1190 Commits

Author SHA1 Message Date
Anthony Minessale
e5f31310db FS-7911 #resolve 2015-09-24 16:00:39 -05:00
Anthony Minessale
43ef01fbbe correct version of proposed patch 2015-09-23 11:58:57 -05:00
Anthony Minessale
f8b19b7485 FS-8190 #resolve [When using nixevent, freeswitch stops sending us certain custom event that were NOT part of the nixevent command] 2015-09-21 18:00:32 -05:00
Michael Jerris
5509a62706 FS-8042, FS-8182: add ping time (in ms) to sip_registrations table, displays as part of the show commands that show registration details, add force_ping=true user var to force options ping on individual registered endpoints 2015-09-21 12:13:28 -05:00
Anthony Minessale
882e6feaf2 FS-6833 FS-6834 add support for X-headers in this 3p mode 2015-09-16 16:12:16 -05:00
Anthony Minessale
eea76c8856 FS-8130
Port video buffer to also support audio and remove original STFU jitter buffer
Add some more resilience to video packet loss
Add codec control mechanism for both call-specific debug and codec/call specfic params
Make opus function better in packet loss and latent situations
Use new codec control prams to make JB lookahead FEC optionally enabled or disabled mid-call
Add Param to allow JB lookahead to be enabled.
2015-09-14 13:30:08 -05:00
Anthony Minessale
2feae3fc69 FS-6833 #comment please test this branch 2015-09-01 16:31:23 -05:00
Mike Jerris
5c59a0159d FS-7966: fix more msvc 2015 warnings. 2015-08-31 17:08:52 -04:00
Brian West
fb383f247b FS-8037 #resolve [zrtp-passthru shouldn't activate unless the zrtp-hash is in the SDP.] 2015-08-25 11:44:05 -05:00
Michael Jerris
58f1272490 FS-7955: [mod_sofia] fix crash caused by invalid contact when using event to send a notify message 2015-08-14 12:51:12 -05:00
karl anderson
46d98d4a19 FS-7759 #resolve added a channel var to suppress setting the completed elsewhere cause 2015-07-02 17:02:47 +01:00
Mike Jerris
40254d322e Merge pull request #245 in FS/freeswitch from ~SAFAROV/freeswitch-mod-radius-cdr_improvement:FS-7311 to master
* commit 'd5cc4a1d87cee1c56b54403affd23feb86cead80':
  FS-7311: Updating display name is disabled when caller_id equal "_undef_"
2015-06-05 14:18:37 -05:00
Sergey Safarov
d5cc4a1d87 FS-7311: Updating display name is disabled when caller_id equal "_undef_" 2015-06-05 21:36:10 +03:00
Anthony Minessale
c9065a85b6 FS-7602 add some of 3b2d00f3e65061393da10a4ba286ac72cdb3c16e from verto to sip and refactor some code to keep sip working like verto 2015-06-02 21:20:03 -05:00
Mike Jerris
95c387315e Merge pull request #33 in FS/freeswitch from ~MOY/freeswitch:sip-watch-headers to master
* commit '3df55b9bb5325ed0f7273576264c5aa94a8a6810':
  Add sip_watched_headers variable to launch events when a SIP message contains a given SIP header
2015-06-02 11:19:05 -05:00
Moises Silva
3df55b9bb5 Add sip_watched_headers variable to launch events when a SIP message contains a given SIP header
FS-6801 #resolve
2015-06-02 00:47:18 -04:00
Anthony Minessale
bc152ed9d8 FS-7500: set 500ms min on retransmit of outdated xml based intraframe request that EVERYTHING still seems to use 2015-05-28 12:47:31 -05:00
Anthony Minessale
a08a89af3d FS-7500: re-enable sip info video refresh 2015-05-28 12:47:30 -05:00
Anthony Minessale
02cac73d37 FS-7499 FS-7513 try to avoid storm of refreshes in heavy usage 2015-05-28 12:47:29 -05:00
Brian West
379950f523 FS-7500: video introp tweaks 2015-05-28 12:47:15 -05:00
Anthony Minessale
d8241a12ea FS-7499: comment out sip based picture update 2015-05-28 12:46:57 -05:00
Michael Jerris
a4d877c189 FS-7460: don't force ice in 3pcc-mode=proxy 2015-04-21 19:58:28 -04:00
Brian West
4ed7b4811a FS-7217: #resolve #comment use upper when you query 2015-01-30 10:53:44 -06:00
Brian West
ded05d1cc9 FS-7211 #comment another exception #resolve 2015-01-28 14:16:12 -06:00
Brian West
e5a711af24 FS-7205 #comment do not url encode unless an at sign is in the uri #resolve 2015-01-27 14:35:18 -06:00
Anthony Minessale
f795acbff2 FS-7193 #resolve 2015-01-26 17:02:03 -06:00
Anthony Minessale
76370f4d17 auto urlencode user portion of sip uri 2015-01-23 21:06:02 -06:00
Jon Bergli Heier
165f54216c mod_sofia: Set sip_to_tag on ringing indication for inbound channels.
When bridging a call, the to-tag used in the outgoing 180 Ringing
message for the inbound channel is unavailable until the channel has
been answered. For the outgoing channel this value is already available
through the sip_to_tag variable via the event socket.

This is solved this by setting sip_to_tag to the local leg's tag when
receiving a ringing indication for inbound channels. This will also make
the variable available in the CHANNEL_PROGRESS event through event
socket.

