similar to: Is this SDP payload Asterisk created valid?

Hello, Is there a way to configure Asterisk so that it doesn't care about the playload type in SDP ? I'm trying to send custom data which has assigned a dynamic PT through RTP so I only need Asterisk to act as a proxy/forwarder but I'm getting "488 Not acceptable here" responses and in console "No compatible codecs, not accepting this offer" Best regards, Nicolae

re-posting the question. ----------- use case: when some one in my pbx calls 100.200, I have translations well defined, Media also (media via asterisk) --Works. when some one calls bob, or for any names I am adding Domain and call is been sent to the other party -- Works, no media... For the cases when it is talking to the external work, I want Astersik not to do anything with the SDP/payload.

Hi, I'd like to contribute to Vorbis and I think this may be of some interest for low bitrate coding. I have been experimenting with low bit-rate coding for the high-band (11 kHz to 22 kHz) and, though I haven't yet started quantizing my coefficients (a gain and an LPC filter), I expect to be able to approximate the whole 11-22 kHz band with around 1000 bits/s per channel (maybe even 500

Ok so I finally realized that you need icecast2 for ogg streaming to work and got the latest version of ices-0.2 installed. I am now unable to ever stream a friggin mp3 never mind an ogg file. The server is up (icecast2) and I see it in the web browser, when I try to run ices-0.2 I get the following errror. I am running ices-0.2 as root and icecast2 as richard (might that be the problemo?) Both

CONTENTS: Meeting days/times & Howto - Mark your calendar's dates; Videos; Hot topics; Opportunities; Announcement Flyers; New webpages ===== Come join in with the Global Free SW HW & Culture community at the BerkeleyTIP/GlobalTIP meeting, via VOIP. Two meetings this month: Sat Nov 7, 12Noon - 3PM Pacific Time (=UTC-8) Sun Nov 22, 12Noon - 3PM Pacific Time (=UTC-8) Mark your

Everytime I post a message this wanker sends me a whole load of shite. Can somone shut him down please?? Begin forwarded message: Date: Sun, 10 Nov 2002 10:20:13 +0200 From: "Slava Shklyar" <sava@techphone.co.il> To: "Richard" <richard@freespeech.org> Subject: Re: [icecast] Re [ogg] <p>qwertyuoupasdfghjklzxcvbnmqwertyuoupasdfghjklzxcvbnmqwertyuoupasdfghj

Excuse my ignorance but how do I do that from the command shell? R <p>On Wed, 09 Oct 2002 19:12:04 +1000 Michael Smith <msmith@labyrinth.net.au> wrote: > At 12:03 AM 10/9/02 -0600, you wrote: > >thanks, this works but its almost impossible to make all my > >programs to add up to an hour exactly so at some stage I would need > >to kill the a song half way through

** Great talks this meeting: (live & on video) ** Ekiga3, Asterisk, GPGPU, GStreamer, Debian Edu, HowTo Present KDE at meetings http://sites.google.com/site/berkeleytip/ Join from anywhere via VOIP conference, with the friendly, educational, productive, BerkeleyTIP people. :) Join the #berkeleytip freenode.net IRC channel for help getting your VOIP working.

You need to check out ogg and vorbis from cvs and install them as well. Refer to the following previous mailling list post for info on what to do: http://www.xiph.org/archives/icecast/2870.html On Wednesday 31 July 2002 05:57 pm, Richard wrote: > Still trying to stream ogg and have not gotten past a make error in > ices2. We www.freespeech.org are an independent media / news > site so

Actually, you don't have to check out ogg and vorbis from CVS. You can just use Vorbis 1.0, if you like. ices was updated to use the Vorbis 1.0 encoding API, and that is the source of the errors the original poster was seing. --- Stan Seibert <p>On Wed, 2002-07-31 at 15:23, D. Anthony Patrick wrote: > You need to check out ogg and vorbis from cvs and install them as well. Refer

Still trying to stream ogg and have not gotten past a make error in ices2. We www.freespeech.org are an independent media / news site so it would be nice to get this running and support this new format. I'd love to convert all of our 700 streaming video files from from realmedia as well but thats another story! Perhaps this is not the correct forum but it seems like there are a lot of

Asterisk receives a 180 Ringing with SDP from the called side, then it sends 180 without SDP to the calling side. We would like asterisk to sends to the calling side the same response that was received from the called side. This is Asterisk cert 13.1, is that a new behavior, is there a setting to change this ? ?

Hello, all. I have an asterisk 2.3.2 and a Sangoma interface through which I connected an external PSTN line. I use it as carrier for VoIP calls. I can make successfully calls, but there's one problem, I receive 200 OK with SDP with delay (sometimes more than 30 seconds). So when I make a call through asterisk I receive intially: - 100 Trying - 183 Session Progress, with SDP when the called

FYI, the Opus RTP payload format is now RFC7587: https://tools.ietf.org/html/rfc7587 Cheers, Jean-Marc -------- Forwarded Message -------- Subject: [payload] RFC 7587 on RTP Payload Format for the Opus Speech and Audio Codec Date: Tue, 30 Jun 2015 16:33:17 -0700 (PDT) From: rfc-editor at rfc-editor.org To: ietf-announce at ietf.org, rfc-dist at rfc-editor.org CC: drafts-update-ref at

Hello list, I have a question here that may be a little bit strange for some of you. I would like to send an INVITE from Asterisk to a given client without any SDP anouncement in it. Indeed, that is pretty useful for Click to call applications for instance, where you have no way to know which codecs are supported by the client you try to reach. Moreover, you let the client decide the

I have been sending OPTIONS requests 1) programatically (my own code), 2)manually via SIP VERIFY PEER x and 3)automatcially by setting verify=yes in sip.conf. The trouble is I do not see anything except an ACK 200 come back from endpoints and it does not contain any SDP/codec info. . My goal is to determine audio and video codec capability in advance of a call INVITE. I notice in both 2 and 3

How does Asterisk (1.2) handle a 180 WITH SDP? I am seeing different behavior when a call is initiated from an Asterisk server and from an alternate point. With Asterisk, I am hearing ringing and with the other origination point, I am getting a message played on the far-end indicating to wait while the call is connected. I am wondering if the ring in the 1st (Asterisk) scenario is being played

I have a box (Genband) expecting the following: 100 trying 180 ringing with SDP Or 100 trying 183 with SDP And asterisk is sending: 100 trying 180 ringing 183 with SDP Any way to modify asterisk to send what he is expecting? Thanks, Dave -------------- next part -------------- An HTML attachment was scrubbed... URL:

Asterisk Project Security Advisory - AST-2013-004 Product Asterisk Summary Remote Crash From Late Arriving SIP ACK With SDP Nature of Advisory Remote Crash Susceptibility Remote Unauthenticated Sessions Severity Major

Asterisk Project Security Advisory - AST-2016-008 Product Asterisk Summary Crash on SDP offer or answer from endpoint using Opus Nature of Advisory Remote Crash Susceptibility Remote