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Timeline

3:56 PM Changeset in sip_ua [70] by alx: Fixed bug with hangup after 1 minute on outgoing calls.

7:58 PM Changeset in sip_ua [69] by alx: Ringback tone in FXS
7:21 PM Changeset in sip_ua [68] by alx: Works with internal freeswitch
11:25 AM Changeset in sip_ua [67] by alx: Transport parameters made configurable. To: domain and From: domain …

12:12 PM Changeset in sip_ua [66] by alx: Accounts created for every endpoint (timeslot)
10:55 AM Changeset in sip_ua [65] by alx: get_route function added (not used yet)

6:22 PM Changeset in sip_ua [64] by alx: DTMF pause in FXO made configurable.
6:19 PM Changeset in sip_ua [63] by alx: DTMF duration in FXO made configurable.
6:16 PM Changeset in sip_ua [62] by alx: Dialtone timeout in FXO made configurable.
6:13 PM Changeset in sip_ua [61] by alx: Ring timeout in FXO made configurable.
6:09 PM Changeset in sip_ua [60] by alx: FXO gets config
3:17 PM Changeset in sip_ua [59] by alx: AGC control fixed
12:32 PM Changeset in sip_ua [58] by alx: Added AGC support
12:10 PM Changeset in sip_ua [57] by alx: Swapped rx-gain and tx-gain parameters. Rx direction is now IP -> TDM, …
11:11 AM Changeset in sip_ua [56] by alx: RTCP parameters logging

7:29 PM Changeset in sip_ua [55] by alx: RXGAIN and TXGAIN made configureble
6:43 PM Changeset in sip_ua [54] by alx: Deleted unused variables
6:41 PM Changeset in sip_ua [53] by alx: Line coding (a-law / u-law) is now configurable
5:55 PM Changeset in sip_ua [52] by alx: Setting real RTP payload types for DTMF events, setting audio RTP …
12:01 PM Changeset in sip_ua [51] by alx: DTMF payload type configuretion added (always 96)

3:52 PM Changeset in sip_ua [50] by alx: max-digits and dial-regexp implemented for FXS.
11:31 AM Changeset in sip_ua [49] by alx: Ring times, dialtone timeout, inter-digit timeout, call timeout made …
10:42 AM Changeset in sip_ua [48] by alx: FXO partially works

7:12 PM Changeset in sip_ua [47] by alx: Start/stop RTP stream from endpoint classes.
3:44 PM Changeset in sip_ua [46] by alx: FXS endpoints are creating according to configfile.
11:34 AM Changeset in sip_ua [45] by alx: fxo module added
11:32 AM Changeset in sip_ua [44] by alx
11:16 AM Changeset in sip_ua [43] by alx: Deleted dead code
10:47 AM Changeset in sip_ua [42] by alx: sendCID added

7:22 PM Changeset in sip_ua [41] by alx: Added parsing local URI and searching endpoint by call_id.
12:32 PM Changeset in sip_ua [40] by alx: Codecs priorities are set from config file.
11:16 AM Changeset in sip_ua [39] by alx: Added rudeconfig library

7:44 PM Changeset in sip_ua [27] by alx: Signaling is now processing by channel objects.
4:46 PM Changeset in sip_ua [26] by alx: Deleted unused members
4:40 PM Changeset in sip_ua [25] by alx: Deleted module threadmutex. Now using Mutex and Interrupt classes.
4:13 PM Changeset in sip_ua [24] by alx
4:09 PM Changeset in sip_ua [23] by alx: Implemented Interrupt and Mutex classes
2:55 AM Changeset in sip_ua [22] by alx: All modules converted to c++.
1:24 AM Changeset in sip_ua [21] by alx: fxs and comcerto modules converted to c++
12:55 AM Changeset in sip_ua [20] by alx: Dep files generation added. sip_ua and log converted to c++.

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