similar to: AudioCodec MP114

I have an MP114 2fxs,2fxo which I would like to use with Trixbox, does anyone have a setup file they can share to help me work this out. Instructions or a link I can follow - thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090726/643e5b96/attachment.htm

Hi, Another newbie question. I'm trying to figure out how to fit vorbis into the Apple AudioCodec API and it asks your codec to be able to answer questions about itself. One I'm not sure about, does vorbis encode a fixed number of frames per packet for any given track? I know it's VBR, but that doesn't necessarily imply variable # of frames per packet. Thanks, -n8 --

I am looking to build a small PBX for an office that has 3 incoming analog lines and less than 10 extensions. For the Asterisk server I am going to use a small form factor PC with no-PCI slots so the FXO interface needs to be either FXO->SIP or USB. Can anyone make suggestions? I am looking at an AudioCodes MP114 FXO or possibly two Sangoma U100's but don't have experience with

I'm currently looking to deploy an Asterisk server using an FXO media gateway to connect to the PSTN and was looking for any user experiences that may aid in selecting a gateway. Specifically i'm looking for a 4-port model under 500 dollars. Within this category exists: MediaTrix 1204 Grandstream GXW-4104 AudioCodes MP114 I've read over Voip-info.org regarding these products and

Hello again Asterisk people. I am running Asterisk 1.42 on an old PowerPC ibook. I have had this deployed for several years now, with pretty good results. Recently I added a callerid service to my landline (qwest). I am using the audiocodes MP114 2fxo/2fxs gateway, which is an outstanding piece of hardware once it's configured (lol). Anyhow, I can see that the gateway is passing

From: Nathaniel Gray <n8gray@gmail.com> Date: Sep 14, 2005 11:30 AM Subject: Newbie q: decoupling vorbis from ogg To: vorbis-dev@lists.xiph.org Hi, Sorry if this is a newbie question. I'm trying to write an OS X AudioCodec for Vorbis using libvorbis. I'm confused about the libvorbis dependency on libogg. I thought the vorbis spec didn't require ogg as the container, but the

Hi, I have a spiffy new gateway which seems quite promising. It's the Audiocodes MP114 FXS_FXO (2 of each). I have got it configured and working reasonably well, but have a couple of issues. 1) Asterisk 1.2.13 voicemail seems to be hung up on by the gateway after 10 seconds. This isn't asterisk saying it's quiet for 10 seconds, it's the gateway deciding it's time to go

Hi List, I am trying to call from my asterisk box (1.2.18) to and audiocodes MP114. I keep getting an error from asterisk of -- Got SIP response 415 "Unsupported Media Type" back from XXX.XXX.XX.XX. Both box's are set up to use G729. Anyone have a hint as to what it may be ? Thanks. Dovid -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, I have something like this: SIPUSER <-sip-> ASTERISK <-oh323-> AUDIOCODEC <-e1-> PSTN After calling from SIP to PSTN (and from PSTN to SIP too) I can't hear anything only in my SIPUSER. At the PSTN side everything is OK. I have another network with another h323/sip (in the place of asterisk) and there everything is OK. In AUDIOCODES logs I see that everything goes

Salve, Imagine ogg vorbis is used to produce radio with free software. A journalist would produce a report end send it with 24kBit/s out from a cricis place somewhere in the world. DRM is going to use MP4 - so his report has to be reconverted with loosing quality :-( Can you imagine to have an free codec someday that would work in embedded radio-reciver like MP4? If yes, should DRM not be open

Hi Guys, So creating a forums application seems like something that rails can handle easily and well, and whatever implementation that came to fruition would be head and shoulders above existing products like vBulliten and phpBB. Even the existing rails forums are using php-based forum products! An insult if you ask me. So my question -- is there any current development of a rails-based

A long time ago (Asterisk 0.x, 1.0.x) my experience is that there were alot of interoperability issues, a common troubleshooting issue was to make sure all endpoints where using the latest version of Asterisk. I have not seen these issues in a while. However I've been working with a customer of mine and this ITSP called IP Communications (IPComms.net) well turns out we have had constant

Razza, I have a MP114 FXO/FXS that I have never got to work , even as an FXS, even though I have several other FXS's that work fine ie Linksys PAP2 etc.. would you put up your config? PDE

Salve, one year ago here was already a discussion about a version of Ogg Vorbis, with low delay enough for VoIP http://www.xiph.org/archives/vorbis/200108/0106.html >all transform codecs. They require a fairly large block of samples, i.e. in >Vorbis you need to input 3072 samples (or is it 4096) from each channel >before you can get any output. This could be reduced, trading off

Salve, one year ago there was already a discussion about a version of Ogg Vorbis, with low delay enough for VoIP http://www.xiph.org/archives/vorbis/200108/0106.html >all transform codecs. They require a fairly large block of samples, i.e. in >Vorbis you need to input 3072 samples (or is it 4096) from each channel >before you can get any output. This could be reduced, trading off

I upgraded to Leopard (version 10.5 of OS X) a few weeks ago. Although I was a bit disappointed, but not surprised, to see that FLAC support isn't built natively into the OS, I was very happy to notice recently that Apple ships source code for a FLAC encoder and decoder codec component in /Developer/Examples/CoreAudio/AudioCodecs/ FLAC.xcodeproj. All that is necessary to build the

Ok, I am back from my thanksgiving holiday, and I find there was a big snow storm here in Seattle. Apparently during the storm there where multiple brown out/black outs. I have struggled since day one to get a high quality PSTN gateway configured with my very long loop and Mac based asterisk. I originally tried the HT-488, which had multiple issues, and was unacceptable. I then purchased

>> recently that Apple ships source code for a FLAC encoder and decoder >> codec component in /Developer/Examples/CoreAudio/AudioCodecs/ >> FLAC.xcodeproj. > > You can find a slightly more detailed review of the Apple's FLAC codec > implementation in my blog post at: > http://barelyfocused.net/blog/2007/10/28/flac-support-in-mac-os-x-105-leopard/ Thanks for the

Hello, I have a problem with a cisco GW, if i only set g711 ulaw or alow as codec in my ata the the GW return a media not acceptable error. but If i add the g729 codec the all is ok. I see in the config of the cisco where to define codec for imcoming call but not for outgoing *Jul 17 15:57:02.604: Received: INVITE sip:0041787518551 at 192.168.0.110 SIP/2.0 Via: SIP/2.0/UDP

Hello guys, Does anyone has an Asterisk server hosted off-site ? Like in those data centers that do web hosting in dedicated servers ? Is there a hosting company that has a special plan to host voip services like this, or usually is hosted in those dedicated servers like the ones I asked above ? What about QoS ? I know that most (if not all) are connected to their switch through a