Displaying 20 results from an estimated 26 matches for "actpass".
2015 Oct 28
2
Receiving Messages and Extensions Config for WebRTC
...ell 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=yes ; Tell...
2016 Jan 20
2
Incoming webrtc call succeeds in Firefox but fails in Google Chrome
...at=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=dynamic...
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?
>
2015 Apr 28
0
hi list need your help
...ive generation 0
a=candidate:1263319685 2 tcp 1518280447 192.168.88.26 0 typ host tcptype
active generation 0
a=ice-ufrag:8nMZ7w8bHdBBoY1a
a=ice-pwd:CFh1JMRfcT5BoH7aoKQuDgDR
a=fingerprint:sha-256
6D:5A:B7:7C:8E:1F:2A:F2:8C:DB:49:E4:8D:8F:BE:A0:84:AF:21:E2:A1:D7:DC:29:B1:EE:C3:0C:2C:AF:6C:04
a=setup:actpass
a=mid:audio
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
a=sendrecv
a=rtcp-mux
a=rtpmap:111 opus/48000/2
a=fmtp:111 minptime=10; useinbandfec=1
a=rtpmap:103 ISAC/16000
a=rtpmap:104 ISAC/32000
a=rtpmap:9 G722/8000
a=rtpm...
2023 Jun 28
1
SDP a=ice-ufrag & a=ice-pwd UNSUPPORTED OR FAILED
...PPORTED OR FAILED.
DEBUG[30891][C-00000000] chan_sip.c: Processing media-level (audio) SDP
a=fingerprint:sha-256
92:6B:C7:79:41:B1:42:78:2B:3A:75:8B:0B:D0:C7:4C:7C:4E:4F:2D:03:A2:DA:D9:BB:CE:B2:39:5D:20:A0:EF...
OK.
DEBUG[30891][C-00000000] chan_sip.c: Processing media-level (audio) SDP
a=setup:actpass... OK.
DEBUG[30891][C-00000000] chan_sip.c: Processing media-level (audio) SDP
a=mid:0... UNSUPPORTED OR FAILED.
DEBUG[30891][C-00000000] chan_sip.c: Processing media-level (audio) SDP
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level... UNSUPPORTED OR
FAILED.
DEBUG[30891][C-00000000] chan_...
2015 May 21
1
asterisk 13 webrtc
...;
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 ipbx>...
2014 Aug 22
0
Asterisk rejects sdp from webrtc client
...1.145.67.22 52821 typ srflx raddr
192.168.0.101 rport 52821 generation 0 //
a=ice-ufrag:QJy1Fslu8ITGYl/d //
a=ice-pwd:Q8N6+0PPj4CUG6leGAie7zaL //
a=ice-options:google-ice //
a=fingerprint:sha-256
CF:30:A7:7F:71:11:D4:5E:B0:E7:E3:F9:D8:C2:F4:60:2A:D0:76:46:F8:3A:97:01:C9:0C:5A:F7:B8:7D:C1:43
a=setup:actpass
a=mid:audio //
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level //
a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time //
a=rtpmap:111 opus/48000/2
a=fmtp:111 minptime=10
a=rtpmap:103 ISAC/16000
a=rtpmap:104 ISAC/32000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:106 C...
2014 Apr 14
1
how to configure callcentric peer
...11.192.161
> t=0 0
> m=audio 50960 RTP/AVP 18 0 8 101
> a=fmtp:18 annexb=no
> a=fmtp:101 0-15
> a=rtpmap:101 telephone-event/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:0 PCMU/8000
> a=rtpmap:18 G729/8000
> a=ptime:20
> a=sendrecv
> a=silenceSupp:off - - - -
> a=setup:actpass
> <------------->
> --- (11 headers 16 lines) ---
> Sending to 204.11.192.161:5060 (NAT)
> Sending to 204.11.192.161:5060 (NAT)
> Using INVITE request as basis request - 18075985-3606475083-968100 at msw2.telengy.net
> No matching peer for '<calling number>' fr...
