9fc8772aef
FS-7463: revert commits c2c04632f3 b33563138b and 6bf74877b5f0. Future pull requests should be verified that they compile before submitting.
2015-05-14 13:55:32 -04:00
64c7fee8ca
conditionally allow intercept of replaced call-id when processing replaces header
...
without this change one leg calls inhaled could not be picked because there is no bridge_uuid.
FS-7463 #resolve
2015-05-14 17:43:09 -05:00
c2c04632f3
Merge pull request #241 in FS/freeswitch from ~LAZEDO/freeswitch-fs-7463:FS-7463 to master
...
* commit 'b33563138b0953693dbe0d32a439c533ea8b4316':
conditionally allow intercept of replaced call-id when processing replaces header
2015-05-14 11:37:51 -05:00
518d74a4bc
FS-7492 fix copy paste typo
2015-04-30 07:35:40 -05:00
6e588b551b
FS-7472
2015-04-28 16:12:37 -05:00
b33563138b
conditionally allow intercept of replaced call-id when processing replaces header
...
without this change one leg calls inhaled could not be picked because there is no bridge_uuid.
FS-7463 #resolve
2015-04-22 20:11:12 +01:00
a4d877c189
FS-7460: don't force ice in 3pcc-mode=proxy
2015-04-21 19:58:28 -04:00
3bbfa9f341
FS-7407: when using proxy, make sure to pick the right transport to the proxy
2015-04-13 22:54:02 +00:00
beeea01724
FS-7242 Waiting for state CS_PARK when receiving REFER.
2015-04-07 22:11:27 -03:00
36e1840d63
Merge pull request #207 in FS/freeswitch from ~MBRANCA/freeswitch:bugfix/FS-6954-freeswitch-adds-additional-m-audio to master
...
* commit 'ca2dd2f3ba80924567472c0516ee8bac2af34d6a':
FS-6954 Detect when we have T.38 in nomedia or proxy media mode and apply same fixups as in media mode.
2015-03-26 10:00:48 -05:00
ca2dd2f3ba
FS-6954 Detect when we have T.38 in nomedia or proxy media mode and apply same fixups as in media mode.
2015-03-26 15:33:30 +01:00
ed0a434b95
Fail the call with 502 bad gateway if the outbound leg returns a redirect with an invalid URI.
...
FS-7385 #resolve
2015-03-25 12:46:59 -07:00
3c4c64a9de
Merge pull request #200 in FS/freeswitch from ~SJTHOMASON/freeswitch:bugfix/FS-7192-expires-header-included-in-gateway to master
...
* commit '54f16398310846a6b793491d21778e0961b48712':
FS-7192: [mod_sofia] do not include Expires header in INVITEs responding to a auth challenge
2015-02-20 13:19:26 -06:00
d562bc601a
Merge pull request #201 in FS/freeswitch from ~SJTHOMASON/freeswitch:bugfix/FS-7308-cannot-disable-sofia-sla-sql-query to master
...
* commit 'cc61f11fb7001ad7a7882e6591b3e89740dce2e9':
FS-7308: [mod_sofia] Only log SLA SQL query SQL when debugging is enabled
2015-02-20 13:12:40 -06:00
74e8db736f
FS-7309: [mod_sofia] Allow removal of User-Agent header
...
In certain deployments it might be desirable to remove the User-Agent header from the SIP message.
Set the sofia profile parameter "user-agent-string" to "none" to suppress this header.
