I’m having the one way audio problem. I´ve read the nat issues on the net, but i don´t use nat. My configuration is not the standard: I have a Asterisk server with two NICs, one connected to the LAN and the other connected to my SIP trunk provider. This second NIC is connected directly to the ISP modem and configured to the SIP proxy from the ISP (The internet service is given by the same ISP, but with another modem connected to my router).
The SIP Trunk has this configuration:
I´ve made a route in my centos to the SipProxy host through the gateway and eth1, and this works fine.
The problem is only with the call made using this SIP trunk. And the LAN phones don´t hear the PSTN phones Audio, but the PSTN phones hear th LAN phones.
I´ve configures the “SIP Asterisk Setting” in the freepbx with:
localnetworks=200.69.XXX.XXX/255.255.255.252 (the IP given by the ISP for the sip trunk connection is this right thing to put?)
I have another IP for Internet, but I didn´t configure it, because i would no connect phones trough the Internet (only LAN/VLAN And ISP SIP Trunk)
AS I use Freepbx, the sip_nat.conf and sip_general_custom.conf are empty and if I change them, the freepbx gives me an error.
I´ve configured the same codecs ulaw, alaw and gsm in the freepbx and the softphone.
I´m using Centos 1.4, Asterisk 22.214.171.124 and freepbx 126.96.36.199
Thanks in advance,