Hi All,<br><br>I have 2 SIP clients configured and connected to Asterisk. When I place a call from SIP1 to SIP2, if both codecs are the same then everything works as expected. I then allowed one of the clients to use alaw instead of ulaw and there were audio problems (couldn't hear the other end, etc). Same thing happened when I tried to use gsm<->alaw/ulaw.<br>
<br>Any ideas? I'm using 1.6.0-beta4.<br><br>Thanks!<br>Gonzalo<br>