similar to: 50 mantis issues marked 'Ready for Testing'

List, Its been 2 weeks since my previous email and this time I am linking all 97 issues marked 'Ready for Testing' [1]. Simply follow the link, view the available patches, download, compile and install. Report your result into the actual issue, we can them continue to triage the issue. The more testers the better. If you have any problems or questions, jump on #asterisk-testing on

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 June 5, 2011, we migrated from Mantis to Jira as the issue tracker for Asterisk [1]. We temporarily left Mantis running in read-only mode to smooth the transition. At 15 months, temporary has turned into semi-permanent. As a part of other infrastructure changes we are making to the community services, we will finally shut down Mantis for good. We will update our DNS servers on the morning of

On June 5, 2011, we migrated from Mantis to Jira as the issue tracker for Asterisk [1]. We temporarily left Mantis running in read-only mode to smooth the transition. At 15 months, temporary has turned into semi-permanent. As a part of other infrastructure changes we are making to the community services, we will finally shut down Mantis for good. We will update our DNS servers on the morning of

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 all, I've installed mantis.noarch and mantis-config-httpd.noarch packages. The problem is I always get the same error in Apache:. You don't have permission to access /mantis on this server. It's curious 'cause: * I work in localhost mode and I've running drupal site and phpMyAdmin without problems. So, Apache works fine. * I've created mantis ddbb according the docs

I am suddenly getting a bunch of OLD (as in 3-9 months old) e-mails from mantis saying things like a note has been added to an issue etc., and yet the issue has not been touched in months and the "new note" it is referring to is also months old. Consequently, I never received these e-mails before either. The e-mail itself shows that "carolina.digium.com" received

Greetings, A few weeks ago I posted a message about the upcoming migration from Mantis to JIRA for issues.asterisk.org [1]. A lot of testing has been done and all known issues have been resolved. We have scheduled the migration for Sunday, June 5th. The issue tracker will be down most of the day as the migration takes place. Once the migration is complete, the issue tracker will be:

Greetings, A few weeks ago I posted a message about the upcoming migration from Mantis to JIRA for issues.asterisk.org [1]. A lot of testing has been done and all known issues have been resolved. We have scheduled the migration for Sunday, June 5th. The issue tracker will be down most of the day as the migration takes place. Once the migration is complete, the issue tracker will be:

Hello listers, I went to pull some CDR's from my PBX, and noticed they were a bit light. I also noticed output on the console about CDR's not being posted. I am currently running 1.4.13, and in looking at the change log, this was a change in behavior as part of mantis 10659. Personally, I think the old behavior was more correct, but obviously at least one person disagrees. I think

Recently, I've been having issues with the URIs returned from e.164.org and toll free calls. It seems that the URIs that are returned from ENUMQUERY and ENUMRESULT are no longer the proper numbering schemes that the poviders use. I've been using the following [enum] template in my outbound route for quite some time with great success until recently. [enum](!) exten =>

Greetings, Does someone have the link to reset your password on bugs.digium.com? I can't seem to find one. Thanks. -- Andrew Thompson http://aktzero.com/ http://dev.asteriskdocs.org/

Hi, My configuration is SipPhone<----->asterisk1 <----->asterisk2. My asterisk version is 1.2.10. I installed chan_h323 according to 'http://astrecipes.net/?n=102'. When i call from asterisk1 to asterisk2, there is no audio. Using 'rtp debug', I can see that rtp packets are being received. Regards, Jason. #------h323.conf for both------------------------ [general]

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

Putting a whole bunch of people into a listen-only/muted Confbridge conference or getting the broadcaster audio into a MOH class and then just having callers attach to that MOH class? Does the the muted side of a Confbridge Room still try to mix in audio from the muted channels or does it just disregard those channels and only run mixes against unmuted channels? Now, if the answer is