VoIP issue - no audio

Hello,

We are attempting to setup 3 sites with a Panasonic TDE100 PBX and Panasonic IP phones.

There are 3 locations, all with static public IPs:
Site A
Site B
Site C

Site A is where the PBX is hosted. PBX is configured with an internal IP (192.168.1.x subnet). PBX is natted to an available IP at Site A for all network traffic (for example, let's say 10.1.1.1) .

Before any Cisco firewalls were implemented, the PBX sat with a public IP configured on it directly (10.1.1.1), and all 3 sites were able to make calls just fine. The issues began when we implemented Cisco firewalls, internal IPs, and started natting stuff.

Site A -> Site B is connected via an IPSec tunnel with Cisco ASA5505's on each end.

Phones at Site B have a 192.168.2.x subnet, and can see the PBX at Site A (via the 192.168.1.x IP over the tunnel) and calls work just fine, both internal extensions and outbound (xxx-xxx-xxxx numbers).

Site C does not have an ASA5505 (international location, ASA not possible at this time). However, the IP phones (same model as Site A and B) have public IP's configured directly on them. Phones at Site C try to connect to PBX at 10.1.1.1 since no VPN tunnel exists.

This setup worked fine before the Cisco's were installed. However, now, Site C phones get dial tone, but once a call is made, there is no audio both ways. Inbound and outbound calls same issue.

Cisco TAC attempted to work on packet capture, and sees all MGCP traffic passing through the ASA to 192.168.1.x PBX IP just fine.

Any ideas?
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SpejAsked:
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PhonebuffCommented:
You now have NAT Traversal in the picture when you are not going through the VPN..

Both the Phones & The IPBX need to be NAT Aware for this to work.

Google MGCP and NAT
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SpejAuthor Commented:
Thanks for pointing me in the right direction.

So it is to my understanding that the internal IP of the PBX in the SIP message is still "192.168.1.x" because the NAT on the ASA can't change that. So the phones at Site C reply back to the IP in the SIP message, which is "192.168.1.x" and it gets dropped once it reaches the WAN interface of the ASA at Site A.

The PBX has NAT Traversal as OFF. I will change that to FIXED IP.

Also, some documentation shows that we require a STUN server to change the IPs in the SIP messages, so that the SIP message sends out the IP of the WAN side of the ASA instead of the local IP of the PBX. Am I following all this correctly?
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PhonebuffCommented:
Yes,  To a Point -- In most IPBXs you set NAT to Yes, but also need to define the "External" (Public Facing IP) and internal (Local) IP Subnets, then since you have a mix you need to set NAT YES/NO on each device.  

I have no idea how you might do this on your Panasonic as I have not worked on any IP aware Panasonic PBXs to date.
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PhonebuffCommented:
PS: To use a STUN server the device has be enabled for that type connection.
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