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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@9643 f38db490-d61c-443f-a65b-d21fe96a405b
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0, this fixes the tv_fix warnings, because we use ast_read to transmit frames to asterisk in jitterbuffer mode, instead of queueing the audio data with ast_queue_frame.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@9186 f38db490-d61c-443f-a65b-d21fe96a405b
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* 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
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all: chan_misdn supports now the mISDN mqueue tree (smp,preemptible,gcc-4 aware\!). Additionally there are some code optimizations, new facility management (Calldeflect works for now).
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@7413 f38db490-d61c-443f-a65b-d21fe96a405b
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f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@7105 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@7063 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@6979 f38db490-d61c-443f-a65b-d21fe96a405b
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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@6910 f38db490-d61c-443f-a65b-d21fe96a405b
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