similar to: Asterisk 13 and WebRTC

Hello, I'm trying to use presence with PJSIP and I have a "issue". I created correctly hint priorities like: exten => 1000,hint,PJSIP/1000 exten => 1001,hint,PJSIP/1001 Extension 1000 can subscribe extension 1001 y vice-versa. The problem is when the extension 1000 make or receive a call. In the softphone where the extension is present on buddy list, the extension appear

Dear All, I use sipml5 to register two users from browser and the two clients are successfully connected. But when I made a call from one of the users, the other user doen'st have call notification and for a while the calling process ended. I check the /var/log/asterisk/messages got the following log: [Dec 17 14:54:11] WARNING[11471][C-00000000] chan_sip.c: Received SAVPF profle in audio

Hello, with qualify_frequency=0 I can't receive calls from others endpoints. Other strange think is if I set mailboxes parameter on the console, when the endpoint registering, i can see: ERROR[2208]: res_pjsip.c:2946 create_out_of_dialog_request: Unable to create outbound NOTIFY request to endpoint 1001 at sip.domain.com WARNING[2208]: res_pjsip_mwi.c:379

Yes, I have tried that too (i.e, exten => +17775551212,1,Log(WARNING, TWILIO)). It does not work and NO error message in CLI. I have also tried http://orourketech.com/elastix-plus-sign-caller-id-messing-things/ since I first emailed this group, but that does not seem to work either. Here is my log: [Dec 2 15:09:28] NOTICE[26464]: res_pjsip_session.c:1920 new_invite: Call from

Humm whats is the diferent? Em 16/12/2015 14:19, "Annus Fictus" <annusfictus at gmail.com> escreveu: > CDR-STATS is for reporting. > > A2Billing is for billing... > > Regards > > El 16/12/2015 a las 11:15, Vitor Mazuco escribi?: > >> Hi everyone! >> >> I'm trying to install CDR-Stats (cdr-stats.org), but it very difficult. >>

Hi, I spent the past week experimenting with webrtc + asterisk 11.9.0-rc1 + opus/vb8 codec patch. This is interesting technology and I try to find out how to connect all the moving parts. Firefox: Neither sipml5 or jssip works with calls to asterisk, audio/video doesn't matter. WARNING[977][C-00000005] chan_sip.c: Rejecting secure audio stream without encryption details: audio 35684

Hello, I'd like share with you my tests about PJSIP channel with the aim of improving the functioning of the channel: * Multi domain support not work correctly: https://issues.asterisk.org/jira/browse/ASTERISK-26026 * Different context subscribe for each endpoint not possible: https://issues.asterisk.org/jira/browse/ASTERISK-25471 * BLF don't work correctly on my tests

Hello, I'd like to do some tests with the StatsD dialplan application but on the last version of Asterisk 13 (13.7.0) I can't find this application. New Features made in this release: ----------------------------------- * ASTERISK-25419 - Dialplan Application for Integration of StatsD (Reported by Ashley Sanders) res_statsd module are correctly compiled y loaded. Any hint?

Hello, I think Device State for Agents don't work correctly My configuration: agents.conf [general] [agent](!) autologoff=15 ackcall=no acceptdtmf=# wrapuptime=5000 musiconhold=default recordagentcalls=no custom_beep=beep [2000](agent) fullname=Fulano [2001](agent) fullname=Zutano [2002](agent) fullname=Mengano queue.conf (Agents Related) member => Agent/2000 member =>

Hi, Yes, we're implementing the dialplan in realtime too. Here the contents of sorcery.conf: [res_pjsip] endpoint=realtime,ps_endpoints aor=realtime,ps_aors contact=realtime,ps_contacts [res_pjsip_endpoint_identifier_ip] identify=realtime,ps_endpoint_id_ips Cheers, Francisco. From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf

Hello List, following this thread: http://asterisk-users.digium.narkive.com/ulR5hd1M/same-pjsip-username-with-differents-domains I tried to configure on the pjsip.conf the same endpoint with different domains like: [1000 at sip.domain.com] type=endpoint [1000 at sip1.domain.com] type=endpoint I can register the two 1000 endpoints using different domain but on the Asterisk console:

Hi, Am 16.02.2017 um 14:19 schrieb Annus Fictus: > And Microsip using PJSIP SIP stack :) Sorry (also, for off-topic), based on my latest experience with PJSIP, I'm not sure if this really is a sign of good quality. Maybe it's a problem with the implementation in Asterisk (I haven't tried PJSIP in other software), but after just five minutes of testing I found several bugs

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 13 it is a feature that definitely asterisk 13 should support . the problems that i faced with this is the following and i hope i could get an advise here. asterisk 13 vanilla version has some issues marking the video packets this complain

Hi All, I am trying to configure webRTC phone example for SIPml5 and i found this info from https://wiki.asterisk.org/wiki/display/AST/Asterisk+WebRTC+Support . I have asterisk 11.9.0 installed and downloaded source of SIPml5 from http://code.google.com/p/sipml5/source/checkout I copied sample code into web root directory and example loaded successfully and also able to register 2 extensions. I

Hello, I am running Asterisk 13.6.0 in an AWS instance, and I set it up with Twilio SIP trunk using pjsip_wizard.conf (nice feature!). I see that the calls actually "reach" the PBX, but for some reason, they are not caught by any of my extensions context. Here's what I observe when I test this from a non-PBX connected E164 number (a landline), say 555-666-1212. My Twilio number is

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 something like ws://123.123.123.123:8088/ws in Expert Mode form (see [1]), I'm getting this error : *2:SecurityError: Failed to construct

Hi everyone! I'm trying to install CDR-Stats (cdr-stats.org), but it very difficult. Is there others optins for billing? Thanks

Hi, I am getting *Can't provide secure audio requested in SDP offer* with sipml5 client hosted on my local system [1060] ; This will be WebRTC client type=friend username=1060 ; The Auth user for SIP.js host=dynamic ; Allows any host to register secret=sameer ; The SIP Password for SIP.js encryption=yes ; Tell Asterisk to use encryption for this peer avpf=yes ; Tell Asterisk to use AVPF

2016-02-18 15:42 GMT+01:00 Marek ?ervenka <cervajs at fpf.slu.cz>: > my experience with pjsip for webrtc > http://lists.digium.com/pipermail/asterisk-users/2015-September/287562.html > > > Yes I saw this post earlier today. Having to fight 14 days scared me a bit ! Did you set sipml5 on your own server or did you use Live demo (

Hello, Can anyone recommend a particular online WebRTC phone for testing with Asterisk? We tried: - JsSIP, but even with the "enable video" checkbox disabled it sends video options in the INVITE SDP and Asterisk rejects it with "Rejecting secure video stream without encryption details". - sipML5, but it won't register, perhaps something to do with not using the Asterisk