I’m headbanging on this from a couple of days, begging here for some help 🙂
I’m configuring tls on asterisk for the first time to experiment with an open (public) service idea about having asterisk accepting any sip user (with the sip.conf option ‘autocreatepeer=yes’)
and call each other on the same server and perhaps to other asterisk servers with the same configuration. Something like ‘skype for poors’ for the ‘average joe’.
I’m using asterisk 10.7.0 on a debian squeeze dedicated server (with public ip).
I’ve followed this tutorial: https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial and got no errors but when dialing a test context:
exten => _X.,1,Answer exten => _X.,n,playback(tt-weasels)
exten => _X.,n,echo exten => _X.,n,Hangup()
i get no audio.
On the client side, I’ve tried with many softphones (bink, jitsi, microsip, phonerlite) on both windows and linux, on two different computers but same result.
I’ve also enabled srtp, checked the sip debug trace, recompiled libsrtp from sources, tried different combination of parameters in sip.conf, enabled and disabled some port forwardings on the client’s router but same result: all looks ok, but i get no audio.
If not using tls (but the usual udp and rtp), audio works full-duplex 🙂
Anyone had a similar problem ?
Any hints ?
Let me know if i can provide more info.
Thanks for supporting, regards and have a nice day, Mike