similar to: Recommended in Asterisk Wiki E-Mail

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

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

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?

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

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

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,

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. > >

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,

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:

On Thu, Oct 20, 2016 at 4:50 AM, marek cervenka <cervajs2 at gmail.com> wrote: > i tested this > > # cat /etc/asterisk/extconfig.conf > [settings] > queue_log => sqlite3,cdrDb > > # cat /etc/asterisk/res_config_sqlite3.conf > [cdrDb] > dbfile = /var/lib/asterisk/realtime.sqlite3 > > sqlite3 /var/lib/asterisk/realtime.sqlite3 > > CREATE TABLE

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:

Some of you have already noticed we've chosen a number of Atlassian tools to provide services to the Asterisk and Asterisk SCF communities (Confluence, Crowd, Crucible, Fisheye and Bamboo). Of course, we're not alone in this since many other open source projects have chosen these tools as well, but I'd just like to state again how happy we are that Atlassian is willing to license these

If you take a look at the safe_asterisk shell script, usually located at /usr/sbin/safe_asterisk (for CentOS at least), you'll be able to find where the core files are located. If it's not located there, then you'll need to look at the Asterisk init script for the scripts location. I hope this helps. Regards; John -----Original Message----- From: asterisk-users-bounces at

Hey all - After much discussion at AstriCon, it became clear that the Asterisk project could use a mailing list dedicated specifically to application development. This new mailing list should be used specifically for discussions regarding the development of applications using AMI, AGI, or ARI - or any other interface exposed by Asterisk in the future. Today, we're pleased to announce the

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

I see that my asterisk is started with the -g option, the core file I cannot find on my system (find / -name core*) -----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Matthew Jordan Sent: Tuesday, March 17, 2015 1:51 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users]