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FS-7500: set 500ms min on retransmit of outdated xml based intraframe request that EVERYTHING still seems to use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
Stan Gor
64060c7dbd
Add sofia gateway parameter "destination-prefix"
...
FS-5497 add sofia gateway parameter destination-prefix in case you need to send Invites to your provider with prefix only to this gateway
2014-08-19 11:54:09 -07:00
Anthony Minessale
3ce4ae962b
FS-6540 #comment please test this patch for the added notify functionality
2014-07-17 22:35:04 -05:00
Travis Cross
1b7360159a
Associate "sending early media" log with session
2014-07-16 04:57:39 +00:00
Patrice Fournier
21ae587063
Disabling Require timer for T.38 re-Invites cause problems
...
Disabling Require timer for T.38 re-Invites tells the remote side it
doesn't need to refresh the session but FreeSwitch will still terminate
the call if the remote session doesn't refresh.
2014-07-08 01:00:52 -04:00
Anthony Minessale
956da6d689
Modify sofia profile to attempt to bind to the interface up to 3 tries with a 5 second wait between attempts.
...
Add new profile params bind-attempts and bind-attempt-interval to modify default behavior.
--NEEDSDOCS
2014-06-02 22:47:26 +05:00
Michael Jerris
4653d78154
CID:1087387 Unused pointer value
2014-05-15 18:30:03 +00:00
