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2016 Aug 15
5
Realtime SIP peers do not register any more after upgrade to Asterisk 13
...19.90.240>' 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.
2015 Oct 28
2
Receiving Messages and Extensions Config for WebRTC
...DP or WebSockets force_avp=yes ; Force Asterisk to use avp. Introduced in Asterisk 11.11 nat=force_rport,comedia accept_outofcall_message=yes outofcall_message_context=messages ;dtlsenable=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 clie...
2014 Sep 08
1
Asterisk removes ice lines in sdp when calling between webrtc clients
...esters.com videosupport no contactpermit contactdeny fullname 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:...
2015 Aug 12
2
webrtc no audio
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? >
2013 Jun 17
1
Has anyone succeeded in making a WebRTC call from Mozilla Nightly to Asterisk?
...ynamic 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=fingerprint:sha-256 48:3E:0C:59:BA:EB:6C:F9:5...
2014 Aug 11
1
Letting rtp profiles be handled by rtpengine instead of Asterisk
...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 ipaddr: PU.BL.IC.IP...
2014 Aug 06
1
From and To headers contain same account in INVITEs
...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: 5060 regseconds: 14073...
2015 May 21
1
asterisk 13 webrtc
...with SIP.js) chan_sip.c:10496 process_sdp: Can't provide secure audio requested in SDP offer " 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...
2015 Aug 11
2
webrtc no audio
...t=mysecret context=default type=friend icesupport=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...
2015 Aug 10
2
webrtc no audio
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
2016 Jan 20
2
Incoming webrtc call succeeds in Firefox but fails in Google Chrome
...0/0.0.0.0 secret=ce93963b0751ed9a88ec1badbc073fce dtmfmode=rfc2833 canreinvite=no context=from-internal host=dynamic trustrpid=yes sendrpid=no type=friend nat=yes port=5060 qualify=yes qualifyfreq=60 transport=wss,ws,udp,tcp,tls avpf=yes icesupport=yes dtlsenable=yes dtlsverify=no dtlssetup=actpass dtlscertfile=/var/lib/asterisk/keys/localhost.crt dtlsprivatekey=/var/lib/asterisk/keys/localhost.key encryption=yes callgroup= pickupgroup= dial=SIP/1001 mailbox=1001 at device permit=0.0.0.0/0.0.0.0 callerid=Usuario Alex <1001> callcounter=yes faxdetect=no With this setup, I can make calls using the S...
2014 Dec 05
2
Inbound call from sip peer to internal webrtc peer fails while internal sip-webrtc calls work
...test amaflags: NULL mailbox: NULL regexten: NULL regserver: fromdomain: 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...
2016 Aug 09
2
Asterisk 11.23.0 on CentOS6 : how to get ICE support ?
...fromuser=770000wrtc secret=987654 disallow=all 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...