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I've got a specific situation where I need to add two SIP accounts to a telephone. One is accessible on the local LAN and the other is only accessible via an OpenVPN connection. The OpenVPN connection works perfectly and I'm only pushing the necessary routes for the phone to access the external SIP account.

The internal local LAN account registers fine with the SIP server (3CX) but there's no audio. Looking at the log files it appears the phone is sending the IP of the OpenVPN tunnel to the local account rather than sending the IP of the LAN interface. Excerpt from the 3CX log file here:

7-Oct-2014 13:56:01.001 NAT/ALG check:L:1927.1[Extn:114] REQUEST 'INVITE' - some of SIP/SDP headers may contain inconsistent information or modified by intermediate hop
SIP contact header is not equal to the SIP packet source(IP:port):
Contact address:
Received from :
Media session IP ('c=' attribute) is not equal to the SIP packet source(IP:port):
Media session IP:
Received from:

This seems to me to be a bug as the phone shouldn't send that IP in the SIP headers unless the traffic is exiting through the VPN interface.


Hi Paulszat,

Currently, if you enable OpenVPN feature then all traffic will be sent through the VPN tunnel. So the two account can't work at a same time.

We will submit the issue to our product department for a future improvement.

But it doesn't though. The OpenVPN client on the phone honours the routes that get pushed through from the OpenVPN server and it only routes traffic through the VPN if the destination IP is in the list of pushed routes. Traffic routing isn't the problem, the problem is with the SIP signalling. Seems like more of a bug in the existing provision rather than a feature request.....
I am having the same issue.
Any progress that has been made on that?

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