similar to: asterisk 11 and no RTP

Hi, For years I've been running a minimal (ish) SIP based Asterisk with the modules based on chan-sip. For various reasons unrelated to Asterisk the machine the latest incarnation of this configuration has been updated to Debian Buster and thus to Asterisk 16. Since this upgrade I have a dependency problem related to res_rtp_asterisk.so. So the old config was: [modules] autoload=no load

On 02/12/2019 10:07, Stefan G. Weichinger via samba wrote: > Am 02.12.19 um 08:47 schrieb L.P.H. van Belle via samba: >> Hai, >> >> Sorry for the late reply. > Never mind, weekend is important ... > >> Here its just apt-get distupgrade --autoremove --purge >> Without the autoremove libldb1 is not removed so libldb2 can be installed. > # apt-get distupgrade

I have a working implementation of the new xaudio2 API in my tree. Microsoft is treating it as a replacement for dsound[1], and as a result, this new API is used by lots of recent games; see Bug 26808[2] for some examples. Most games that use the xaudio2 API use a version of Microsoft's WMA codec. Wine doesn't currently have the ability to decode this audio for playback. In order for the

Here is what installs: apt-get install sernet-samba-ad Reading package lists... Done Building dependency tree Reading state information... Done The following extra packages will be installed: libavahi-client3 libavahi-common-data libavahi-common3 libcups2 libdm0 libfam0 libwbclient0 sernet-samba sernet-samba-client sernet-samba-common sernet-samba-libs sernet-samba-libsmbclient0

I was able to get h.264 support on www.youtube.com/html5 on fedora 21 by installing gstreamer1-libav but that package does not exist in C6 (or it's popular third party repos) What is the name of the package which adds h.264 to firefox on youtube/html5 in C6?

https://gist.github.com/oneman/5394332 This is a little hack for sending -live encoded- WebM streams to Icecast from STDIN. An example (I was using libav from git, params likely different between libav, ffmpeg etc programs) avconv -v debug -f video4linux2 -s 320x240 -r 10 -i /dev/video0 -vcodec libvpx -threads 2 -vb 128k -r 10 -f webm pipe:1 | krad_pipe2 europa.kradradio.com 8008

L'octidi 18 pluviôse, an CCXXV, Miscellaneous a écrit : > > The RLP draft mentioned on the libvorbis docs page, are there sample > > implementations (in C) of this yet anywhere?  > Sorry that should say RTP, not RLP I am not sure I understand exactly what you are asking, but libavformat has a RTP packetizer that works for both Vorbis and Theora, implemented in rtpenc_xiph.c.

Hello, What does it mean when "rtp set debug ip" shows RTP packets that have been send, but there is no audio ? There was no audio on my call in both directions, but "rtp set debug" shows that there were RTP packets send. There is no firewall active on my Asterisk server : [root at sip asterisk]# /sbin/service iptables status iptables: Firewall not running. Kind

Hallo all, i didn't found any deb packages for ptalarbvorm so i did it by my self. currently i do it in my playgrond here https://launchpad.net/~bug-track/+archive/ppa probably it will be better to add ffmpeg2theora too. I stated write scrip to do it regularity without big work overhead. any comments? Regards, Alexey

Hi, I'm just trying to install the DPMA on my Asterisk. I already made the upgrade from Asterisk 1.8.5 to Asterisk 1.8.11-cert2. This is what i did: *mv /usr/lib/asterisk/modules /usr/lib/asterisk/modules-185 * *compiling Asterisk-Cert2 1.8.11* *./configure make make install make config * Afther that i register the DPMA license, and finally copied the * res_digium_phone.so* to

Hello, 2 asterisk servers 16.8.0 version running on Debian 10.3 On one of them, I can't compile asterisk having error    [CC] res_rtp_asterisk.c -> res_rtp_asterisk.o res_rtp_asterisk.c:2674:3: error: ‘pj_ice_sess_cb’ {aka ‘struct pj_ice_sess_cb’} has no member named ‘on_valid_pair’   .on_valid_pair = ast_rtp_on_valid_pair,    ^~~~~~~~~~~~~ res_rtp_asterisk.c:2674:19: warning:

Hello! I'm facing completely choppy sound. The wireshark trace shows, that there are a lot of codec changes without any trigger (means no options or reinvite or any other package). Background: The call is initiated by asterisk and is received by the same asterisk conference room via Phone extension -> asterisk -> provider A -> provider B -> asterisk. Asterisk initially sends

I am trying to enable full WebRTC support on asterisk-11.15 for installation on a CentOS 5 machine. Currently the distro cannot be upgraded to any later CentOS series. This CentOS series ships with openssl-0.9.8e, which lacks DTLS-SRTP support required for WebRTC. So I decided to build a parallel install of openssl. I chose the Fedora 21 package, openssl-1.0.1j, and built it on CentOS 5. The

Hi, compiling the current git version on Centos 7 gives me: [CC] res_statsd.c -> res_statsd.o res_rtp_asterisk.c:2669:2: error: unknown field ‘on_valid_pair’ specified in initializer .on_valid_pair = ast_rtp_on_valid_pair, ^ res_rtp_asterisk.c:2669:2: warning: initialization from incompatible pointer type [enabled by default] res_rtp_asterisk.c:2669:2: warning: (near initialization

I have been using 11.4.0 for some time. All was fine. I downloaded 11.5, extracted, run ./configure, make, make install. I got a message about res_rtp_asterisk.so was not compiled in the 11.5 Sure enough I have rss_rtp_asterisk.c but not .o file and no .so file. I then looked in the config.log and nothing is in there about res_rtp_asterisk What's up? jerry -------------- next part

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

On 08/21/2015 12:19 AM, Andrew Udvare wrote: > So basically I think you would have to propose inclusion of a version of > ffmpeg into Wine's source to get the green-light (and update as > necessary). Otherwise there will be a lot of new bug reports of crashes > landing in a probably stripped copy of the system's ffmpeg or libav, > with users not knowing how to give good

Hi I'm trying to get the rebuilt parking functionality to work in Asterisk 12.0.0. In Asterisk 11.6.0 I managed to get a call to get parked by adding a dynamic feature in features.conf for the DMTF sequence *# which called a macro in extensions.conf, which then runned the ParkAndAnnounce application, and the call got parked. The syntax for ParkAndAnnounce I used was this (I don't

Hello, I am trying to get MP4/H.264 playback in Firefox to work on my CentOS laptop (for vimeo). I installed the gstreamer plugins as described here: https://wiki.centos.org/TipsAndTricks/MultimediaOnCentOS7 (No, I did not install Flash, VLC and all the other stuff. I only want HTML5 MP4 playback..) I enabled the nux repos. I did install all available gstreamer plugins, i.e. -good -bad

Le 13/03/2020 à 13:30, Joshua C. Colp a écrit : > On Fri, Mar 13, 2020 at 9:27 AM Administrator <admin at tootai.net > <mailto:admin at tootai.net>> wrote: > > Hello, > > 2 asterisk servers 16.8.0 version running on Debian 10.3 On one of > them, > I can't compile asterisk having error > >     [CC] res_rtp_asterisk.c ->