63544 typ srflx
a=candidate:Hc0a80005 2 UDP 2130706430 192.168.0.5 19363 typ host
a=candidate:Sbca2dde4 2 UDP 1694498814 188.162.221.228 63544 typ srflx
a=sendrecv
a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:dVQWpjH/5WjkzQA2k2oFi+TKMxRDdIg/IpQD4Jd/
SIPml-api.js:1
SEND: ACK sip:10...@192.168.0.5:5060;transport=WS SIP/2.0
Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKlAm795NAMph38uaiOSsf;rport
From: <sip:10...@telecor.ru>;tag=ilM7AJCt5andtC2JXLyG
To: <sip:10...@telecor.ru>;tag=as560ec8ac
Contact: "1060"<sip:10...@df7jal23ls0d.invalid;rtcweb-breaker=yes;transport=ws>;+g.oma.sip-im;+sip.ice;language="en,fr"
Call-ID: 2a2777d0-17d9-ad5e-c3c5-b941b54ff22f
CSeq: 27485 ACK
Content-Length: 0
Route: <sip:192.168.0.5:5060;lr;sipml5-outbound;transport=udp>
Max-Forwards: 70
Authorization: Digest username="1060",realm="telecor.ru",nonce="7baccc98",uri="sip:10...@192.168.0.5:5060;transport=WS",response="2d3c877726dcbf1fb49564198056c908",algorithm=MD5
User-Agent: IM-client/OMA1.0 sipML5-v1.2013.02.26
Organization: Doubango Telecom
SIPml-api.js:1
__on_open SIPml-api.js:1
__on_state_change SIPml-api.js:1
onSetRemoteDescriptionError SIPml-api.js:1
- SetRemoteDescription failed. SIPml-api.js:1
==session event = m_early_media call.htm:720
==session event = connected
when i call from x-lite to chrome i have error in asterisk console.
asterisk log:
== Using SIP RTP CoS mark 5
[Mar 1 17:45:57] ERROR[32506][C-00000016]: netsock2.c:269 ast_sockaddr_resolve: getaddrinfo("df7jal23ls0d.invalid", "(null)", ...): Name or service not known
[Mar 1 17:45:57] WARNING[32506][C-00000016]: chan_sip.c:15763 __set_address_from_contact: Invalid host name in Contact: (can't resolve in DNS) : 'df7jal23ls0d.invalid'
2013/2/28 Mamadou <diopm...@doubango.org>
Mamadou DIOP
Mar 1, 2013, 10:26:47 AM3/1/13
to doub...@googlegroups.com
This is not a SDP from a patched Asterisk. When you say "patch has been successfully installed, but still no sound" it's not correct at all.
Максим Буйлин
Mar 1, 2013, 10:39:51 AM3/1/13
to doub...@googlegroups.com
is this patch http://sipml5.googlecode.com/svn/trunk/asterisk/asterisk_379070.patch ?
2013/3/1 Mamadou DIOP <diopm...@doubango.org>
Mamadou DIOP
Mar 1, 2013, 10:49:38 AM3/1/13
to doub...@googlegroups.com
yes and for sure you haven't applied it correctly.
Максим Буйлин
Mar 1, 2013, 11:01:51 AM3/1/13
to doub...@googlegroups.com
i uninstall asterisk, patch,configure and found error
root@IASRPI:/usr/src/asterisk# ./configure --with-crypto --with-ssl --with-srtp --prefix=$PREFIX
checking build system type... x86_64-unknown-linux-gnu
checking host system type... x86_64-unknown-linux-gnu
checking for gcc... gcc
checking whether the C compiler works... yes
checking for C compiler default output file name... a.out
checking for suffix of executables...
