Bug#272259: Same Problem with SIP

[Tim Ruehsen]

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'?

Tim