[Asterisk-Users] Oneway voice

[Jerry Rasmussen]

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: asterisk-users-bounces at lists.digium.com 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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