similar to: Video through IAX2 trunks

I am trying to send SMS from my grandstream GXV3240 Asterisk receives the message in a NOTIFY block. How can I get asterisk to run dialplan code when receiving these Notify SMS Message Blocks. I can then route them to my SMS provider. Any ideas are appreciated. Below is debug of a message sent from the phone when received no dialplan code is triggered. I am wounding if I need to

hi, do you someone test this http://www.grandstream.com/y-gxv3000.htm? video works? (it's have H264 video codec) i want this topology gxv-3000 - nat -{Internet}- Asterisk -{Internet}- nat - gxv-3000 --------------------------------------- Marek Cervenka LCNA - http://lcna.slu.cz =======================================

https://bugs.freedesktop.org/show_bug.cgi?id=47846 Bug #: 47846 Summary: Nouveau -> overscan using HDMI Classification: Unclassified Product: xorg Version: git Platform: x86 (IA32) OS/Version: Linux (All) Status: NEW Severity: normal Priority: medium Component: Driver/nouveau

Hi, I am new to the list, so I hope my questions aren't too stupid. I am using Asterisk 1.4.21.2 and already set it up to use realtime, so a CDR for an incoming SIP call is written in my mysql database. This works fine. The problem is that I don't want to have my phone ringing all the time. I just need a CDR of everyone how is calling me and to read out the CDR from my PHP script. I

On 29/04/2019 22.01, Michal Privoznik wrote: > On 4/29/19 1:06 PM, lameventanas@gmail.com wrote: >> I want to run libvirtd as a special user, and allowing users that belong >> to a special group to connect via qemu+unix:///system (eg: unix socket). >> >> I did everything necessary to do so: created a libvirt user and group, >> added the libvirt user to the kvm

I am trying to set up a Grandstream GXV-3000 Video phone to Asterisk ver 1.2.21.1. The problem I'm having is that it can call other SIP phones, but not vice versa. Can someone tell me where is the problem? TIA! Here's part of my configurations: ---------- sip.conf ---------- ; 113 is the Grandstream phone [113] type=friend username=113 secret=secret context=default dtmfmode = rfc2833

Has anyone managed to get these two phones to make a video call to each other ? If so, care to share how the hell you managed ? the GXV is at the latest firmware, and xlite the latest download Asterisk 1.4 trunk TIA Julian

Dear, I'm having problems with the configuration of this gateway(GrandStream GXW 4108), I used the instructions from GrandStream but it doesn't work. Someone has a good configuration for this gateway? Thanks in advance, Nelson -------------- next part -------------- An HTML attachment was scrubbed... URL:

I want to run libvirtd as a special user, and allowing users that belong to a special group to connect via qemu+unix:///system (eg: unix socket). I did everything necessary to do so: created a libvirt user and group, added the libvirt user to the kvm group, added my normal user to the libvirt group, and made sure the socket is owned by libvirt:libvirt with permissions set to 770. libvirtd starts

https://bugs.freedesktop.org/show_bug.cgi?id=51579 Bug #: 51579 Summary: Xv video support is black on NVidia NVa8 Chipset in latest git Classification: Unclassified Product: xorg Version: git Platform: x86-64 (AMD64) OS/Version: All Status: NEW Severity: critical Priority:

Hi, I use a simple scheme: SIP video phone A (h264/Asterisk 1.8.11) <---IAX2 trunk----> SIP video phone B (h264/Asterisk 11.7.0) When calls from A to B and vice versa drop on pickup. On B side: [Oct 24 16:33:49] DEBUG[15590][C-00000012] res_rtp_asterisk.c: Setting the marker bit due to a source update [Oct 24 16:33:49] DEBUG[15590][C-00000012] res_rtp_asterisk.c: Setting the marker bit

Hi, i am trying to install the Asterisk test suite on my ubuntu system , i have followed all the installlation steps as mentioned in the link ( http://blogs.asterisk.org/2010/04/29/installing-the-asterisk-test-suite/) , but when i am trying to run the script some of the test cases are PASSED and most of them are FAILED and SKIPPPED. So please help me out to do the testing correctly. The

Hello all. We have an asterisk working perfectly but we need a sollution for the PA system. Before Asterisk PBX we had an expensive analog PBX which plugged an extension into an audio amplifier, and that was the PA system. Now, the Asterisk server is quite far from the audio amplifier and it has no audio card. So my idea is to plug the audio card of another linux server, which is over the

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

https://bugs.freedesktop.org/show_bug.cgi?id=82118 Priority: medium Bug ID: 82118 Assignee: nouveau at lists.freedesktop.org Summary: nouveau tearing with xvideo QA Contact: xorg-team at lists.x.org Severity: normal Classification: Unclassified OS: Linux (All) Reporter: lameventanas at gmail.com

Hello, Could you help me to summarize what is SIP Early Media useful for ? I was thinking of: - Passing error messages to caller, - Custom ringing tones to caller. Did I miss something ? Best regards -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20160203/33dec62b/attachment.html>

Hi, One of my clients, wants to use * box to run weekly meetings between remote locations over the internet. What would be the best configuration for this? We are talking about two conference rooms. I am referring to the actual hardware/software and bandwidth requirements for this to work well. I have run two software video phones and I had marginal results with it when displayed on large LCDs,

Has anyone created any docker images I might be able to use on EC2 for load testing an asterisk platform? I started an instance this morning and was about to load sipp and other tools, and then thought surely someone must have done this already. I'd like to hammer a platform we have created with multiple EC2 images until it breaks, to test capacity. Cheers, j

Hi! This weekend we had a problem with our Asterisk Box which ran flawlessly for nearly 4 weeks. The Asterisk server sits between the PSTN and a Siemens PBX and bridges 2 BRI lines. No calls, not incoming, not outgoing. The admin rebooted the Dell Box and then everything worked fine again. Now, I'm analyzing log files to find the cause. During the Asterisk outage the logfiles only show

Hi there, I've been grappling with this for some time, so I'm finally breaking down and trying this list. I'm trying to integrate libvorbisfile and libao to create a simple sound file player, a la ogg123. To do this, I borrowed heavily from ao_example.c and vorbisfile_example.c. I feed the buffer from ov_read into ao_play in a simple loop. It usually works, but occasionally all it