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  1. 09 9月, 2006 4 次提交
    • Anthony Minessale's avatar
      add uuid to originate · 7142a03c
      由 Anthony Minessale 提交于 
      git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2591 d0543943-73ff-0310-b7d9-9358b9ac24b2
      7142a03c
    • Anthony Minessale's avatar
      dox · 95249aff
      由 Anthony Minessale 提交于 
      git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2590 d0543943-73ff-0310-b7d9-9358b9ac24b2
      95249aff
    • Anthony Minessale's avatar
      dox · 5e575f14
      由 Anthony Minessale 提交于 
      git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2589 d0543943-73ff-0310-b7d9-9358b9ac24b2
      5e575f14
    • Anthony Minessale's avatar
      Adding bugs to the core · ae9d56e2
      由 Anthony Minessale 提交于 
      This is the primary commit to add bugs to the core (media bugs that is)
Media bugs are kind of like what ChanSpy is in Asterisk only cooler (I wrote ChanSpy too so I can say that)

Here is an example of using them to record a call by the higher level switch_ivr functionality passed
up to the dialplan via mod_playback.

The call will be recorded while the some.wav plays then stop for the rest of the call (when some_other.wav plays)

The bugs may have bugs since this is 1 day's work so happy hunting ......

<extension name="42">
  <condition field="destination_number" expression="^42$">
   <action application="set" data="RECORD_TITLE=recording test"/>
   <action application="set" data="RECORD_ARTIST=FreeSWITCH"/>
    <action application="record_session" data="/tmp/rtest.wav"/>
    <action application="playback" data="/tmp/some.wav"/>
    <action application="stop_record_session" data="/tmp/rtest.wav"/>
    <action application="playback" data="/tmp/some_other.wav"/>
  </condition>
</extension>



git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2588 d0543943-73ff-0310-b7d9-9358b9ac24b2
      ae9d56e2
  3. 08 9月, 2006 16 次提交
  5. 07 9月, 2006 12 次提交
  7. 06 9月, 2006 5 次提交
  9. 05 9月, 2006 3 次提交
    • Brian West's avatar
      RTP profile for audio and video conferences with minimal control. · b7ded398
      由 Brian West 提交于 
      This memorandum is a revision of RFC 1890 in preparation for advancement
from Proposed Standard to Draft Standard status. Readers are encouraged to
use the PostScript form of this draft to see where changes from RFC 1890 are
marked by change bars.

"G722 is specified in ITU-T Recommendation G.722, "7 kHz audio-coding within
64 kbit/s". The G.722 encoder produces a stream of octets, each of which
SHALL be octet-aligned in an RTP packet. The first bit transmitted in the
G.722 octet, which is the most significant bit of the higher sub-band
sample, SHALL correspond to the most significant bit of the octet in the RTP
packet.

Even though the actual sampling rate for G.722 audio is 16000 Hz, the RTP
clock rate for the G722 payload format is 8000 Hz because that value was
erroneously assigned in RFC 1890 and must remain unchanged for backward
compatibility. The octet rate or sample-pair rate is 8000 Hz."




git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2523 d0543943-73ff-0310-b7d9-9358b9ac24b2
      b7ded398
    • Brian West's avatar
      cleanups · 7c11839c
      由 Brian West 提交于 
      git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2522 d0543943-73ff-0310-b7d9-9358b9ac24b2
      7c11839c
    • Brian West's avatar
      Thanks stkn · 55156a46
      由 Brian West 提交于 
      git-svn-id: http://svn.freeswitch.org/svn/freeswitch/trunk@2521 d0543943-73ff-0310-b7d9-9358b9ac24b2
      55156a46