similar to: Audio Convert Module

I'm running SIP between my Lucent TNT acting as a gateway, and an asterisk server. We have a PRI coming into the Lucent. Basically the problem I'm having is mostly on inbound calls but some outbound calls as well. I hear echo and sometimes some weird artifacting on calls coming in from the lucent. Everything routed over IAX to VoIP Jet or Nufone sounds fine. It seems like every 3

Does anyone have a SEP<MAC>.cnf.xml file that works with asterisk? I have the SIP firmware loaded on my Cisco 7970 but the status log shows errors parsing the config. I copied a config that was posted to the list but it didn't seem to work. Any help would be appreciated. Jeremiah -- ______________________________________________________________ Rock River Internet

Trying to use: Asterisk 1.4.2 Zaptel 1.4.0 chan_zap won't compile in asterisk 1.4.2 when used with zaptel 1.4.0. The changelog has this entry: * channels/chan_zap.c, configure, configure.ac: If we receive ZT_EVENT_REMOVED, destroy the specified channel. (issue #7256, tzafrir) Also, update the configure script to make sure that we don't try to build chan_zap

We wrote and submitted a patch to do this. Just modify app_directory.c and recompile. It adds a new flag "b" to the directory( ) app where you can have it use both first and last name. -= Info about application 'Directory' =- [Synopsis] Provide directory of voicemail extensions [Description] Directory(vm-context[|dial-context[|options]]): This application will present

Hello everyone, Now that astlinux-trunk has been coming along very nicely, I thought I would try to add support for hard realtime capabilities to AstLinux. If everything works (and there are no problems with zaptel), with a little tweaking this should improve the audio quality on systems with high loads (and probably any system at that) - especially if it is finely tuned and has zaptel

Hello everyone, (I first sent this several hours ago, it appears it got lost. Thought I'd give it a second try). Now that astlinux-trunk has been coming along very nicely, I thought I would try to add support for hard realtime capabilities to AstLinux. If everything works (and there are no problems with zaptel), with a little tweaking this should improve the audio quality on systems

Does anyone know what application I should place this function in? For example with the DB function I currently do something like this to add an entry to the asterisk database: exten => s,n,Set(DB(AGENT/${MACRO_EXTEN:1})=${CALLERID(num)}) To delete the entries I do something like this: exten => s,n,DBDel(AGENT/${MACRO_EXTEN:1}) DBDel is marked as deprecated in favor of the DB_DELETE

I've downloaded: asterisk-1.4.1 zaptel-1.4.0 I've compiled and installed zaptel. When I go to install asterisk I do: ./configure make menuselect I then take a look under the codec selection menu and I see that codec_zap can not be compiled. *************************************

I currently have two Cisco phones, a 7960 and 7970. The 7960 has a SIP OS on it and the 7970 has a SCCP. When the 7960 powers up it loads OS79XX.TXT, SIPDefault.cnf, SIP000E3875266C.cnf, RINGLIST.DAT, and dialplan.xml. I have a Cisco SmartNet agreement with the phone so I have access to download the firmware. I recently purchased a Cisco 7970 phone and was in the process of configuring

I'm trying to change the ring tone on my 7960 from the dialplan. I've tried the example on the wiki but it doesn't seem to work. Something like: exten => 3010,1,SetVar(ALERT_INFO=<Bellcore-dr1>) ; selects Ringer exten => 3010,2,Dial(SIP/3010,15) I'm not sure what the Bellcore-dr1 ringer is supposed to be. I've tried replacing ALERT_INFO with another ring tone

Mark, While these samples are pretty good they do not work "out of the box" - there are a couple of issues: 1. the samples are 44100 samples/second and Asterisk needs them to be at 8000 samples/second. This is what happens if you prune out all of the Amercian voicemail prompts and substitute yours: Asterisk 1.2.7, Copyright (C) 1999 - 2006 Digium, Inc. and others. Created by Mark

Hey everyone, A while back I worked on a project to measure call quality. I've finally gotten around to releasing it and I'm calling it recqual (Real Call Quality). There isn't much to it and it should be considered alpha quality. I'm hoping some of the bright minds on the list can help me out with it. I'll include the intro text from the README in the tarball: ----

I think it's because the website uses Flash as first player instead of using it as a fallback for when the browser does not support modern web standards like the audio tag. I am running a desktop without flashplayer and this is the screenshot of your web radio 2016-01-20 18:55 GMT+01:00 Jeremiah Rogers <jeremiahzrogers at gmail.com>: > I use iOS to listen to my Icecast all the

.PLS files should work for iOS devices. See sample text: [playlist] numberofentries=1 File1=http://91.121.91.172:31090/LG73 Title1=LG73 Vancouver's Best Music Mix! Length1=-1 Version=2 On Wed, Jan 20, 2016 at 1:27 PM, Holden Stanford <hstanford1 at gmail.com> wrote: > You can create a .m3u or .mp3 container linking to the stream URL. I can't > remember the exact tags

On Sat, 2016-12-03 at 21:25 +0200, Aki Tuomi wrote: > > On December 3, 2016 at 9:11 PM "Jeremiah C. Foster" <jeremiah at jerem > > iahfoster.com> wrote: > > > > On Sat, 2016-12-03 at 12:23 +1000, Noel Butler wrote: > > > On 03/12/2016 12:08, Jeremiah C. Foster wrote: > > > > > > > On Fri, 2016-12-02 at 10:48 +0200, Aki Tuomi

> On December 3, 2016 at 9:11 PM "Jeremiah C. Foster" <jeremiah at jeremiahfoster.com> wrote: > > > On Sat, 2016-12-03 at 12:23 +1000, Noel Butler wrote: > > On 03/12/2016 12:08, Jeremiah C. Foster wrote: > > > > > On Fri, 2016-12-02 at 10:48 +0200, Aki Tuomi wrote: > > > On 02.12.2016 10:45, Jonas Wielicki wrote: On Freitag, 2.

Hello everyone, As I promised at eTel last week, I have finished up work on my "Asterisk Native Sounds" project. Here's a little diddy from astlinux.org: ----------------------------------- Asterisk Native Sounds are a collection of audio prompts for Asterisk. They will improve quality, reduce CPU usage, reduce latency, and (in some cases) eliminate the need for G729

Thank's Philipp and Dmitrijs. That got it fixed. I just pointed the web and admin settings where they belong and all's working great. Thanks for the sighup help, Dmitrijs. Jeremiah Rogers Cell: 704-996-5334 Email: jeremiahzrogers at gmail.com Social Networking: /jzrogers > On Nov 2, 2015, at 08:35, Philipp Schafft <lion at lion.leolix.org> wrote: > > Good afternoon, >

You can create a .m3u or .mp3 container linking to the stream URL. I can't remember the exact tags required but this should work, just use your favorite text editor to make it. On Wed, Jan 20, 2016 at 1:23 PM, Edoardo Putti <edoardo.putti at gmail.com> wrote: > I think it's because the website uses Flash as first player instead of > using it as a fallback for when the browser

Ok folks, I need to find a method for my listeners to listen to my radio station with their Android Phones - - -I think that I need to bypass the flash player - - -can anyone help me out there. I would imagine that many of you are broadcasting to people with Android phones - - -any help would be appreciated Regards Gary Hudson Home: 817-710-6367 Cell: 0959476691 From: