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
...
* 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FS-7500: set 500ms min on retransmit of outdated xml based intraframe request that EVERYTHING still seems to use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
...
* 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
...
* 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
...
* 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
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
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
3ce4ae962b
FS-6540 #comment please test this patch for the added notify functionality
2014-07-17 22:35:04 -05:00
1b7360159a
Associate "sending early media" log with session
2014-07-16 04:57:39 +00:00
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
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
4653d78154
CID:1087387 Unused pointer value
2014-05-15 18:30:03 +00:00
607247397c
FS-6413 update presence_epoch to lock to midnite
2014-05-02 23:49:46 +05:00
4828ecd7fd
remove unused variable
2014-04-28 15:00:22 -04:00
1affd78204
we only call this with values, and it would crash if you passed it null anyways, just assert to make it 100% clear
2014-04-26 15:41:34 -04:00
a4a792488b
add generic keepalive system and implement it in sofia to send MESSAGE or INFO packets in-dialog at specified interval.
...
Adds app: enable_keepalive 0|<seconds>
This app can be run in the dialplan or with execute_on_* type variables for B-legs.
Adds sofia param: keepalive-method : defaults to MESSAGE can also be "INFO"
This param sets which SIP method to use.
2014-04-16 06:10:25 +05:00
7151d6acea
FS-6402 part 2
2014-04-02 03:21:37 +05:00
5c0cff70b3
FS-6402 --resolve
2014-04-02 01:20:19 +05:00
c02a5e67b8
FS-6413 --resolve with this patch you will need to make sure the boxes have the clocks synced and both started inside the same occurence of the most recent new year.
2014-03-31 15:23:50 -05:00
f3acb03dc0
S-6341:make sure to unlock too
2014-03-19 16:38:53 -04:00
6e3f4d667c
add missing break, please review
2014-03-19 07:58:13 +08:00
340b697e1b
FS-6341: --resolve add 3pcc invite w/o sdp support for 100rel/PRACK
2014-03-17 12:27:42 -04:00
804ef7709d
change from sqlite hash to newly added one
2014-03-09 00:37:17 +05:00
e5b291514c
FS-5755
...
rtp_secure_media=mandatory
rtp_secure_media=optional
rtp_secure_media=mandatory:AES_CM_256_HMAC_SHA1_80,AES_CM_256_HMAC_SHA1_32
rtp_secure_media=optional:AES_CM_256_HMAC_SHA1_80
rtp_secure_media=forbidden
true implies mandatory
false implies forbidden
not set implies optional
rtp_secure_media_inbound or rtp_secure_media_outbound take precedence and are treated the same way based on leg direction
2014-03-06 07:34:47 +05:00
2c1a25d5f8
add sip_force_nat_mode so you can engange nat mode manually
2014-03-01 04:43:07 +05:00
5646957c5b
FS-5937
2014-02-26 04:06:59 +05:00
8bf70dcf47
FS-6164 I can see from this jira that this should be strcasecmp so SIP or sip are caught
2014-02-20 13:50:04 -06:00
6e7d5d0897
update copyright header for 2014
2014-02-12 12:08:56 -06:00
10d2dd3e73
use portable switch_inet_ntop
2014-02-12 03:31:21 +05:00
900db14f1d
FS-6203 --resolve
2014-02-07 22:34:34 +05:00
b65d2a9a78
FS-5396 --resolve add gethost function to call gethostbyname as desired
2014-02-05 01:08:31 +05:00
3eb645a336
FS-6093 --resolve
2014-01-09 14:30:41 -05:00
ccaa3ae732
FS-5959
2013-11-21 01:38:21 +05:00
bb9adff511
rework media code
2013-11-08 03:48:00 +05:00
25e7be1c60
FS-2731 apply patch
2013-10-16 10:56:01 -04:00
4a172402d2
some refactoring and plumbing for 1.4
2013-10-16 02:24:32 +05:00
6521de294c
FS-5717 --resolve
2013-09-19 10:15:02 -05:00
a7ab24ef4c
FS-2731 refactor a bit to make this actually usable and more cross-endpoint compatible
2013-09-18 16:00:03 -04:00
38cd508a12
FS-5772 --resolve this is sort of a hack but less painful than reporting bugs to polycom
2013-09-10 23:14:30 +05:00
5bf27df0a4
FS-5752 --resolve
2013-09-04 21:05:13 +05:00
7d6d688e53
FS-5731 --resolve
2013-08-26 20:25:14 +05:00
863e6cfa3f
FS-2731 significantly reworked version of Emmanuel's patch to allow subscribing and notifying for the as-feature-event events. we still need a module to handle the FS events for this automatically... coming soon to a repository near you
2013-08-23 15:54:49 -04:00
b76c39ade1
FS-5700 --resolve
2013-08-15 21:19:57 +05:00
4efeced015
FS-5682 --resolve there was one other change to allow them to copy over by setting them on the A leg. Exporting the sip_multipart variable or setting it in {} on the dial string should already work. This patch should make it work with setting it too.