2015-02-20 08:36:44 -08:00
cc61f11fb7
FS-7308: [mod_sofia] Only log SLA SQL query SQL when debugging is enabled
2015-02-19 15:52:44 -08:00
54f1639831
FS-7192: [mod_sofia] do not include Expires header in INVITEs responding to a auth challenge
2015-02-19 15:36:54 -08:00
302a339fdf
FS-7294: Enable -Werror when building with clang compiler #resolve
2015-02-17 12:20:33 -05:00
5db8a0c16b
FS-7228 - Max Register Refresh before the Register is challenged for credentials
2015-02-06 10:16:36 -08:00
f135232ef5
FS-7196: add recording_follow_transfer support for semi attended transfers
2015-02-02 15:48:26 +01:00
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
441dd91288
FS-7204, Allow to force a per-account send-message-query-on-register
2015-01-27 17:14:54 +01:00
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
01dcb74f33
FS-7187 add switch_cache_db_create_schema() to test for SCF_AUTO_SCHEMAS flag
2015-01-22 15:41:22 -05:00
6afc2b5a2e
FS-6688: #resolve fix resubscribe through proxy with record route when the resub does not have a record route and the route has uri params
2015-01-07 17:41:19 -05:00
25404c121e
Merge pull request #153 in FS/freeswitch from ~JONHEIER/freeswitch-fs-7137:master to master
...
* commit '165f54216c47a5343ac0c7a6ac62fd6a9de57b5f':
mod_sofia: Set sip_to_tag on ringing indication for inbound channels.
2015-01-06 10:32:21 -06:00
c460c00b55
FS-7134 #resolve
2015-01-06 16:29:38 -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
067cb0f0f2
FS-7100: make buffer for sub contact big enough
2014-12-22 17:19:08 -05:00
c96d64e7e1
invalid type cast correction
2014-12-19 11:13:55 +03:00
5376e82669
FS-6688: don't update the contact when the original had fs_path, the new one doesnt, and the contact didn't change. This will fix the normal case of record route from a proxy without breaking normal changing of a contact
2014-12-10 10:33:43 -05:00
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
bad5dc3022
FS-7037 #resolve
2014-12-01 15:10:00 -06:00
5bbef7f1e5
FS-7015 #comment please test
2014-11-25 17:02:10 -06:00
5127b64df8
FS-7014: [mod_sofia] don't touch the tech_pvt when a call has just ended, leaving us with a null tech_pvt
2014-11-18 10:22:13 -06:00
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
82aa33140e
FS-6531: #resolve set to tag on auto answer 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 21:37:14 -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
9e9c4378de
Merge branch 'master' of ssh://stash.freeswitch.org:7999/fs/freeswitch
2014-11-06 18:26:28 -06:00
415f82fe93
FS-6954 #resolve #comment please test
2014-11-06 18:26:16 -06:00
f66f2cae8c
FS-6890 #comment please test
2014-11-06 17:13:02 -06:00
1190e59adf
FS-6965 #resolve
2014-11-06 09:47:54 -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
9c1e6037c9
FS-6954 #comment we fixed another bug and this is the side effect which is completely valid, too bad you can never fix broken t38 endpoints. Can you please test this patch
2014-11-05 11:51:30 -06:00
a4971693d3
FS-6890 #comment please test
2014-11-05 11:35:16 -06:00
32a9ff3d39
Merge pull request #60 in FS/freeswitch from ~SJTHOMASON/freeswitch:FS-6823 to master
...
* commit 'afb00b2ecc8a9b049801f3f475c80e1111070fa8':
Force rport on ADTRAN TA Devices
2014-11-04 07:36:36 -06:00
39be877760
One place we said Failed Registration, the other we said Registration Failed, lets try to be consistent.
2014-10-30 10:40:52 -05:00
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
75815877d6
FS-6688 #resolve
2014-10-27 14:14:21 -05:00
f772b400bf
FS-6939 please do pull request next time. ;)
2014-10-27 14:12:55 -05:00
26af9c3d67
FS-6939 #resolve
2014-10-27 12:23:44 -05:00
f1ee4ba4d7
Merge branch 'master' of ssh://stash.freeswitch.org:7999/fs/freeswitch
2014-10-16 17:03:57 -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
2e10407336
actual fix for commit cff5209ca3 which was in the wrong place
2014-10-16 16:04:15 -05:00
3bdbdd4abf
revert cff5209ca3
2014-10-16 14:39:59 -05:00
beb1d17921
FS-6400 Improve sip ping generation by distributing them across an interval
2014-10-14 14:24:21 +02:00
cff5209ca3
fix leak of nua handle due to reference counting that must be between 3 to 7 years old. Effects all calls with auth/challenge on INVITE
2014-10-13 18:06:32 -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
b5294c53d6
Fix crash on transport=tls with non-TLS profile
...
