[Asterisk-Users] No congestion

[Moises Silva]

check incominglimit and outgoinglimit in sip.conf http://www.voip-info.org

On 1/20/06, Kristian Larsson <kristian at netatonce.se> 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
> 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 would is for phone 150 to receive a busy
> tone since phone 100 is already speking with
> someone else, how would I go about doing this?
>
>    Kristian.
>
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