[asterisk-dev] Add new option to Queue function
Nguyen Hoang Son
nhson at vasc.com.vn
Sun Apr 27 21:08:14 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 9:02 PM
To: asterisk-dev at lists.digium.com
Subject: asterisk-dev Digest, Vol 117, Issue 172
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. [Code Review] 3479: chan_pjsip: Call pickup test. (Joshua Colp)
2. [Code Review] 3478: chan_pjsip: Add call pickup support.
(Joshua Colp)
3. Add new option to Queue function (Nguyen Hoang Son)
4. Re: encoding issues in Asterisk 11.9.0 Now Available
(Matthew Jordan)
5. Re: [Code Review] 3478: chan_pjsip: Add call pickup support.
(Matt Jordan)
----------------------------------------------------------------------
Message: 1
Date: Fri, 25 Apr 2014 13:05:57 -0000
From: "Joshua Colp" <reviewboard at asterisk.org>
To: "Joshua Colp" <reviewboard at asterisk.org>, "Joshua Colp"
<jcolp at digium.com>, "Asterisk Developers"
<asterisk-dev at lists.digium.com>
Subject: [asterisk-dev] [Code Review] 3479: chan_pjsip: Call pickup
test.
Message-ID: <20140425130557.5072.51019 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/
-----------------------------------------------------------
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/20140425/60e43d1
1/attachment-0001.html>
------------------------------
Message: 2
Date: Fri, 25 Apr 2014 13:06:00 -0000
From: "Joshua Colp" <reviewboard at asterisk.org>
To: "Joshua Colp" <reviewboard at asterisk.org>, "Joshua Colp"
<jcolp at digium.com>, "Asterisk Developers"
<asterisk-dev at lists.digium.com>
Subject: [asterisk-dev] [Code Review] 3478: chan_pjsip: Add call
pickup support.
Message-ID: <20140425130600.5072.92187 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/3478/
-----------------------------------------------------------
Review request for Asterisk Developers.
Repository: Asterisk
Description
-------
While configuration exists to place PJSIP channels into pickup and call
groups the functionality to actually perform a call pickup does not exist.
This change adds it.
Diffs
-----
/branches/12/res/res_pjsip_session.c 413007
/branches/12/channels/chan_pjsip.c 413007
Diff: https://reviewboard.asterisk.org/r/3478/diff/
Testing
-------
Ran test and confirmed failed on normal 12. Applied change. Re-ran test and
confirmed fixed.
Thanks,
Joshua Colp
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
<http://lists.digium.com/pipermail/asterisk-dev/attachments/20140425/e774ec9
3/attachment-0001.html>
------------------------------
Message: 3
Date: Fri, 25 Apr 2014 20:59:41 +0700
From: "Nguyen Hoang Son" <nhson at vasc.com.vn>
To: <asterisk-dev at lists.digium.com>
Subject: [asterisk-dev] Add new option to Queue function
Message-ID: <016c01cf608e$9ac55580$d0500080$@com.vn>
Content-Type: text/plain; charset="utf-8"
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
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
<http://lists.digium.com/pipermail/asterisk-dev/attachments/20140425/6d6999f
1/attachment-0001.html>
------------------------------
Message: 4
Date: Fri, 25 Apr 2014 09:06:33 -0500
From: Matthew Jordan <mjordan at digium.com>
To: Asterisk Developers Mailing List <asterisk-dev at lists.digium.com>
Subject: Re: [asterisk-dev] encoding issues in Asterisk 11.9.0 Now
Available
Message-ID:
<CAN2PU+6DF3PF2cqjnSjqhNjLr7kRdsdL5Yp=6Eon2FPhekwVMA at mail.gmail.com>
Content-Type: text/plain; charset=ISO-8859-1
On Fri, Apr 25, 2014 at 4:32 AM, Walter Doekes
<walter+asterisk-dev at osso.nl> wrote:
> On 23/04/14 18:52, Asterisk Development Team wrote:
>>
>> --===============4365525224653466459==
>> Content-Type: text/plain; charset="us-ascii"
>> Content-Transfer-Encoding: 8bit
>
> ...
>>
>> * ASTERISK-23028 - [patch] Asterisk man pages contains unquoted
>> minus signs (Reported by Jeremy Lain??)
>
> ...
>>
>> * ASTERISK-19499 - ConfBridge MOH is not working for transferee
>> after attended transfer (Reported by Timo Ter??s)
>
> ...
>
> Could you update the `charset` param to "utf-8" the next time?
>
> Thanks!
>
Sure - sorry about that!
--
Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com & http://asterisk.org
------------------------------
Message: 5
Date: Fri, 25 Apr 2014 14:14:15 -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] 3478: chan_pjsip: Add call
pickup support.
Message-ID: <20140425141415.10829.15378 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/3478/#review11741
-----------------------------------------------------------
Ship it!
/branches/12/res/res_pjsip_session.c
<https://reviewboard.asterisk.org/r/3478/#comment21534>
Blob.
- Matt Jordan
On April 25, 2014, 8:06 a.m., Joshua Colp wrote:
>
> -----------------------------------------------------------
> This is an automatically generated e-mail. To reply, visit:
> https://reviewboard.asterisk.org/r/3478/
> -----------------------------------------------------------
>
> (Updated April 25, 2014, 8:06 a.m.)
>
>
> Review request for Asterisk Developers.
>
>
> Repository: Asterisk
>
>
> Description
> -------
>
> While configuration exists to place PJSIP channels into pickup and call
groups the functionality to actually perform a call pickup does not exist.
This change adds it.
>
>
> Diffs
> -----
>
> /branches/12/res/res_pjsip_session.c 413007
> /branches/12/channels/chan_pjsip.c 413007
>
> Diff: https://reviewboard.asterisk.org/r/3478/diff/
>
>
> Testing
> -------
>
> Ran test and confirmed failed on normal 12. Applied change. Re-ran test
and confirmed fixed.
>
>
> Thanks,
>
> Joshua Colp
>
>
-------------- next part --------------
An HTML attachment was scrubbed...
URL:
<http://lists.digium.com/pipermail/asterisk-dev/attachments/20140425/709ecd8
a/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 172
**********************************************
More information about the asterisk-dev
mailing list