<div dir="ltr">how to download phonebook in asterisk</div><div class="gmail_extra"><br><br><div class="gmail_quote">On Mon, Dec 2, 2013 at 6:58 PM, <span dir="ltr"><<a href="mailto:asterisk-dev-request@lists.digium.com" target="_blank">asterisk-dev-request@lists.digium.com</a>></span> wrote:<br>
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Today's Topics:<br>
<br>
1. Re: [Code Review] 3036: res_pjsip_transport_websocket: Fix<br>
crash with security events and improve implementation (Joshua Colp)<br>
2. SIP request (Stas Kobzar)<br>
3. Re: SIP request (Joshua Colp)<br>
4. Re: SIP request (Stas Kobzar)<br>
5. WebRTC over SRTP-DTLS (nitesh bansal)<br>
6. Re: WebRTC over SRTP-DTLS (nitesh bansal)<br>
<br>
<br>
----------------------------------------------------------------------<br>
<br>
Message: 1<br>
Date: Sun, 01 Dec 2013 19:56:42 -0000<br>
From: "Joshua Colp" <<a href="mailto:reviewboard@asterisk.org">reviewboard@asterisk.org</a>><br>
To: "Matt Jordan" <<a href="mailto:mjordan@digium.com">mjordan@digium.com</a>>, "Joshua Colp"<br>
<<a href="mailto:reviewboard@asterisk.org">reviewboard@asterisk.org</a>>, "Joshua Colp" <<a href="mailto:jcolp@digium.com">jcolp@digium.com</a>>,<br>
"Asterisk Developers" <<a href="mailto:asterisk-dev@lists.digium.com">asterisk-dev@lists.digium.com</a>><br>
Subject: Re: [asterisk-dev] [Code Review] 3036:<br>
res_pjsip_transport_websocket: Fix crash with security events and<br>
improve implementation<br>
Message-ID: <20131201195642.12391.39676@sonic.digium.api><br>
Content-Type: text/plain; charset="utf-8"<br>
<br>
<br>
-----------------------------------------------------------<br>
This is an automatically generated e-mail. To reply, visit:<br>
<a href="https://reviewboard.asterisk.org/r/3036/" target="_blank">https://reviewboard.asterisk.org/r/3036/</a><br>
-----------------------------------------------------------<br>
<br>
(Updated Dec. 1, 2013, 1:56 p.m.)<br>
<br>
<br>
Status<br>
------<br>
<br>
This change has been marked as submitted.<br>
<br>
<br>
Review request for Asterisk Developers.<br>
<br>
<br>
Changes<br>
-------<br>
<br>
Committed in revision 403256<br>
<br>
<br>
Bugs: ASTERISK-22897<br>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-22897" target="_blank">https://issues.asterisk.org/jira/browse/ASTERISK-22897</a><br>
<br>
<br>
Repository: Asterisk<br>
<br>
<br>
Description<br>
-------<br>
<br>
The attached change fixes/tweaks a few things:<br>
<br>
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.<br>
<br>
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.<br>
<br>
<br>
Diffs<br>
-----<br>
<br>
/branches/12/res/res_pjsip_transport_websocket.c 403236<br>
/branches/12/res/res_pjsip/security_events.c 403236<br>
/branches/12/res/res_pjsip/pjsip_options.c 403236<br>
/branches/12/res/res_pjsip/location.c 403236<br>
/branches/12/res/res_pjsip.c 403236<br>
/branches/12/include/asterisk/res_pjsip.h 403236<br>
<br>
Diff: <a href="https://reviewboard.asterisk.org/r/3036/diff/" target="_blank">https://reviewboard.asterisk.org/r/3036/diff/</a><br>
<br>
<br>
Testing<br>
-------<br>
<br>
Connected using JsSIP, confirmed no crash and that traffic is sent out the proper connection.<br>
<br>
<br>
Thanks,<br>
<br>
Joshua Colp<br>
<br>
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Message: 2<br>
Date: Sun, 1 Dec 2013 20:49:51 -0500<br>
From: Stas Kobzar <<a href="mailto:stas.kobzar@modulis.ca">stas.kobzar@modulis.ca</a>><br>
To: <a href="mailto:asterisk-dev@lists.digium.com">asterisk-dev@lists.digium.com</a><br>
Subject: [asterisk-dev] SIP request<br>
Message-ID:<br>
<CAMYcjooWTZY6Or9KFaRbg1uT91x0JN_Aoxqz7H=<a href="mailto:BHPJH0L5skQ@mail.gmail.com">BHPJH0L5skQ@mail.gmail.com</a>><br>
