Oneway Voice

Hi All,

I’m testing free PBX these days and got stuck with outbound calling due to one way voice.

my topology is like below,

Eyebeam softphone —>FreePBX—>Gendband SBC—>mobile operator

I’ve set the SIP trunk and I’m able to call B party mobile operator but only A–>B RTP flow is ok but A party can not hear the B party.

If someone can provide a clue on this I would be really grateful.
Thanks in advance for the help.

Well, FreePBX is just a web interface for Aserisk.

Asterisk SIP and NAT is one of the most popular topics, their are thousands of posts on hundreds of message boards about how to deal with NAT and SIP.

Of course you did not tell us anything about the network topology, so I am making an assumption that this is a NAT issue because:

If you are not using NAT you can’t have one way audio since you have to have a duplex path to setup a call!

Remember SIP and RTP are two different protocols, one is signaling the other media. Just because the signaling works does not mean the media is being NAT’d properly.

Thanks a lot for your expertise advices and clue.
I’ll look around for SIP and NAT post’s to find out the solution.
If still unable to find it I’ll explain my topology to you.

Thanks again.

Follow these instructions and pay special attention to Step 5.