similar to: mp3 stream relay without reen-/transcoding (SLIMP3)

To force guest access to a share, you need both guest ok = yes guest only = yes Otherwise it will try and authenticate you first before dropping you to guest. If this doesn't work, possibly you don't have a valid guest account set. If no account is set as the guest account (on the linux machine) it will use the user "nobody". > -Larry Engleman > The Burgiss Group >

I've been trying to setup AddQueueMember() as a replacement for the deprecated AgentCallbackLogin(), but I get _tree_ Queue()'s. Massaged extensions.conf (can provide the original if need be): ----- s n i p ----- [default] include => agent-loginout include => local ; ---------- [agent-loginout] exten => _100.,n,Macro(queue-addremove,I${EXTEN:3},dispatch,10)

Hi, I am new to this mailing list. I am adding FLAC support to the SLIMP3 player (http://www.slimdevices.com) and I had two questions about utilizing the format. The SLIMP3 player is a network mp3 player with an open source server written in perl. All audio must be re-encoded to the mp3 format before the player can use it. 1. Do any perl modules exist to read and/or write the metadata in a FLAC

--- Jason Holtzapple <jasonholtzapple@yahoo.com> wrote: > Hi, I am new to this mailing list. > > I am adding FLAC support to the SLIMP3 player excellent... keep us posted. > 1. Do any perl modules exist to read and/or write the metadata in a > FLAC file? not that I know of. the metadata parsing is not too bad but writing data back can get hairy. probably the best thing

Hello, While trying to debug a problem with the open-source slimp3 server, which uses oggdec, I found that the tarball for vorbis-tools 1.0 contains an old version of oggdec, containing at least one bug in the command line handling code (a missing break statement). The latest code in CVS appears to be fixed. Are you planning to release a new tarball for vorbis-tools? Ori Pessach --- >8

Greetings, I have some confusion. I've seen it mentioned, somewhere in some docs, that Icecast2 does not (as yet) support Push relaying. Is this actually the case? Should I be looking at icecast1 (which does support push relay, I believe) until such capability is implemented on icecast2? Or should I be creating a combination of both? Or what? Here's my situation: I have a customer

I have configured call pickup, and this works fine. Although there are 2 problems, perhaps anyone would know a solution to this; - When I pickup a call from another set, the *8 code keeps being displayed in my screen (Snom 220). I would like it to show the phonenumber of the person calling me. - When a caller that I've answered through Call-Pickup disconnects, my phone does not close

Hello Samber's, Sorry my question, but I'm a newbie on the list. I have a network with about 57 Win2K workstations, and 2 Linux(RH7.3) Servers. Samba is in one Linux box, controlling the workgroup, but now my boss wants to put the network as a domain (internal domain, e.g. wpc.wpc). All Win2K boxes make a domain login, and bla bla bla... Can anyone help me where I can find a

Hello fellows, I have this kind of configuration: one PC acting as samba server (PDC) with RH7.3 and three workstations with win2k. Today I've installed an Epson Stylus C62 printer on one of the Win2k machines (shared by the name of epsonlpt, and now I want to know if it is possible to use this printer from the RH Server? If yes, can you tell me how? Thanks in advance, Rapazito PT A

https://bugzilla.mindrot.org/show_bug.cgi?id=2153 Bug ID: 2153 Summary: ~/.profile not sourced anymore after upgrade to 6.2p1 Product: Portable OpenSSH Version: 6.2p1 Hardware: amd64 OS: Linux Status: NEW Severity: minor Priority: P5 Component: sshd Assignee: unassigned-bugs

Hello I use asterisk realtime, and I can set the order of codec preference on my realtime allow column. If I could disable transcoding, then I can always ensure a passthrough of the common codec from origin to destination without transcoding (expensive on CPU) - and more or less, force the codec to use by setting the codec preference So, can I disable transcoding?

It appears that there are no transcoders from g723 to anything else in Asterisk 10.7.1. Does anybody know how to fix that?

Hi All, I have 2 SIP clients configured and connected to Asterisk. When I place a call from SIP1 to SIP2, if both codecs are the same then everything works as expected. I then allowed one of the clients to use alaw instead of ulaw and there were audio problems (couldn't hear the other end, etc). Same thing happened when I tried to use gsm<->alaw/ulaw. Any ideas? I'm using

Hi, I am trying to configure asterisk to translate between rfc2833 and inband DTMF. I have a cisco gateway which is configured as a trunk, and a cisco IP phone which is registered to asterisk. The gateway does not support rfc2833 and the IP phone does. I tried changing directrtpsetup to "no", and that didn't help. I tried changing "canreinvite" to "no", but that

I'd like to store all of my music in FLAC form; however, I would also like to stream the library as Vorbis. Is there a tool to do on the fly stream encoding to Vorbis from FLAC? Basically, I'm looking for something that is almost like a DJ application that lets you pull in the FLAC files, do whatever DSP algorithms you need, and then code them into Vorbis and shove them out a stream.

There is a serious bug in Sangoma transcoding cards. The card has an internal, small jitter buffer and it drops samples from the audio stream when there is high jitter in the network. The bandwidth is cheap now so for me the only reason to use transcoding is where I have low-bandwidth-high-jitter links. Sangoma said they will not fix it and we had to go back to software transconding. Do you have

Hi, I have written a small app in C# that uses oggdsf to transcode to Ogg Theora on Windows. I thought some of you might find it useful. You can download it from: http://www.a2ii.com/tech/directx/TransTheora.zip Cheers, -daniel ps. it will only handle files with a single audio and/or video stream _________________________________________________________________ Win a trip for four to a

Is there a way for voice media clients (like SIP phones and POTS/PSTN phones) that connect their call legs to Asterisk to negotiate a common codec that they both use at their end, so Asterisk doesn't have to transcode? Asterisk would know which codecs each client can use, and which each prefers, then find the one they each have in common so the fewest legs need Asterisk to transcode to their

Sorry for a possible retransmit: the first was sent from an incorrect email address. I'm trying to use the Polycom SoundStation IP 7000 with Confbridge. But the transcoding from siren14 to slin32 is via slin. First, it seems odd that there's no transcoder directly to slin32 since anything else will lower fidelity. But, more importantly, there is transcoding from siren14 to slin16 and

Can't find an answer to this, but maybe I've not looked hard enough ... Does MixMonitor work without transcoding? ie. if I have a g729 stream passing through and I'm recording it with e.g. MixMonitor(/dump/filename.g729,b) and specify g729 in the filename, does MixMonitor transcode both legs of the stream to a format it can then "mix" then transcode it back to g729 to