[asterisk-users] Problems With Playback of Audio On SIP Only System

[Brian Alexander]

Mark,

Thanks - that was the problem I was having. Is there somewhere I could have
looked to have discovered the problem on my own? I would never have guessed
that on my own and my searches had not found it either.

Thanks again,
-Brian


On Mon, Jan 19, 2009 at 7:00 PM, Mark Michelson <mmichelson at digium.com>wrote:

> Brian Alexander wrote:
> > I have been installing Asterisk as a SIP only system (no Digium
> > Hardware) for demonstration purposes. SIP users can connect to menus and
> > voicemail fine but the audio quality is terrible. The stock voicemail
> > problems are bad but basically understandable - voice menus recorded
> > through the asterisk-gui-2.0 are difficult to even understand.
> >
> > The phone I am testing with is a Polycom SountPoint IP 430 SIP. I have
> > configured the phone for ulaw to be it primary codec and set disallow
> > all and allow ulaw in the users.conf.
> >
> > When that did not work I guessed that something was wrong with
> > dahdi_dummy but dahdi_test is showing results around 99.987%.
> >
> > Here are the details of what software I have been using:
> > asterisk-1.4 (r168975)
> > dahdi-linux-complete 2.1.0 (r 5662)
> > asterisk-gui-2.0 (r4446)
> >
> > The linux kernel is 2.6.24.6 built with 1000 Hz timer.
> >
> > Thank you for your help, I am a stumped.
> >
> > -Brian
>
> If you are using gsm prompts and gcc version 4.2 or higher, then you may be
> experiencing the optimizer bug that gcc has with gsm audio. The workarounds
> for
> this are to use a different format for sounds or to set the DONT_OPTIMIZE
> flag
> in menuselect. If you want an optimized build and gsm formatted sounds,
> then you
> could always attempt downgrading your gcc version to 4.1 or earlier.
>
> Mark Michelson
>
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