Displaying 13 results from an estimated 13 matches for "dtlscertfile".
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tlscertfile
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...