2013-08-12 19:05:04 +05:00
e9adbb7ac6
FS-5687 this patch adds a debouncer to that info message. if you still have problems try altering switch_rtp.c:5469 to add 0 condition so it skips that whole block and retest eg... if (0 && switch_rtp_test_flag(rtp_session, SWITCH_RTP_FLAG_VIDEO)) {
2013-08-12 00:57:47 +05:00
39ad799683
FS-4932 FS-5090 --resolve
2013-07-29 10:18:05 -05:00
b22a693e9d
FS-5561 --resolve
...
Merge remote-tracking branch 'moy/blind-transfer-response'
2013-07-24 16:15:16 -05:00
57a13694f7
use this and send {sip_gethostbyname=true} in the dial string to force this behaviour
2013-07-12 13:54:50 -05:00
647f3fd3fb
FS-5396 --resolve
2013-07-12 13:36:04 -05:00
5324063f44
Added support for blind transfer notification of negative responses
2013-06-26 23:17:49 -04:00
32adc789f6
make noexpandtab explicit in all vim modelines other than xml files
2013-06-25 11:50:17 -05:00
2fac3a8e83
refactor some of the video passthru code
2013-06-17 19:52:37 -05:00
b74ad2a6bb
FS-5503 --resolve indeed, thx
2013-06-11 09:31:16 -05:00
4240526ce3
add some device-state mechinism to FS to allow tracking of device-specific states where they may have more than one call from the same device
2013-06-05 11:20:11 -05:00
d27a95a50e
Possible leak if someone were to try and add extra headers to a channel that had either an inbound info on that session or the session were to be deflected. I don't see a way that these extra headers could be added outside of a C module subscribing to the session states, but just to make sure I'm adding these so that just like other extra header locations it's cleaned up.
2013-05-18 17:00:41 -07:00
c4e7c302f9
FS-5351 --resolve
2013-04-27 08:19:18 -05:00
7a42ec06df
RTP stats were not being logged. There is no JIRA on this sigh
2013-04-26 09:31:28 -05:00
88a13a7580
FS-5316 --resolve
2013-04-25 13:48:02 -05:00
e1c325dcb5
FS-5341 --resolve
2013-04-23 14:49:07 -05:00
c8a6c99f84
FS-5322 --resolve
2013-04-18 16:27:45 -05:00
7aa23c4b21
FS-5299 --resolve
2013-04-17 15:38:37 -05:00
0aa2009d1d
fix case for call_completed_elsewhere var set
2013-04-15 16:45:32 -04:00
8adeab7d5a
move callstate changes down to take place after the core states have run
2013-04-15 10:43:32 -05:00
b934891f49
sla cid tweaks
2013-04-05 00:38:57 -05:00
3cc441485b
revert
2013-04-05 00:38:09 -05:00
900f2aab74
sla cid tweaks
...
Conflicts:
src/mod/endpoints/mod_sofia/sofia_glue.c
2013-04-05 00:36:44 -05:00
dd878da49d
clean up proxy media mode
2013-03-31 21:27:27 -05:00
45bd5952b1
fix some regressions and improve candidate selection
2013-03-31 21:27:26 -05:00
4b3aa396fb
fix some regessions and sync namespace of some vars
2013-03-31 21:27:25 -05:00
c4b221c750
trying to get mozilla to work using sipml5, got stun/dtls/rtp flowing and its dead silent. c'mon guys... why do you have to do everything differently?????
2013-03-31 21:27:25 -05:00
ad1226b5c4
change name to match steves changes in master
2013-03-31 21:27:23 -05:00
5c90322e8c
MILESTONE: kind of works on media and signaling
2013-03-31 21:27:20 -05:00
1c120dd2c8
not too shabby audio and vid work both directions but browswer hates our srtcp ...
