I am running Asterisk on a machine that is also my NAT/firewall, samba server, email server, apache webserver, etc.
I found that I have to run Asterisk at realtime priority (asterisk -pg) when Asterisk is bridging calls from my internal SIP/misdn clients to my SIP-provider. Without realtime priority the sound is quite bad (I hear the other side quite clear, but they can almost not hear me).
Without running in realtime priority, everything works fine. Except for the sound quality.
When I run Asterisk at realtime priority however, I have the following problem on my mISDN connected phones (HFC-based card in NT-mode):
When I pick up the handset, I have no dialtone. I can type a number, the call is set up (e.g. I call my own cellphone, it rings and I pick it up). However I do not get any sound (both ways: no sound out of the mISDN-connected phone, also I hear nothing in my cellphone).
Receiving phone calls with Asterisk in realtime priority still works fine: The mISDN-connected phone rings, I pick it up and both sides have good sound quality.
It seems to me like a bug in mISDN, chan_misdn and/or Asterisk. Where should I post this bug? Or maybe it is not a bug? Suggestions for isolating the problem? How to make a more detailed ‘bug report’?