We use the transport of the Contact header of the remote UAC to decide
which of our own Contact addresses we should use when replying to a
SUBSCRIBE or sending a presence NOTIFY.
If TLS is not enabled on a Sofia profile, then the TLS Contacts for
that profile are NULL. Unfortunately we were using these NULL values
uncritically when the remote UAC sent us a Contact header with a TLS
transport and our own Sofia profile did not have TLS enabled.
With this commit we fall back to our TCP Contact address when the
remote Contact is TLS and our Sofia profile does not have TLS enabled.
2014-10-10 18:36:37 +00:00
855cc4b4e0
add 908-retry-seconds gateway param to set reg retry time when getting a 908 for backup interfaces to connect quickly
2014-10-09 14:43:23 -04:00
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
6860b41763
Merge pull request #83 in FS/freeswitch from ~MKVONARX/freeswitch-fs-6710:FS-6710 to master
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* commit '490efb7177ddcd3e61018f02c1435362937e8b15':
FS-6710: fix incorrect comparison for Min-SE values between SIP INVITE and local configuration
2014-10-07 11:50:19 -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
490efb7177
FS-6710: fix incorrect comparison for Min-SE values between SIP INVITE and local configuration
2014-10-07 10:41:36 +02:00
b2ae5f4cc2
few bugs on recent new features
2014-10-03 15:36:23 -05:00
bde2e2da51
FS-6889 #resolve
2014-10-03 11:34:42 -05:00
ae5d86515a
FS-6884 #comment these were mostly simple warnings
2014-10-02 19:20:35 -05:00
10a3fa55ef
%FEATURE add bypass_media_resume_on_hold and bypass_media_after_hold variables to be set to true to enable these functions on a per channel basis
2014-10-02 17:49:09 -05:00
43733a6166
FS-6886 #comment addition of ignoring unhold as well
2014-10-02 15:48:29 -05:00
afb00b2ecc
Force rport on ADTRAN TA Devices
...
ADTRAN Total Access devices do not support sending the rport parameter in
the Via header. This allows us to detect the device and force rport when
using the "safe" parameter, enabling the device to be used behind NAT.
FS-6823 #resolve
2014-10-02 13:09:15 -07:00
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
9e9175321a
FS-6886 #resolve
2014-10-02 11:30:13 -05:00
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
789e1481ed
FS-6880 #resolve #comment I would think that in real life once the call agreed on a codec it would only offer the negotiated codecs but we can fix this to always filter for good measure. I am not sure what the ramifications are of filtering responses but I think this patch will do so as well.
2014-10-01 13:03:50 -05:00
644b41f792
FS-6874 #resolve
2014-09-30 17:05:06 -05:00
24084adf77
%FEATURE Add new feature to filter the SDP on bypass_media calls to remove or limit codecs.
...
VARIABLE: bypass_media_sdp_filter
Can be set globally or per leg on the inbound side of a bypass_media bridge.
VALID FILTERS:
remove(): Removes the specified codec if it exists in the SDP.
only(): Removes all codecs besides the one specified (providing that it exists in the sdp) (will not remove telephone-event))
EXAMPLE 1 (remove everything leaving only g729):
<action application="set" data="bypass_media_sdp_filter=only(g729)"/>
<action application="set" data="bypass_media=true"/>
<action application="bridge" data="sofia/internal/1238@conference.freeswitch.org"/>
EXAMPLE 2 (remove everything leaving only g729 and also remove dtmf):
<action application="set" data="bypass_media_sdp_filter=only(g729)|remove(telephone-event)"/>
<action application="set" data="bypass_media=true"/>
<action application="bridge" data="sofia/internal/1238@conference.freeswitch.org"/>
EXAMPLE 3 (remove alaw and speex):
<action application="set" data="bypass_media_sdp_filter=remove(pcma)|remove(speex)"/>
<action application="set" data="bypass_media=true"/>
<action application="bridge" data="sofia/internal/1238@conference.freeswitch.org"/>
2014-10-01 01:28:10 +05:00
5bd35471f7
Add var to suppress `Privacy: none` header
...
