similar to: [OT] IP Phone with Braille console for blind reseptionist

-----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 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

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, I made some tests with asterisk-13.2.0 and chan_pjsip this weekend myself, and came to the same conclusion: some peerstatus events are missing (eg. when contacts become unreachable / unavailable, IIRC), and I could not find a way to get contacts status through AMI. It looks a bit similar to issues 23172, 23173: PJSip missing functionalities.

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Le 20/05/2015 00:50, Joshua Colp a ?crit : > It looks like this is an incoming leg, in which case that information > isn't available. There is no association of an AOR and Contact on > incoming legs (it MAY be possible to deduce but it certainly wouldn't > work in all cases). Since you specify one explicitly on outgoing, that

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, I'm trying to use CHANNEL(aor) and CHANNEL(contact) on PJSIP channel, on asterisk-13.3.2, but they don't return anything. Is this a bug, or did I miss something? Here is my test dialplan: exten => *98,1,Answer same => n,NoOp(Channel=<${CHANNEL(name)}>,type= <${CHANNEL(channeltype)}>) same =>

-----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 Thanks for your reply Larry. Le 27/07/2015 01:22, Larry Moore a ?crit : > I think the "488 Not acceptable here" is occurring because the channel > connecting through is not T.38 capable, that will be the IAX channel > from iaxmomdem. This is what T38gateway is supposed to do. And I'm very happy to report that after one more

-----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)

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

-----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

-----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.

Hi list, Last Friday, an Asterisk server became unresponsive after ~8,5 months of smooth operation (~320000 calls). Server did reply to pings, but no ssh, no more console login. Also Asterisk no longer took calls, but ISDNguard watchdog was still alive. Looking at the logs after reboot, I could not find anything significant, except in a file created by the following command via a cron job:

-----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

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:

Hi list, I'm using asterisk to send alerts via phone or fax, by using the spool functionnality (writing files in /var/spool/asterisk/outgoing). The system works, but sometimes a file in the outgoing directory is simply ignored by asterisk (nothing on the cli); the spool file is correct. It happenned only twice since the system is installed (about two months), but since it is an alert

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi list, I'm trying to use CEL to display channel information in real time. It works fine for simple calls, blind transfers, or SIP attended transfers (initiated from the SIP phone). My problem is for Asterisk attended transfers (atxfer as configured in features.conf). The scenario is: . phone 107 calls phone 100, . 100 dials the atxfer code,

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