similar to: MozPhone

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!

Hi Guys! After 1 week or looking for answers about CTI and Asterisk, I havent been able to find the necessary applications to do what I want to do. Maybe you guys have more insight on this. I tried installing asttapi, and works great! Can make outbound calls from outlook, etc.. Nice work! For incoming calls.. Ive been trying software like ascendis callerid, call alert and identapop pro, but

Anyone aware of any SIP softphones that might virtualize well with Citrix presentation server? I suspect I know the answer already as I have been researching softphones that work with Cisco CallManager that can be virtualized if you will with Citrix and have come to learn that its not something that seems to be doable at this time. I have to assume that the issues affecting the virtualization of

Hi everybody Im the IT Manager for a new call center and my bosses has assing to me a very dificult task i have to configure the call center using Hp 5520 thin clients, asterisk and some kind of dialer that allows outbound calls. I triyed using terminal services but it dind worked because the lack on the sound and the microphone do not work on the thin clients using terminal services, we tried

Does anyone know a softphone that can accept URLs during a call and open that page in the default browser when the call is answered? I Know DIAX and the IDEFISK, only pro version.I need another ones. It can be using the cmd SetURL Regards. -- Bruno de Assump??o Loureiro msn: loureiro_bruno@hotmail.com

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, - --

A Polycom 550 and a IAX client (Mozphone and ZoIPer were used) are having a conversation. Call quality is reported as good except for an echo with a 3 second delay. Most of my searches are saying echo happens only on the PSTN piece, but there isn't one here. Can someone point me in the right direction? Asterisk 1.4.21.2 Under 40 users Quad-Core AMD Opteron(tm) Processor 2352 and 4G RAM

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:

-----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

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 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

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

-----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

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?

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