SIP no sound - Multi pfsense boxes and Asterisk



  • We have recently configured the following:

    where we separated voice and other data traffic.

    All firewalls and the router are pfsense.

    the Asterisk server is located at 192.168.1.0 network, where as both 192.168.1.0 and 10.10.1.0 networks have respective IP phones.

    We separated voice and data by deploying a pfsense router (configured to route OPT1 and LAN)

    We are able to get IP phone running SIP protocol register with Asterisk, able to dial, but was not able to get the voice working. IP phone calls another IP phone rings, and responded to off hook, but no voice.

    Anyone out there able to help?

    Rgds,
    chiacy



  • What are your firewall policies? Which ports are open?



  • @Monoecus:

    What are your firewall policies? Which ports are open?

    At the moment we allow everything to pass thru



  • Are you sure that you have the right NAT policies for the necessary incoming ports? You should allow the ports 5060 and 7070:7079 from the WAN to your SIP interface.



  • On the router, we have tried allowing wan to pass thru SIP and RTP ports. SIP is working but no sound.

    I was suspecting whether do we need source natting on LAN side.


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