Timeline
Nov 30, 2011:
- 3:56 PM Changeset in sip_ua [70] by
- Fixed bug with hangup after 1 minute on outgoing calls.
Nov 29, 2011:
- 7:58 PM Changeset in sip_ua [69] by
- Ringback tone in FXS
- 7:21 PM Changeset in sip_ua [68] by
- Works with internal freeswitch
- 11:25 AM Changeset in sip_ua [67] by
- Transport parameters made configurable. To: domain and From: domain …
Nov 28, 2011:
- 12:12 PM Changeset in sip_ua [66] by
- Accounts created for every endpoint (timeslot)
- 10:55 AM Changeset in sip_ua [65] by
- get_route function added (not used yet)
Nov 25, 2011:
- 6:22 PM Changeset in sip_ua [64] by
- DTMF pause in FXO made configurable.
- 6:19 PM Changeset in sip_ua [63] by
- DTMF duration in FXO made configurable.
- 6:16 PM Changeset in sip_ua [62] by
- Dialtone timeout in FXO made configurable.
- 6:13 PM Changeset in sip_ua [61] by
- Ring timeout in FXO made configurable.
- 6:09 PM Changeset in sip_ua [60] by
- FXO gets config
- 3:17 PM Changeset in sip_ua [59] by
- AGC control fixed
- 12:32 PM Changeset in sip_ua [58] by
- Added AGC support
- 12:10 PM Changeset in sip_ua [57] by
- Swapped rx-gain and tx-gain parameters. Rx direction is now IP -> TDM, …
- 11:11 AM Changeset in sip_ua [56] by
- RTCP parameters logging
Nov 24, 2011:
- 7:29 PM Changeset in sip_ua [55] by
- RXGAIN and TXGAIN made configureble
- 6:43 PM Changeset in sip_ua [54] by
- Deleted unused variables
- 6:41 PM Changeset in sip_ua [53] by
- Line coding (a-law / u-law) is now configurable
- 5:55 PM Changeset in sip_ua [52] by
- Setting real RTP payload types for DTMF events, setting audio RTP …
- 12:01 PM Changeset in sip_ua [51] by
- DTMF payload type configuretion added (always 96)
Nov 23, 2011:
- 3:52 PM Changeset in sip_ua [50] by
- max-digits and dial-regexp implemented for FXS.
- 11:31 AM Changeset in sip_ua [49] by
- Ring times, dialtone timeout, inter-digit timeout, call timeout made …
- 10:42 AM Changeset in sip_ua [48] by
- FXO partially works
Nov 22, 2011:
- 7:12 PM Changeset in sip_ua [47] by
- Start/stop RTP stream from endpoint classes.
- 3:44 PM Changeset in sip_ua [46] by
- FXS endpoints are creating according to configfile.
- 11:34 AM Changeset in sip_ua [45] by
- fxo module added
- 11:32 AM Changeset in sip_ua [44] by
- 11:16 AM Changeset in sip_ua [43] by
- Deleted dead code
- 10:47 AM Changeset in sip_ua [42] by
- sendCID added
Nov 21, 2011:
- 7:22 PM Changeset in sip_ua [41] by
- Added parsing local URI and searching endpoint by call_id.
- 12:32 PM Changeset in sip_ua [40] by
- Codecs priorities are set from config file.
- 11:16 AM Changeset in sip_ua [39] by
- Added rudeconfig library
Nov 19, 2011:
- 12:27 PM Changeset in sip_ua [38] by
- RudeConfig library added
Nov 17, 2011:
- 8:44 PM Changeset in sip_ua [37] by
- Outgoung calls now work
- 7:10 PM Changeset in sip_ua [36] by
- Cancel incoming call bug fixed.
Nov 16, 2011:
- 8:33 PM Changeset in sip_ua [35] by
- Fixed codec parameters check
- 8:08 PM Changeset in sip_ua [34] by
- Dummy codec module added
Nov 15, 2011:
- 8:09 PM Changeset in sip_ua [33] by
- call_id bug fixed
- 7:41 PM Changeset in sip_ua [32] by
- Added dialing support
Nov 7, 2011:
- 7:56 PM Changeset in sip_ua [31] by
- Dialtone timeout added
- 7:25 PM Changeset in sip_ua [30] by
- Tone generation made
Nov 6, 2011:
- 12:22 AM Changeset in sip_ua [29] by
- 12:10 AM Changeset in sip_ua [28] by
- abstract_channel.h added
Note:
See TracTimeline
for information about the timeline view.