if it is properly into Trunk, I mean into extensions.conf then it shud work
properly

On 6/6/07, Crazy Boy <[EMAIL PROTECTED]> wrote:

Hi Friends,
I want to use 5062 port for SIP protocol. I made the below modifications
in my server to use 5062 port.
Polycom phone: port=5062
Trunk settings: port=5062
sip.conf: bindaddr=5062
Extension configuration details: 5062
Our VoIP provider told me that they are allowing the SIP traffic through
5060 to 5064. I observed on my server console that my server is registered
with our VoIP provider with 5062 port. But, I am unable to make outgoing
calls.
Do I need to modify any other settings in Asterisk?
Look forward to your response. Thank you.
Regards,
Chandra.

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