ZyWall pbx on wlan alias no incoming

[TheCondor]

Hi all, i've a problem with my ziwall usg20w that i don't succeed to fix in this two days.... i post here, hope someone can help me.

I've a pbx sistem running on my router (provided by my ISP, so any config on this box is possible..) with ip 192.168.0.254
I configured the wan of the ziwall in pppoe (required by my ISP for get the public fixed ip assigned and all working well, on the pppoe i get the public ip and the VPN tunnel is working.
For reach the private ip of the router i create an alias interface on the WAN of the zywall, with ip 192.168.0.1.
I can reache from my lan the web gui of the router, and also i successfully register on the pbx.
The problem comes during the voip call, the audio incoming doesn't works.
This is the scheme of what i need to do.

wan pppoe (publicip)
asteriskpbx (192.168.88.10) ---> zywal (192.168.88.254) --->
wanalias (192.168.0.1) ->pbx (192.168.0.254)

I can place call from 192.168.88.10 out through 192.168.0254 with the wan_alias, they can hear me but i cannot hear them. Any advice is much appreciated, thanks in advance!

 

[alphamatter]

I'm not sure I understand why you don't have the Asterisk on a 192.168.0.10 IP address instead of behind/in front of the Zywall. If you call Station User to Station User from the Asterisk on the same Subnet, does that work?

On the PBX 192.168.0.254(What is this PBX?), if you have Users/SIP phones setup on that same Subnet, 192.168.0.x, can you call, Phone to Phone?

If so, I'd have to see the Router/Gateway settings, so as to know if you need to create Static Routes between these networks.

One way audio usually means a Router/Gateway NAT issue or Routes need to be created. The Call Setup utilized TCP (connection assured) packets, then, move to UDP (Connectionless) and there is no confirmation that the Voice Packets are allowed or being routed to the correct destination.

 

[TheCondor]

I'm not sure I understand why you don't have the Asterisk on a 192.168.0.10 IP address instead of behind/in front of the Zywall. If you call Station User to Station User from the Asterisk on the same Subnet, does that work?

On the PBX 192.168.0.254(What is this PBX?), if you have Users/SIP phones setup on that same Subnet, 192.168.0.x, can you call, Phone to Phone?

If so, I'd have to see the Router/Gateway settings, so as to know if you need to create Static Routes between these networks.

One way audio usually means a Router/Gateway NAT issue or Routes need to be created. The Call Setup utilized TCP (connection assured) packets, then, move to UDP (Connectionless) and there is no confirmation that the Voice Packets are allowed or being routed to the correct destination.

Thanks alphamatter for your super speedy answer, if i connect directly to the pbx on 192.168.0.254 all works fine so the problem is some routes missed in the zywall. I need to build a route similar to this:

from 192.168.88.4 -> pass all to -> 192.168.0.254
from 192.168.0.254 -> pass all to_ 192.168.88.4

I tried to setup a nat roule but without success, in the route policy i canno select as interface wlan0:0 (the alias)
I cannot move my asterisk on the other subnet 192.168.0.X because i've other trunk witch need the internet link

 

[alphamatter]

Mmmm... NAT Passthrough will cause some problems, sure. The only way to diagnose that is with Port Mirroring and some software to take that info and point you in the right direction, such as Wireshark.

However, I would suggest to not use an IP Masquerade, as you are now. Easiest to do is just create Static Routes, in a small network environment, this should be good enough.

Can you create a Static Route Table such as:
Zywall Route Table Entry (192.168.88.254)
192.168.0.0 --> Netmask-255.255.255.0-->Gateway-192.168.88.250(Whatever that Gateway/IP address you use to route traffic between the (2) Subnets)

On the Other Router (192.168.0.?)
192.168.88.0-->Netmask-255.255.255.0-->Gateway-192.168.0.250(Whatever that Gateway/IP address you use to route traffic between the (2) Subnets)

 

[TheCondor]

fixed in the hard way: added a second nic on the pbx and wired directly to the other pbx... thanks anyway for your kind help!

 

[sfx2000]

Sounds like just SIP is being opened up - this is why you get the rings (sip invite and response), but the RTP ports are not open - hence no audio...

Make sure the correct ports are opened up... for traditional VOIP, SIP carries the control signaling, and RTP carries the data...

SIP is default as TCP/UDP 5060, and for SIP-TLS TCP 5061.

RTP is is UDP traffic, and is typically somewhere within the range of UDP 16384 thru UDP 32768 - depending on configuration of the SIP gateway...