2013-03-31 21:27:18 -05:00
7510bccd6c
MILESTONE: working inbound calls
2013-03-31 21:27:17 -05:00
0fffc839e4
pass1
2013-03-31 21:27:14 -05:00
eb81042a6f
wip
2013-03-31 21:27:14 -05:00
d6af3a1fdd
omfg
2013-03-31 21:27:14 -05:00
1ab88581d7
rename
2013-03-31 21:27:14 -05:00
0a246d1cb4
pull srtp negotiation stuff into the core stage 1
2013-03-31 21:27:13 -05:00
330f68d946
convert rtp flags to arrays
2013-03-31 21:27:13 -05:00
9fc4beeaa5
FS-3576 --resolve
2013-03-07 08:44:32 -06:00
9ccdf30ed2
sqlite can't handle bi-directional mutexing
2013-03-06 18:02:01 -06:00
fc1316601b
FS-5118 take 2
2013-02-27 09:04:55 -06:00
cf7c9428fb
FS-5118 --resolve
2013-02-26 16:37:33 -06:00
8987e96744
add dedicated mutex for select based sql stuff
2013-02-25 15:42:26 -06:00
18cd797065
cleanup
2013-02-18 13:54:34 -06:00
5144700927
Changed the ..._image_sdp routine to ..._udptl_image_sdp in preparation for
...
adding other kinds of image (specifically RTP and TCP/TPKT).
2013-02-03 16:28:20 +08:00
c7c4bda847
fix pnp
2013-01-09 18:18:07 -05:00
b78ffffed5
FS-3951 --resolve
2013-01-07 12:36:58 -06:00
f2428b08f9
FS-4779 --resolve
2013-01-07 10:27:28 -06:00
7e8e99e52c
fix answer contention issue
2012-12-06 11:11:07 -06:00
b35ac6b3bd
fix auto_simplify
2012-11-29 18:34:27 -06:00
685da11d0d
FS-4885 --resolve this is not the best solution but it will work
2012-11-29 12:37:46 -06:00
d4f8a79299
try to improve audio stream sync
2012-11-28 23:12:35 -06:00
d48e3de7c8
make it so when you put someone on hold and then they attended transfer you, the resulting bridge will still be playing hold music
2012-11-15 21:07:58 -06:00
2f2a351067
add initial_callee_id_name/number
2012-11-14 09:20:52 -06:00
feb38cece2
FS-4779 give this a shot
2012-11-07 08:14:32 -06:00
23bb963d17
FS-4779 I don't have this labbed up to try so can you try this patch and repost logs if it doesn't work
2012-10-31 08:52:00 -05:00
b94ef869e3
FS-4769 try this patch and put {drop_dtmf=true} in your dial string
2012-10-26 14:46:36 -05:00
b29412054a
fold reconfig_sofia into config_sofia so we don't have 2 difft config funcs
2012-10-25 12:30:08 -05:00
68e0b7e859
optimize sofia sql by using new core transaction processor we will no longer support databases that do not support transactions
2012-10-25 11:31:47 -05:00
fbb99f60b4
really fix FS-3726 this time
2012-10-18 15:56:06 -05:00
986fede6b0
make a way to get the jitter buffer down to the codecs
2012-10-18 15:29:00 -04:00
a20aec182a
add rtp_manual_video_rtp_bugs
2012-10-16 18:43:00 -04:00
7eafa85de5
add execute_on_media_timeout var
2012-10-16 12:48:16 -04:00
ea599992e1
FS-4656 --resolve with this patch the bug is fixed but also the old behaviour of setting this variable on the A leg expecting it to auto-set it on the B leg is no longer supported. It must be exported or put in the dialstring with {} to put it on the B leg going forward
2012-10-03 07:42:05 -05:00
06a3266360
FS-4655 its not a bug its a behaviour request, try this patch
2012-09-27 14:27:28 -05:00
9c97ae3ccc
FS-4640 please test
2012-09-21 14:45:30 -05:00
bf4bb0028e
Merge branch 'master' into FS-4558
2012-09-08 02:29:59 -04:00
8ef13dd8a2
FS-4558 --resolve
2012-09-08 02:28:35 -04:00
bfc46567c1
add uuid_media_reneg api command to tell a channel to send a re-invite with optional list of new codecs
2012-09-07 15:33:09 -05:00
455b22480c
FS-3971 --resolve
2012-09-04 12:30:23 -05:00
a5e9f65e31
FS-4582 --resolve
2012-09-01 11:46:45 -05:00
6ed3218f8a
FS-4558: add uuid
2012-08-31 16:35:13 -04:00
6d2b3256c3
another efficiency pass on sofia and refactoring of 3c685bff25 to be more efficient
2012-08-30 15:26:13 -05:00
04924f1694
FS-4079 FS-4540 regression from 5213c764bb please update ASAP if you are running this one
2012-08-23 14:56:37 -05:00
5213c764bb
FS-4079 FS-4540 minor refactoring to fix some minor side effects of recent changes
2012-08-23 11:10:03 -05:00
2a8841ab66
change mod_sofia to use new core based recovery engine
2012-08-22 16:27:07 -05:00
Sergey Safarov