similar to: MP3Player and Ringing (long)

Hiya, I sent this bugfix to the asterisk-dev mailing list, and modified it as I noticed side effects, but now it appears to be finished. Nobody seemed to notice it there, so I thought I'd post here, as it seems to be something that will be needed as people update to the latest CVS version. So...read on :) Ted programmer_ted@hotmail.com P.S. Read to the very end. The original bugfix

Hi, I was messing around with FireFly last night and got asterisk to crash hard. It looks like the bug is a division by zero in chan_iax2.c. I reproduced it and here are some infos I got from gdb: [Switching to Thread 245775 (LWP 23251)] 0x41154918 in calc_timestamp (p=0x816b710, ts=0, f=0x424eef24) at chan_iax2.c:2896 2896 int diff = ms % (f->samples / 8);

Hi all, I could use MP3Player to play local sound (e.g: /usr/sound/abc.mp3) but I could not use it to run a remote stream, if I use mpg123 in command line, I can hear the audio ( /usr/bin/mpg123 http://...), but the same remote mp3 file could not be replay with asterisk. I would appreciate with any suggestion. Phuong Here are the log message: -- Starting simple switch on 'Zap/3-1'

Hi All, I'm having problem with MP3Player app. I want the caller to hear mp3 when he is waiting until I answer my phone. -- from extentions.conf -- exten => 200,1,Answer() exten => 200,2,MP3Player(/home/user200/mp3/hanna-hais.mp3) exten => 200,3,Dial(SIP/200|20|tTrR) exten => 200,4,Hangup() -- end -- here is debug from CLI: -- Executing

Hi I have kind of an odd problem. When dialing in from an outside line via a TE410P card it seems like MusicOnHold and MP3Player doesn't work properly (for me anyway). The remote end which is calling * doesn't hear the music but just keeps ringing. But if I insert a Playback("file_which_dont_exist") just before the Moh or MP3Player I can hear the music. Actually I observed the

Hi all, Here another guy working with ztdummy and having problems with music. MP3Player does not work for me both from a telephone entering through a passive ISDN-adapter as well as a SIP-client in the LAN. Ztdummy works with conferences. Here are my messages. I have seen several discussion threads in the mail-archive regarding this subject, but could not find an answer. Is this something I

There is a patch to mplayer that allows it to suppress stdout messages and instead output pcm data to stdout. I managed to get it working with app_mp3.c and seems like it is working fine. All that was needed was a change in the execl line and a slight increase in timeout value. I have only done limited testing. If mplayer is able to replace mpg123 without issues, this opens up a whole lot of

My experience told me that mp3player (command) was rather unreliable with shoutcast. I heard nothing. If I use madplay in musiconhold.conf, everything is ok. But, using mp3player in extensions.conf is better than using MoH. Pls kindly advise if we could use madplay in extensions.conf. Could we put it into system command? Best Rgds, ????????????????????????????????????????

Hi there, I am using bri-stuff.0.1.0-RC2k and it seems that things didn't become better. I have got lots of dropouts on the IAX2 link (no matter if jitter buffers are enabled). Further the MP3Player application does not playback streams like http://somestreamserver/somestream. It stops saying: -- Executing MP3Player("SIP/27870-ba4f",

Hi all! I downloaded right mpg123, chabged path to mpg123 binary in app_mp3.c, rebuilt app_mp3.so, and got MusicOnHold to work. But MP3Player refuses to do properly: -- Accepting AUTHENTICATED call from x.x.x.x, requested format = 1024, actual format = 1024 -- Executing Answer("IAX2/maxhome@maxhome/3", "") in new stack -- Executing

I am running CVS HEAD (on a Linux-PPC machine.) My current dialplan generates an error at the console in asterisk when I attempt to issue the MP3Player command -- I can't figure out why it's not playing the actual audio file? The rest of the dialplan works great. Here's what I see in the console: -- Executing MP3Player("IAX2/income-in-01@IP",

Hi, I've successfully installed Asterisk and placed test calls using the MP3Player application. However I notice that my call quality varies drastically depending on which MP3 I use. Since I'm not changing any settings I'm assuming that the encoding of the file makes a difference. I've tried with both the g729 and alaw codec and they both give the same replicable results.

I installed Asterisk 1.0.9 in a Freebsd 5.4 ( with no zaptel card); I have 2 zoom x5v and works great ( in extensions 123 and 321 ) but I was trying to test cmd Playback, MusicOnHold, MP3Player but when I call to extension 100 I don't hear the sound ( mp3 or gsm that I put) , I only hear noise If I leave a message in a mailbox the same, all the record is noise --------- extensionns.conf

If there is anyone on this list running Asterisk 1.0.7 beta 3 under 10.4.1 can you please tell me if you are able to execute the Asterisk cmd MP3Player(filename) I can't figure out how to get a working MP3 player to trigger the audiofile when I replace filename with an actual path to a file. Basically, when I put an OSX compiled version of MPG123 in /usr/bin Asterisk will not startup.

Getting the proper version of MPG123 (v0.59r) setup under OS X was unusually complicated for me. Sorry for pestering the list with the MP3Player() issues I kept posting about. Your inbox will be happy to know that I finally got the correct file at: http://www.macupdate.com/info.php/id/6275 What I did was: (1) First, download the installer at:

Hi, I'm running the latest Asterisk (built last Saturday) and can't get mp3's to playback on my handsets (this includes music on hold). I setup a couple of extensions, 901 and 902 to playback an mp3 I loaded on, and the sample moh that is included with Asterisk. When I attempt to call either extension I don't hear any sound, and the following displays on the console:

Sometimes in huge call volume am facing this type of error, [Jun 4 08:42:46] WARNING[8459][C-000079fa]: channel.c:5075 ast_write: Codec mismatch on channel Local/8038 at xss-call-out-00004774;1 setting write format to slin from ulaw native formats (ulaw) [Jun 4 08:43:04] WARNING[8285][C-000079da]: channel.c:5075 ast_write: Codec mismatch on channel Local/6513 at xss-call-out-00004775;1 setting

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Okay, I've looked around for a FAQ on this, and it's kinda driving me up the wall. I recently installed a new Asterisk system, and built mpg123 v. 0.59s from sources, and the audio comes out waaaay overmodulated. Any thoughts?

Hi list! I have 4 numbers on my Asterisk 1.8. 3 work perfectly, the 4.th not. I'm not sure, when it finish to work, since a month ago it runs without any problem... Well, if I will be called on this number I can't hear anything and in Asterisk I see these: [Jul 15 18:59:55] WARNING[8752]: channel.c:5060 ast_write: Codec mismatch on channel SIP/00493514977290-000001d1 setting write format