[asterisk-users] PJSIP Real-time Text (T.140)

Hi, is the support of real-time text limited to the SIP channel driver only? Somehow Asterisk is not offering T.140 to the called party when initiating a call and including real-time text. In my pjsip.conf I allowed T.140 and enabled text support. Regards, Simon Hohberg

[asterisk-users] Empty user string on pjsip inbound trunk

Hi, I try to setup an inbound trunk using pjsip_wizard.conf. Now, when I receive a call from that trunk with an empty user string, and I try to match it with 's' in the dial plan, Asterisk reports that the extension was not found in the context. * pjsip_wizard.conf: [example] type = wizard

Re: [asterisk-users] PJSIP - Video Support for WebRTC

On 07/26/2016 03:15 PM, Olivier wrote: Matthew Jordan digium.com> writes: On Mon, Mar 23, 2015 at 8:55 AM, Gosmac gmail.com> wrote: Hey i have an interesting topic to discuss here. The main goal here is to be able to make a video call between two WebRTC endpoints registered on asterisk 1

Re: [asterisk-users] PJSIP Multipart Body

On 06/27/2016 12:09 PM, Joshua Colp wrote: Simon Hohberg wrote: Hi, I want to pass a part of a SIP INVITE multipart body. I found a quite old patch here: https://issues.asterisk.org/jira/browse/ASTERISK-14510?jql=text%20~%20%22body%20part%22 But this patch is for the SIP channel driver not

[asterisk-users] PJSIP Multipart Body

Hi, I want to pass a part of a SIP INVITE multipart body. I found a quite old patch here: https://issues.asterisk.org/jira/browse/ASTERISK-14510?jql=text%20~%20%22body%20part%22 But this patch is for the SIP channel driver not PJSIP, right? Is it even possible without a patch? What do I have

Re: [asterisk-users] Asterisk 13 and WebRTC. Is wiki page still valid ?

Is it implied here that both HTTPS and WSS must also come from the same server (Same Origin Policy) ? No, the same origin policy does not apply to web sockets. Then, can I also install my own WebRTC demo page on my own private Asterisk server and access this demo page through HTTPS ? If I'm

Re: [asterisk-users] Asterisk 13 and WebRTC. Is wiki page still valid ?

Hi Oliver, On 02/18/2016 12:10 PM, Olivier wrote: Hello, I'm trying to have my first calls with WebRTC. My server has asterisk 13.7.0. I'm following the instructions from the wiki [1]. So I'm using [2] live demo from a Chrome navigator (v48) on Debian Jessie station. Whenever I type somethi

[asterisk-users] Delayed start of video with WebRTC - Missed FIR due to DTLS?

Hi, I am using Asterisk 13.7.0 with PJSIP. I set up Asterisk for use with WebRTC SIP clients. After I managed to get video working, I noticed, that the calling party receives no video till 90s (or so) have passed. After 90s both parties receive video perfectly. I am suspecting that this is