similar to: 97 issues marked 'Ready for Testing'

List, Over the last few months we have managed to bring the total number of issue on the tracker from 610+ to 537 (as of writing). While this is good news, we still have a number of open issues that require testers to help move them along. Below, I have posted the oldest 50 issues that are in the 'Ready for Testing' state. Basically, we are looking for more people to step-up and test

Hi, I submitted a bug report, 0004193 and did a forum post: http://centos.org/modules/newbb/viewtopic.php?topic_id=24748&forum=42 regarding a bug / problem I keep coming up against with CentOS 5.x's kernel. I've done a load of searching on the subject but can't see any references on any CentOS list or mantis, so I'm trying to follow up and see if anyone can work on it,

ok it works i had a problem with a syntax: i had to wrire: exten =>_!X.,n(external),Dial(SIP/011212664800450 at pstn2,,S(20)) thanks ________________________________ De : Adil Zaaraoui <adilzeaaraoui at yahoo.fr> ? : Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users at lists.digium.com> Envoy? le : Jeu 8 juillet 2010, 19h 41min 15s Objet : Re :

First I've tryed with the version 1.4.36 But it didn't worked so I supposed it should be ok with the last version 1.6.2... but not => I will create a new issue for this if you think it should be. Just hope it will not be too long to have a correction. Thanks a lot. Sebastien On Fri, Sep 24, 2010 at 9:11 AM, IMS <ims77.dev at gmail.com> wrote: > No ideas ? > Just give me

On Wed, Jan 28, 2015 at 12:23 PM, Ishfaq Malik <ish at pack-net.co.uk> wrote: > Hi > > We're using 1.8.23.1 on CentOS 5 and are trying to get accurate stats for > queues. > > For a particular customer, when I run queue show <queue_name> I get the > following numbers: > > <queue_name> has 0 calls (max unlimited) in 'ringall' strategy (17s

On Tue, Jan 27, 2015 at 4:14 PM, symack <symack at gmail.com> wrote: > Hello Everyone, > > I am required to write a java program that will get our asterisk to: > > * Query the database for phone numbers > * Loop through numbers and dial > * Play message > * Get dial pressed response > - If 1 = Yes > - If 2 = No > - If 3 = Connect to Agent

On Wed, Mar 4, 2015 at 12:47 AM, Jarrod Cuzens <jarrod at mogl.com> wrote: > For those that were interested I have attached the kamailio.cfg which we > have working with Kamailio 4.2.1 and Asterisk 1.8.23/32. Specifically, the > following yum packages: > > kamailio.x86_64 4.2.1-4.1 > @home_kamailio_v4.2.x-rpms > kamailio-auth-ephemeral.x86_64

On Thu, Mar 5, 2015 at 2:29 AM, Dmitry Melekhov <dm at belkam.com> wrote: > Hello! > > Just installed asterisk 13.2.0 and see many such messages in log, I see them > in console during calls, really something like this: > > > -- Executing [6166 at kanbaikal:2] Dial("OOH323/kanbaikal-6", > "SIP/6166 at asterisk") in new stack > == Using SIP

Hi I am new to mailing list ,please correct me if the way of posting is not correct Relpying to : Re: make asterisk do something when an outgoing call is picked up (lee) For making asterisk do something on outgoing call Dial application is itself used Like for Playing an announcement to the caller on pick up the is an option A(x) where x is the file to play to the called party. Also

On 8/27/10, asterisk-users-request at lists.digium.com <asterisk-users-request at lists.digium.com> wrote: > Send asterisk-users mailing list submissions to > asterisk-users at lists.digium.com > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.digium.com/mailman/listinfo/asterisk-users > or, via email, send a message with subject or body

Thank you Paul. I have downloaded the code. How out-of-call messaging can be configured in the Dialplan? Regards, Rajib Deka SIEMENS Ltd. Robert V Chandran Tower, First Floor, West Wing, #149, Velechery Tambaram Main Road, Pallikaranai, Chennai-100, INDIA. www.siemens.com Mob: +91-9176780669 | E-Mail: rajib.deka at siemens.com Date: Thu, 07 Apr 2011 10:14:37 -0400 From: Paul Belanger

I need to change the sip port from 5060 to 5061 actually we already used 5060 for proxy to sip any idea to change 5060 to 5061 so all can acces the sip using this port please help........................ On 4/8/11, asterisk-users-request at lists.digium.com <asterisk-users-request at lists.digium.com> wrote: > Send asterisk-users mailing list submissions to > asterisk-users at

Since it is Saturday evening (7PM EST) I am asking this on the list in case someone who knows sees it and can answer. Astricon is in my back yard for the first time, and I could hit you with a rock. I would always like to attend, and spoke at the 2007 Astricon in Phoenix but don't have the idle cycles. Question: Can I just go to Astricon and take the dCAP exam only? In and out? Cost? I

Good morning, There is a new patch up on reviewboard[1] right now for the removal of app_rpt and chan_usbradio from Asterisk trunk. As it stands right now these two modules do not appear to be maintained in this repository and have out-of-date code. Russellb's patch will see these to modules removed from asterisk trunk (asterisk 11). If a large part of the community wishes to help

Hello, I have a big problem which I’m just not able to solve. I created the following mean value from the following dataset structure: Id |value 1 | 2 1 | 3 1 | 4 2 | 2 2 | 1 3 | 5 4 | 3 etc.|etc. with the command: mean_rating <- tapply(ratok$value, ratok$project_id , mean,simplify = FALSE) this gives me a ragged array: > mean_rating [1] $`14`

please send me the ways to change asterisk port from 5060 to 5061 i need to configure it because we are already using 5060 port in router then we cant use it again we have to configure other sip server so please suggest me a way.......................... On 4/10/11, asterisk-users-request at lists.digium.com <asterisk-users-request at lists.digium.com> wrote: > Send asterisk-users mailing

Dear all, First off, I am very new to asterisk so forgive me if any of my comments or questions seem trivial. Thanks to [this post](http://blog.polybeacon.com/2010/10/17/asterisk-1-8-and-google-voice/) and [this post](http://www.davidvossel.com/?p=28), I have GV set up on asterisk through jabber.conf and gtalk.conf. I can successfully dial out from asterisk. I'm trying to set up an

I''m on the downhill side of a large project that requires an additional integer field to be added to the tag.rb in acts_as_taggable. I feel I have a good understanding of ActiveRecord and have performed the correct migrations. (As a short background for those following this thread) when one wants to add tags to a model they call the ''tag_with'' method that jumps into

Hi, We''re trying to rewrite veeeery old Rails app that we successfully managed to move to 1.2.6 and now we''re trying with 2.1.0. One of the problems is caused by :include in AR.find. Rails 1.2.6 generates totally different SQL query than 2.1.0. Here''s Rails find query and generated SQL queries: Foo.find(:all, :include => :ticket, :conditions =>

I''m having a problem with belongs_to and has_and_belongs_to_many. Here''s a brief summary of the models involved: class Member < ActiveRecord::Base set_primary_key ''member_id'' has_and_belongs_to_many :projects, :join_table => ''projects__members'' has_many :projects, :foreign_key => ''created_by'' end class