You're probably way better off asking this on the users's mailing list ([email protected]<mailto:[email protected]>) instead of asking it here. It would also be more likely responded to if you CREATE A BRAND NEW EMAIL TREAD instead of responding digest that seems to have nothing to do with your question.
-Justin ________________________________ From: [email protected] [mailto:[email protected]] On Behalf Of raj singh Sent: Monday, December 02, 2013 3:12 PM To: [email protected] Subject: Re: [asterisk-dev] asterisk-dev Digest, Vol 113, Issue 2 how to download phonebook in asterisk On Mon, Dec 2, 2013 at 6:58 PM, <[email protected]<mailto:[email protected]>> wrote: Send asterisk-dev mailing list submissions to [email protected]<mailto:[email protected]> To subscribe or unsubscribe via the World Wide Web, visit http://lists.digium.com/mailman/listinfo/asterisk-dev or, via email, send a message with subject or body 'help' to [email protected]<mailto:[email protected]> You can reach the person managing the list at [email protected]<mailto:[email protected]> When replying, please edit your Subject line so it is more specific than "Re: Contents of asterisk-dev digest..." Today's Topics: 1. Re: [Code Review] 3036: res_pjsip_transport_websocket: Fix crash with security events and improve implementation (Joshua Colp) 2. SIP request (Stas Kobzar) 3. Re: SIP request (Joshua Colp) 4. Re: SIP request (Stas Kobzar) 5. WebRTC over SRTP-DTLS (nitesh bansal) 6. Re: WebRTC over SRTP-DTLS (nitesh bansal) ---------------------------------------------------------------------- Message: 1 Date: Sun, 01 Dec 2013 19:56:42 -0000 From: "Joshua Colp" <[email protected]<mailto:[email protected]>> To: "Matt Jordan" <[email protected]<mailto:[email protected]>>, "Joshua Colp" <[email protected]<mailto:[email protected]>>, "Joshua Colp" <[email protected]<mailto:[email protected]>>, "Asterisk Developers" <[email protected]<mailto:[email protected]>> Subject: Re: [asterisk-dev] [Code Review] 3036: res_pjsip_transport_websocket: Fix crash with security events and improve implementation Message-ID: <[email protected]> Content-Type: text/plain; charset="utf-8" ----------------------------------------------------------- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/3036/ ----------------------------------------------------------- (Updated Dec. 1, 2013, 1:56 p.m.) Status ------ This change has been marked as submitted. Review request for Asterisk Developers. Changes ------- Committed in revision 403256 Bugs: ASTERISK-22897 https://issues.asterisk.org/jira/browse/ASTERISK-22897 Repository: Asterisk Description ------- The attached change fixes/tweaks a few things: Security events now determine the transport type using a saner method (by looking at the transport type on the message itself), which includes WebSocket based connections. This means no having to create a container of configured transports and no having to iterate them. Connection handling now uses the built-in PJSIP transport manager for figuring out what active transport/connection to use. This is based on the target IP address/port of the active WebSocket connection. Diffs ----- /branches/12/res/res_pjsip_transport_websocket.c 403236 /branches/12/res/res_pjsip/security_events.c 403236 /branches/12/res/res_pjsip/pjsip_options.c 403236 /branches/12/res/res_pjsip/location.c 403236 /branches/12/res/res_pjsip.c 403236 /branches/12/include/asterisk/res_pjsip.h 403236 Diff: https://reviewboard.asterisk.org/r/3036/diff/ Testing ------- Connected using JsSIP, confirmed no crash and that traffic is sent out the proper connection. Thanks, Joshua Colp -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-dev/attachments/20131201/a7d21e6b/attachment-0001.html> ------------------------------ Message: 2 Date: Sun, 1 Dec 2013 20:49:51 -0500 From: Stas Kobzar <[email protected]<mailto:[email protected]>> To: [email protected]<mailto:[email protected]> Subject: [asterisk-dev] SIP request Message-ID: <CAMYcjooWTZY6Or9KFaRbg1uT91x0JN_Aoxqz7H=bhpjh0l5...@mail.gmail.com<mailto:[email protected]>> Content-Type: text/plain; charset="iso-8859-1" Hello list, I am trying to develop my own Asterisk module. I need to create and send PUBLISH SIP message with special headers and/or message body. I found in that in include folder there is a sip_api.h (Asterisk 11), an API for INFO method. But I can not figure out how to access to other methods. Is it possible to use chan_sip methods in other modules? If yes, could you, please, give me a hint where to look? Thank you, -- Stas Kobzar -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-dev/attachments/20131201/e572cadc/attachment-0001.html> ------------------------------ Message: 3 Date: Sun, 01 Dec 2013 22:02:49 -0400 From: Joshua Colp <[email protected]<mailto:[email protected]>> To: Asterisk Developers Mailing List <[email protected]<mailto:[email protected]>> Subject: Re: [asterisk-dev] SIP request Message-ID: <[email protected]<mailto:[email protected]>> Content-Type: text/plain; charset=ISO-8859-1; format=flowed Stas Kobzar wrote: > Hello list, > > I am trying to develop my own Asterisk module. > I need to create and send PUBLISH SIP message with special headers > and/or message body. > > I found in that in include