b410p - dahdi - asterisk - sip - one way audio

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


Did you try leaving a voicemail? Leave a voicemail via outside, and the softphone, download the two .wav files off of the asterisk server and play them back. (Or you could have asterisk email you the voicemail)

or turn sip debugging on and see of the sip-phone is asking for RTP data to be sent to the correct IP address.

Better yet, do an echo test.