[SR-Users] One way communications in Kamailio 3.1.4

[Timo Klecker]

Hello,

the disconnect after 32 seconds is sure to be a lost ACK. Maybe the ACK is
not send E2E but using the Kamailio. In this case you would have to redirect
it.

In the content of INVITE and 200, 183 or ACK you should be able to see the
IPs for RTP data. Compare these with your phones.
O=root 12312331 12312331 IN IP4 XXX.XXX.XXX.XXX

Be sure, the RTP ports are open on your firewall, too. You can see the ports
in the content, too.
M=audio XXXXX RTP/AVP 8 101

Greetings
Timo

-----Ursprüngliche Nachricht-----
Von: sr-users-bounces at lists.sip-router.org
[mailto:sr-users-bounces at lists.sip-router.org] Im Auftrag von CaptWho
Gesendet: Mittwoch, 14. September 2011 05:01
An: users at lists.kamailio.org
Betreff: [SR-Users] One way communications in Kamailio 3.1.4


I'm running 3.1.4 on centos and I'm having some trouble with voice only
going one way.  

Both extensions will ring each other, but after they connect, voice will
only travel in one direction.  One extension hears fine, but can't talk.  

I've totally opened up the firewalls (including port 5060) and I'm still
having the trouble.  I'm not behind NAT. 

I've tried it using X-Lite, VoIP phones and ATAs in a number of
combinations. The problem is across the board.  X-Lite sometimes
automatically disconnects after 32 seconds.

Does anyone have any suggestions on where to look? ANY help appreciated. 
This has been going on for weeks.

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