[asterisk-users] Asterisk peer definition registration
Hi, I am using Asterisk currently as a SIP proxy to a SIP provider. Inside my real-time, I would set the SIP credential based on what the user has provided. For example [name] type=peer defaultuser=USER_PROVIDED secret=USER_PROVIDED host=USER_PROVIDED When I reset Asterisk, Asterisk will attempt to register with the sip provider. And if there are sufficiently amount of records with invalid credentials, I'll get blocked by the SIP provider as they might think that I'm brute forcing. Just a question to check if there's any chance I could ask Asterisk not to register when I reset. Or is there any other possible solution for this? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk peer definition registration
Is there a way that I could set the configuration for reloading after ITSP brute force timer expiration? On Sun, Aug 17, 2014 at 3:51 AM, Paul Belanger paul.belan...@polybeacon.com wrote: On Sat, Aug 16, 2014 at 11:21 AM, Steve Ng steveng.1...@gmail.com wrote: Hi, I am using Asterisk currently as a SIP proxy to a SIP provider. Inside my real-time, I would set the SIP credential based on what the user has provided. For example [name] type=peer defaultuser=USER_PROVIDED secret=USER_PROVIDED host=USER_PROVIDED When I reset Asterisk, Asterisk will attempt to register with the sip provider. And if there are sufficiently amount of records with invalid credentials, I'll get blocked by the SIP provider as they might think that I'm brute forcing. Just a question to check if there's any chance I could ask Asterisk not to register when I reset. Or is there any other possible solution for this? No, only reload after your ITSP brute force timer has expired. -- Paul Belanger | PolyBeacon, Inc. Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode) Github: https://github.com/pabelanger | Twitter: https://twitter.com/pabelanger -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk websocket with Nginx 502 Gateway error
Hi all, I am using the following set up: SIPML5 -- Nginx -- Asterisk, where NGINX as a reverse proxy, main purpose is to take in wss and route to Asterisk's ws. However, I am facing this issue recently where Nginx will return 502 gateway error of 8018#0: *24183 upstream prematurely closed connection while reading response header from upstream, client: 116.15.31.xxx, server: asteriskstage.xxx.yy, request: GET / HTTP/1.1, upstream: http://127.0.0.1:8088/ws;, host: asteriskstage.xxx.yy Anyone has any idea why? Here's my nginx config: http://pastebin.com/UU0G3YLh Regards, Steve -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] WSS over Asterisk
Hi, Have anyone tried using SIPML5 to connect to Asterisk over wss? I'm having the error as shown below Connecting to 'wss://54.xxx.xxx.xxx:8080/ws wss://54.254.228.251:8080/ws' SIPml-api.js?svn=224:1 ==stack event = starting SIPml-api.js?svn=224:1 __tsip_transport_ws_onerror SIPml-api.js?svn=224:1 __tsip_transport_ws_onclose SIPml-api.js?svn=224:1 ==stack event = failed_to_start Where if I'm connecting through ws://54.xxx.xxx.:8080/ws, it works fine. Any idea why? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] WSS over Asterisk
I am using Asterisk v12.3. As far as DTLS, I understand that applying the following Javascript will temporarily fix for SIPML5 to Asterisk: https://gist.github.com/steve-ng/14b9b88af43f92db1e46 WS works for me, its just wss which I'm stuck currently. On Thu, Jun 12, 2014 at 4:37 AM, Miguel Molina mfmolina-lis...@millenium.com.co wrote: El 11/06/2014 1:52 p. m., Matthew Jordan escribió: On Wed, Jun 11, 2014 at 1:32 PM, William Hetherington w...@willwh.com wrote: Chrome 35 broke all of this you need to be using DTLS now I believe. I had working secure web sockets with asterisk 12.2.x and chrome 34 and then google broke eveything :) I have not yet got around to test out DTLS etc. with chrome 35 Just so I don't waste too much time when I go to test, does anyone know if all that's required for DTLS on the asterisk side is the following in sip.conf? dtlsenable=yes dtlsverify=yes dtlsrekey=60 dtlscafile=/usr/local/share/ca-certificates/myCA.crt dtlscertfile=/etc/ssl/mycert.com.pem dtlssetup=actpass I assume I also need TLS configs in http.conf Signalling is independent of the media; DTLS only affects the media. However, there are known issues with Chrome's negotiation of DTLS and Asterisk - see https://issues.asterisk.org/jira/browse/ASTERISK-22961 -- Matthew Jordan Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com http://asterisk.org It is broken in Chrome (firefox never had SDES) because the WebRTC standard favoured the DTLS SRTP implementation instead of the SDES one. The thing is that although Asterisk supports DTLS implementation, it only supports SHA-1 hashing but both Firefox and Chrome work with SHA-256. The patch proposed in ASTERISK-22961 is an effort to solve this issue. Best regards -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