Content-Type: text/plain; charset="iso-8859-1"<br>
<br>
Hello list,<br>
<br>
I am trying to develop my own Asterisk module.<br>
I need to create and send PUBLISH SIP message with special headers and/or<br>
message body.<br>
<br>
I found in that in include folder there is a sip_api.h (Asterisk 11), an<br>
API for INFO method. But I can not figure out how to access to other<br>
methods.<br>
<br>
Is it possible to use chan_sip methods in other modules? If yes, could you,<br>
please, give me a hint where to look?<br>
<br>
Thank you,<br>
--<br>
Stas Kobzar<br>
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Message: 3<br>
Date: Sun, 01 Dec 2013 22:02:49 -0400<br>
From: Joshua Colp <<a href="mailto:jcolp@digium.com">jcolp@digium.com</a>><br>
To: Asterisk Developers Mailing List <<a href="mailto:asterisk-dev@lists.digium.com">asterisk-dev@lists.digium.com</a>><br>
Subject: Re: [asterisk-dev] SIP request<br>
Message-ID: <<a href="mailto:529BEA49.6060003@digium.com">529BEA49.6060003@digium.com</a>><br>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed<br>
<br>
Stas Kobzar wrote:<br>
> Hello list,<br>
><br>
> I am trying to develop my own Asterisk module.<br>
> I need to create and send PUBLISH SIP message with special headers<br>
> and/or message body.<br>
><br>
> I found in that in include folder there is a sip_api.h (Asterisk 11), an<br>
> API for INFO method. But I can not figure out how to access to other<br>
> methods.<br>
><br>
> Is it possible to use chan_sip methods in other modules? If yes, could<br>
> you, please, give me a hint where to look?<br>
<br>
There is no way to do this. It doesn't provide any APIs to extend it.<br>
Any additional functionality has to be built into chan_sip itself.<br>
<br>
In Asterisk 12 the new PJSIP based modules DO provide various APIs to<br>
allow you to do exactly this.<br>
<br>
Cheers,<br>
<br>
--<br>
Joshua Colp<br>
Digium, Inc. | Senior Software Developer<br>
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA<br>
Check us out at: <a href="http://www.digium.com" target="_blank">www.digium.com</a> & <a href="http://www.asterisk.org" target="_blank">www.asterisk.org</a><br>
<br>
<br>
<br>
------------------------------<br>
<br>
Message: 4<br>
Date: Sun, 1 Dec 2013 21:10:30 -0500<br>
From: Stas Kobzar <<a href="mailto:stas.kobzar@modulis.ca">stas.kobzar@modulis.ca</a>><br>
To: Asterisk Developers Mailing List <<a href="mailto:asterisk-dev@lists.digium.com">asterisk-dev@lists.digium.com</a>><br>
Subject: Re: [asterisk-dev] SIP request<br>
Message-ID:<br>
<<a href="mailto:CAMYcjoocjBJpydqDV5KvLh7CgyxzWAtGFVy50LvVsNRDU1XOjA@mail.gmail.com">CAMYcjoocjBJpydqDV5KvLh7CgyxzWAtGFVy50LvVsNRDU1XOjA@mail.gmail.com</a>><br>
Content-Type: text/plain; charset="iso-8859-1"<br>
<br>
Thank you!<br>
<br>
<br>
On Sun, Dec 1, 2013 at 9:02 PM, Joshua Colp <<a href="mailto:jcolp@digium.com">jcolp@digium.com</a>> wrote:<br>
<br>
> Stas Kobzar wrote:<br>
><br>
>> Hello list,<br>
>><br>
>> I am trying to develop my own Asterisk module.<br>
>> I need to create and send PUBLISH SIP message with special headers<br>
>> and/or message body.<br>
>><br>
>> I found in that in include folder there is a sip_api.h (Asterisk 11), an<br>
>> API for INFO method. But I can not figure out how to access to other<br>
>> methods.<br>
>><br>
>> Is it possible to use chan_sip methods in other modules? If yes, could<br>
>> you, please, give me a hint where to look?<br>
>><br>
><br>
> There is no way to do this. It doesn't provide any APIs to extend it. Any<br>
> additional functionality has to be built into chan_sip itself.<br>
><br>
> In Asterisk 12 the new PJSIP based modules DO provide various APIs to<br>
> allow you to do exactly this.<br>
><br>
> Cheers,<br>
