Commit Graph
4925 Commits
Author SHA1 Message Date
Brian West ddb345636a FS-3220: more than just typos 2011-04-01 17:38:58 -05:00
Anthony Minessale 8c5586b2bc add option for from-domain to be set to auto-aleg in gateway config 2011-04-01 14:22:43 -05:00
Anthony Minessale e177d377aa FS-3214 try this 2011-04-01 13:20:35 -05:00
Anthony Minessale 7c143da409 FS-3214 try this patch 2011-03-31 18:17:52 -05:00
Anthony Minessale 31273b428d pass failure across in T.38 passthru mode 2011-03-30 11:35:19 -05:00
Anthony Minessale 8312d74121 FS-2819 --comment-only please try this patch 2011-03-30 11:26:19 -05:00
Anthony Minessale 7dcbe7bda6 FS-3189 ok, patch added, but have you discovered why you trigger this because its not typical and is the sign of a problem I would assume? 2011-03-29 21:05:04 -05:00
Anthony Minessale 7e52acf8ea reduce flags to buy time. Solaris thinks enum should be int32 not uint32 and cries about overflow in enum shifted by 31 2011-03-28 22:18:47 -05:00
Anthony Minessale 9d8e54b500 FS-2751 --comment-only Please try latest patch with profile param NDLB-force-rport set to server-only or disabled. The parameter you were hacking on only applies to inbound calls not outbound calls. Its confusing but in in sip lingo client and server are outbound and inbound call direction respectively. 2011-03-28 18:31:46 -05:00
Jeff Lenk c735e28a55 FS-3190 2011-03-28 10:27:06 -05:00
Brian West 4827374574 FS-3192 FS-3191 2011-03-27 22:35:26 -05:00
Anthony Minessale 7556ec57e9 FS-3187 2011-03-25 16:35:30 -05:00
Anthony Minessale 3e4957c0b3 revert 4f6d888152 2011-03-25 16:30:16 -05:00
Marc Olivier Chouinard 81bfe43589 mod_sofia: Correct a problem where restarting profile would cause some profile hash entry to remain. 2011-03-25 15:50:52 -04:00
Michael Jerris 294b077977 FS-3182: fix mod_dingaling/iksemel/gnutls link error when using newer autotools 2011-03-25 12:17:25 -05:00
Brian West 4f6d888152 Here try this 2011-03-24 21:29:55 -05:00
Anthony Minessale e657e32fca FS-3172 this also fixes the incorrect usage of L16 on payload 10 which may or may not break interop with other sip devices if we do it right. also added rtp_disable_byteswap variable that can be set to false to disable byteswap when a device is encountered that is incompat (inluding all precious version of FS up till now) 2011-03-21 14:31:16 -05:00
Moises Silva 07d574a662 mod_portaudio: use default global configuration when configuring streams
add tons of comments to default portaudio.conf.xml for streams and endpoints
2011-03-20 21:45:51 -04:00
Moises Silva e335a876cf mod_portaudio: fix endpoint reads 2011-03-20 03:11:13 -04:00
Moises Silva dbe4a4850a mod_portaudio: do not destroy codec and timers if there is a call in progress! 2011-03-20 02:36:46 -04:00
Moises Silva e7a58ab233 mod_portaudio: use the read timer for endpoints 2011-03-20 02:04:19 -04:00
Moises Silva 43551c6503 mod_portaudio: do not set the global codec for endpoints 2011-03-20 01:57:35 -04:00
Moises Silva 667507bda9 mod_portaudio: release the endpoint on hangup 2011-03-20 01:33:08 -04:00
Moises Silva dc98b03b4c mod_portaudio: set read/write codec to L16 2011-03-20 01:16:55 -04:00
Moises Silva 3814eb13df mod_portaudio: initialize read/write endpoint timers per call
fix pablio multiplexing
2011-03-19 23:43:40 -04:00
Moises Silva 739ff9d35a mod_portaudio: implement endpoint writes and multiplex read/writes in pablio streams 2011-03-19 20:09:18 -04:00
Moises Silva fbce9061a3 mod_portaudio: implement endpoint reads 2011-03-19 19:55:12 -04:00
Moises Silva 877b4cf53b mod_portaudio: create the actual shared stream 2011-03-19 19:23:11 -04:00
Moises Silva e4b24e841c mod_portaudio: XML parsing of endpoints 2011-03-19 16:01:11 -04:00
Moises Silva 5e4911ff25 mod_portaudio: added XML parsing and CLI commands for configuration of streams 2011-03-19 15:06:43 -04:00
Moises Silva b05965c88d mod_portaudio: update to support multiple io buffers 2011-03-19 13:24:02 -04:00
Michael Jerris c885d09f23 don't strcasecmp on null string 2011-03-19 10:14:50 -04:00
Anthony Minessale db7933e72b jitter buffer sanity checks 2011-03-17 22:29:16 -05:00
Moises Silva df43e51ca5 initial reworking for mod_portaudio multiple enpoint support 2011-03-17 21:46:52 -04:00
Anthony Minessale 4c435ec530 change text of error message to be more descriptive 2011-03-14 11:54:08 -05:00
Anthony Minessale 69a5b30061 FS-3150 --comment-only this looks like an unhandled parse error, try this patch, though the call will likely fail but we can see what it doesn't like about the sdp now 2011-03-14 11:43:48 -05:00
Anthony Minessale 24a972925b pass header in X-FS headers on attended transfer CID update to indicate specific situation to flip callee/caller id when targeting a 1 legged call 2011-03-11 13:00:55 -06:00
Anthony Minessale 59f6654e96 send another presence event on calls that were cancelled from LOSE_RACE to fix winnable race in Broadsoft SCA 2011-03-10 22:02:45 -06:00
Anthony Minessale 4832d26a3a put this back to 0 2011-03-10 15:32:09 -06:00
Anthony Minessale 9e89f607c8 FS-3140 --comment-only please try this patch 2011-03-10 00:18:06 -06:00
Anthony Minessale 59da356d06 fix mistake from earlier commit and improve flow of dtmf through a bridge when timer is disabled 2011-03-09 20:06:32 -06:00
Anthony Minessale 2a35dfb51e add rtp-notimer-during-bridge (alternative to rtp-autoflush-during-bridge 2011-03-09 15:17:26 -06:00
Anthony Minessale 8727e568e8 alter implementation of renegotiate codec on hold feature to still take other sdp elements into consideration 2011-03-08 10:37:16 -06:00
Anthony Minessale bfd0ba9798 do not renegotiate codecs on hold re-invites 2011-03-07 13:02:41 -06:00
Anthony Minessale 89592a86e5 fix issue with polycom changing to 1 way audio on hold 2011-03-07 12:15:46 -06:00
Anthony Minessale 8c3651fa66 FS-640 --comment-only can you see if this patch helps, I think it should really be fixed in sofia but this shold keep it at bay 2011-03-06 14:49:39 -06:00
Anthony Minessale 8fe24a2914 FS-3121 this is less of a bug and more of a feature request but here you go, that's your quota for the month 2011-03-04 12:28:41 -06:00
Anthony Minessale 3eeb49950f FS-3117 --comment-only try this patch 2011-03-03 10:14:52 -06:00
Anthony Minessale 01073a796e add sip_jitter_buffer_during_bridge which you can set to true to keep a jitter buffer on both ends of the call when you are NormT 2011-03-02 19:11:29 -06:00
Anthony Minessale f398dc1f3a fix issue reported from mailing list and not JIRA =/ 2011-03-01 13:42:14 -06:00