atcomsystems.ca/forum Forums VOIP Phones & IP PBX VoIP & Cloud Phones, Hosted PBX, General IP Phones One way audio

I've been trying for several weeks to figure out VOIP with an Iwatsu ECS 4.0 and ISA 2004. I have server publishing rules set up for TCP 50000 traffic and UDP 6004-6019 traffic. I can ring the station and get dial tone but I'm only getting one way audio. I don't hear anything on the VOIP phone, but I do on the cell phone (other end)

Is this most likely a phone config setting or a firewall or router issue?

Something is messing with the RTP stream, have you done any packet captures? Is the phone on the LAN side? Is it on a different VLAN?

The VOIP phone is at my house using the NonVPN mode option in the Iwatsu. I'm not really familiar with capturing packets. The audio is flipflopped now. I can hear on the VOIP side now so anything said into the outside line (cell phone) can be heard on the VOIP phone. It's like the voice on the VOIP isn't transmitting.

My UPD handshakes are working correctly on ISA though.

from the PBX out: are you on a dedicated circuit or is it shared? Are you passing through a Firewall or directly to the Internet? the switch port at both ends are they set for full duplex? Don't know anything about Iwatsu however it's just dial tone that needs to be routed, are you using MicroSoft ISA server as your Firewall or is ISA something else for Iwatsu?

If you place an ip phone at the office on the same switch in the same segment does it work there? (same Vlan)

if you are on a dedicated internet connection can you place an ACL that allows any any for udp traffic and on your home traffic allow any any to your ip phone. Not a best practice but if you aren't comfortable with packet sniffing this will give you a stat.

on a cisco switch if you are able to manipulate the ACL then try one where udp any any log and tcp any any log and watch the terminal to se what ports the router think you should be using.