similar to: Free Music for MOH from Digium?

hi using the following in sip.conf, codec preferences aren't set, and asterisk uses alaw whatever I do, except force it to one specific in the [user] [general] disallow=all allow=g726 allow=g729 allow=gsm allow=alaw then, from 'sip show peer something' it tells me Codecs : 0x11a (gsm|alaw|g726|g729) Codec Order : (none) can someone please explaing why? this is

Hi On the www.asterisk.org main page it says "Music provided by Freeplay Music" with a link - Where is the music in the *config? I cant find any supplied music - is there any?

Hi, I am wondering if vorbis has a standard tag for general music type. I don't mean pop, jazz, etc., but something more in line with describing the mood of the mucic. I am thinking of games which have a soundtrack in vorbis format which could use the user's own repository of .ogg files during gameplay, and use the tag to choose appropriate music for that part of the game. For

The Asterisk development team is pleased to announce the release of Asterisk Core Sounds version 1.4.15, Extra Sounds 1.4.8, and Freeplay Music On Hold sound files. These sound files are available at http://downloads.digium.com/pub/telephony/sounds/. Future versions of Asterisk will do this automatically from the Makefile (when the sounds are enabled in menuselect). Jean-Marc Valin (from

(Note: This announcement originally went out with an incorrect version number mentioned for the Extra sounds. It should have went out as Extra Sounds 1.4.9 and has been corrected in this announcement. Thank you for your understanding.) The Asterisk development team is pleased to announce the release of Asterisk Core Sounds version 1.4.15, Extra Sounds 1.4.9, and Freeplay Music On Hold sound

We are running * V1.0.9 on a demo box. We have set up everything in our dialplan and we have a directory where we store individual extension settings. That directory is called "extensions-phones.d" and it contains a number of .conf files. In my extensions.conf file I have put a #include "extensions-phones.d/*.conf" in my [globals] context If we reload and restart *, and

Hello *'s, First Of all Marry Christmas, I want to setup asterisk at large means "my main asterisk server placed in my office(in Pakistan), and some offices outside Pakistan and i want to connect these locations to my main * server (in Pakistan) on remote locations i'll used asterisk can i do this or may be i changed my plans kindly guides me. Thanks In Advance. Adnan Ahmed.

Hi, Is there any Royalty one needs to pay when using the inbuilt exisimg asterisk on hold music or when using any other mp3 from a music album. I think we need to pay for the later, but I am not sure if we need to pay for the inbuilt asterisk(freepbx) on hold music. -- Deepak --------------------------------- Yahoo! Answers - Get better answers from someone who

Hi, maybe this has already been discussed already.. I _did_ search the archives, found no solution... I have a 4.1 channel track, I rip it to 5 separate mono wav files. Is it possible to encode these independant files to one single multi-channel flac file with flac? I heard a bit about ripping to wav, converting all mono tracks to flac separately (i had to use --channel-map=none to get this to

Is there a way of calling the method of the root class directly? Here is an example explaing my question. Everything relates to S3/UseMethod classes and methods. I have an object x of class MicroarrayMatrix, which inherits from class Matrix. The "core" object of x is a simple matrix. In other words, I basically have x <- matrix(logratios, ncol=nbrOfSpots, nrow=nbrOfSlides)

On Sun, Dec 21, 2014 at 7:08 PM, Marc Muehlfeld <mmuehlfeld at samba.org> wrote: > Am 21.12.2014 um 04:14 schrieb Dongsheng Song: >> After put domain users in the remote destop group in the DC, these >> users still can't logon workstation via remote desktop, it's a very >> strange design. Maybe we can make samba-tool fix it by certain >> parameter ? >

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Hello, When I tried to install chan_h323-1.0.1-module.i386 RPM i got these errors. Failed dependencies: libh323_linux_x86_r.so.1 is needed by chan_h323-1.0.1-module.i386 libpt_linux_x86_r.so.1 is needed by chan_h323-1.0.1-module.i386 But i found the same files in /usr/lib/libh323_linux_x86_r.so.1 /usr/lib/libpt_linux_x86_r.so.1 What to do for asterisk to detect the same

On Tue, May 14, 2019 at 04:55:05PM +1000, Ben Skeggs wrote: > On Sun, 12 May 2019 at 04:23, Peteris Rudzusiks > <peteris.rudzusiks at gmail.com> wrote: > > > > nv50_head_atomic_duplicate_state() makes a copy of nv50_head_atom > > struct. This patch adds copying of struct member named "or", which > > previously was left uninitialized in the duplicated

From: Eric Dumazet <edumazet at google.com> commit 3fa2a1df909 (virtio-net: per cpu 64 bit stats (v2)) added a race on 32bit arches. We must use separate syncp for rx and tx path as they can be run at the same time on different cpus. Thus one sequence increment can be lost and readers spin forever. Signed-off-by: Eric Dumazet <edumazet at google.com> Cc: Stephen Hemminger

From: Eric Dumazet <edumazet at google.com> commit 3fa2a1df909 (virtio-net: per cpu 64 bit stats (v2)) added a race on 32bit arches. We must use separate syncp for rx and tx path as they can be run at the same time on different cpus. Thus one sequence increment can be lost and readers spin forever. Signed-off-by: Eric Dumazet <edumazet at google.com> Cc: Stephen Hemminger

Can anyone recommend CRM software with a link into Asterisk? I would like a pop up on caller ID if possible. I've played with the FOP and SugarCRM but can't get them working together. Regards Lee -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050818/e71c34a3/attachment.htm

Is it true that asterisk 1.2.6 does not use mpg123? I just installed asterisk 1.2.6 and while I do have music on hold (through format_mp3?) I do not have an mpg123 process running. I seem to be having serious audio issues when going through one of my providers (and just through that provider) when using mp3 for hold music, however when using wav files it is fine. The processor is only at about

Hi, If I try to transfer a user (user listens to MOH while I transfer) to eg. a queue, and the transfer occour while the MOH in the queue is playing, the MOH will stop, and the user hears nothing but scilence, but is in the queue. If I make the transfer to the queue, while still listening to the announcement, the user will hear the announcement, and then the MOH will start. Using latest CVS

I'm running into some errors I can't figure out. Situation: I have a Sparc IPX (2.6 Solaris), a Mac OS X box and an Ultra Sparc 1 running Solaris 9. I am trying to rsync a web site between the IPX and Ultra 1. All boxes are running 2.5.5 rsync built from source. Problem: Can't get a a sync between the IPX -> Ultra. IPX = kashmir Ultra = vortac Mac = archer Assumptions: a).