search for: dtlssetup

...or '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.

...le=yes ; Tell Asterisk to enable DTLS for this peer ;dtlsverify=no ; Tell Asterisk to not verify your DTLS certs ;dtlscertfile=/etc/asterisk/keys/asterisk.pem ; Tell Asterisk where your DTLS cert file is ;dtlsprivatekey=/etc/asterisk/keys/asterisk.pem ; Tell Asterisk where your DTLS private key is ;dtlssetup=actpass ; Tell Asterisk to use actpass SDP parameter when setting up DTLS [1061] ; This will be the legacy SIP client type=friend username=1061 host=dynamic secret=fee50 ;encryption=yes ; Tell Asterisk to use encryption for this peer avpf=yes ; Tell Asterisk to use AVPF for this peer icesupport=ye...

...ullname 660 win8 hasvoicemail subscribemwi dtlsenable yes dtlsverify no dtlscertfile /etc/asterisk/keys/asterisk.pem dtlsprivatekey /etc/asterisk/keys/asterisk.pem dtlssetup actpass sippasswd md5pwd rpid domain testers.com sippasswd2 and my sip.conf: [general] bindport = 5070 bindaddr = PU.BL.IC.IP udpbindaddr = PU.BL.IC.IP tcpenable = yes limitonpeers = yes rtcachefriends =...

Dne 11.8.2015 v 12:18 Joshua Colp napsal(a): > Vinicius Fontes wrote: >> I'm having the same issue! The difference in my case is Asterisk server >> has a public IPv4 and the browser is behind a single NAT. >> >> I'm forwarding my configuration below (which I posted previously on >> asterisk-users). >> >> How can we debug ICE negotiation? >

...a=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=fingerprint:sha-256 48:3E:0C:59:BA:EB:6C:F9:5D:65:BF:08:54:63:C3:EA:AF:A9:60:9D:39:47:A...

...fullname: NULL cid_number: NULL callingpres: NULL mohinterpret: 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...

...fullname: NULL cid_number: NULL callingpres: NULL mohinterpret: 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: 50...

...hold=yes nat=yes [1000] deny=0.0.0.0/0.0.0.0 secret=6ff108122cce3b0b45e0abf374c14ef4 dtmfmode=rfc2833 canreinvite=no context=from-internal host=dynamic trustrpid=yes sendrpid=no type=friend nat=yes port=5060 qualify=yes qualifyfreq=60 transport=udp avpf=no icesupport=no dtlsenable=no dtlsverify=no dtlssetup=actpass encryption=no callgroup= pickupgroup= dial=SIP/1000 mailbox=1000 at device permit=0.0.0.0/0.0.0.0 callerid=Usuario 1 elx4 <1000> callcounter=yes faxdetect=no [1001] deny=0.0.0.0/0.0.0.0 secret=ce93963b0751ed9a88ec1badbc073fce dtmfmode=rfc2833 canreinvite=no context=from-internal host...

...WSS   Codecs       : (alaw|g729|gsm)   Useragent    : SIP.js/0.10.0   Reg. Contact : sip:u79mer6v at 1u7hp86jdg67.invalid;transport=ws   RTP Engine   : asterisk   Encryption   : Yes   RTCP Mux     : Yes avpf = yes force_avp =yes icesupport = yes dtlsenable = yes dtlsverify = fingerprint dtlssetup = actpass dtlsfingerprint = sha-256 Why is there "UNSUPPORTED OR FAILED" in the log when processing "a=ice-ufrag" and "ice-pwd" ?? Asterisk gives no "a=ice-ufrag" and "ice-pwd" in its "SIP/2.0 200 OK" response to the INVITE and thus s...

...ffer " sip.conf (important parts) [vr1a882] ... nat=force_rport,comedia canreinvite=no encryption=yes avpf=yes force_avp=yes icesupport=yes directmedia=yes transport=wss,ws dtlsrekey=60 dtlsverify=no dtlscertfile=/etc/pki/tls/certs/rapidssl.crt dtlsprivatekey=/etc/pki/tls/private/rapidssl.key dtlssetup=actpass sip dump <--- SIP read from WS:2.2.2.2:8558 ---> INVITE sip:887 at ipbx SIP/2.0 Via: SIP/2.0/WSS df7jal23ls0d.invalid;branch=z9hG4bKyhVsmJdAtwRvuOWz9BHiNo1DGcqp4grQ;rport From: "cervenka"<sip:vr1a882 at vhXXX.example.com>;tag=RDmpGm2Mubc5xQQ8NMli To: <sip:887 at...

...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 Cervenka wrote: > >> hello, &...

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

I need to get hooked up with this class, I could have students doing projects for homework :) Interested in RTCP? j On 6/26/23 7:45 PM, TTT wrote: > > I’m in training, so I have to demonstrate something SIP related.  I > figure it would be cool to hack a call, hanging it up while in > progress from outside Asterisk.  Doing so will demonstrate > use/knowledge of ARI, AMI, SIP,

...romdomain: testers.com videosupport: no contactpermit: NULL contactdeny: NULL fullname: 660 win8 subscribemwi: NULL dtlsenable: yes dtlsverify: no dtlscertfile: /etc/asterisk/keys/asterisk.pem dtlsprivatekey: /etc/asterisk/keys/asterisk.pem dtlssetup: actpass sippasswd: a84a4ddcda13d13c9573d5294047b6a2 rpid: NULL domain: testers.com sippasswd2: 5c4671ae1043e6116118fed39bee091a callbackextension: NULL insecure: NULL cheers, Olli -------------- next part -------------- An HTML attachment was scrub...

...anreinvite=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 178.119.146.190:36940 [Aug 9 22:12:00] &g...