Apparently the MetaSwitch guys incorrectly interpret `Privacy: none`
as `Privacy: id`.
ref: RFC 3325
Reported-by: Stéphane Alnet <stephane@shimaore.net>
FS-6817 #resolve
2014-09-11 19:56:19 +00:00
7144b25254
obey sip_copy_custom_headers on bye
2014-09-12 00:37:19 +05:00
77c99b6306
FS-6806 #resolve #comment off by 1 error in last fix
2014-09-10 20:32:36 +05:00
69d5cda6d6
resolve FS-6809
2014-09-09 15:33:19 +02:00
98c8a9b508
Merge pull request #38 in FS/freeswitch from ~ALEXDG/freeswitch-event-for-gateway-ping:master to master
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* commit '388e9638de7c14e00272777245dacc87cf09fc1c':
F-5946 add the patches. if in the sofia gateway config the param pin-monitoring is true, then every ping result raise an sofia::gateway-state event
2014-09-08 14:35:47 -05:00
a73583b5f3
FS-6806 #resolve
2014-09-09 00:09:31 +05:00
388e9638de
F-5946 add the patches. if in the sofia gateway config the param pin-monitoring is true, then every ping result raise an sofia::gateway-state event
2014-09-05 10:57:01 +02:00
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
5a209a9680
Remove misleading `tport` example from configs
...
As an example of using mod_sofia's gateway parameter `contact-params`
we'd used the value `tport=tcp`. Looking around, it's clear this has
misled people into believing you can specify `tport=tcp` to make the
gateway use TCP or `tport=tls` to make the gateway use TLS. This does
not work.
The actual contact parameter is named `transport` rather than `tport`,
and you shouldn't use `transport` in `contact-params` because we
automatically add a `transport` to the Contact: based on the value of
`register-transport` (even if the gateway is set to not register).
It's clear why this would be confusing, so we'll just remove this as
an example.
2014-08-27 23:15:45 +00:00
1dc44067cd
FS-6770 #resolve
2014-08-27 13:28:15 -05:00
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
e3e84a7820
FS-6679 #resolve
2014-08-09 02:13:00 +05:00
5075d4af0d
fix typo that can lead to seg
2014-07-30 22:17:47 +05:00
49700ecf0f
fix extraneous < in sip url parser
2014-07-30 22:13:57 +05:00
940171c623
FS-6670 #resolve #comment Doh, see this one is dangerous
2014-07-21 16:04:36 -05:00
c15663cba2
FS-6670 #resolve #comment This should resolve this fun stuff.
2014-07-21 15:58:32 -05:00
83a4305003
Merge pull request #6 in FS/freeswitch from ~FRANCOIS/fs-6568:master to master
...
* commit 'e3b0d7219e1ea269ccd7c5f04747ad21f62c73b2':
Fix BLF confusion when dealing with the same extension in multiple domains
2014-07-18 12:27:19 -05:00
3ce4ae962b
FS-6540 #comment please test this patch for the added notify functionality
2014-07-17 22:35:04 -05:00
91d405a2c1
call recovery_track on recovering channels once the recovery has completed and fix race condition with repeated recovery
2014-07-16 21:40:23 +05:00
1b7360159a
Associate "sending early media" log with session
2014-07-16 04:57:39 +00:00
c58e64bf6d
add some more headers to sip messages
2014-07-16 00:55:46 +05:00
6dca24bb2a
Merge pull request #5 in FS/freeswitch from ~KATHLEEN.KING/freeswitch-fork:clang-warnings-werror to master
...
* commit '1affff9db4aaa29ab66f9f5db76f575eeabd86b5':
Fixed a clang-3.5 missing-prototype warning and added doxygen documentation for switch_channel_set_presence_data_vals. #doxygen
Fixed dead code.
