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

Justin Killen jkillen at allamericanasphalt.com
Mon Dec 2 17:24:08 CST 2013 

You're probably way better off asking this on the users's mailing list (asterisk-users at lists.digium.com<mailto:asterisk-users at lists.digium.com>) 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: asterisk-dev-bounces at lists.digium.com [mailto:asterisk-dev-bounces at lists.digium.com] On Behalf Of raj singh
Sent: Monday, December 02, 2013 3:12 PM
To: asterisk-dev at lists.digium.com
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, <asterisk-dev-request at lists.digium.com<mailto: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<mailto:reviewboard at asterisk.org>>
To: "Matt Jordan" <mjordan at digium.com<mailto:mjordan at digium.com>>, "Joshua Colp"
        <reviewboard at asterisk.org<mailto:reviewboard at asterisk.org>>, "Joshua Colp" <jcolp at digium.com<mailto:jcolp at digium.com>>,
        "Asterisk Developers" <asterisk-dev at lists.digium.com<mailto: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<mailto:stas.kobzar at modulis.ca>>
To: asterisk-dev at lists.digium.com<mailto:asterisk-dev at lists.digium.com>
Subject: [asterisk-dev] SIP request
Message-ID:
        <CAMYcjooWTZY6Or9KFaRbg1uT91x0JN_Aoxqz7H=BHPJH0L5skQ at mail.gmail.com<mailto: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<mailto:jcolp at digium.com>>
To: Asterisk Developers Mailing List <asterisk-dev at lists.digium.com<mailto:asterisk-dev at lists.digium.com>>
Subject: Re: [asterisk-dev] SIP request
Message-ID: <529BEA49.6060003 at digium.com<mailto: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<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 <stas.kobzar at modulis.ca<mailto:stas.kobzar at modulis.ca>>
To: Asterisk Developers Mailing List <asterisk-dev at lists.digium.com<mailto:asterisk-dev at lists.digium.com>>
Subject: Re: [asterisk-dev] SIP request
Message-ID:
        <CAMYcjoocjBJpydqDV5KvLh7CgyxzWAtGFVy50LvVsNRDU1XOjA at mail.gmail.com<mailto: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<mailto: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<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>
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Message: 5
Date: Mon, 2 Dec 2013 12:29:04 +0100
From: nitesh bansal <nitesh.bansal at gmail.com<mailto:nitesh.bansal at gmail.com>>
To: Asterisk Developers Mailing List <asterisk-dev at lists.digium.com<mailto:asterisk-dev at lists.digium.com>>
Subject: [asterisk-dev] WebRTC over SRTP-DTLS
Message-ID:
        <CAOLsin5qn4MwCEY72e+v6dCAZO99yLTthZz=QxOL0_vO0ptoNQ at mail.gmail.com<mailto: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<mailto:nitesh.bansal at gmail.com>>
To: Asterisk Developers Mailing List <asterisk-dev at lists.digium.com<mailto: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<mailto: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<mailto: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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--
Thanks & Regards
R.S.Parihar
+919650049450
rajendraparihar781 at gmail.com<mailto:rajendraparihar781 at gmail.com>
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