[asterisk-users] sipml5
I have a machine that is completely NOT on the internet - closed network. Can sipml5 work there ? how ? It cannot use LetsEncrypt or anything. can self sign certs work ? IS there another way. Thanks Jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] sipml5
I have my asterisk 18 working with
https://www.doubango.org/sipml5/call.htm?svn=252#
I then tried to take the 15 lines of javascript library API (below) and
when it runs I get
asterisk console message about "failed to authenticate".I took ALL the
same settings I was using in the above URL - and plugged into the
javascript function below
The console log says 403 forbidden.
Is there a trick to get the API working ?
Any pointers to share ? Thanks.
Jerry
SIPml.init(
function(e){
var stack = new SIPml.Stack({realm: 'example.org',
impi: 'bob', impu: 'sip:b...@example.org', password: 'mysecret',
events_listener: { events: 'started', listener:
function(e){
var callSession =
stack.newSession('call-audiovideo', {
video_local:
document.getElementById('video-local'),
video_remote:
document.getElementById('video-remote'),
audio_remote:
document.getElementById('audio-remote')
});
callSession.call('alice');
}
}
});
stack.start();
}
);
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
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New to Asterisk? Start here:
https://wiki.asterisk.org/wiki/display/AST/Getting+Started
asterisk-users mailing list
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[asterisk-users] sipml5 how many video connections
Hi - Any one using SIPML5 ? How many video connections can a "normal" asterisk server box (2.2G 8GIG ram) handle in a SINGLE video session ? Thanks, Jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIPml5
Hi All - I am playing with SIPML5. I was getting an error about wss I fixed that by doing : cat privkey.pem > asterisk.pem cat fullchain.pem >> asterisk.pem with my letsencrypt certificate. and setting tlscertfile=/etc/letsencrypt/live/mypath/asterisk.pem But now when I use the https://www.doubango.org/sipml5/call.htm and I click Login... I get this error on asterisk /var/log/asterisk/messages [Sep 27 14:58:08] NOTICE[46486] chan_sip.c: Registration from '"SIPMl5"< sip:jerry.g...@somewhere.com>' failed for 'IP:37994' - Wrong password my sip.conf entry is this: [jerry.g...@somewhere.com] type=friend defaultname=jerry.g...@somewhere.com defaultuser=jerry.g...@somewhere.com secret=(matches that on the website page) encryption=yes avpf=yes force_avpf=yes dtlsenable=yes dtlsverify=fingerprint dtlscertfile=/etc/letsencrypt/live/mypath/asterisk.pem dtmfmode=RFC2833 host=dynamic description=Test context=sipml5 qualify=yes rtptimeout=60 rtpholdtimeout=60 rtpkeepalive=60 callerid="Jerry.Geis" qualify=no canreinvite=yes timezone=1 nat=force_rport,comedia disallow=all allow=ulaw allow=alaw allow=gsm Not sure what I dont have that is not accespting the PW. I am using 18.6.0 and still on the OLD chan_sip.c (which is working fine for everything else). Thanks Jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] sipML5, Ast12 and WebRTC: not acceptable here
Hi All. I'm running some tests with the latest Asterisk SVN-branch-12-r410493M compiled with fresh github pjsip and srtp 1.4.2 on an i386 centOS machine (2.6.32-358.18.1.el6.i686). As a client I'm using the sipMLP WebRTC javascript softphone running on Chrome 33.0.1750.146 m. I have the softphone correctly registered on the Asterisk machine but as soon as I try to start a new call from the softphone, Asterisk answers with a 488 not acceptable here. I'm probably missing something but I'm not able to find what and where. Is there someone able to point me to the right direction? Below is my configuration. The sofpthone is registered as 1060. Thanks in advance. Marco Signorini. pjsip.conf: [transport-tls] type=transport protocol=tls bind=0.0.0.0 cert_file=/etc/asterisk/sslcert.pem method=tlsv1 [1060] type=endpoint transport=transport-tls context=from-internal use_avpf=yes media_encryption=sdes disallow=all allow=alaw allow=ulaw aors=1060 auth=1060 [1060] type=auth auth_type=userpass password=1060 username=1060 [1060] type=aor max_contacts=10 [204] http.conf: enabled=yes bindaddr=10.10.5.49 bindport=8088 CLI pjsip show endpoints Endpoint: 1060 Not in use0 of inf InAuth: 1060/1060 Aor: 1060 10 Contact: 1060/sip:1060@10.10.5.106:54083;transport=ws;rt Unknown nan Transport: transport-tls tls 0 0 0.0.0.0:5061 Endpoint: 204 Not in use0 of inf InAuth: 204/204 Aor: 2041 Contact: 204/sip:204@10.10.5.120:5066;transport=udp Unknown nan Transport: transport-udp udp 0 0 0.0.0.0:5060 *CLI pjsip show transport transport-tls Transport: TransportId Type cos tos BindAddress = Transport: transport-tls tls 0 0 0.0.0.0:5061 ParameterName : ParameterValue == async_operations : 1 bind : 0.0.0.0:5061 ca_list_file : cert_file : /etc/asterisk/sslcert.pem cipher : cos: 0 domain : external_media_address : external_signaling_address : external_signaling_port: 0 local_net : method : tlsv1 password : priv_key_file : protocol : tls require_client_cert: No tos: CS0 verify_client : No verify_server : No And this is the relevant SIP data exchange (with public IP hidden): *CLI --- Received SIP request (2420 bytes) from WS:10.10.5.106:54411 --- INVITE sip:204@10.10.5.49 SIP/2.0 Via: SIP/2.0/WS df7jal23ls0d.invalid;branch=z9hG4bKiBw81ooU7ybSRbRqr8TOqWkMPQRdkMXo;rport From: John Doe (101)sip:1060@10.10.5.49;tag=heMv1HvlT7DeQxPxuqcq To: sip:204@10.10.5.49 Contact: John Doe (101)sip:1060@df7jal23ls0d.invalid;rtcweb-breaker=yes;click2call=no;transport=ws;impi=1060;ha1=0b2413e6f3c96a0517b4413a6f6ce7ae;+g.oma.sip-im;+sip.ice;language=en,fr,it Call-ID: 636a5d79-5fda-f79a-cc4b-9ba18d060edc CSeq: 38718 INVITE Content-Type: application/sdp Content-Length: 1827 Max-Forwards: 70 User-Agent: IM-client/OMA1.0 sipML5 v=0 o=- 365893986064703740 2 IN IP4 127.0.0.1 s=Doubango Telecom - chrome t=0 0 a=group:BUNDLE audio a=msid-semantic: WMS dXVhxyOSxULu3iClZayhTeEBzH2voboiJJ28 m=audio 37874 RTP/SAVPF 111 103 104 0 8 106 105 13 126 c=IN IP4 85.0.XXX.XXX a=rtcp:37874 IN IP4 85.0.XXX.XXX a=candidate:296123718 1 udp 2113937151 10.10.5.106 63858 typ host generation 0 a=candidate:296123718 2 udp 2113937151 10.10.5.106 63858 typ host generation 0 a=candidate:3103388307 1 udp 1845501695 85.0.XXX.XXX 37874 typ srflx raddr 10.10.5.106 rport 63858 generation 0 a=candidate:3103388307 2 udp 1845501695 85.0.XXX.XXX 37874 typ srflx raddr 10.10.5.106 rport 63858 generation 0 a=candidate:1596293558 1 tcp 1509957375 10.10.5.106 0 typ host generation 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 minptime=10 a=rtpmap:103 ISAC/16000 a=rtpmap:104 ISAC/32000 a=rtpmap:0
