Re: [Asterisk-Users] No congestion

2006-01-21 Thread Moises Silva
check incominglimit and outgoinglimit in sip.conf http://www.voip-info.org On 1/20/06, Kristian Larsson <[EMAIL PROTECTED]> wrote: > Hey! > > I'm having a small problem. I'm using Realtime to > store SIP account information. Dialing works just > fine, but when dialing a person already on the > pho

[Asterisk-Users] No congestion

2006-01-20 Thread Kristian Larsson
Hey! I'm having a small problem. I'm using Realtime to store SIP account information. Dialing works just fine, but when dialing a person already on the phone I don't get a busy tone. Eg, Phone 100 calls 200 and they chat with each other phone 150 calls 100, and gets a regular ringing tone what I