Commit Graph

891 Commits

Author SHA1 Message Date
Anthony Minessale 73279f01bf FS-3166 --resolve 2011-04-22 16:43:29 -05:00
Anthony Minessale 4b706dac51 FS-3227 --resolve this looks like sane changes. My only complaint was the formatting. Watch for whitespace indentation by looking at the code in emacs or vi where it should be tabed properly. 2011-04-04 11:55:05 -05:00
Anthony Minessale fda2283bbd auto-aleg-full and auto-aleg-domain for from_domain field in gateway 2011-04-03 12:03:29 -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 7556ec57e9 FS-3187 2011-03-25 16:35:30 -05:00
Anthony Minessale 3e4957c0b3 revert 4f6d888152 2011-03-25 16:30:16 -05:00
Brian West 4f6d888152 Here try this 2011-03-24 21:29:55 -05:00
Anthony Minessale db7933e72b jitter buffer sanity checks 2011-03-17 22:29:16 -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 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 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 d59d41d7b4 add param to jb to try to recapture latency (disabled by default) 2011-02-25 11:59:45 -06:00
Anthony Minessale 0dcdd78cb5 FS-3054 --comment-only try latest commit, I can guess what probably causes the seg based on my last patch 2011-02-22 17:22:01 -06:00
Anthony Minessale e7acd4d138 FS-3054 re-open if this does not fix it. 2011-02-21 20:17:58 -06:00
Anthony Minessale 4e60f14a4d FS-3072 2011-02-21 11:02:42 -06:00
Anthony Minessale a2c0da53f3 add centralized registration db to core db and use it from mod_sofia 2011-02-11 23:10:12 -06:00
Anthony Minessale 88d410d314 fix uuid_jitterbuffer edge case debugging a non-existant jb causing a seg 2011-02-11 20:15:06 -06:00
Anthony Minessale 2401fec54b minor regression from 4ae8282e6c (sofia_contact with no args from cli caused seg) 2011-02-08 13:01:42 -06:00
Anthony Minessale f0a31e1bff default to 10 2011-02-07 14:35:56 -06:00
Anthony Minessale 4ae8282e6c fix possible bad pointer in global vars (please test) 2011-02-02 15:43:26 -06:00
Anthony Minessale 52bf0423e2 try to fix SOA problem with early and answer audio with dissimilar sdp 2011-02-01 11:23:32 -06:00
Daniel Swarbrick 0e0431ecc6 update mod_sofia management interface OID 2011-01-26 20:07:33 +01:00
Travis Cross 7eceff48a2 update sofia usage string for flush_inbound_reg 2011-01-16 22:33:50 +00:00
Anthony Minessale 7b01cbbca4 add send-presence-on-register (true|false|first-only) param to sofia and api command sofia global debug [presence|sla|none] 2011-01-14 13:58:21 -06:00
Anthony Minessale 54de293b05 fix seg related to ptime mismatch + CNG + PLC (if you ever get purple ptime mismatch warnings you want this patch) 2011-01-12 16:05:08 -06:00
Anthony Minessale 181b543b0c add auto-jitterbuffer-msec param and auto-disable the jitterbuffer when briding to another channel who also has a jitterbuffer so both legs will disable during a bridge 2011-01-05 16:25:14 -06:00
Brian West 3734f4cd44 bump copyright date and fix some email and typos from diego. 2011-01-05 10:09:04 -06:00
Anthony Minessale 0920645d1f update 2010-12-29 15:04:19 -06:00
Anthony Minessale 650393fb90 add recovery_refresh app and api and use it in mod_conference to send a message to the channel telling it to sync its recovery snapshot 2010-12-29 13:15:14 -06:00
Anthony Minessale 81608da006 refactor sofia_contact to try the profile_name first then the domain to resolve the profile then fall back to querying every profile to reduce confusion with multi-homers (d'oh) also special profile name * will force a search-all situation 2010-12-29 12:28:12 -06:00
Anthony Minessale 668763f490 prevent race on codec change mid-call 2010-12-17 17:27:23 -06:00
Anthony Minessale 1e0df408cf oops 2010-12-17 15:28:19 -06:00
Anthony Minessale 8f452bc519 cid logic changes for calle[re] 2010-12-17 14:35:53 -06:00
Anthony Minessale 93cc3dc556 normalize tests for outbound channels to use switch_channel_direction instead of testing for CF_OUTBOUND 2010-12-15 20:59:42 -06:00
Anthony Minessale e9958c5b0c more jb work, add debug command and logging (sorry jlenk if this breaks win32) 2010-12-14 23:46:26 -06:00
Anthony Minessale 321013efe7 have mod_sofia always elect to be the session refresher so we know it will work, also make the session-expires set to 0 imply 100% disabled session timers 2010-12-13 14:02:46 -06:00
Anthony Minessale d547096164 dramatic jitterbuffer changes 2010-12-10 17:47:46 -06:00
Brian West a669f76f78 Fix issue when fs_path is used so we pick the correct media IP in our outbound invite this was soemthing that wouldn't work correctly over ATT on the iphone. 2010-11-30 17:43:13 -06:00
Anthony Minessale 143949941c add presence-probe-on-register sofia param to send a probe on register instead of presence to deal with some broken phones and add some general improvements to allow multi homed presence 2010-11-24 21:39:08 -06:00
Anthony Minessale 10119e9e88 FS-2824 2010-11-08 10:13:35 -06:00
Michael Jerris 40ac860aaa fix missing name and potential segfault in gateway status 2010-11-05 20:24:31 -04:00
Anthony Minessale 2043d5a671 fix display of timeout 2010-11-03 12:22:01 -05:00
Anthony Minessale 97d80d924a same as last one in another place 2010-11-02 11:50:04 -05:00
Anthony Minessale 35676e7e04 parse static route in sip uri in notify by event 2010-11-01 18:47:09 -05:00
Anthony Minessale e10bc0a965 allow {dtmf_type=none} to work in oubound dial strings 2010-10-26 15:43:14 -05:00
Anthony Minessale dfa78985b4 Change codec behaviour
channel_variable: sdp_m_per_ptime
Adds a new m= line for each distinct ptime in codec list.

When this variable is not set:
	When mixing codecs with various ptime in a codec list, they will now be allowed to co-exist in the sdp but it will send no ptime attr.
		This means the ptime preferences on the offer will be ignored when mixing codecs with various ptimes.
	When receiving a codec list with no ptime attr, the ptime will be chosen from local preference instead of assuming 20ms
		This means if offer contains PCMU with not ptime and FS has PCMU@40i

Dynamic payloads will now start at 98 and increment per additional dynamic codec per call.
	So now you can add CELT@32000h,CELT@48000h and each one will be auto-assigned a dynamic paylaod type.
2010-10-13 19:28:20 -05:00
Anthony Minessale f13fa0c1a5 FS-2763 2010-10-08 15:38:01 -05:00
Anthony Minessale 8f13eb8966 FS-2762 2010-10-06 15:17:48 -05:00