similar to: No subject

... Thanks, anyway for telling as at least, it reflects your needs. > > > You want NT PtMP and i second that, > not being limited on the asterisk > side is a must in the > telephony ecosystem, since the legacy PABX aren't alwsys easy to > reconfigure. > > _______________________________________________ > -- Bandwidth and Colocation Provided by

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

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?

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 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

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

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, Has anyone information about BRI hardware supported by 1.6 libpri ? In another thread, I was told a basic BRI card with HFC chipset (Bewan Gazel 128) was supported but I would delighted to lear about other harwarde (and specifically about Digium B410P). Regards -------------- next part -------------- An HTML attachment was scrubbed... URL:

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

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

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

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 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

-----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, - --

-----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.

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi list, I'm looking for the best / recommended solution for automatic discovery of phone numbers for a multiple Asterisk system. This would be for an administration, with many branches (~30), but a common infrastructure (DNS, LDAP). Most branches would have Asterisk servers for various reasons (location, administrative). All contacts would be in

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi list, It looks like Call Completion Supplementary Services is not available for PJSIP channels, am I right? Is there another solution? Thanks, - -- Jean-Denis Girard SysNux Syst?mes Linux en Polyn?sie fran?aise http://www.sysnux.pf/ T?l: +689 40.50.10.40 / GSM: +689 87.79.75.27 -----BEGIN PGP SIGNATURE-----

-----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 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