folder there is a sip_api.h (Asterisk 11), an > API for INFO method. But I can not figure out how to access to other > methods. > > Is it possible to use chan_sip methods in other modules? If yes, could > you, please, give me a hint where to look? There is no way to do this. It doesn't provide any APIs to extend it. Any additional functionality has to be built into chan_sip itself. In Asterisk 12 the new PJSIP based modules DO provide various APIs to allow you to do exactly this. Cheers, -- Joshua Colp Digium, Inc. | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: www.digium.com<http://www.digium.com> & www.asterisk.org<http://www.asterisk.org> ------------------------------ Message: 4 Date: Sun, 1 Dec 2013 21:10:30 -0500 From: Stas Kobzar <[email protected]<mailto:[email protected]>> To: Asterisk Developers Mailing List <[email protected]<mailto:[email protected]>> Subject: Re: [asterisk-dev] SIP request Message-ID: <camycjoocjbjpydqdv5kvlh7cgyxzwatgfvy50lvvsnrdu1x...@mail.gmail.com<mailto:camycjoocjbjpydqdv5kvlh7cgyxzwatgfvy50lvvsnrdu1x...@mail.gmail.com>> Content-Type: text/plain; charset="iso-8859-1" Thank you! On Sun, Dec 1, 2013 at 9:02 PM, Joshua Colp <[email protected]<mailto:[email protected]>> wrote: > Stas Kobzar wrote: > >> Hello list, >> >> I am trying to develop my own Asterisk module. >> I need to create and send PUBLISH SIP message with special headers >> and/or message body. >> >> I found in that in include folder there is a sip_api.h (Asterisk 11), an >> API for INFO method. But I can not figure out how to access to other >> methods. >> >> Is it possible to use chan_sip methods in other modules? If yes, could >> you, please, give me a hint where to look? >> > > There is no way to do this. It doesn't provide any APIs to extend it. Any > additional functionality has to be built into chan_sip itself. > > In Asterisk 12 the new PJSIP based modules DO provide various APIs to > allow you to do exactly this. > > Cheers, > > -- > Joshua Colp > Digium, Inc. | Senior Software Developer > 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA > Check us out at: www.digium.com<http://www.digium.com> & > www.asterisk.org<http://www.asterisk.org> > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-dev mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-dev > -- Stas Kobzar VoIP Developer 514 284 2020 www.modulis.ca<http://www.modulis.ca> -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-dev/attachments/20131201/2e8ea161/attachment-0001.html> ------------------------------ Message: 5 Date: Mon, 2 Dec 2013 12:29:04 +0100 From: nitesh bansal <[email protected]<mailto:[email protected]>> To: Asterisk Developers Mailing List <[email protected]<mailto:[email protected]>> Subject: [asterisk-dev] WebRTC over SRTP-DTLS Message-ID: <CAOLsin5qn4MwCEY72e+v6dCAZO99yLTthZz=qxol0_vo0pt...@mail.gmail.com<mailto:[email protected]>> Content-Type: text/plain; charset="iso-8859-1" Hello everybody, I want to setup a basic Demo of WebRTC using Asterisk as WebServer and SRTP-DTLS. I got the demo setup using SRTP-DES with chrome, chrome is porpoising both DTLS and DES, Asterisk responds with DES abd call is connected. But i want asterisk to propose DTLS also in its response, can you please tell me if asterisk supports DTLS and if yes, is there a wiki page with the documentation? I could not find any relevant wikipage. Regards, Nitesh -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-dev/attachments/20131202/76bd0472/attachment-0001.html> ------------------------------ Message: 6 Date: Mon, 2 Dec 2013 14:32:16 +0100 From: nitesh bansal <[email protected]<mailto:[email protected]>> To: Asterisk Developers Mailing List <[email protected]<mailto:[email protected]>> Subject: Re: [asterisk-dev] WebRTC over SRTP-DTLS Message-ID: <CAOLsin6HUGJo3E98Vf73SmY+TJEtiGXdnDF+B7=U8V=3_Wt1=q...@mail.gmail.com<mailto:[email protected]>> Content-Type: text/plain; charset="iso-8859-1" Sorry, i forgot to mention Asterisk version, i am using Asterisk 11.4 Regards, Nitesh On Mon, Dec 2, 2013 at 12:29 PM, nitesh bansal <[email protected]<mailto:[email protected]>>wrote: > Hello everybody, > > I want to setup a basic Demo of WebRTC using Asterisk as WebServer and > SRTP-DTLS. > I got the demo setup using SRTP-DES with chrome, chrome is porpoising both > DTLS and DES, > Asterisk responds with DES abd call is connected. > But i want asterisk to propose DTLS also in its response, can you please > tell me if asterisk supports DTLS and if yes, is there a wiki page with the > documentation? > I could not find any relevant wikipage. > > Regards, > Nitesh > -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-dev/attachments/20131202/a139a132/attachment.html> ------------------------------ _______________________________________________ --Bandwidth and Colocation Provided by http://www.api-digital.com-- AstriCon 2010 - October 26-28 Washington, DC Put in your talk proposal: http://www.bit.ly/speak-astricon2010 asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev End of asterisk-dev Digest, Vol 113, Issue 2 ******************************************** -- Thanks & Regards R.S.Parihar +919650049450 [email protected]<mailto:[email protected]>
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