similar to: improved logging

Hi. I am having a strange problem where the source times out during anicecast2 stream -- I was using mp3pro at 32khz and even some other speeds. I have set the source timeout for 1 minute, but it still times out or says connection reset by peer. Now the strange thing is that I also had a windows media stream on the same source machine going to a wm server and it had no problems! So can someone

on Friday 08/15/2003 Michael Smith(msmith@xiph.org) wrote > On Friday 15 August 2003 02:01, John Covici wrote: > > Hi. I am having a strange problem where the source times out during > > anicecast2 stream -- I was using mp3pro at 32khz and even some other > > speeds. I have set the source timeout for 1 minute, but it still > > times out or says connection reset by

But the source computer runs the other os, so I don't have too much choice as to what I can do and the timeout problems remain with mp3 as well. The persons involved don't want to use .ogg for now so that is also not an option. Is this an icecast bug, or some other problem? on Thursday 08/14/2003 Arc(arc@indymedia.org) wrote > On Thu, Aug 14, 2003 at 12:01:11PM -0400, John Covici

diff -u --recursive icecast/src/admin.c icecast-new/src/admin.c --- icecast/src/admin.c 2003-07-18 16:29:23.000000000 -0400 +++ icecast-new/src/admin.c 2003-08-06 19:18:32.000000000 -0400 @@ -213,7 +213,7 @@ html_write(client, "HTTP/1.0 200 OK\r\n" "Content-Type: text/html\r\n" "\r\n"); - DEBUG1("Sending XSLT

As I said ogg is not an option here, what I am trying to find out is whether this is an icecast problem which it certainly seems to be, or what else might be going on. I know Windows is not too great, but that is what they have and they don't have the bandwidth to be the server themselves -- that is where I thought icecast could do the streaming. on 08/14/2003 Lee

Hi. I just checked out the latest icecast from svn and I am now getting errors in format_vorbis.c amoung which are the following: Note: These were detected at link time. format.o(.text+0x9d): In function `format_get_plugin': /usr/src/icecast/src/format.c:69: undefined reference to `format_ogg_get_plugin' format_vorbis.o(.text+0x296): In function `get_buffer_audio':

Rowland penny <rpenny at samba.org> wrote: > On 29/12/15 15:44, covici at ccs.covici.com wrote: > > Rowland penny <rpenny at samba.org> wrote: > > > >> On 29/12/15 13:59, covici at ccs.covici.com wrote: > >>> Hi. I am having problems accessing subdirectories on a samba share. I > >>> am using windows 10 build 10586 and linux kernel

Rowland penny <rpenny at samba.org> wrote: > On 29/12/15 13:59, covici at ccs.covici.com wrote: > > Hi. I am having problems accessing subdirectories on a samba share. I > > am using windows 10 build 10586 and linux kernel 4.1.15-gentoo and samba > > 4.2.7. I have two shares, one called audio and the other called > > myshare. I cannot access the subdirectories

Oh, I forgot to mention that Asterisk 15 went End-Of-Life last Thursday. :) You should use Asterisk 16. On Mon, Oct 7, 2019 at 5:58 AM George Joseph <gjoseph at digium.com> wrote: > > > On Fri, Oct 4, 2019 at 1:19 PM John Covici <covici at ccs.covici.com> wrote: > >> Hi. I am trying to install asterisk 15.7.4 from git onto a Debian 10 >> system and I am

How did you install these packages -- make sure you do ./configure and if needed make menuselect in each one of these before the make and make install. This is the only thing I can think of -- check whether there are any built-in modules as well. on Monday 03/12/2007 Asterisk Asterisk(asteriskbunnies@yahoo.com) wrote > Hey! > > Thanks for your interest, i checked the modules and i

Hi. I am having problems accessing subdirectories on a samba share. I am using windows 10 build 10586 and linux kernel 4.1.15-gentoo and samba 4.2.7. I have two shares, one called audio and the other called myshare. I cannot access the subdirectories in the myshare share. Here are the definitions. [myshare] comment = root directory path = / #fake oplocks = yes writable = yes printable =

Hi. For several months now asterisk will mysteriously stop inserting records into cdr database. I am using mysql and the asterisk addons 1.6.2 to accomplish this. Sometimes there is a strange error about column names, but often there is no error, it just stops. I just have to restart asterisk to get things going again, so I am stumped as to what is happening, or even how to troubleshoot. I

I am using viatalk as my voip provider and they use dtmf=rfc2833, but asterisk is not seeing any of the dtmf. I am using CVShead as of 8/26/05. Nothing in the logs indicates a dtmf is being seen. If I use my pots line it sees it fine. Any assistance would be appreciated. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici

Hi. I am using asterisk 1.8 and everything was working fine when I was at svn 342661. I then upgraded to vrsion 349339 and discovered the following problem -- one of the end points is a freeswitch box which offers a number of codecs, including PCMU. However, when I tried to make a call I got a 488 response and a message "multiple audio streams not supported" in the log. Is this by

Hi. I am using asterisk 11 svn r401076M and I am getting this warning at times. I can't find much doing a google search, so anyone with any ideas? I have looked at the logs, but can find no particular pattern to indicate where this is happening and the system appears to be otherwise working, but I am still wondering if something is wrong. I am also using freepbx in case there are known

I am using asterisk 1.6.0 and -- not all the time -- when a caller comes in and my ivrdials an extension, the ring he gets sounds like a modem handshake instead of the normal ring tone and it only sounds once even if the phone is not picked up. Anyone seeing this -- the logs look fine as far as I can tell. -- Your life is like a penny. You're going to lose it. The question is: How do you

Hi. I have been using asterisk-1.6.2 and if I update the version -- using svn -- to around May 19 or after, when I dial a digit on my fxs port which is on an X400p card, asterisk seg faults. If I go back before about this date, this problem does not occur. The dahdi version is svn 7445. Any ideas would be appreciated. -- Your life is like a penny. You're going to lose it. The question

Hi. I have a SIP provider which wants RFC2833 for the dtmfmode, however I would like to increase the duration of the tone, its pretty short and some IVR's are unhappy or don't detect it. I did poke around, but it looks like when RFC2833 is used, it actually generates rtp packets of some sort, so I have no idea how to increase that duration. Any assistance would be appreciated. -- Your

Hi. I am having a very strange problem --aren't they all -- with the release candidate. I have softphone which talks to asterisk from behind nat -- the asterisk is on a public ip -- and when I hit mute on the softphone, all rtp traffic ceases! Now, a version which does work is r281875, this does not happen in that vrsion, but right after that this strange thing starts and is not fixed in

I downloaded the latest 1.8 (27922) but got some segmentation faults. The first one was when it loaded cdr_odb, and so I changed menuselect not to compile that one, but the second one was when it tried to load chan_agent and so I stopped there to see if anyone else was seeing this. The agents.conf is all commented out except for [general] . Anyone know what is happening? Thanks. P.S. I deleted