search for: dtlscafile

...#39; failed for '78.119.140.190:5060' - Wrong password Is this a known problem ?? Second question I have : can I get the complete list of columns that can be used in realtime database for sip peers somewhere (update for Ast 13) ? Are columns like dtlsenable, dtlsverify, dtlscertfile, dtlscafile, dtlssetup possible ?? Thanks for the help. Kind regards. Jonas.

...eventually Firefox) type=user host=dynamic hassip=yes transport=ws,wss directmedia=no ; proxy the media icesupport=yes ; needed for webrtc avpf=yes ; needed for webrtc context=default encryption=yes dtlsenable=yes dtlsverify=no dtlsrekey=60 dtlscafile=/opt/asterisk/keys/ca.crt dtlscertfile=/opt/asterisk/keys/asterisk.pem dtlssetup=actpass insecure=invite Here is the SDP offered by Nightly: v=0 o=Mozilla-SIPUA-24.0a1 25687 1 IN IP4 0.0.0.0 s=Doubango Telecom - firefox t=0 0 a=ice-ufrag:7194cbcc a=ice-pwd:e57c14491015e529b84c5a6baf6d7b67 a=finger...

...rpret: NULL mohsuggest: NULL hasvoicemail: NULL subscribemwi: NULL vmexten: NULL rtpkeepalive: NULL directrtpsetup: yes dtlsenable: yes dtlsverify: no dtlsprivatekey: /etc/asterisk/keys/asterisk.pem dtlssetup: actpass dtlscertfile: /etc/asterisk/keys/asterisk.pem dtlscafile: /etc/asterisk/keys/ca.crt sippasswd: md5ofmypwd rpid: NULL domain: testers.com sippasswd2: NULL This is how all other clients are currently defined: id: 7 name: 771 ipaddr: PU.BL.IC.IP port: 5060 regseconds: 1407748788 d...

...rpret: NULL mohsuggest: NULL hasvoicemail: NULL subscribemwi: NULL vmexten: NULL rtpkeepalive: NULL directrtpsetup: yes dtlsenable: yes dtlsverify: no dtlsprivatekey: /etc/asterisk/keys/asterisk.pem dtlssetup: actpass dtlscertfile: /etc/asterisk/keys/asterisk.pem dtlscafile: /etc/asterisk/keys/ca.crt sippasswd: a84a4ddcda13d13c9573d5294047b6a2 rpid: NULL domain: testers.com sippasswd2: NULL id: 8 name: 700 ipaddr: 1.1.1.1 port: 5060 regseconds: 1407323638 defaultuser: 700 fullcontact: sip:70...

...port=yes directmedia=no disallow=all allow=ulaw qualify=yes [6001] host=dynamic secret=mysecret context=default type=friend encryption=yes avpf=yes force_avp=yes icesupport=yes directmedia=no disallow=all allow=ulaw dtlsenable=yes dtlsverify=fingerprint dtlscertfile=/etc/asterisk/keys/asterisk.pem dtlscafile=/etc/asterisk/keys/ca.crt dtlssetup=actpass *extensions.conf:* [default] exten => _6XXX,1,Dial(SIP/${EXTEN}) *rtp.conf:* [general] rtpstart=10000 rtpend=20000 icesupport=yes stunaddr=stun.l.google.com:19302 2015-08-10 12:35 GMT-03:00 Joshua Colp <jcolp at digium.com>: > Marek Ce...

hello, i'm facing strange problem asterisk13.5 + chan_sip wss transport + SIPML5 1.5.230 person1 to person3 are behind different NATs audio devices double checked call from person1(chrome) to person2(chrome) works call from person1(chrome) to person 3(chrome) - no audio on both side (RTP flowing only in one direction) call from person2(chrome) to person 3(chrome) - no audio on both side

...l allow=alaw ;allow=gsm qualify=yes canreinvite=no dtmfmode=rfc2833 amaflags=billing context=testwebrtc nat=force_rport,comedia transport=udp,ws,wss encryption=yes avpf=yes force_avp=yes icesupport=yes directmedia=no dtlsenable=yes dtlsverify=fingerprint dtlscertfile=/etc/asterisk/keys/asterisk.pem dtlscafile=/etc/asterisk/keys/ca.crt dtlssetup=actpass SIP registration works fine : [Aug 9 22:12:00] == WebSocket connection from '178.119.146.190:36940' for protocol 'sip' accepted using version '13' [Aug 9 22:12:00] -- Registered SIP '770000wrtc' at 178.119.146.1...