Fixed trucation of value warning.
Removed a useless called to abs.
Removed an autological-pointer-compare from src/switch_utils.c.
Fixed trucation of value warning.
2014-07-14 17:38:21 -05:00
a0a8a66525
FS-6420
2014-07-10 10:17:54 -05:00
36fd87a67a
Merge pull request #7 in FS/freeswitch from ~PFOURNIER/freeswitch:bugfix/t38_require_timer to master
...
* commit '21ae587063eb3b55c690f47cf9299a539ec25415':
Disabling Require timer for T.38 re-Invites cause problems
2014-07-09 10:49:04 -05:00
28e90addf2
FS-5501 fire event on profile start
2014-07-09 08:53:14 -05:00
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
e3b0d7219e
Fix BLF confusion when dealing with the same extension in multiple domains
2014-07-06 16:57:50 +02:00
85699480b3
Fixed trucation of value warning.
...
There was a parameter mismatch between abs(), which expects an int,
and atol() which returns a long. Since max_drift is defined as an int,
there is no need to pars q as a long rather than an int.
2014-07-03 14:41:24 -07:00
2423fc7d0f
FS-6540 This should proxy all refers now, the param has changed to proxy-refer
2014-06-30 12:53:22 -05:00
52fb53cba4
Allow reincarnation from mod_sofia's shutdown-on-fail
...
mod_sofia's parameter shutdown-on-fail now accepts the value
"reincarnate-now". This will cause the switch to exit immediately
with a non-zero exit code so that the supervisor can recover the
switch. For this to work you have to pass in -reincarnate or
-reincarnate-reexec to freeswitch.
2014-06-27 22:18:39 +00:00
80ed14ab3e
Ensure mod_sofia params can be unset or reset
...
This is the result of auditing each mod_sofia profile parameter to
ensure that it can be unset or reset after being set. One use-case
for this being done correctly is so a later parameter in a
configuration file can reliably override an earlier one, which is
useful for setups with layered include files.
2014-06-26 09:04:56 +00:00
75feed68fc
Avoid leaking memory on sofia profile reload
...
We were leaking the string values of apply-candidate-acl parameters.
2014-06-26 08:39:55 +00:00
f7b4ec617a
Allow SIP UPDATE messages without display updates
...
Previously if send-display-update was set to false we would also
remove UPDATE from our Allow: headers. This is unnecessary. The
UPDATE message is useful in SIP transactions even if we're not sending
display updates.
With this commit, we add a new boolean profile flag, allow-update. If
set to true we'll send Allow: UPDATE. If set to false, we will not.
If there is a conflict with another setting that requires UPDATE
support, the allow-update parameter will win and a warning will be
printed.
ref: RFC 3311
2014-06-25 20:43:29 +00:00
8f47f403fd
FS-6612: Don't use rport in tcp or tls
2014-06-20 10:23:05 -05:00
49ee6781a5
FS-6540 please test
2014-06-05 23:49:50 +05:00
3ab1044d6e
FS-6562 --resolve
2014-06-03 15:16:45 -04:00
6722e3dfea
fix regression from regression
2014-06-02 19:25:04 -05:00
f358c35b67
remove debug line
2014-06-02 19:21:02 -05:00
a7b9329a08
FS-6446 regression fixed
2014-06-02 19:09:48 -05:00
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
7c2b4381cf
FS-6446 --resolve
2014-05-27 20:40:00 +05:00
ed6dd184f7
FS-6482 Commit 7e15c938e2 broke MWI for device having no transport parameters (plain UDP connection)
2014-05-23 08:42:16 -04:00
bedf8327b6
FS-6418 FS-6427 add profile param NDLB-proxy-never-patch-reinvites to solve both issues
2014-05-21 03:31:50 +05:00
c19aa584cc
CID: 1214222
...
sofia_glue_get_url_from_contact to strdup the value so before
overwriting the value signal to sofia that we have finished with the
raw header.