2014 Sep 08
1
Asterisk removes ice lines in sdp when calling between webrtc clients
...0 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 = no
tos_s...
2015 Mar 23
2
PJSIP - Video Support for WebRTC
Hey i have an interesting topic to discuss here.
The main goal here is to be able to make a video call between two WebRTC endpoints registered on asterisk 13 it is a feature that definitely asterisk 13 should support .
the problems that i faced with this is the following and i hope i could get an advise here.
asterisk 13 vanilla version has some issues marking the video packets this complain
2015 May 04
0
Asterisk proxying a REFER
...263319685 2 tcp 1518280447 192.168.88.26 0 typ host tcptype
> active generation 0
> a=ice-ufrag:8nMZ7w8bHdBBoY1a
> a=ice-pwd:CFh1JMRfcT5BoH7aoKQuDgDR
> a=fingerprint:sha-256
> 6D:5A:B7:7C:8E:1F:2A:F2:8C:DB:49:E4:8D:8F:BE:A0:84:AF:21:E2:A1:D7:DC:29:B1:EE:C3:0C:2C:AF:6C:04
> a=setup:actpass
> a=mid:audio
> a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
> a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
> a=sendrecv
> a=rtcp-mux
> a=rtpmap:111 opus/48000/2
> a=fmtp:111 minptime=10; useinbandfec=1
> a=rtpmap:103 ISAC/16000
> a=rtpm...
2013 Jun 17
1
Has anyone succeeded in making a WebRTC call from Mozilla Nightly to Asterisk?
...; 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:A5:41:6B:...
2014 Aug 06
1
From and To headers contain same account in INVITEs
...dd423d 1 UDP 2130706431 1.1.1.1 18740 typ host
a=candidate:S5bdd423d 1 UDP 1694498815 1.1.1.1 18740 typ srflx
a=candidate:H5bdd423d 2 UDP 2130706430 1.1.1.1 18741 typ host
a=candidate:S5bdd423d 2 UDP 1694498814 1.1.1.1 18742 typ srflx
a=connection:new
a=setup:actpass
a=fingerprint:SHA-256
CE:EE:D9:28:EA:B0:6E:D0:CE:4F:5A:9A:FB:53:66:74:83:47:18:37:2F:76:C1:6D:10:C0:EE:FF:A4:56:F4:05
a=sendrecv
Here's the dialplan, nothing special:
exten => _XXX,1,NoOp(general : Dialed ${EXTEN})
same => n,Dial(SIP/${EXTEN},3600,rt)
same => n,Han...
2023 Jun 27
1
Get channel variables via ARI/AMI
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,
2014 Mar 14
0
sipML5, Ast12 and WebRTC: not acceptable here
...ation 0
a=candidate:1596293558 2 tcp 1509957375 10.10.5.106 0 typ host generation 0
a=ice-ufrag:l8AWdK4ft+AnAYGl
a=ice-pwd:3tLKvT97tf0GQr+e8v8bKncd
a=ice-options:google-ice
a=fingerprint:sha-256
89:D4:EB:6E:9C:41:36:03:A1:44:CD:A2:08:78:CD:86:FE:EC:30:09:53:0F:77:CE:BA:8E:DE:8C:1B:A1:41:10
a=setup:actpass
a=mid:audio
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=sendrecv
a=rtcp-mux
a=crypto:0 AES_CM_128_HMAC_SHA1_32
inline:dYMEPP1zoNS/W70Ln6cnBCtHXDCq6ciLZmHDHdFj
a=crypto:1 AES_CM_128_HMAC_SHA1_80
inline:Gr23SpFGDiukOKyrrfAauWssQ+3pYjD0jwkK9hOo
a=rtpmap:111 opus/48000/2
a=fmtp:111 minpt...
