i have “one way audio” problems with asterisk.
the sip phones are all in the same subnet with the asterisk-server.
asterisk 1.6 server is connected to ISDN via Dahdi with a digium b410p.
i can hear the sip-phone on my cell when calling from outside.
the sip-phone can’t hear me. there is no firewall or nat at all.
where would i start debugging?
PSTN -> ISDN -> b410p -> dahdi -> asterisk -> sip-phone