[asterisk-dev] asterisk-dev Digest, Vol 117, Issue 173

Fri Apr 25 20:44:14 CDT 2014

Hi White, 
It is no problem. This is a small function which is customized by myself for
internal calls only in my company. It is not commercial activity. So , it is
not something violation. 

Best regards,
NHSON 

-----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: Friday, April 25, 2014 9:08 PM
To: asterisk-dev at lists.digium.com
Subject: asterisk-dev Digest, Vol 117, Issue 173

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: Add new option to Queue function (jonathan white)
   2. Re: [Code Review] 3479: chan_pjsip: Call pickup test.
      (Matt Jordan)

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

Message: 1
Date: Fri, 25 Apr 2014 15:15:20 +0100
From: jonathan white <jw at uvacity.com>
To: Asterisk Developers Mailing List <asterisk-dev at lists.digium.com>
Subject: Re: [asterisk-dev] Add new option to Queue function
Message-ID:
	<CAC5vyGNX_ckz+yyUd_gHcTHR2FYWbe9KL1TBFeJBx9mZZXpJkw at mail.gmail.com>
Content-Type: text/plain; charset="utf-8"

Just something I know which may restrict what can be done. Avaya have many
patents for call distribution. This includes call distribution to agents
who have spent the least amount of time on the phone and taken the lowest
number of calls.
On 25 Apr 2014 15:00, "Nguyen Hoang Son" <nhson at vasc.com.vn> wrote:

>  Hi all,
> I'm using Queue function of Asterisk to arrange calls which is coming to
> my agents. I want to customize the way asterisk arrange coming call, in
> other word, is it possible to create a new option instead of using the
> existing: RINGALL, ROUNDROBIN,... . For example: The incoming call should
> come to the argent who has the most waiting time (idle time). I find out
> that the algorithm of each option of Queue is defined in "app_queue.c" in
> the source code but I don't know how to change, how to add the waiting
time
> as a new option to sort by.
>
> This question is quite related to the development of asterisk, so please
> help if you have any idea or experience on that. Thank you very much.
>
> ---------------------------
>
> *NGUY?N HO?NG S?N*
>
> M-Commerce Center
>
> VASC Software and Media Company - VNPT
>
> Addr: No.97 Nguyen Chi Thanh Street, Dong Da District, Hanoi, Vietnam
>
> Cell phone: +84 912998101
>
> Skype: hoangsonk49
>
> E-mail: nhson at vasc.com.vn
>
>
>
>
>
>
>
> --
> _____________________________________________________________________
> -- 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/20140425/c210fe8
d/attachment-0001.html>

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

Message: 2
Date: Fri, 25 Apr 2014 14:19:55 -0000
From: "Matt Jordan" <reviewboard at asterisk.org>
To: "Joshua Colp" <jcolp at digium.com>, "Asterisk Developers"
	<asterisk-dev at lists.digium.com>, "Matt Jordan"
	<reviewboard at asterisk.org>
Subject: Re: [asterisk-dev] [Code Review] 3479: chan_pjsip: Call
	pickup test.
Message-ID: <20140425141955.11866.65097 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/3479/#review11742
-----------------------------------------------------------

/asterisk/trunk/tests/channels/pjsip/call_pickup/run-test
<https://reviewboard.asterisk.org/r/3479/#comment21536>

    2014

/asterisk/trunk/tests/channels/pjsip/call_pickup/run-test
<https://reviewboard.asterisk.org/r/3479/#comment21535>

    Are you sure you're Jonathan Rose?

/asterisk/trunk/tests/channels/pjsip/call_pickup/run-test
<https://reviewboard.asterisk.org/r/3479/#comment21543>

    These are always used as regular expressions. Why not just compile them
here and use them as such everywhere else?

/asterisk/trunk/tests/channels/pjsip/call_pickup/run-test
<https://reviewboard.asterisk.org/r/3479/#comment21537>

    Since this is using PJSIP, there's no need to support previous versions
of Asterisk. Just the bridging model for 12 is sufficient.

/asterisk/trunk/tests/channels/pjsip/call_pickup/run-test
<https://reviewboard.asterisk.org/r/3479/#comment21540>

    And just use 12 here as well

/asterisk/trunk/tests/channels/pjsip/call_pickup/run-test
<https://reviewboard.asterisk.org/r/3479/#comment21541>

    No spaces between parameters and their values:

    channel="Local/test_out at pickuptest"

/asterisk/trunk/tests/channels/pjsip/call_pickup/run-test
<https://reviewboard.asterisk.org/r/3479/#comment21538>

    Just use the Asterisk 12 logic

/asterisk/trunk/tests/channels/pjsip/call_pickup/run-test
<https://reviewboard.asterisk.org/r/3479/#comment21539>

    PEP8 Guidelines: no spaces between equals in parameters passed to a
function.

    You may want to pass this through pylint to catch anything else as well.

/asterisk/trunk/tests/channels/pjsip/call_pickup/run-test
<https://reviewboard.asterisk.org/r/3479/#comment21542>

    Why is the Local channel shouting at me? :-)

- Matt Jordan

On April 25, 2014, 8:05 a.m., Joshua Colp wrote:
> 
> -----------------------------------------------------------
> This is an automatically generated e-mail. To reply, visit:
> https://reviewboard.asterisk.org/r/3479/
> -----------------------------------------------------------
> 
> (Updated April 25, 2014, 8:05 a.m.)
> 
> 
> Review request for Asterisk Developers.
> 
> 
> Repository: testsuite
> 
> 
> Description
> -------
> 
> This is a modified version of the normal call pickup test which uses
chan_pjsip instead of chan_sip to test call pickup functionality.
> 
> 
> Diffs
> -----
> 
>   /asterisk/trunk/tests/channels/pjsip/call_pickup/test-config.yaml
PRE-CREATION 
>   /asterisk/trunk/tests/channels/pjsip/call_pickup/run-test PRE-CREATION 
>   /asterisk/trunk/tests/channels/pjsip/call_pickup/configs/ast2/pjsip.conf
PRE-CREATION 
>
/asterisk/trunk/tests/channels/pjsip/call_pickup/configs/ast2/extensions.con
f PRE-CREATION 
>   /asterisk/trunk/tests/channels/pjsip/call_pickup/configs/ast1/pjsip.conf
PRE-CREATION 
>
/asterisk/trunk/tests/channels/pjsip/call_pickup/configs/ast1/features.conf
PRE-CREATION 
>
/asterisk/trunk/tests/channels/pjsip/call_pickup/configs/ast1/extensions.con
f PRE-CREATION 
> 
> Diff: https://reviewboard.asterisk.org/r/3479/diff/
> 
> 
> Testing
> -------
> 
> I tested the test by running the test.
> 
> 
> Thanks,
> 
> Joshua Colp
> 
>

-------------- next part --------------
An HTML attachment was scrubbed...
URL:
<http://lists.digium.com/pipermail/asterisk-dev/attachments/20140425/8cd3aa4
0/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 117, Issue 173
**********************************************