PJSIP, DAHDI And Fanvil Phones

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I am having a problem with Fanvil phones (X3) when they make a call through DAHDI. Pure SIP calls flow normally but when a call goes through a DANDHI interface to the PSTN we only get one way audio. This is Asterisk 13.10.0 (bundled pjsip) and Dahdi 2.11.1 with an Openvox A400 card (4 port FXO). We also have Aastra phones and those do not have any problem making callsto the PSTN. All phones are on the internal network and there is no NAT. If I configure a SIP trunk to PSTN audio works both ways, only when going through dahdi do we lose audio.

I have never used Fanvil before today so I really do not know their best configuration settings for Asterisk. Has anyone experienced this problem with Fanvil phones? Any recommendations? A SIP debug show proper invites and the correct IP for both phone and Asterisk, RTP flows both ways between Asterisk and the phone but only outgoing audio (from phone) is heard and there is no incoming (from pstn).

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