similar to: Asterisk 10 EOL Approaching

Hello everyone! On December 15th, 2013, Asterisk 10 officially reached its End of Life [1]. As a Standard Release, Asterisk 10 received one year of bug fix support, followed by one year of security fix support. Users of Asterisk 10 should consider moving to Asterisk 11 at their earliest possible convenience. Asterisk 11 is a Long Term Support (LTS) Release, and will continue to receive bug fix

Hello everyone! On December 15th, 2013, Asterisk 10 officially reached its End of Life [1]. As a Standard Release, Asterisk 10 received one year of bug fix support, followed by one year of security fix support. Users of Asterisk 10 should consider moving to Asterisk 11 at their earliest possible convenience. Asterisk 11 is a Long Term Support (LTS) Release, and will continue to receive bug fix

Hey everyone! This is a friendly reminder that Asterisk 12 will be entering security fix only mode soon. As a Standard release of Asterisk, Asterisk 12 received one year of maintenance fixes, and will receive one year of security fixes. Asterisk 12 was first released on 2013-12-20 - the one year anniversary of which is just around the corner! After 2014-12-20, additional releases of Asterisk 12

recreate Echo, if that is possible. trying to recode all dialplan to stasis application > On 22 May 2015, at 19:29, Scott Griepentrog <sgriepentrog at digium.com> wrote: > > Nick- > > Are you wanting to recreate the dialplan Echo() application in stasis? > > Why not just send the call to Echo() instead of Stasis()? > > On Fri, May 22, 2015 at 11:25 AM, Matthew

I'm pretty sure there isn't a way to do that currently. ?My best guess would be that a new special type of bridge technology could be created that would implement the per-channel echo (no audio bridged between channels in the bridge). That would require new C code in Asterisk for the bridge, and then the usual methods of moving channels in to bridges with ARI could be used.? On Sat, May

Nick- Are you wanting to recreate the dialplan Echo() application in stasis? Why not just send the call to Echo() instead of Stasis()? On Fri, May 22, 2015 at 11:25 AM, Matthew Jordan <mjordan at digium.com> wrote: > On Fri, May 22, 2015 at 4:41 AM, Nick Awesome <jleed at me.com> wrote: > > Can anyone tell me how can I create echo test using ARI stasis > application?

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

2013/9/19 Matthew Jordan <mjordan at digium.com> > > On Thu, Sep 19, 2013 at 9:02 AM, Olivier <oza_4h07 at yahoo.fr> wrote: > >> Hi, >> >> Asterisk 11 doc says CDR(src) value is read-only (see >> https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Function_CDR). >> >> For various reasons, I would appreciate to change its value so that it

2014-08-13 19:06 GMT+02:00 Matthew Jordan <mjordan at digium.com>: > On Wed, Aug 13, 2014 at 12:01 PM, Olivier <oza.4h07 at gmail.com> wrote: >> After installing various packages, here is what I did: >> >> TDIR=/usr/src >> cd $TDIR >> PJOPTIONS="--prefix=/usr --enable-shared --disable-sound >> --disable-resample --disable-video

Well, it breaks audio for all NAT endpoints, how can I fix this? > On 18 Mar 2015, at 15:48, Matthew Jordan <mjordan at digium.com> wrote: > > On Wed, Mar 18, 2015 at 7:41 AM, Nick Awesome <jleed at me.com <mailto:jleed at me.com>> wrote: >> Hey guys, >> >> have issues with reinvite, no matter what endpoint is calling asterisk >> always tries

On Thu, Feb 12, 2015 at 10:38 AM, David M. Lee <dlee at digium.com> wrote: > On Feb 12, 2015, at 9:11 AM, Joshua Colp <jcolp at digium.com> wrote: > > Justin Sherrill wrote: > > I would love to run Asterisk on a BSD system. I do not know of any > developers actively working on Asterisk on a BSD platform, though my > knowledge isn't comprehensive. > >

NAT endpoint calling local endpount - switching to native_rtp then no audio, both of them have direct_media=no, Verbose log: -- Executing [99 at dialmap:1] AGI("PJSIP/304-00000022", "/pbx/agi.php") in new stack -- Launched AGI Script /pbx/agi.php -- AGI Script Executing Application: (Dial) Options: (PJSIP/99/sip:99 at 192.168.1.73:5060,20) -- Called

It means, AMI application is no more running or crashed or lost network connection with asterisk server. In such cases call is neither answered nor disconnected by Asterisk. I want to detect such state and jump to next dial plan to answer or reject the calls Regards Amit Patkar On September 20, 2016 8:07:23 PM GMT+05:30, Matthew Jordan <mjordan at digium.com> wrote: >On Sat, Sep 17,

WTF is a jitterbuffer? Sent from my Verizon Wireless 4G LTE smartphone -------- Original message -------- From: Matthew Jordan <mjordan at digium.com> Date: 01/29/2015 10:41 AM (GMT-05:00) To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users at lists.digium.com> Subject: Re: [asterisk-users] JITTERBUFFER function On Thu, Jan 29, 2015 at 4:56 AM,

On Tue, Mar 10, 2015 at 5:22 PM, Toufic Khreish (Gmail) <toufic.khreish at gmail.com> wrote: > Thank you, I needed a starting point to start my post. > > 1. Asterisk 12.8.1 (IAX2 voice issues) no video issues. > Voice issues on IAX2 Trunks, All extensions are SIP. > The IAX2 trunks on Asterisk 12.8.1 produces only one error out of : iax2 > set debug trunk on >

I'm willing to bet you are forcing the use of G729. 7940 and 7960 phones can only do a single G729 channel, and if you require G729 for the second leg of a conference, it will fail. On Tue, Jan 20, 2015 at 10:03 AM, Jordan Cook - Gyron Networks < jordan.cook at gyron.net> wrote: > Possibly slightly off topic, has anyone ever had Cisco 79xx Series > phones come up with ?cannot

Next step is packet capture to see if there is a clue as to the cause of the failure in the SIP signalling. On Tue, Jan 20, 2015 at 10:41 AM, Jordan Cook - Gyron Networks < jordan.cook at gyron.net> wrote: > We were using G722 - I thought similarly and tried a call with alaw. Same > problem occurred, any other ideas? > > > I'm willing to bet you are forcing the use of

On Tue, Mar 10, 2015 at 5:00 AM, Dmitry Melekhov <dm at belkam.com> wrote: > 05.03.2015 11:42, Dmitry Melekhov ?????: > >> 05.03.2015 11:29, Dmitry Melekhov ?????: >>> >>> 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: >>>

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello Everyone, On April 30th, FreeBSD 7.0 will reach its End of Life and will no longer be supported by the FreeBSD Security Team. Users of FreeBSD 7.0 are strongly encouraged to upgrade to FreeBSD 7.1 before that date. Note that the End of Life date for FreeBSD 7.0 was originally announced as being February 28, but was delayed by two months in

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello Everyone, On April 30th, FreeBSD 7.0 will reach its End of Life and will no longer be supported by the FreeBSD Security Team. Users of FreeBSD 7.0 are strongly encouraged to upgrade to FreeBSD 7.1 before that date. Note that the End of Life date for FreeBSD 7.0 was originally announced as being February 28, but was delayed by two months in