similar to: SIP Softphone on Thinclient?

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!

Did anybody tried MozIAX extension? It is Mozilla IAX2 soft-phone. http://moziax.mozdev.org/ I tried it yesterday on eee pc, connected to asterisk on local LAN and the performance is terrible! The delay is about 2sec or 3sec. and very bad echo. I think it is the implementation of their IAX2 in their add on, as I have tried external mic. and the same delay problem. As a comparison I've tried

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, Is anyone using mozPhone? If so any feedback you can provide? Thanks, Glenn

i have a lot of questions, maybe you've got some answers. I've got the following setup : 1 central server ( linux ) , 17 servers ( LTSP ) on different locations, where each server has a number of thinclients connected to it ( max 6 ) The central server / 17 LTSP servers are on the internet and share 1 VPN ( freeswan ). It would be nice if the 17 LTSP servers could use asterix on one

Hello all, Please send a Cc: to me in addition to the list. I am not subscribed and this will make it easier for me to follow the thread and reply. Thanks. My scenary is as follows: I have a Linux server with OpenSSH 3.7.1p2 installed and this server acting as an LTSP. Some PCs (all Pentium+ classes) acting as X-Terminals connects to this server via PXE boot --> DHCP --> booting

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

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 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

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

-----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, - --

On 22/11/18 10:09 pm, Philipp Marek wrote: > if it happens that your local terminal emulation is not available > on the remote machine(s), what would be the right place to fix it? Is it a trick question?? Isn't the remote machine the only place that you can fix ?? Setting TERM on the local machine won't magically make a Wyse 60 understand VT220 control codes. Why not wrap ssh

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

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:

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, someone know a good webphone, possibily a free one Thx -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060627/0e83bc29/attachment.htm

-----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

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

-----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)

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