2014-05-16 17:01:03 -07:00
388e119e55
CID: 1214221,1214222
...
sip_header_as_string returns new memory alloc'd on the nua_handle's memory pool. This memory would not be freed until
the handle was destroyed. Since there is no usage of the call-info header after it is added to the event as a header value,
we should make sure to both su_free the sofia returned string, but also more cleanly create the event header value to reduce a strdup.
2014-05-16 16:35:49 -07:00
3666029ff0
add auth-subscriptions flag. sofia ua can now optionally challenge subscrition requests
2014-05-16 19:05:35 +00:00
7d5434f57c
CID: 1214223-1214230 If sent an invalid registration authentication header with multiple params of the same name, then only use the first of the params instead of leaking memory
2014-05-16 11:50:18 -07:00
ebc0a99f1c
CID:1024250 Dereference before null check. CID:1024251 Dereference before null check, we don't need tto check these, as they are already used above in all cases
2014-05-15 18:34:25 +00:00
4653d78154
CID:1087387 Unused pointer value
2014-05-15 18:30:03 +00:00
0a2ba8869a
sofia is treating all notify sending as if they were natted rather than when nat is detected. This causes issues on some polycoms that use strange ports in their contact for getting notifies that do not match the port the calls come from
2014-05-06 04:10:19 +05:00
607247397c
FS-6413 update presence_epoch to lock to midnite
2014-05-02 23:49:46 +05:00
d06b137abd
FS-6498 --resolve
2014-05-01 15:20:28 -05:00
af32ca00e3
Correction to commit 0ba59aa6c2 using the correct variable in tel uri param parsing for mod_sofia
2014-04-29 15:23:24 -07:00
4828ecd7fd
remove unused variable
2014-04-28 15:00:22 -04:00
cc06d9181b
FS-4779 FS-6467 the behaviour from 4479 is now masked behind the param 3pcc-reinvite-bridged-on-ack FS-6467 please test master
2014-04-28 22:10:14 +05:00
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
abd7882d18
contact->m_url is an array, don't check array for zero as its useless
2014-04-26 15:35:27 -04:00
0ba59aa6c2
Expanding on 052638186b to support boolean tel url params in mod_sofia
2014-04-25 13:49:59 -07:00
7e15c938e2
FS-6482 --resolve
2014-04-24 14:05:18 -05:00
175c9eebca
Remove extraneous abort(3) when sending chat
...
This appears to have been accidentally added in commit
79ebcb104b
2014-04-19 19:26:27 +00:00
79ebcb104b
add enable-chat sofia profile param which defaults to true when not present, set to false to turn off the chat interface in sofia
2014-04-19 12:29:10 -05:00
bce51017fb
FS-6462 --resolve
...
I found a problem here but it may not completely match your expectations.
I reviewed the RFC 4028 and checked against the code and I discovered we should not be putting a Min-SE in any response at all besides a 422:
section 5:
The Min-SE header field MUST NOT be used in responses except for
those with a 422 response code. It indicates the minimum value of
the session interval that the server is willing to accept.
I corrected this problem and implemented the 422 response so if you request a value lower than the minimum specified for the profile.
If the value is equal or higher to the minimum, it will be reflected in the Session-Expires header in the response and no Min-SE will be present.
2014-04-17 04:26:43 +05:00
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
052638186b
add parsing of user params on invite
2014-04-15 12:48:07 -04:00
12af80d6b2
Condense styling a bit
2014-04-12 05:06:51 +00:00
3f8cf4a59d
Fix memory leak in subscription handling
...
We were leaking one event (~539 bytes) for every subscribe packet
received with both an "event: as-feature-event" and an authorization
header.
