[asterisk-dev] asterisk-dev Digest, Vol 113, Issue 2

Mon Dec 2 17:12:23 CST 2013

how to download phonebook in asterisk

On Mon, Dec 2, 2013 at 6:58 PM, <asterisk-dev-request at lists.digium.com>wrote:

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> 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" <reviewboard at asterisk.org>
> To: "Matt Jordan" <mjordan at digium.com>, "Joshua Colp"
>         <reviewboard at asterisk.org>, "Joshua Colp" <jcolp at digium.com>,
>         "Asterisk Developers" <asterisk-dev at lists.digium.com>
> Subject: Re: [asterisk-dev] [Code Review] 3036:
>         res_pjsip_transport_websocket: Fix crash with security events and
>         improve implementation
> Message-ID: <20131201195642.12391.39676 at sonic.digium.api>
> 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
>
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> ------------------------------
>
> Message: 2
> Date: Sun, 1 Dec 2013 20:49:51 -0500
> From: Stas Kobzar <stas.kobzar at modulis.ca>
> To: asterisk-dev at lists.digium.com
> Subject: [asterisk-dev] SIP request
> Message-ID:
>         <CAMYcjooWTZY6Or9KFaRbg1uT91x0JN_Aoxqz7H=
> BHPJH0L5skQ at mail.gmail.com>
> 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
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> ------------------------------
>
> Message: 3
> Date: Sun, 01 Dec 2013 22:02:49 -0400
> From: Joshua Colp <jcolp at digium.com>
> To: Asterisk Developers Mailing List <asterisk-dev at lists.digium.com>
> Subject: Re: [asterisk-dev] SIP request
> Message-ID: <529BEA49.6060003 at digium.com>
> 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 <stas.kobzar at modulis.ca>
> To: Asterisk Developers Mailing List <asterisk-dev at lists.digium.com>
> Subject: Re: [asterisk-dev] SIP request
> Message-ID:
>         <
> CAMYcjoocjBJpydqDV5KvLh7CgyxzWAtGFVy50LvVsNRDU1XOjA at mail.gmail.com>
> Content-Type: text/plain; charset="iso-8859-1"
>
> Thank you!
>
>
> On Sun, Dec 1, 2013 at 9:02 PM, Joshua Colp <jcolp at digium.com> 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
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> ------------------------------
>
> Message: 5
> Date: Mon, 2 Dec 2013 12:29:04 +0100
> From: nitesh bansal <nitesh.bansal at gmail.com>
> To: Asterisk Developers Mailing List <asterisk-dev at lists.digium.com>
> Subject: [asterisk-dev] WebRTC over SRTP-DTLS
> Message-ID:
>         <CAOLsin5qn4MwCEY72e+v6dCAZO99yLTthZz=
> QxOL0_vO0ptoNQ at mail.gmail.com>
> 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
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> ------------------------------
>
> Message: 6
> Date: Mon, 2 Dec 2013 14:32:16 +0100
> From: nitesh bansal <nitesh.bansal at gmail.com>
> To: Asterisk Developers Mailing List <asterisk-dev at lists.digium.com>
> Subject: Re: [asterisk-dev] WebRTC over SRTP-DTLS
> Message-ID:
>         <CAOLsin6HUGJo3E98Vf73SmY+TJEtiGXdnDF+B7=U8V=3_Wt1=
> Q at mail.gmail.com>
> 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 <nitesh.bansal at gmail.com
> >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
> >
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> ------------------------------
>
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> End of asterisk-dev Digest, Vol 113, Issue 2
> ********************************************
>

-- 
Thanks & Regards
R.S.Parihar
+919650049450
rajendraparihar781 at gmail.com
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