* removed the state handling from release_chan, and simplified the ast_hangup/ast_queue_hangup stuff
* added pp_l2_check option, for pp lines where the pbx does not initially gets the L2 up
* simplified and fixed a bug in the pid generation code
* fixed a bug in empty_chan, which might cause segfaults and memorry corruptions
* added prepare_bc function, which is sort of the opposite of empty_bc
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@37172 f38db490-d61c-443f-a65b-d21fe96a405b
* fixed tone handling after ast_hangup was called
* optimized the tone_indication function
* removed warnings in favour of log debugs
* improved the round_robin method
* added logs for channel setting/emptying
* fixed channel forgot to set bug
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@36082 f38db490-d61c-443f-a65b-d21fe96a405b
* added early bridge-hook, so we know if we need to generate ringing or
can take it from the far end chan_misdn channel (if available)
* fixed the issue, that we may not activate the bchannel on PTMP,
when we receive ALERTING/PROCEEDING/PROGRESS, only on CONNECT. There might
be other PTMP devices and we might disturb their bchannel.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@34552 f38db490-d61c-443f-a65b-d21fe96a405b
* added more code for connected party number handling
* fixed the portinfo program, it can now be used to test mISDN again
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@17562 f38db490-d61c-443f-a65b-d21fe96a405b
* added statefullness for bchannel activation/deactivation
* fixed a lot PCM bridging issues
* some debugging logs are now on a higher loglevel
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@17128 f38db490-d61c-443f-a65b-d21fe96a405b
* fixed "RETRIEVE does not work" bug
* fixed some NT Mode bugs
* removed some // comments
* added configureable jitterbuffer
* removed own tone-generator, and use asterisks instead, to support
asterisks indications
* added more support for hw-bridging, we bridge now every possible call
* fixed some hdlc mode issues, with a patch for chan_zap we can make
data calls between chan_zap and chan_misdn now
* completely reworked the config engine, works like a charm now
* fixed SetCallerPres - bug
* added Progress and Proceeding passing
* optimized Ringing Indication handling
* added full ast_send_text support (you can setup nice menus with the dialplan
now)
* added support to read /etc/misdn-init.conf to clarify the NT+PTP Problem
* we compile now channels/misdn if mISDNuser is installed systemwide
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@9114 f38db490-d61c-443f-a65b-d21fe96a405b
crichter