[asterisk-dev] asterisk-dev Digest, Vol 138, Issue 3 - Asterisk - 13.6.0 - Problem in sharing data between EXE and SO file:

Wed Jan 6 22:49:46 CST 2016

Hi All,

Do anyone having idea on this case?. Waiting for your valuable responses to
take it forward.

The scenario is explained in previous mail content.

Regards,
Boobalan

-----Original Message-----
From: asterisk-dev-bounces at lists.digium.com
[mailto:asterisk-dev-bounces at lists.digium.com] On Behalf Of
asterisk-dev-request at lists.digium.com
Sent: Wednesday, January 06, 2016 11:30 PM
To: asterisk-dev at lists.digium.com
Subject: asterisk-dev Digest, Vol 138, Issue 3

Send asterisk-dev mailing list submissions to
	asterisk-dev at lists.digium.com

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
	asterisk-dev-request at lists.digium.com

You can reach the person managing the list at
	asterisk-dev-owner at lists.digium.com

When replying, please edit your Subject line so it is more specific
than "Re: Contents of asterisk-dev digest..."

Today's Topics:

   1. Re: Equivalent to svnview for asterisk? (Joshua Colp)
   2. Re: Equivalent to svnview for asterisk? (Joshua Colp)
   3. Re: Equivalent to svnview for asterisk? (Tony Mountifield)
   4. Re: RTP/SAVP & TLS (Ross Beer)
   5. AppKonference 2.7 (Paul Albrecht)
   6. Re: How to use a DAHDI kernel driver in linux using The
      Bridging Framework Tecnology in the Asterisk 13 (Richard Mudgett)

----------------------------------------------------------------------

Message: 1
Date: Wed, 06 Jan 2016 08:49:44 -0400
From: Joshua Colp <jcolp at digium.com>
To: Asterisk Developers Mailing List <asterisk-dev at lists.digium.com>
Subject: Re: [asterisk-dev] Equivalent to svnview for asterisk?
Message-ID: <568D0D68.1070404 at digium.com>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed

Tony Mountifield wrote:
> It's a while since I've looked at the Asterisk code repositories,
> and I see that svnview is no more, because Asterisk has moved to gerrit.
>
> Is there any equivalent to the old svnview, where I can browse the
> source code for different versions without having to download or clone
> the complete packages?

We also mirror the repositories onto Github[1] so you can use all of the 
browsing, statistics, etc features available there (or even clone from 
them). That's what I use when I want to browse in a browser or link 
people to things.

Cheers,

[1] https://github.com/asterisk

-- 
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org

------------------------------

Message: 2
Date: Wed, 06 Jan 2016 08:52:27 -0400
From: Joshua Colp <jcolp at digium.com>
To: Asterisk Developers Mailing List <asterisk-dev at lists.digium.com>
Subject: Re: [asterisk-dev] Equivalent to svnview for asterisk?
Message-ID: <568D0E0B.1010908 at digium.com>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed

Joshua Colp wrote:
> Tony Mountifield wrote:
>> It's a while since I've looked at the Asterisk code repositories,
>> and I see that svnview is no more, because Asterisk has moved to gerrit.
>>
>> Is there any equivalent to the old svnview, where I can browse the
>> source code for different versions without having to download or clone
>> the complete packages?
>
> We also mirror the repositories onto Github[1] so you can use all of the
> browsing, statistics, etc features available there (or even clone from
> them). That's what I use when I want to browse in a browser or link
> people to things.

And to respond to myself if you want to stick to asterisk.org things and 
like Atlassian products there's also a FishEye instance running[1].

[1] https://code.asterisk.org/

-- 
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org

------------------------------

Message: 3
Date: Wed, 6 Jan 2016 13:02:34 +0000 (UTC)
From: tony at softins.co.uk (Tony Mountifield)
To: asterisk-dev at lists.digium.com
Subject: Re: [asterisk-dev] Equivalent to svnview for asterisk?
Message-ID: <assp.081383216c.n6j39a$f1e$1 at softins.softins.co.uk>

In article <568D0E0B.1010908 at digium.com>, Joshua Colp <jcolp at digium.com>
wrote:
> Joshua Colp wrote:
> > Tony Mountifield wrote:
> >> It's a while since I've looked at the Asterisk code repositories,
> >> and I see that svnview is no more, because Asterisk has moved to
gerrit.
> >>
> >> Is there any equivalent to the old svnview, where I can browse the
> >> source code for different versions without having to download or clone
> >> the complete packages?
> >
> > We also mirror the repositories onto Github[1] so you can use all of the
> > browsing, statistics, etc features available there (or even clone from
> > them). That's what I use when I want to browse in a browser or link
> > people to things.
> 
> And to respond to myself if you want to stick to asterisk.org things and 
> like Atlassian products there's also a FishEye instance running[1].
> 
> [1] https://code.asterisk.org/

Thanks for the pointers!

