how to download phonebook in asterisk
On Mon, Dec 2, 2013 at 6:58 PM, <[email protected]>wrote: > Send asterisk-dev mailing list submissions to > [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] > > You can reach the person managing the list at > [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]> > To: "Matt Jordan" <[email protected]>, "Joshua Colp" > <[email protected]>, "Joshua Colp" <[email protected]>, > "Asterisk Developers" <[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]> > To: [email protected] > Subject: [asterisk-dev] SIP request > Message-ID: > <CAMYcjooWTZY6Or9KFaRbg1uT91x0JN_Aoxqz7H= > [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]> > To: Asterisk Developers Mailing List <[email protected]> > Subject: Re: [asterisk-dev] SIP request > Message-ID: <[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 & www.asterisk.org > > > > ------------------------------ > > Message: 4 > Date: Sun, 1 Dec 2013 21:10:30 -0500 > From: Stas Kobzar <[email protected]> > To: Asterisk Developers Mailing List <[email protected]> > Subject: Re: [asterisk-dev] SIP request > Message-ID: > < > 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]> 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 & 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 > -------------- 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]> > To: Asterisk Developers Mailing List <[email protected]> > Subject: [asterisk-dev] WebRTC over SRTP-DTLS > Message-ID: > <CAOLsin5qn4MwCEY72e+v6dCAZO99yLTthZz= > [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]> > To: Asterisk Developers Mailing List <[email protected]> > Subject: Re: [asterisk-dev] WebRTC over SRTP-DTLS > Message-ID: > <CAOLsin6HUGJo3E98Vf73SmY+TJEtiGXdnDF+B7=U8V=3_Wt1= > [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] > >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]
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