* You are viewing Posts Tagged ‘wave audio’

ControlPlayback Unable To Play MixMonitor Files, Premature EOF

I have a strange problem with MixMonitor and ControlPlayback,

I am recording wav files

MixMonitor(/path/filename.wav,b)

When I try to play them back, I get no audio, dialplan continues to the next line as if the sound file is 0 seconds long

ControlPlayback(/path/filename)

I can play the file in other programs and verify there is sound there

If I touch the file with sox, it will start working

sox /path/filename.wav /path/filename-trim.wav trim 1

ControlPlayback(/path/filename-trim) ; Works

Both files look the same

/path/filename.wav: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono 8000 Hz
/path/filename-trim.wav: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono 8000 Hz

However the sox command above gives a “Premature EOF” warning message for the input wav file. I suspect this is the problem. Why would I be creating files using MixMonitor that have a premature end of file and what can I do to avoid this?

Using Asterisk 1.8.19.

Thanks

Not able to play wav files in asterisk

Hello all,

I am trying to play .wav file using asterisk 1.8.7.1.  I tried playing the wav files having different properties.

The properties of the file can be seen using ‘file’ command or ‘ffmpeg -i

1.  Filname: miss_audio.wav

[root@localhost en]# file miss_audio.wav

miss_audio.wav: RIFF (little-endian) data, WAVE audio, ITU G.711 mu-law, mono 8000 Hz

[root@localhost en]# ffmpeg -i miss_audio.wav

Stream #0:0: Audio: pcm_mulaw ([7][0][0][0] / 0x0007), 8000 Hz, 1 channels, s16, 64 kb/s

I am getting the following warning in this case.

[Dec 24 15:08:47] WARNING[26513]: format_wav.c:92 check_header_fmt: Not a wav file 7
[Dec 24 15:08:47] WARNING[26513]: file.c:376 fn_wrapper: Unable to open format wav

2. Filename: msg0000.wav

This file is recorded using asterisk voicemail() application .

[root@localhost en]# file msg0000.wav
msg0000.wav: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono 8000 Hz

[root@localhost en]# ffmpeg -i msg0000.wav

Stream #0:0: Audio: pcm_s16le ([1][0][0][0] / 0x0001), 8000 Hz, 1 channels, s16, 128 kb/s

And in this case  asterisk is trying to play the file in slin format.

  — Playing ‘msg0000.slin’ (language ‘en’)
[Dec 24 15:14:10] WARNING[26566]: app_playback.c:475 playback_exec: ast_streamfile failed on SIP/phone1-00000000 for msg0000
    — Auto fallthrough, channel ‘SIP/phone1-00000000′ status is ‘UNKNOWN’

[Dec 24 15:14:10] DEBUG[26566]: channel.c:2679 ast_softhangup_nolock: Soft-Hanging up channel ‘SIP/phone1-00000000′

Why does asterisk start playing the file in slin format.

I would be thankful if someone can guide me to the steps for playing the wav file in asterisk.

Regards,

Shalu

Polycom SoundPoint IP 650 freezes on boot after adding just one custom ringtone

Hi I’m new to this list, so please forgive me off-topic or RTFM-questions.

I have an asterisk/elastix driven phone-environment using Polycom
SoundPoint IP 650 as extensions. When adding just one custom ringtone
(~57KB) in a proper format (ML.wav: RIFF (little-endian) data, WAVE
audio, ITU G.711 mu-law, mono 8000 Hz) the phone boots well. But after I
have chosen the custom ringtone as my ringtone the phone works without
problems until next reboot. I have to rename my custom ringtone for that
it is not found on boottime, change the ringtone to a default one and
reboot the phone to make it work again.

My questions:
1. Where are configurations done with the Webserver of the phone stored?
I guess must be somewhere in the tftpboot-dir on my asterisk/elastix
server. But I can’t recognize any file changes (compared timestamps)
2. Where can I find out how many space is left on my phone (some
PDF-Guides from polycom say that about 160KB of custom ringtones or
about 120 items in contact directory are fine for the phone – I have 7
items in my contact directory and just one (57KB also tried 38KB) custom
ringtone
3. What else could be the problem for this behaviour?

Thank your for helping me gettng started with asterisk

Marco