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[cursor at telecomabmex...]

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).


--
Telecomunicaciones Abiertas de México S.A. de C.V.
Carlos Chávez
+52 (55)9116-91161


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