Unity 2.4.6 w/Call Manager 2.x and 3.0(8). Users Can Hear Prompts, But Can't Record Resolution - First, check the NT event log for wave record errors. If you have any, Then it's a Unity wave problem. Contact Support with the event log Messages. If not, then it could be a network and/or firewall Configuration problem. In Call Manager-land, audio streams across RTP (which is different from The skinny messages used for call setup, etc.) Thus, the routers / Firewalls / etc must allow RTP streams to go in both directions. These RTP streams typically use UDP ports 16384 and above, so the network must Allow traffic on these UDP ports to go *both* ways. While trying to record, push the "info" button twice on the IP phone. This will bring up a bunch of things, one of which is a "Tx packet Count" If that is not increasing as you speak, then you know that no RTP data is being sent. Sometimes, though, the firewalls will silently Drop packets and so just because the audio gets sent from the phone Doesn't always mean it'll make it to Unity. Which brings us to a more Definitive test: Place a phone on the same hub / switch / whatever as the Unity box and See if audio makes it both ways between that phone and the other phones. If not, then something in the middle is blocking that traffic. Also, While you're at it, see if you can record messages from that phone to Unity. (You should be able to, because this takes away all the Intermediary firewalls, etc.)
I have the same or similar problem. I am using a 2600 with an FXO port in it. I am using mgcp and I have the above mentioned command in the router. I have a def route and IP routing is turned on. If I dial into the FXO port from the PSTN I can get to voicemail and it will response to touch tones from the phone, but it does not hear me. I can dial an extension and the 7960 IP phone can receive the call. I can hear the person on the IP phone but they cannot hear me. Everything is on the same subnet, so there are no routers or firewall that I have to go through, so that is not the issue. Any ideas?
I had a similar problem with vg200's and 3620's. Here's what corrected ours: First of all, we changed the IOS versions to the newest 12.1.5T release (ie. 12.1.5T4 for 3620's and 12.1.5T7 for vg200's at the time). Then under the ethernet config we added:
h323-gateway voip bind srcaddr "ethernet ip-address" (we're obviously using h323)
then under general config term:
voice rtp send-recv
These commands along with the IOS change fixed our one-way voice.
How are you connecting to the VG200(H.323, MGCP)? What version of IOS are you running?
Also, check the configuration of the VG200, if you see "no ip routing", then this is your problem. Although the VG200 is not really 'routing' anything, 'ip routing' needs to be enabled so that the RTP stream can be fast-switched. If it is disabled, one-way audio will result.
If you are using MGCP, ensure you have command below configured:
mgcp dtmf-relay codec all mode out-of-band
If you are using H.323, configure (under dial-peer voice xx voip):
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