similar to: Outgoing spool file ignored

Does anybody know if there is an app that will cause similar to occur on users PC? I have a scenario where users will have snom phones on their desks. Ideally when their phone receives a call I need to popup a web browser with a specific url. Any ideas appreciated. Neil on 5/7/05 10:52 PM, Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> wrote:

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?

Thanks for reply, but this is irrelevant, I'm looking for an *Iridium* gateway. Regards, -- Jean-Denis Girard SysNux Syst?mes Linux en Polyn?sie fran?aise https://www.sysnux.pf/ T?l: +689 40.50.10.40 / GSM: +689 87.797.527 Le 03/04/2018 ? 16:05, albert zhang a ?crit?: > http://www.dinstar.cn/en/index.php/GSM/ > > 2018-04-04 10:01 GMT+08:00 Jean-Denis

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]

I am considering using iax.cc (sixtel) and wondering if anyone had opinions, good or bad. Are there outages with any regularity? How responsive are tech support? How is packet loss? I am particularly interested in termination to the UK, but will accept any comments people have. Thanks -- Trixter http://www.0xdecafbad.com Bret McDanel UK +44 870 340 4605 Germany +49 801 777 555 3402 US

-----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, 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:

does anyone have knopsterisk for download, I assume that because its GPL the creator of that iso cant restrict spreading it. A friend wanted it to play on a box and the only thing I can find with google is the knopsterisk.com site which wants $10 to get a copy and does not provide (as far as I can tell) any free distribution access which is his/hers/its/them/they/whatever right (being politically

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 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)

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 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, - --

Asterisk handling My Skype Calls This is for me, once more, Asterisk as the Future of Telephony. Today I've integrated my Skype Account as SIP extension in my * Box. This has been possible using "Uplink Skype to SIP Adapter", available for free at http://www.nch.com.au/skypetosip/index.html . Main features that any one can easily integrate into Asterisk: - Route skype incoming

Any means of killing a .call file that is in progress? James Taylor MetroTel 3505 Summerhill Road Suite 11 Texarkana, Tx 75503 903-793-1956

-----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

2015-05-21 17:59 GMT+02:00 Jean-Denis Girard <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 CCSS in any way. > > Is CCSS support planned for PJSIP? chan_sip is in

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 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, 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.