checking whether we are cross compiling... no
checking for suffix of object files... o
checking whether we are using the GNU C compiler... yes
checking whether gcc accepts -g... yes
checking for gcc option to accept ISO C89... none needed
checking how to run the C preprocessor... gcc -E
checking for grep that handles long lines and -e... /bin/grep
checking for egrep... /bin/grep -E
checking for ANSI C header files... yes
checking for sys/types.h... yes
checking for sys/stat.h... yes
checking for stdlib.h... yes
checking for string.h... yes
checking for memory.h... yes
checking for strings.h... yes
checking for inttypes.h... yes
checking for stdint.h... yes
checking for unistd.h... yes
checking minix/config.h usability... no
checking minix/config.h presence... no
checking for minix/config.h... no
checking whether it is safe to define __EXTENSIONS__... yes
checking for uname... /bin/uname
checking for g++... g++
checking whether we are using the GNU C++ compiler... yes
checking whether g++ accepts -g... yes
checking how to run the C preprocessor... gcc -E
checking how to run the C++ preprocessor... g++ -E
checking for a sed that does not truncate output... /bin/sed
checking for egrep... grep -E
checking for ld used by gcc... /usr/bin/ld
checking if the linker (/usr/bin/ld) is GNU ld... yes
checking for gawk... no
checking for mawk... mawk
checking for a BSD-compatible install... /usr/bin/install -c
checking whether ln -s works... yes
checking for ranlib... ranlib
checking for GNU make... make
checking for egrep... (cached) /bin/grep -E
checking for strip... strip
checking for ar... ar
checking for bison... :
checking for cmp... /usr/bin/cmp
checking for flex... :
checking for grep... (cached) /bin/grep
checking for python... /usr/bin/python
checking for find... /usr/bin/find
checking for compress... :
checking for basename... /usr/bin/basename
checking for dirname... /usr/bin/dirname
checking for sh... /bin/bash
checking for ln... /bin/ln
checking for doxygen... :
checking for dot... :
checking for wget... /usr/bin/wget
checking for curl... /usr/bin/curl
checking for rubber... :
checking for catdvi... :
checking for kpsewhich... :
checking for xmllint... :
checking for xmlstarlet... :
checking for git... :
checking for ldconfig... /sbin/ldconfig
checking for sha1sum... /usr/bin/sha1sum
checking for openssl... /usr/bin/openssl
checking for bison that supports parse-param...
checking for soxmix... no
checking for md5... no
checking for md5sum... md5sum
checking for the pthreads library -lpthreads... no
checking whether pthreads work without any flags... no
checking whether pthreads work with -Kthread... no
checking whether pthreads work with -kthread... no
checking for the pthreads library -llthread... no
checking whether pthreads work with -pthread... yes
checking for joinable pthread attribute... PTHREAD_CREATE_JOINABLE
checking if more special flags are required for pthreads... no
checking for gawk... (cached) mawk
checking for curl-config... no
checking whether libcurl is usable... no
checking for size_t... yes
checking for working alloca.h... yes
checking for alloca... yes
checking for dirent.h that defines DIR... yes
checking for library containing opendir... none required
checking for ANSI C header files... (cached) yes
checking for sys/wait.h that is POSIX.1 compatible... yes
checking arpa/inet.h usability... yes
checking arpa/inet.h presence... yes
checking for arpa/inet.h... yes
checking fcntl.h usability... yes
checking fcntl.h presence... yes
checking for fcntl.h... yes
checking for inttypes.h... (cached) yes
checking libintl.h usability... yes
checking libintl.h presence... yes
checking for libintl.h... yes
checking limits.h usability... yes
checking limits.h presence... yes
checking for limits.h... yes
checking locale.h usability... yes
checking locale.h presence... yes
checking for locale.h... yes
checking malloc.h usability... yes
checking malloc.h presence... yes
checking for malloc.h... yes
checking netdb.h usability... yes
checking netdb.h presence... yes
checking for netdb.h... yes
checking netinet/in.h usability... yes
checking netinet/in.h presence... yes
checking for netinet/in.h... yes