Tony
-- 
Tony Mountifield
Work: tony at softins.co.uk - http://www.softins.co.uk
Play: tony at mountifield.org - http://tony.mountifield.org

------------------------------

Message: 4
Date: Wed, 6 Jan 2016 14:07:53 +0000
From: Ross Beer <ross.beer at outlook.com>
To: Asterisk Developers Mailing List <asterisk-dev at lists.digium.com>
Subject: Re: [asterisk-dev] RTP/SAVP & TLS
Message-ID: <SNT151-W790682BB06B71265FC0BD4FFF40 at phx.gbl>
Content-Type: text/plain; charset="iso-8859-1"

> Date: Wed, 6 Jan 2016 08:22:34 -0400
> From: jcolp at digium.com
> To: asterisk-dev at lists.digium.com
> Subject: Re: [asterisk-dev] RTP/SAVP & TLS
> 
> Ross Beer wrote:
> > Hi Dev,
> >
> > In Asterisk 1.8 Snom phones accept calls when RTP/SAVP is set to
> > 'mandatory' which means that the RTP/SAVP options appear in the SDP 'm'
> > lines. However in Asterisk 13 chan_pjsip, no such lines exist when using
> > 'SDES' encryption.
> 
> The "media_encryption=sdes" option turns on SRTP support and thus makes 
> the media RTP/SAVP. You can also turn on optimistic SRTP support as well 
> using "media_encryption_optimistic=yes" which will use RTP/AVP but 
> include a crypto line. I just checked the testsuite tests for SDP 
> offer/answer and they are passing, I also manually enabled it and 
> confirmed it is RTP/SAVP. You may have a configuration error. Snom devices
work correctly when 'media_encryption_optimistic=no', when this is set to
yes the RTP/SAVP is replaced: Set to No = "m=audio 41988 RTP/SAVP 8 0 3 101"
Set to Yes = "m=audio 36240 RTP/AVP 8 0 3 101" I have updated my
configuration to not use the optimistic setting.
> 
> >
> > Therefore Snom phones require this option to be set to 'off'. Should
> > Asterisk 13 be offering RTP/SAVP in the same way as chan_sip did?
> >
> > With regards to TLS, devices reject calls if a 'transport=transport-tls'
> > is specified. Is this also a bug as it appears that Asterisk doesn't
> > re-use an active connection in this situation?
> 
> This is a bug in PJSIP which has an issue on our side[1]. If an explicit 
> transport is specified PJSIP will not reuse a connection.
> 
> [1] https://issues.asterisk.org/jira/browse/ASTERISK-22658
>  Great, I can work around this until a fix is in place. Thank you for your
assistance.
> -- 
> Joshua Colp
> Digium, Inc. | Senior Software Developer
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - US
> 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

-------------- next part --------------
An HTML attachment was scrubbed...
URL:
<http://lists.digium.com/pipermail/asterisk-dev/attachments/20160106/eb3ec3c
c/attachment-0001.html>

------------------------------

Message: 5
Date: Wed, 6 Jan 2016 11:11:40 -0600
From: Paul Albrecht <paul at glccom.com>
To: Asterisk Developers Mailing List <asterisk-dev at lists.digium.com>
Subject: [asterisk-dev] AppKonference 2.7
Message-ID: <9BA7BE66-72A8-4B12-83D0-D9A8AABF6743 at glccom.com>
Content-Type: text/plain; charset="us-ascii"

I have released an updated AppKonference. You can download the latest code
from source forge:
sourceforge.net/projects/appkonference<http://sourceforge.net/projects/appko
nference>
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
<http://lists.digium.com/pipermail/asterisk-dev/attachments/20160106/2d4c895
2/attachment-0001.html>

------------------------------

Message: 6
Date: Wed, 6 Jan 2016 11:47:22 -0600
From: Richard Mudgett <rmudgett at digium.com>
To: Asterisk Developers Mailing List <asterisk-dev at lists.digium.com>
Subject: Re: [asterisk-dev] How to use a DAHDI kernel driver in linux
	using The Bridging Framework Tecnology in the Asterisk 13
Message-ID:
	<CALD46g30R9jSYX3CxQ7kDRH57_waRhqUkUs5wXmhORsaPyNuQw at mail.gmail.com>
Content-Type: text/plain; charset="utf-8"

On Wed, Jan 6, 2016 at 6:27 AM, Di?genes Vila Nova Pereira <
dvnp at cesar.org.br> wrote:

> Hi Folks,
>
> I'm newbie in Asterisk developement tecnology. I had read and seen
> documentation that the Asterisk supports new bridging framework tecnology
> that has a pluggable interface, allowing a native bridging to be written
in
> a separate module and selected based on criteria it presents to the core.
>
> I have a scenary that following this way: I have a PABX where there's a
> digital matrix controled by DSP card that controls the TDM networks
> channels commutation between cards FX0, FXS, E1 and Media Gateway for RTP
> audio.
>
> How to configure the Asterisk and what level implement/modify/customize a
> DAHDI kernel module that does possible to use Asterisk by a native bridge
> to control and permit two audio channels commutes direct by DSP without
the
> interference and just so to monitor this until hangup complete of calling
> by the Asterisk.
>

I think you are mixing up the various software layers involved.  Since you
are talking about
implementing a DAHDI kernel module you need to follow the rules within
DAHDI to implement
your native bridge.  There is already an Asterisk level native DAHDI bridge
technology
implemented that uses the DAHDI API to setup the native bridge.

1) Asterisk bridging framework where channels can freely be moved between
bridges.
2) Asterisk bridging technology (holding, simple, softmix, native_rtp,
native_DAHDI) - The technology determines how media frames are exchanged
between channels.
3) DAHDI itself (Directly interfaces with hardware.)

Richard
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
<http://lists.digium.com/pipermail/asterisk-dev/attachments/20160106/1443f4b
4/attachment-0001.html>

------------------------------

_______________________________________________
--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

End of asterisk-dev Digest, Vol 138, Issue 3
********************************************