><br>
> --<br>
> Joshua Colp<br>
> Digium, Inc. | Senior Software Developer<br>
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA<br>
> Check us out at: <a href="http://www.digium.com" target="_blank">www.digium.com</a> & <a href="http://www.asterisk.org" target="_blank">www.asterisk.org</a><br>
><br>
> --<br>
> _____________________________________________________________________<br>
> -- Bandwidth and Colocation Provided by <a href="http://www.api-digital.com" target="_blank">http://www.api-digital.com</a> --<br>
><br>
> asterisk-dev mailing list<br>
> To UNSUBSCRIBE or update options visit:<br>
> <a href="http://lists.digium.com/mailman/listinfo/asterisk-dev" target="_blank">http://lists.digium.com/mailman/listinfo/asterisk-dev</a><br>
><br>
<br>
<br>
<br>
--<br>
Stas Kobzar<br>
<br>
VoIP Developer<br>
514 284 2020<br>
<a href="http://www.modulis.ca" target="_blank">www.modulis.ca</a><br>
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Message: 5<br>
Date: Mon, 2 Dec 2013 12:29:04 +0100<br>
From: nitesh bansal <<a href="mailto:nitesh.bansal@gmail.com">nitesh.bansal@gmail.com</a>><br>
To: Asterisk Developers Mailing List <<a href="mailto:asterisk-dev@lists.digium.com">asterisk-dev@lists.digium.com</a>><br>
Subject: [asterisk-dev] WebRTC over SRTP-DTLS<br>
Message-ID:<br>
<CAOLsin5qn4MwCEY72e+v6dCAZO99yLTthZz=<a href="mailto:QxOL0_vO0ptoNQ@mail.gmail.com">QxOL0_vO0ptoNQ@mail.gmail.com</a>><br>
Content-Type: text/plain; charset="iso-8859-1"<br>
<br>
Hello everybody,<br>
<br>
I want to setup a basic Demo of WebRTC using Asterisk as WebServer and<br>
SRTP-DTLS.<br>
I got the demo setup using SRTP-DES with chrome, chrome is porpoising both<br>
DTLS and DES,<br>
Asterisk responds with DES abd call is connected.<br>
But i want asterisk to propose DTLS also in its response, can you please<br>
tell me if asterisk supports DTLS and if yes, is there a wiki page with the<br>
documentation?<br>
I could not find any relevant wikipage.<br>
<br>
Regards,<br>
Nitesh<br>
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Message: 6<br>
Date: Mon, 2 Dec 2013 14:32:16 +0100<br>
From: nitesh bansal <<a href="mailto:nitesh.bansal@gmail.com">nitesh.bansal@gmail.com</a>><br>
To: Asterisk Developers Mailing List <<a href="mailto:asterisk-dev@lists.digium.com">asterisk-dev@lists.digium.com</a>><br>
Subject: Re: [asterisk-dev] WebRTC over SRTP-DTLS<br>
Message-ID:<br>
<CAOLsin6HUGJo3E98Vf73SmY+TJEtiGXdnDF+B7=U8V=3_Wt1=<a href="mailto:Q@mail.gmail.com">Q@mail.gmail.com</a>><br>
Content-Type: text/plain; charset="iso-8859-1"<br>
<br>
Sorry, i forgot to mention Asterisk version, i am using Asterisk 11.4<br>
<br>
Regards,<br>
Nitesh<br>
<br>
<br>
<br>
On Mon, Dec 2, 2013 at 12:29 PM, nitesh bansal <<a href="mailto:nitesh.bansal@gmail.com">nitesh.bansal@gmail.com</a>>wrote:<br>
<br>
> Hello everybody,<br>
><br>
> I want to setup a basic Demo of WebRTC using Asterisk as WebServer and<br>
> SRTP-DTLS.<br>
> I got the demo setup using SRTP-DES with chrome, chrome is porpoising both<br>
> DTLS and DES,<br>
> Asterisk responds with DES abd call is connected.<br>
> But i want asterisk to propose DTLS also in its response, can you please<br>
> tell me if asterisk supports DTLS and if yes, is there a wiki page with the<br>
> documentation?<br>
> I could not find any relevant wikipage.<br>
><br>
> Regards,<br>
> Nitesh<br>
><br>
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</blockquote></div><br><br clear="all"><div><br></div>-- <br><div>Thanks & Regards</div>
<div>R.S.Parihar</div>
<div>+919650049450</div><div><a href="mailto:rajendraparihar781@gmail.com" target="_blank">rajendraparihar781@gmail.com</a></div>
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