[Asterisk-Users] Remote SIP Connection using Asterisk // Cisco7940's

[Adam M. Dobrin]

sounds like the inbound (out from the phone, into the local net) RTP 
packets are getting dropped..

just a guess here.. whats the output of "iptables -L -v"

Ross Overstreet wrote:

>Yes, I can call the phones, they ring, etc, and call call out, just no outbound audio.  Is their any difference in the inbound & outbound audio streams in Asterisk that could cause it, e.g., different ports, protocols, connection/discovery methods, etc?
>
>Thanks,
>
>Ross
>
>
>---------- Original Message ----------------------------------
>From: "Carlos Alperin" <calperin at senecacom.net>
>Date:  Sat, 9 Jul 2005 19:33:31 -0400
>
>  
>
>>Some way you should have a udp filter between you box and your phones. I see
>>that before.
>>
>>Can you call those phones?
>>
>>Carlos Alperin
>>
>>-----Original Message-----
>>From: asterisk-users-bounces at lists.digium.com
>>[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Ross
>>Overstreet
>>Sent: Saturday, July 09, 2005 2:04 PM
>>To: asterisk-users at lists.digium.com
>>Subject: [Asterisk-Users] Remote SIP Connection using Asterisk //
>>Cisco7940's
>>
>>Asterisk/phones work perfectly within our LAN.  Asterisk box has a public IP
>>- no NAT or firewalls.   When I take the phones to a remote location (again,
>>public IP - no NAT or firewalls that I know of) the outgoing audio does not
>>work.  I can hear the other party, my phones ring, I can dial out, etc, but
>>the other party cannot hear me (even if I dial #'s, etc).
>>
>>Any ideas?
>>
>>Thanks,
>>
>>Ross 
>>
>>
>>
>>
>>
>>
>>                  
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>>    
>>
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