Problems with MusicOnHold

Asterisk Asterisk Support
1

Dear Team,

i use Asterisk 1.8 on a Debian Wheezy Installation.

Here are my Configuration-Files:

[code]/etc/asterisk/extensions.conf:
…
exten => _4XXXXXXXXXXXX,1,Answer()
exten => _4XXXXXXXXXXXX,2,Wait(1)
exten => _4XXXXXXXXXXXX,3,Dial(SIP/10,60,m(default))
…

/etc/asterisk/musiconhold.conf:
[general]

[default]
mode=custom
directory=/var/lib/asterisk/mohmp3
application=/usr/bin/mpg123 -q -r 8000 -f 8192 -b 2048 --mono -s

/etc/asterisk/features.conf:
…
parkedmusicclass=default
…
[/code]

Now, if i call the Number, i get the MusicOnHold (the SIP 10 rings correctly), but the MOH Plays only a few Seconds (but the SIP 10 rings without Problems).

Now my question:
how i can fix them?

i have created a MP3 file with 128 KBits and a length of 60 Seconds…

My second question:
how i can Change the dial tone to Europe?

can you help me?

thanks

many greets, markus

2

You need to change the dialtone in the calling SIP phone (I’m guessing that is what you have), or in the PSTN to SIP gateway. Dialtone is only generated by Asterisk for locally connected analogue FXS devices.

Although a bit strange for a phone, it is possible that you destination phone is generating early media. You need to produce a CLI trace at at least verbosity 5 and be prepared to provide a SIP debugging trace.

There is really no point in using MP3 for static music on hold. The telephone system cannot reproduce it accurately. You would be better to pre-convert it to each of the codecs actually used for your phone calls.

The other problem you might be having is that the caller is silence suppressing and you haven’t configured a timing source.

3

[quote=“david55”]You need to change the dialtone in the calling SIP phone (I’m guessing that is what you have), or in the PSTN to SIP gateway. Dialtone is only generated by Asterisk for locally connected analogue FXS devices.

Although a bit strange for a phone, it is possible that you destination phone is generating early media. You need to produce a CLI trace at at least verbosity 5 and be prepared to provide a SIP debugging trace.

There is really no point in using MP3 for static music on hold. The telephone system cannot reproduce it accurately. You would be better to pre-convert it to each of the codecs actually used for your phone calls.

The other problem you might be having is that the caller is silence suppressing and you haven’t configured a timing source.[/quote]

Hello!

Thanks for your reply,

now i have tried to put in this folder .gsm files they are working perfect with (playback) and also tested with wav files. after a few seconds, the musiconhold is silent…

also i have the problem that the music on hold ,runs not from beginning…

how i can do that, that each caller get the music on hold from start?

can anybody help me?

thanks!

many greets