similar to: terrible MeetMe sound with 1.6.2.9

Hello Everyone, What kind of hardware setup would I need to do if I want a T1 connection to PSTN and have 48 users in office with analog phones. Will something work if I have a T410P card in asterisk and have one T1 going to PSTN and other two to a channel bank. I would then connect the 48 phones (FXS) to the channel bank! Thanks. Deepak

The author of CBMysql mentioned in his readme that it would be better if MeetMe read everything directly from the database. I was bored so I put this together in a few hours. This patch will allow app_meetme to dynamically lookup conference numbers and pins in a database (MySQL). This allows for easy creation of web-based conference administration. INSTRUCTIONS: Apply the patch. Create a new

Hi, Is this card the same as the T410P, after all, it's made by Digium. There's one prior reference on the mailint list[1] but it didn't answer the question. There was also an SS7 status report[2] last June but it's doesn't seem to have lead anywhere either. There was post saying an SS7 release was immenent last September[3], but then silence. Any info anyone would like to

Hi! I'd like to make a stereo 8-bit wav file from a stereo (16 bit) ogg vorbis file (in delphi, with the vorbisfile). I use: ret := ov_open_callbacks(filein,vf,nil,0,ops_callbacks); if ret = 0 Then begin fileOut := TFileStream.Create(savedlg.FileName,fmCreate); repeat ret := ov_read(VF, pcmout, BufferSize, 0, 1, 1, nil); //'cause of the 8-bit

Hi, I've tried two SIP clients so far and both have unusable outgoing audio quality. Skype app sounds fine, and recording the same mic sounds fine, so I can only assume there is an issue with the clients themselves. Both clients allow you to register and make calls via SIP with any abitrary provider and credentials, so they'll work with Asterisk. I've tried them with two good

Hello. I want to write my notes in Sweave in my own language (spanish). But my language has accents and when I run Sweave in R to translate my Snw file into the tex file the accents are translated into unrecognizable characters. For example, the word "cami?n" (truck) is translated into "cami??n" Somebody knows how can I do it? One solution I don't like is using a

Hi all, I think it's Gentoo's bug, but I want to know if anyone has the same issue here. I am compiling LLVM trunk on Gentoo (gcc 4.5.2), and I get error message below, --- make[4]: Entering directory `/nfs_home/chenwj/llvm-3.1/svn/build/tools/clang/lib/Frontend' llvm[4]: Compiling InitHeaderSearch.cpp for Release+Asserts build

Per https://buildd.debian.org/build.php?pkg=r-base, we see the Alpha platform having trouble with deriv.c : gcc -I. -I../../src/include -I../../src/include -DHAVE_CONFIG_H -mieee-with-inexact -fpic -std=gnu99 -O3 -pipe -g -c deriv.c -o deriv.o deriv.c: In function 'simplify': deriv.c:267: error: unrecognizable insn: (insn 2103 64 65 9 ../../src/include/Rinlinedfuns.h:86 (set

> To provide you with a proper answer, I'd have to read what you do to > the text data to encode it in binary. Haven't got the time for that > right now, sorry. But if it is some binary encoding and not just a > gzip-type compression, maybe application/x-kate may indeed be > appropriate. This would however mean that some specialised application > is required to decode the

Greetings I did another quick server install on an old rackmount Compaq Proliant 1850R with hardware RAID5 today. would someone please shed some of their experience and light on how they deal with deleting or keeping the default configs on these items please when i do a chkconfig --list | grep :on | more nfslock 0:off 1:off 2:off 3:on 4:on 5:on 6:off readahead

Hi, With drm-nouveau-next branch on top of 3.13-rc2 and a few sound commits I get following errors in kernel log (repeating much more often). (once there was a corrupted fbcon with matrix-like strings of unrecognizable glyph-strings hanging down from screen top border) These errors do stop once I switch to Xorg. [ 13.825627] nouveau E[ PFB][0000:02:00.0] trapped write at 0x0000408280 on

Hi all, Has anyone got any experience with hooking Asterisk up with a Siemens EWSD 13 switch over a E1/PRI ? We're located in Belgium (Europe) and one of our telecom partners uses this switch. We connected one of our TE410P ports with their switch, but the status light on the TE410P card keeps blinking red. On their side they are getting a DSA (distance service alarm) error, so this

Hi Ilia, On Thu, 5 Dec 2013 10:55:06 -0500 Ilia Mirkin wrote: > On Thu, Dec 5, 2013 at 10:17 AM, Bruno Pr?mont wrote: > > With drm-nouveau-next branch on top of 3.13-rc2 and a few sound commits > > I get following errors in kernel log (repeating much more often). > > (once there was a corrupted fbcon with matrix-like strings of > > unrecognizable glyph-strings hanging

On Wed, Apr 21, 2004 at 10:52:38PM +0200, Jack Moffitt wrote: > > Does it matter if the stream is mp3 or ogg? > > No. It could be WAV, FLAC, or some 2-bit per sample mono format where > the music is unrecognizable. You'd still have to pay the royalties. It DOES matter if it's mp3. You have to pay the publisher royalties regardless, but in addition, there's a 2%

> /nfs_home/chenwj/llvm-3.1/svn/llvm/tools/clang/lib/Frontend/InitHeaderSearch.cpp:197:1: internal compiler error: in extract_insn, at recog.c:2103 I think that this is a GCC ICE. --Sean Silva On Wed, Sep 12, 2012 at 10:53 PM, 陳韋任 (Wei-Ren Chen) <chenwj at iis.sinica.edu.tw> wrote: > Hi all, > > I think it's Gentoo's bug, but I want to know if anyone has the same

I am setting up Asterisk for use in a low bandwidth environment. As bandwidth is precious and our ATA's support it, the decision was made to use the g723 codec. I have been working on this for a few days and have not been successful. The issue that I am having is garbled noise at the client on calls whose RTP streams are terminated by Asterisk system. This is the case for all the dialplan

See attachments: config.log Error on command line: llvm[4]: Compiling InitHeaderSearch.cpp for Release+Asserts build /home/jamitch/Downloads/llvm-3.2.src/tools/clang/lib/Frontend/InitHeaderSearch.cpp: In member function ‘void<unnamed>::InitHeaderSearch::AddMinGWCPlusPlusIncludePaths(llvm::StringRef, llvm::StringRef, llvm::StringRef)’:

The Asterisk Development Team has announced the release of Asterisk 1.6.2.9. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.6.2.9 resolves several issues reported by the community, and would have not been possible without your participation. Thank you! The following are a few of the issues resolved by community

The Asterisk Development Team has announced the release of Asterisk 1.6.2.9. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.6.2.9 resolves several issues reported by the community, and would have not been possible without your participation. Thank you! The following are a few of the issues resolved by community

Hello, I have found a "bug", I think in the way TDM400P cards handle FXO interface disconnect/re-connect problems. Normally I do keep all the wires connected from my CO / PABX quite securely, but I had a need to re-route the cable from one side of the desk to another, and I simply disconnected the RJ-45 connector and plugged it back in. THIS PROMPTLY RESULTED IN VERY VERY SCRATCHY AUDIO