–cBz73V41rX4G2GXAJWymOOACf5HfIvaNf Content-Type: text/plain; charset=utf-8Content-Language: en-USContent-Transfer-Encoding: quoted-printable ,Im having a strange problem when using pjsip wizard and reloading(pjsip reload on CLI): some data (specifica..
Author : Jean-Denis Girard
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–Wzca4TvHXCFHSh8czzo74PZJeI6j7deHq Content-Type: text/plain; charset=utf-8Content-Language: fr-FRContent-Transfer-Encoding: quoted-printableWojciech,The IAX2 RFC is available here:https://tools.ietf.org/html/rfc5456I once developped an IAX2 softph..
, Using Asterisk 16.1.1, with PJSIP, Im 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,..
, I have a request to integrate Iridium in a Asterisk system. A quick search didnt return much: I expected to find products similar to GSM gateways, but this does not seem to exist. so Id be very interested about possible solutions. Has it be done alrea..
, Ive 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-De..
, 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 Im still having the same issues. In the trace below, Im sending a fax from Hylafax server through iaxmodem on Asterisk..
, Im trying to receive fax from PSTN, with the following setup: Fax machine — PSTN — *11 — *13 — IAXmodem + Hylafax Fax machine is connected to the PSTN, call arrives via ISDN on Asterisk 11.16.0 used as gateway, chan_sip relays the call to Aster..
, It looks like Call Completion Supplementary Services is not available for PJSIP channels, am I right? Is there another solution? Thanks, – — Jean-Denis Girard SysNuxSystèmes Linux en Polynésie française http://www.sysnux.pf/ Tél: +689 40.50.10..
, Im trying to use CHANNEL(aor) and CHANNEL(contact) on PJSIP channel, on asterisk-13.3.2, but they dont return anything. Is this a bug, or did I miss something? Here is my test dialplan: exten => *98,1,Answer same => n,NoOp(Channel=,type= ) same..
, Im currently working on a project where the customer is employing a blind person as receptionist: she has an Alcatel phone with extensions and a Braille console connected via serial port. Ive searched for something similar for IP telephony, but fo..