dc8f2b2044
FS-6833 FS-6834 found a few missing content-types in requests/resonses with sdp that were outside the norm
2015-10-12 12:38:25 -05:00
e5f31310db
FS-7911 #resolve
2015-09-24 16:00:39 -05:00
43ef01fbbe
correct version of proposed patch
2015-09-23 11:58:57 -05:00
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
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
882e6feaf2
FS-6833 FS-6834 add support for X-headers in this 3p mode
2015-09-16 16:12:16 -05:00
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
2feae3fc69
FS-6833 #comment please test this branch
2015-09-01 16:31:23 -05:00
5c59a0159d
FS-7966: fix more msvc 2015 warnings.
2015-08-31 17:08:52 -04:00
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
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
46d98d4a19
FS-7759 #resolve added a channel var to suppress setting the completed elsewhere cause
2015-07-02 17:02:47 +01:00
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
d5cc4a1d87
FS-7311: Updating display name is disabled when caller_id equal "_undef_"
2015-06-05 21:36:10 +03:00
c9065a85b6
FS-7602 add some of 3b2d00f3e6 from verto to sip and refactor some code to keep sip working like verto
2015-06-02 21:20:03 -05:00
95c387315e
Merge pull request #33 in FS/freeswitch from ~MOY/freeswitch:sip-watch-headers to master
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* 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
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
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
a08a89af3d
FS-7500: re-enable sip info video refresh
2015-05-28 12:47:30 -05:00
02cac73d37
FS-7499 FS-7513 try to avoid storm of refreshes in heavy usage
2015-05-28 12:47:29 -05:00
379950f523
FS-7500: video introp tweaks
2015-05-28 12:47:15 -05:00
d8241a12ea
FS-7499: comment out sip based picture update
2015-05-28 12:46:57 -05:00
a4d877c189
FS-7460: don't force ice in 3pcc-mode=proxy
2015-04-21 19:58:28 -04:00
4ed7b4811a
FS-7217: #resolve #comment use upper when you query
2015-01-30 10:53:44 -06:00
ded05d1cc9
FS-7211 #comment another exception #resolve
2015-01-28 14:16:12 -06:00
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
f795acbff2
FS-7193 #resolve
2015-01-26 17:02:03 -06:00
76370f4d17
auto urlencode user portion of sip uri
2015-01-23 21:06:02 -06:00
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
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
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
65502293cf
FS-6890 #comment revert
2014-11-12 13:09:39 -06:00
a279bf38af
FS-6890 #comment please test
2014-11-11 12:56:40 -06:00
f66f2cae8c
FS-6890 #comment please test
2014-11-06 17:13:02 -06:00
78cab12dd2
Merge pull request #48 in FS/freeswitch from ~ANTONIO/freeswitch-fs-6809:master to master
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* commit '69d5cda6d67074d6e5c1b7038b4dd7cab0adf60f':
resolve FS-6809
2014-11-05 16:05:00 -06:00
a4971693d3
FS-6890 #comment please test
2014-11-05 11:35:16 -06:00
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
3b9f0c32e6
FS-6927 #comment allow sub millisecond resolution for option ping times
2014-10-29 16:01:28 -05:00
443ab8a8db
FS-5949 #resolve
2014-10-28 13:38:06 -05:00
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
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
34bc98cafa
Merge pull request #47 in FS/freeswitch from ~FLAVIO/freeswitch-fs-5106:master to master
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* 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
2051a86df2
FS-6889 #resolve
2014-10-07 13:47:44 -05:00
d4929443f9
Merge pull request #59 in FS/freeswitch from ~SJTHOMASON/freeswitch:FS-5868 to master
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* commit '747322dcc6f4db1bffc985c9bcff0bd32a2682a9':
Remove Contact header from BYE and CANCEL requests.
2014-10-07 11:47:40 -05:00
bde2e2da51
FS-6889 #resolve
2014-10-03 11:34:42 -05:00
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
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
69d5cda6d6
resolve FS-6809
2014-09-09 15:33:19 +02:00
a73583b5f3
FS-6806 #resolve
2014-09-09 00:09:31 +05:00
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
Anthony Minessale