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

[Mark Michelson]

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