Timeline



Nov 30, 2011:

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

Nov 29, 2011:

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 …

Nov 28, 2011:

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)

Nov 25, 2011:

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

Nov 24, 2011:

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)

Nov 23, 2011:

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

Nov 22, 2011:

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

Nov 21, 2011:

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

Nov 19, 2011:

12:27 PM Changeset in sip_ua [38] by alx
RudeConfig library added

Nov 17, 2011:

8:44 PM Changeset in sip_ua [37] by alx
Outgoung calls now work
7:10 PM Changeset in sip_ua [36] by alx
Cancel incoming call bug fixed.

Nov 16, 2011:

8:33 PM Changeset in sip_ua [35] by alx
Fixed codec parameters check
8:08 PM Changeset in sip_ua [34] by alx
Dummy codec module added

Nov 15, 2011:

8:09 PM Changeset in sip_ua [33] by alx
call_id bug fixed
7:41 PM Changeset in sip_ua [32] by alx
Added dialing support

Nov 7, 2011:

7:56 PM Changeset in sip_ua [31] by alx
Dialtone timeout added
7:25 PM Changeset in sip_ua [30] by alx
Tone generation made

Nov 6, 2011:

12:22 AM Changeset in sip_ua [29] by alx
12:10 AM Changeset in sip_ua [28] by alx
abstract_channel.h added
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