FS-7137 #resolve
2015-01-06 17:20:22 +01:00
Michael Jerris
21458f85cc FS-7062: [mod_sofia] on redirect, when uri are passed in without <> with multiple uris, automatically add the q= header param in decending order. This should make 300 Multiple Choices work well with devices that require the q param. If you would like to specify explicit q-values, please use the syntax of redirect where you specify the entire header using the <> 2014-12-08 10:47:47 -05:00
Michael Jerris
75473a70b6 FS-6531: #resolve set to tag on uuid_phone_event notify to make grandstream happy, even tho they could have matched the dialog fine off the from tag like every other phone does. 2014-11-12 21:55:31 -06:00
Anthony Minessale
65502293cf FS-6890 #comment revert 2014-11-12 13:09:39 -06:00
Anthony Minessale
a279bf38af FS-6890 #comment please test 2014-11-11 12:56:40 -06:00
Anthony Minessale
f66f2cae8c FS-6890 #comment please test 2014-11-06 17:13:02 -06:00
Mike Jerris
78cab12dd2 Merge pull request #48 in FS/freeswitch from ~ANTONIO/freeswitch-fs-6809:master to master
* commit '69d5cda6d67074d6e5c1b7038b4dd7cab0adf60f':
  resolve FS-6809
2014-11-05 16:05:00 -06:00
Anthony Minessale
a4971693d3 FS-6890 #comment please test 2014-11-05 11:35:16 -06:00
Anthony Minessale
52ae551d1a FS-6954 #resolve #comment technically the new way is more correct but there is no hope for making fax endpoints follow a real spec. This should take care of it. 2014-10-30 10:15:10 -05:00
Brian West
3b9f0c32e6 FS-6927 #comment allow sub millisecond resolution for option ping times 2014-10-29 16:01:28 -05:00
Anthony Minessale
443ab8a8db FS-5949 #resolve 2014-10-28 13:38:06 -05:00
Brian West
15e9e68064 FS-6927 #resolve #comment This display option ping times in the gateway status on sofia status gateways or individual gateway status output 2014-10-16 17:03:37 -05:00
Anthony Minessale
e4e9b1b9f9 have resume media on hold not send invite back out at the holder but rather enable media in the 200ok 2014-10-10 16:09:43 -05:00
Mike Jerris
34bc98cafa Merge pull request #47 in FS/freeswitch from ~FLAVIO/freeswitch-fs-5106:master to master
* commit '56535519043201c723467c66c772d7519a2b6f62':
  FS-5106 fire an event when a sip client doesn't respond to option-ping
2014-10-07 14:06:34 -05:00
Anthony Minessale
2051a86df2 FS-6889 #resolve 2014-10-07 13:47:44 -05:00
Mike Jerris
d4929443f9 Merge pull request #59 in FS/freeswitch from ~SJTHOMASON/freeswitch:FS-5868 to master
* commit '747322dcc6f4db1bffc985c9bcff0bd32a2682a9':
  Remove Contact header from BYE and CANCEL requests.
2014-10-07 11:47:40 -05:00
Anthony Minessale
bde2e2da51 FS-6889 #resolve 2014-10-03 11:34:42 -05:00
Spencer Thomason
747322dcc6 Remove Contact header from BYE and CANCEL requests.
Per rfc3261 the Contact header is not applicable and MUST not appear in
the request.

FS-5868 #resolve
2014-10-02 12:24:46 -07:00
Flavio Grossi
5653551904 FS-5106 fire an event when a sip client doesn't respond to option-ping
When all-reg-options-ping is enabled, this adds a new custom event to mod_sofia
(sofia::sip_user_state), which is fired when a client stops responding to such
ping packets (or when it is reachable again).

Add two needed new columns to the sip_registrations table:
  - ping_status, which is "Reachable" or "Unreachable" depending on the client
    status;
  - ping_count, which tracks the number of ping responses received and is used
    to provide some kind of hysteresis to avoid firing the event in case of
    transitory network failures.

Then ping_count is checked against two threshold values, sip-user-ping-min
and sip-user-ping-max in a similar fashion as the ping-{max,min} options for
the gateways. These two values are configurable in the profile's xml
configuration file.

Also, if unregister-on-options-fail is enabled, the client is unregistered
based on the number of OPTIONS failure which is also checked against the
sip-user-ping-{min,max} values.
2014-10-02 12:34:47 +02:00
Antonio
69d5cda6d6 resolve FS-6809 2014-09-09 15:33:19 +02:00
Anthony Minessale
a73583b5f3 FS-6806 #resolve 2014-09-09 00:09:31 +05:00
Travis Cross
5c29d8d4fa Show gateway uptime in seconds
In `sofia status gateway ...` let's show the uptime in seconds rather
than in microseconds.  We'll output the uptime in microseconds in
`xmlstatus` and we'll label it as such.
2014-09-04 05:39:26 +00:00
Steven Ayre
93bd5833c2 Add uptime property to mod_sofia gateways
The 'UP' status indicates a gateway is online as determined by
registration and/or SIP OPTIONS pinging.

The time the gateway has been in the 'UP' status is recorded,
and can be monitored using 'sofia status' and 'sofia xmlstatus'.

This can be used to detect and graph when there are outages.

ref: FS-6772

Reviewed-by: Travis Cross <tc@traviscross.com>
2014-09-04 03:43:36 +00:00