2015 Sep 15
3
Asterisk 13 WebRTC Status report
...rmines whether res_pjsip will use and enforce
dtls_verify=no ; Verify that the provided peer certificate is valid
(default:
dtls_rekey=0 ; Interval at which to renegotiate the TLS session and rekey
dtls_cert_file=/etc/pki/tls/certs/pbx.crt
dtls_private_key=/etc/pki/tls/private/pbx.key
dtls_setup=actpass
ice_support=yes ;This is specific to clients that support NAT traversal
media_use_received_transport=yes
[auth-userpass](!)
type=auth
auth_type=userpass
[aor-single-reg](!)
type=aor
remove_existing=yes
max_contacts=1
;===============DEVICES
[webrtc1](endpoint-basic)
auth=webrtc1
aors=webrtc1...
2019 May 10
4
Asterisk 13.26.0 webRTC: Asterisk not passing along video
...-cost 10
[May 10 10:45:24] a=ice-ufrag:y8md
[May 10 10:45:24] a=ice-pwd:nyjEuDKhDVeu8B+OyvuEp6le
[May 10 10:45:24] a=ice-options:trickle
[May 10 10:45:24] a=fingerprint:sha-256
C9:33:B0:E9:7C:F4:F2:39:98:A6:5C:AE:16:7F:5E:18:99:8F:9F:EB:DC:C6:E3:D5:EA:5B:AE:CD:DE:75:79:0B
[May 10 10:45:24] a=setup:actpass
[May 10 10:45:24] a=mid:audio
[May 10 10:45:24] a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
[May 10 10:45:24] a=sendrecv
[May 10 10:45:24] a=rtcp-mux
[May 10 10:45:24] a=rtpmap:111 opus/48000/2
[May 10 10:45:24] a=rtcp-fb:111 transport-cc
[May 10 10:45:24] a=fmtp:111 minptime=10;useinban...
2014 Aug 25
0
WebRTC / Rejecting secure audio stream errors
...:1 1 UDP 2113667327 192.168.1.161 52081 typ host
-a=candidate:2 1 UDP 2113667327 195.8.117.161 54978 typ host
-a=candidate:0 2 UDP 2113667326 10.10.1.144 58499 typ host
-a=candidate:1 2 UDP 2113667326 192.168.1.161 33161 typ host
-a=candidate:2 2 UDP 2113667326 195.8.117.161 36491 typ host
+a=setup:actpass
+a=candidate:0 1 UDP 2122252543 10.10.1.90 60221 typ host
+a=candidate:1 1 UDP 1686110207 195.8.117.200 60221 typ srflx raddr
10.10.1.90 rport 60221
+a=candidate:2 1 UDP 8388607 195.8.117.59 51390 typ relay raddr
195.8.117.59 rport 51390
+a=candidate:3 1 UDP 2122187007 192.168.150.1 38505 typ host...
2014 Dec 05
2
Inbound call from sip peer to internal webrtc peer fails while internal sip-webrtc calls work
...ap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=ice-ufrag:75a7e84431d15f682bd728ee10bd867d
a=ice-pwd:028c19574216643c12188a8530f278f8
a=candidate:H5bdd423d 1 UDP 2130706431 PU.BL.IC.IP 15662 typ host
a=candidate:H5bdd423d 2 UDP 2130706430 PU.BL.IC.IP 15663 typ host
a=connection:new
a=setup:actpass
a=fingerprint:SHA-256
CE:EE:D9:28:EA:B0:6E:D0:CE:4F:5A:9A:FB:53:66:74:83:47:18:37:2F:76:C1:6D:10:C0:EE:FF:A4:56:F4:05
a=sendrecv
Here the call fails (sip peer 201 calls from outside the server to webrtc
peer 660):
Invite that Asterisk receives:
PU.BL.IC.IP:5060 > PU.BL.IC.IP:5070: SIP, length...
2015 Aug 11
2
webrtc no audio
...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,
>>...