search for: actpass

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, >>...