Bug#272259: Same Problem with SIP

[Alessandro Polverini]

On Thu, 2004-11-25 at 15:30, Tim Ruehsen wrote:
> Hi,
> 
> i experienced 'no sound' for SIP as well. This has been tested between several 
> clients to asterisk set up on localhost (dialled to sip:1234@localhost). This 
> leads to the demo (voicemail) application. While I did not hear anything, I 
> could see asterisk (-vvv) 'playing' voicefiles. After a while the demo app 
> goes into recording and sends the recorded message to my email. The content 
> is correct (my voice coming over the micro).
> 
> I experienced this with KPhone, linphone, gnomemeeting (H.323) and a VoIP 
> Telephone (Snom100, firmware 1.9).
> 
> After limiting the clients to one codec, I could hear the voice files from 
> asterisk. I tested gsm, pcm-ulaw and pcm-alaw. All worked fine.
> 
> My biggest problem is the Snom100, where I can't limit the codecs to one of 
> many...
> 
> This seems to be a very annoying bug in asterisk (or an underlying component). 
> It made me (and maybe many other people) wasting lot of time to find out what 
> is going on. Maybe you can raise the level of this bug to 'important'?

Hello Tim,
I never used the Snom 100, anyway, instead of disabling some codecs, try
to change the preferred one.

Also, it's possible you need to set the "allow" and "disallow"
parameters in sip.conf.

Alex