similar to: Yuck! Error in buffer handling

> > > $ gdb /path/to/ices /path/to/coredump > > > > > #0 0x402b708d in III_dequantize_sample () from > > #/usr/local/lib/libmp3lame.so.0 1 0x402d0bc0 in ispow () from > > #/usr/local/lib/libmp3lame.so.0 > > Cannot access memory at address 0x1 > > Could this be an alpha-version of LAME? Switching to the latest > stable version (3.93-1 afair)

> > ices plays for hours, then core dumps. I replayed the last song it was playing, and it played fine. > > Use the coredump for a backtrace: > > $ gdb /path/to/ices /path/to/coredump > #0 0x402b708d in III_dequantize_sample () from /usr/local/lib/libmp3lame.so.0 #1 0x402d0bc0 in ispow () from /usr/local/lib/libmp3lame.so.0 Cannot access memory at address 0x1 Neat -

According to the AppDB EU2 1.09 has a platinum rating, but it's constantly crashing on me. I installed the game from the CD, updated to the latest version (1.09), and the game launches just fine. When it gets into the menu I notice that it's running a lot slower than it should be. I can start a single player game and it still runs slow, and then crashes. It doesn't matter what I do in

Hi All Somebody knows as resolv the error below? Already I compiled asterisk-addons-1.2.3, but exactly thus it reports this error, could help me? -- Executing WaitMusicOnHold("SIP/3205-d9ef", "30") in new stack -- Started music on hold, class 'default', on SIP/3205-d9ef Jun 23 02:14:21 WARNING[24960]: interface.c:215 decodeMP3: Junk at the beginning of frame

Please try a stable version of LAME. I don't really know what to do with this backtrace. It would also help immensely if you could reproduce the crash reliably (eg on the same song). On Sunday, 03 August 2003 at 08:34, Carl Karsten wrote: > > > > $ gdb /path/to/ices /path/to/coredump > > > > > > > #0 0x402b708d in III_dequantize_sample () from > > >

Hello, I am currently using MusicOnHold(mpg123), and it works just fine, but every once in a while I will get a flurry of warnings in the CLI like those below and Asterisk will freeze completely, and the only way to come out of it is with a kill -9 . Is mpg123 causing my problem? Is there a specific format of MP3 that should be used/avoided to not have errors like these? Any help would be greatly

Checking the earlier mails, it stated that the phones were $75 (100) & $85 (102) ref :- http://lists.digium.com/pipermail/asterisk-users/2003-June/013483.html Well, I just called Ovislink/dgtimes and was quoted $90 & $100 and the person said there was no price change. Anyone on this list actually bought them at the $75 & $85 rate ??? Regards...Martin -- Too much is just enough.

Hi ! :) Does someone have an idea of the reason why my MusicOnHold (with clean mp3 downloaded from http://aussievoip.com.au/wiki-MOH) is always having interruptions and micro-cuts ? No high load of the system, no swap, no hard disk r/w, mpg123 seems running well... nothing ! Except a little message at startup : "Warning, flexibel rate not heavily tested!" I'm getting

Hello, I just can't seem to get the MusicOnHold function to work out ok. I' managed to get the MP3Player app to work out fine, but when I run the MusicOnHold all i get is siliece. I can see that Asterisk executes mpg123 properly (I think) #ps axuww|grep mp gk 4383 0.0 0.4 3736 552 pts/4 S 15:06 0:00 /usr/bin/mpg123 -q -s --mono -r 8000 -b 2048 -f 8192 sample-hold.mp3

Type 'mpg123' at the Linux CL. (no quotes) If the version is anything other than 59r, you problem is solved. Go to the Wiki and search for Music On Hold. Do the install of version 59r ONLY as described in the docs. Cheers, Wiley -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Neil A. Hillard

Does anyone know how to rewind the music on hold? Thanks Dan Journo -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060202/ab26b18f/attachment.htm

In migrating to Samba on FreeBSD from Win(yuck)NT, I?ve run into this hitch. Let us say I have 9 users named User1, User2, User3, ? User9 User1 is a member of group wheel User2 & User3 have the administrative task (add/change/delete) of managing the content of the directory Dirc1 and all subordinate objects (files and directories). Dirc1 is the directory /usr/Shared/Dirc1. Only User1

Jack, thanks for the heads up. Could you possibly tell me what this error means: Warning: Encountered error while transfering /serve/ftp/mp3s/various_artists/Various_Artists-BLLLEEEEAAAUUURRRRGGHHH!-The_Record_7''/06-Nuclear_Roach-Pessimism-Fcs.mp3. [Libshout reported send error: Libshout socket error.] Libshout communication problem, trying to reconnect to server Failed reconnecting to

Hello, I'm new to the list, just want to say thanks a head of time to who ever can help me out with some streaming problems Hareware/software: Athlon 700 128megs of RAM 256megs swap FreeBSD 4.2-RELEASE 100mbit connection icecast 1.3.10 ices (beta) My setup: I am running 2 spawns of icecast and 2 spawns of ices. one ices process is reencoding to 24bit and the other to 128bit and both are

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 I just noticed that the Skype API for linux seems to be available. I've read before a number of posts where people were talking about implementing a chan_skype with the skype API. I wonder if there is any progress in that direction, and if anyone is working on it. /B - -- * GPG-Key: http://evil.gnarf.org/mrbk.pgp A: Because we read from top to

I'm an asterisk newbie, so initially tried to install it on a Mandrake 10.1 (see config below) and with a bit of messing about using sample config, have been able to make the test call to device 1000, and also through to the IAX test number at Digium. However, operation is extremely flaky - I cannot reliably startup and use the system on a regular basis. I have several problems listed below

I am really new to R and would appreciate some help to sort out a problem with data extraction from a large file. I have entered the following command to filter data from a large data set called "upanddown" such that those records in upanddown corresponding to blockType=3'UTR are transferred to a new file called "new": new<-sqldf("select * from upanddown where

Hello, I've done a very straightforward install of Asterisk, and can't seem to get it started. This is a proof-of-concept installation, and currently does not have any T1/E1 or FXO/FXS cards in it. I just want to use it as an internal SIP server for now. However, when I try to start Asterisk, it dies with the following messages: Junk at the beginning 49443303 Warning, flexibel

Hi all, I'm new to asterisk. I have it installed and 1 x100p card. I'm trying to use kphone to connect out. But, when I start * it gives me sound card busy error. I've checked ps aux and nothing seems to have the sound card. Any ideas? Does starting asterisk automatically take the sound card? Also, can you use the phone connected into the XFO card w/ asterisk or do you need an XFS card

Address lookup canonical name asterisk.org. aliases addresses 216.27.40.102 Service scan FTP - 21 Error: TimedOut SMTP - 25 Error: ConnectionRefused HTTP - 80 Error: ConnectionRefused POP3 - 110 Error: TimedOut NNTP - 119 Error: TimedOut digium.com is ok though Address lookup canonical name digium.com. aliases addresses 216.207.245.1 Service scan FTP - 21 Error: TimedOut SMTP - 25