similar to: ARI libraries?

In article <301a2e78-d490-3805-e30f-41b668aac5c1 at sysnux.pf>, Jean-Denis Girard <jd.girard at sysnux.pf> wrote: > > Hi Tony, > > Le 20/07/2019 à 06:29, Tony Mountifield a écrit : > > Are there any other languages/libraries I should be considering? > > Same here, after years of AGI / AMI, I recently made my first project > using ARI on Asterisk-16. I love

Hi list, I have a request to integrate Iridium in a Asterisk system. A quick search didn't return much: I expected to find products similar to GSM gateways, but this does not seem to exist. so I'd be very interested about possible solutions. Has it be done already, how? Thanks, -- Jean-Denis Girard SysNux Syst?mes Linux en Polyn?sie fran?aise

On Mon, Mar 7, 2016 at 2:53 PM, Jean-Denis Girard <jd.girard at sysnux.pf> wrote: > Hi, > > Le 07/03/2016 09:28, George Joseph a ?crit : > > PLEASE TRY THIS!! I'd love some feedback BEFORE 13.8.0 is released. > > I have tried GIT-master-ee5a944M on my Fedora 23 test server, and got: > > [pjproject] Unpacking /tmp/pjproject-2.4.5.tar.bz2 > [pjproject]

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Le 21/05/2015 00:16, Joshua Colp a ?crit : > If CCSS is needed then the only option is to use chan_sip. The > chan_pjsip module does not implement CCSS in any way. Is CCSS support planned for PJSIP? chan_sip is in "extended" state in asterisk-13, so chan_pjsip should be preferred for new installations, ri ght? Thanks, - --

Hi list, I'm having a strange problem when using pjsip wizard and reloading ("pjsip reload" on CLI): some data (specifically endpoint/pickup_group) is not modified. For example, initially I have empty pickup group: tiare*CLI> pjsip show endpoint xxx ... pickup_group : ... Then, I add endpoint/pickup_group = 0,3 to pjsip_wizard.conf, and reload:

Hi list, Using Asterisk 16.1.1, with PJSIP, I'm asked to build a SIP trunk to a system that uses exclusively tel: uri on inbound and outbound calls. I could not find documentation or sample config about tel:uri. Is this doable? If not possible with PJSIP, is chan_sip a better option? Any pointer would be greatly appreciated. Thanks, -- Jean-Denis Girard SysNux Systèmes

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Le 18/02/2016 11:03, Richard Mudgett a ?crit : > I've been using Grandstream phones for more than 10 years, but onl y > yesterday tried to use Early Dial... and I failed. What is needed on the > Asterisk side to reply 484 to INVITE? Phones are talking to chan_p jsip > on Asterisk-13.7.1. > > > Look into the

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi Bryant, Thanks for your reply. It didn't work immediately, I had to create a second context, or else it was looping between the second and first line. This seems to work: [earlydial] ; Test Early Dial exten => _.,1,Set(l_Extension=${EXTEN}) exten => _.,n,Goto(earlydial2,${l_Extension},1) [earlydial2] exten => _.,n,Goto(noMatch,1)

On Fri, Jul 7, 2023 at 6:40 PM Jean-Denis Girard <jd.girard at sysnux.pf> wrote: > There seems to be a problem with the tar.gz archive on github. It's > correct on downloads.asterisk.org. Can you be more specific? They are identical and the same tarball. I just downloaded both from each place and confirmed that, and confirmed they both extract fine. -- Joshua C. Colp Asterisk

Le 07/07/2023 à 12:49, Joshua C. Colp a écrit : > On Fri, Jul 7, 2023 at 6:40 PM Jean-Denis Girard <jd.girard at sysnux.pf > <mailto:jd.girard at sysnux.pf>> wrote: > > There seems to be a problem with the tar.gz archive on github. It's > correct on downloads.asterisk.org <http://downloads.asterisk.org>. > > > Can you be more specific?

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi list, I'm getting a strange problem with a Mediatrix 3631 Gateway connected to the PSTN via an E1 PRI link configured for Euro ISDN signaling. The Mediatrix sends incoming calls from the PSTN to an Asterisk server via SIP: this works fine. But when the caller hangs up, the Mediatrix doesn't send "Bye" to Asterisk, so the call is

Hello Jean-Denis. I believe the idea is that you answer the survey for each type of scenarios you are running. So one for call centre, another one for ivr, etc... Regards, Marcelo On Mon, 11 Mar 2019, 02:10 Jean-Denis Girard, <jd.girard at sysnux.pf> wrote: > Hi Matt, > > I would have loved to participate to the survey, but I feel it does > apply to my situation: as an

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi list, I've been using Grandstream phones for more than 10 years, but only yesterday tried to use Early Dial... and I failed. What is needed on the Asterisk side to reply 484 to INVITE? Phones are talking to chan_pjsip on Asterisk-13.7.1. Thanks, - -- Jean-Denis Girard SysNux Syst?mes Linux en Polyn?sie fran?aise

Hey All, For those of you that do not know me, my name is Matthew Fredrickson and I’m the project lead for the Asterisk project. First off, I wanted to thank all of you that contribute in various ways to the project – whether it be at a developmental level, answering questions on forums and mailing lists, contributing documentation, or just generally advocating for it within your sphere of

Has anyone had any luck with MozIAX (Mozphone) on FC4 with Firefox 1.6? jslib and moziax install through Firefox correctly - at least that is the message I get. I am able to log into the IAX Phone on Windows, however I get an error stating: -------------------------------------------------- FATAL ERROR: no connection to "network_client". MozPhone will stop now!

I am trying to use dahdi complete 2.11.1 with a 4.13 kernel. - NOT working for know reasons. I tried applying two patches but still get compile errors. AHHH! How do I just use git to get the latest with the fixes ???? This command did not work - I still get the errors. git clone git://git.asterisk.org/dahdi/linux dahdi-linux Thanks, Jerry -------------- next part -------------- An HTML

Hello. Asterisk 13.2, PJSIP. Problem: I do not get any AMI events when changing the status of the contact. When using chan_sip I got "peerstatus" event. When using res_pjsip and devices (endpoint configuration) I got "peerstatus" event. When using res_pjsip.so and OUTBOUND REGISTRATION WITH OUTBOUND AUTHENTICATION i got "registry" event. When using

Ludovic Gasc wrote: > 2015-05-21 17:59 GMT+02:00 Jean-Denis Girard <jd.girard at sysnux.pf > <mailto:jd.girard at sysnux.pf>>: > > -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA1 > > Le 21/05/2015 00:16, Joshua Colp a ?crit : > > If CCSS is needed then the only option is to use chan_sip. The > > chan_pjsip module does not implement

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi list, 2 weks ago I asked questions about PJSIP and T.38 but got no replies. I upgraded Asterisk to git as of yesterday (309dd2a), and I'm still having the same issues. In the trace below, I'm sending a fax from Hylafax server through iaxmodem on Asterisk-13 (tiare) to a second Asterisk (11.18.0 t0gw) connected to the PSTN via ISDN; the

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, I'd like to know what is the current situation with regard to AOC-E, when Asterisk is inserted between the telco and an existing PBX, using E1 / EuroISDN. Can Asterisk pass the AOC-E information received from the telco to the PBX, so that billing system still works? The system would be for a hotel, so breaking billing system is not possible.