If your connection to the internet is being nated you may need to add this entry to your sip.conf externip=210.x.x.x
________________________________ From: [EMAIL PROTECTED] on behalf of ram Sent: Thu 3/9/2006 12:16 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [Asterisk-Users] Oneway voice Hi all I have installed AAH 2.6 created extension, and created Trunk created outbound routing iam able to make calls out and configured incoming, also working fine with the extension I have problem here I ahve extension sitting in same network where the AAH installed My provider support canreinvite=yes when iam making calls, its not consuming any b/w and voice quality is good in sip_additional.conf i have made in extension also canreinvite=yes another extension sitting another Country and he is behind nat here also made extension caninvite=yes i get one way Voice, later i have made the extension config( out side country extension) canreinvite=no the voice quality is good, but its taking 128Kb b/w how can i resolve this problem using g729 codec and save b/w thanks any suggestions ram
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