similar to: how to incorporate amr codec with asterisk 11

Hi All, I would like to configure AMR codec in asterisk 1.8.9.1. After lots of research i found " http://sourceforge.net/projects/asterisk-amr/files/" thie link, and follow steps to configure amr. codec_amr.so successfully compiled and load in asterisk. *> core show translation * Translation times between formats (in microseconds) for one second of data Source

Has anyone successfully compiled the AMR codec into an Asterisk install, and if so, what steps did you take? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100825/b3db9f37/attachment.htm

Hi all, I have a project for the 3G related, AMR and AMR-WB support. I'm using the client develop suite from the PortSIP(http://www.portsip.com), as their said support the AMR, AMR-WB with RFC4867. Now I have to setup a SIP server/SIP PBX in our Lab for test, does the Asterisk support these codecs and RFC4867 ? If no, there has any plugin to support this ? Also, any other Server/PBX which

In order to develop a Voip application, today i should make it robust to bit-errors over wireless networks. This is an actually lack of Speex, infact what i've understood is that if a packet arrives corrupted, i must pass NULL to the decoder in order to let it know. My target is to use UDP (with checksum field disabled) and exploit also corrupted packets giving them to the AMR codec. Otherwise

Just noticed that packman has precompiled versions of amr codec. Both wideband and narrowband. Can these be used for asterisk? Heard some nice about AMR (in general) If so, any one around with experience with either?? hw

Hi, I'm having problems with an asterisk server that's not offering Codecs for ulaw and alaw as it should. I've three servers in total: a1, a2 and "b" A1 and A2 have pretty much the same config files, except IP address info changes Server B is configured to accept all inbound invites. Calls from A1 to B, all work fine, and in a sip debug session I can see

> On Thu, May 10, 2018 at 11:44:14AM -0700, Steve Edwards wrote: >> I receive an INVITE/SDP containing: >> >> m=audio 11310 RTP/AVP 3 0 101 >> >> which I interpret as gsm, ulaw, rfc2833. >> >> and I reply with an OK/SDP containing: >> >> m=audio 15884 RTP/AVP 0 3 101 >> >> which I interpret as ulaw, gsm, rfc2833. >>

Hi! my asterisk-12.2.0 with pjsip-2.2.0 does not translate codecs any more. I tried every combination. silent on both sides. I compiled pjsip with no resample in pjsip. ./configure --prefix=/usr --enable-shared --disable-sound*--disable-resample* --disable-video --disable-opencore-amr is there a way to force asterisk back to do the codec translation? Attachment: sip show channel of the

Hi Christian, Thanks for publishing this. Do you happen to have some numbers on what the expected bit error rates are for a "typical AMR-WB usage scenario"? I'm curious about that because you note that the Opus bitstream wasn't designed to be inherently robust against a bit error at any point in the packet (and some places will obviously effect it worse than others) - however

I just noticed something very strange on one of our Dell PE1750 servers. It is running FreeBSD 4-STABLE on dual CPU's with the embedded Dell Raid controller (amr driver). Attached are 3 disks of 145GB. On a RAID5 logical drive this gives me ~280GB storage. Up until the last reboot (35 days ago) the 'amrcontrol' status utility gave me: Logical drive 0 Stipes blah Size blah

Rethinking to BER in UMTS, it seems acceptable to use speex on it do you agree? ----- Messaggio originale ----- Da: marzullo maio <marzullo63@yahoo.it> A: Jean-Marc Valin <jean-marc.valin@usherbrooke.ca> Cc: speex-dev@xiph.org Inviato: Sabato 5 maggio 2007, 23:06:40 Oggetto: Re: [Speex-dev] AMR vs Speex on wireless networks. NB for CSD/HSCSD and WB for UMTS/HSDPA. The UMTS/HSDPA

So the call used Alaw as Codec. > Am 13.06.2020 um 17:23 schrieb Luca Bertoncello <lucabert at lucabert.de>: > > Am 13.06.2020 um 13:47 schrieb Michael Keuter: > > Hi > >> Try "sip show peer <peername>" for a phone. > > So: > > mobile phone: > bpi*CLI> sip show peer 0049177xxxxxxx > > > > > * Name :

Hi, according to http://www.freebsd.org/cgi/cvsweb.cgi/src/sys/dev/amr/amr.c?only_with_tag=RELENG_6 it seems MegaMGR for linux now can work. Any experience? -- Cris, member of G.U.F.I Italian FreeBSD User Group http://www.gufi.org/

Hi! We are in process of setting up an audio guide that will cover notable places of our capital Riga, Latvia. The target audience are tourists that dials a free phone number from a mobile handset to listen to a 3 minute introduction to historic place. All audio, 10+ languages are recorder in studio at 44KHz. The audio is stored on server in A-law 8KHz because we'll be pushing it through E1

Am 13.06.2020 um 13:47 schrieb Michael Keuter: Hi > Try "sip show peer <peername>" for a phone. So: mobile phone: bpi*CLI> sip show peer 0049177xxxxxxx * Name : 0049177xxxxxxx Description : Secret : <Set> MD5Secret : <Not set> Remote Secret: <Not set> Context : default Record On feature : automon

On Thu, Feb 20, 2020 at 9:38 PM Patrick Wakano <pwakano at gmail.com> wrote: > Hello list, > Hope you are all doing well! > > I am facing a problem when compiling Asterisk 16.8.0 in a CentOS 6 box and > I wonder if someone can put some light on it. > Log history short, install_prereq fails to install the packages (not sure > how important they actually are....):

Hi everyone, Here's a follow-up on the two BoFs we've had about doing royalty-free codecs at the IETF. Well, the good new is that the proposal is now in IETF last call until January 20th (see below). No final decision has been made, so it's important to get as much support as possible for the Working Group proposal. You can see the ongoing discussion on the mailing list archive:

Hi everyone, Here's a follow-up on the two BoFs we've had about doing royalty-free codecs at the IETF. Well, the good new is that the proposal is now in IETF last call until January 20th (see below). No final decision has been made, so it's important to get as much support as possible for the Working Group proposal. You can see the ongoing discussion on the mailing list archive:

Hello all, I am currently investigating the JitterBuffer struct provided in the Speex library, and I am actually thinking about using it with two different codecs: namely, Speex-NB and AMR-NB. From looking at the code, it seems that JitterBuffer is capable of working for any codec. Both Speex-NB and AMR-NB (and probably also other narrowband codecs) produce 20 ms frames and the sampling frequency

Hello list, Hope you are all doing well! I am facing a problem when compiling Asterisk 16.8.0 in a CentOS 6 box and I wonder if someone can put some light on it. Log history short, install_prereq fails to install the packages (not sure how important they actually are....): speexdsp-devel, gmime-devel, uriparser-devel, iksemel-devel, uw-imap-devel, hoard Then, I am running the following commands