One way audio on SIP Trunks?

[BlackSunshine]

I have set up SIP trunks with provider.  I am getting one way audio.  When I call out via SIP Trunk the person outside the MITEL can hear me however I can't hear them on mitel also I'm not getting ringback while call is ringing.  I am not using MBG.  My IT Dept is NAT'ing via a Palo Alto firewall.  It seems like it is a maybe a RTP issue.  Just curious could I see one way audio in a SIP Trace, if so what would it look like.  Any help appreciated

[x-man]

It'll either be SIP-ALG running on the firewall; needs turning off  or you don't have all the required ports open on said firewall.

[ralph]

I think the hardest thing for people to wrap their heads around is the fact that the 3300 is only used for call setup.
Once the endpoint answers the call the PBX drops out of it and the RTP will go from end point to end point - or in your case firewall to end point.   The firewall rules have to allow for communication to the entire subnet, not just the PBX.

Ralph

[BlackSunshine]

Still can't get but one way audio.  Do I need to open RTP ports 50000-50255 between Mitel and SIP provider?

[x-man]

Depends on what your provider needs.

[BlackSunshine]

We have opened every ports between provider and 3300.   Still getting one way audio.  When call is up we see no traffic to from provider flowing through firewall.

[VinceWhirlwind]

Check your signalling - what IP address is your call setup telling the remote set to use for RTP?

[sunspark]

No gateway IP address
programmed on E2T.

[billbry66]

"We have opened every ports between provider and 3300.   Still getting one way audio.  When call is up we see no traffic to from provider flowing through firewall. "

you need to allow the ports not just to the 3300 but to the whole voice subnet

voice will stream from handset to sip provider ( unless you are using a proxy)

all handsets MUST have a gateway address that allows this as well

i way or no speech is almost always ports or routing