checking stddef.h usability... yes
checking stddef.h presence... yes
checking for stddef.h... yes
checking for stdint.h... (cached) yes
checking for stdlib.h... (cached) yes
checking for string.h... (cached) yes
checking for strings.h... (cached) yes
checking sys/event.h usability... no
checking sys/event.h presence... no
checking for sys/event.h... no
checking sys/file.h usability... yes
checking sys/file.h presence... yes
checking for sys/file.h... yes
checking sys/ioctl.h usability... yes
checking sys/ioctl.h presence... yes
checking for sys/ioctl.h... yes
checking sys/param.h usability... yes
checking sys/param.h presence... yes
checking for sys/param.h... yes
checking sys/socket.h usability... yes
checking sys/socket.h presence... yes
checking for sys/socket.h... yes
checking sys/time.h usability... yes
checking sys/time.h presence... yes
checking for sys/time.h... yes
checking syslog.h usability... yes
checking syslog.h presence... yes
checking for syslog.h... yes
checking termios.h usability... yes
checking termios.h presence... yes
checking for termios.h... yes
checking for unistd.h... (cached) yes
checking utime.h usability... yes
checking utime.h presence... yes
checking for utime.h... yes
checking arpa/nameser.h usability... yes
checking arpa/nameser.h presence... yes
checking for arpa/nameser.h... yes
checking sys/io.h usability... yes
checking sys/io.h presence... yes
checking for sys/io.h... yes
checking for tgetent in -ltermcap... yes
checking for tgetent in -ltinfo... yes
checking for initscr in -lcurses... yes
checking curses.h usability... yes
checking curses.h presence... yes
checking for curses.h... yes
checking for initscr in -lncurses... yes
checking for curses.h... (cached) yes
checking for uuid_generate_random in -luuid... no
configure: error: *** uuid support not found (this typically means the uuid development package is missing)
2013/3/1 Mamadou DIOP <diopm...@doubango.org>
Максим Буйлин
Mar 1, 2013, 11:06:43 AM3/1/13
to doub...@googlegroups.com
I solved this problem
and configure
2013/3/1 Максим Буйлин <builin...@gmail.com>
Максим Буйлин
Mar 4, 2013, 1:36:24 AM3/4/13
to doub...@googlegroups.com
does anyone know how to solve this problem?
[Mar 4 10:10:01] ERROR[24589][C-00000002]: netsock2.c:269 ast_sockaddr_resolve: getaddrinfo("df7jal23ls0d.invalid", "(null)", ...): Name or service not known
[Mar 4 10:10:01] WARNING[24589][C-00000002]: chan_sip.c:15894 __set_address_from_contact: Invalid host name in Contact: (can't resolve in DNS) : 'df7jal23ls0d.invalid'
== Spawn extension (default, 1060, 50005) exited non-zero on 'SIP/1002-00000004'
I found a lot of questions on this issue,but did not find a suitable solution.
2013/3/1 Максим Буйлин <builin...@gmail.com>
Muhammad Shahzad
Mar 4, 2013, 6:50:49 AM3/4/13
to doub...@googlegroups.com
It basically means you don't have webrtc support available or enable in Asterisk. You need to following instructions exactly as given on below link to get it working,
Thank you.
Muhammad Shahzad
-----------------------------------
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +49 176 99 83 10 85
MSN: shari...@hotmail.com
Email: shahe...@googlemail.com
-----------------------------------
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +49 176 99 83 10 85
MSN: shari...@hotmail.com
Email: shahe...@googlemail.com
Максим Буйлин
Mar 5, 2013, 1:14:07 AM3/5/13
to doub...@googlegroups.com
after installing webrtc2sip audio works.
but the error stayed:
Connected to Asterisk 11.2.1 currently running on IASRPI (pid = 17035)
== WebSocket connection from '192.168.0.86:1463' for protocol 'sip' accepted using version '13'
== Using SIP RTP CoS mark 5
== Using SIP RTP CoS mark 5
[Mar 5 09:52:44] ERROR[17081][C-00000000]: netsock2.c:269 ast_sockaddr_resolve: getaddrinfo("df7jal23ls0d.invalid", "(null)", ...): Name or service not known
[Mar 5 09:52:44] WARNING[17081][C-00000000]: chan_sip.c:15779 __set_address_from_contact: Invalid host name in Contact: (can't resolve in DNS) : 'df7jal23ls0d.invalid'
== Spawn extension (default, 1061, 50005) exited non-zero on 'SIP/1002-00000000'
2013/3/4 Muhammad Shahzad <shahe...@gmail.com>
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