2014-04-12 05:00:41 +00:00
19e3175518
FS-6422: --resolve obvious copy and paste error
2014-04-03 07:20:41 -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
1cde5f01e0
FS-6386 --resolve
2014-03-23 16:15:49 -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
65c58a7ceb
FS-6341: fix compiler warning from set but unused var
2014-03-17 18:42:02 -04:00
340b697e1b
FS-6341: --resolve add 3pcc invite w/o sdp support for 100rel/PRACK
2014-03-17 12:27:42 -04:00
3c95a52ff6
we are intentionally using a value outside of the enum here as a special value. The compiler will always make this an int anyways, so this is safe. cast to remove the warning
2014-03-15 22:47:55 -04:00
8f703c7a78
FS-6353 regression from 7ba257ecd6
2014-03-14 18:34:08 -04:00
89f9490710
FS-6350 --resolve Parse params out of user portion of caller_id and save them to sip_name_params
2014-03-14 13:13:22 -05:00
e37616e36d
do not retry so fast on failed outbound subs
2014-03-14 12:55:50 -05:00
7a6e8f4672
FS-6354 --resolve, add realm to sofia::expires event
2014-03-13 11:13:10 -05:00
7ba257ecd6
FS-6353 --resolve
2014-03-13 10:23:13 -05:00
70f2908976
more nat fix
2014-03-11 16:12:43 -05:00
5c8a3b12fd
Convert some behaviors to optional.
...
These all were set before but now are disabled by default because they are not proper to have enabled without a good reason.
tcp-unreg-on-socket-close
If registration is TCP or TLS and socket connection is interrupted, unregister the client.
tcp-always-nat
Treat all TCP connections as NATTED, this causes all traffic back to the phone to reuse the existing TCP socket but may
cause problems when not used with TCP keepalive.
tls-always-nat
Same as tcp-always-nat but only for TLS connections.
2014-03-11 15:17:02 -05:00
804ef7709d
change from sqlite hash to newly added one
2014-03-09 00:37:17 +05:00
07272e864e
FS-6321 --resolve
...
Copy URI params from Refer-To header into custom header in subsequent INVITE
sip_h_X-FS-Refer-Params is set to show up in the INVITE as X-FS-Refer-Params
2014-03-07 12:56:58 -06:00
f87ae15fc6
don't refuse invites on established sessions due to session limits
2014-03-07 23:46:26 +05:00
0da8c6331d
don't kick in nat mode on polycom tcp unless its not in the local network
2014-03-07 08:34:26 +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
7cb91467e0
FS-5814 --resolve
2014-03-06 00:02:40 +05:00
fd38a255f8
FS-6167 --resolve
2014-03-04 13:41:34 -08:00
2c1a25d5f8
add sip_force_nat_mode so you can engange nat mode manually
2014-03-01 04:43:07 +05:00
1d36f5b219
FS-5915 --resolve
2014-02-28 16:09:59 -06:00
2dc71d2d26
add sip_refer_to_params
2014-03-01 00:13:35 +05:00
b82df8a7da
FS-6287 this should either prevent the problem or spell out what is causing it. please test
2014-02-28 23:23:30 +05:00
45e19b75dd
FS-6270 add client_port to SIP Registration events
2014-02-25 21:49:35 -06:00
5646957c5b
FS-5937
2014-02-26 04:06:59 +05:00
9f98ccc105
FS-5115 --resolve
2014-02-24 20:25:31 -05:00
97e136c12b
FS-5724 include realm in sofia::unregister Event
2014-02-22 18:31:23 -06:00
19dbd07ff8
Allow access to headers from INVITE
...
This adds a sip profile parameter parse-all-invite-headers that when
set parses all headers from an INVITE into channel variables. The
headers are converted to lowercase, underscores are replaced with
dashes, and the result is prefixed with sip_i_. Headers than exist
more than once are set as arrays.
FS-6075 --resolve
Thanks-to: Peter Olsson <peter@olssononline.se>
2014-02-21 20:45:40 +00:00
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
bd72cd9cc1
Allow setting late margin on register expirations
...
When an endpoint registers to us we internally mark the expiration as
some seconds longer than the actual registration. Previously this
value was fixed at 60 seconds.
Some people need this value to be shorter so they can meet their SLA
by taking a different action when a device doesn't re-register when
expected.
This commit adds a SIP profile parameter sip-expires-late-margin which
allows setting the margin value we apply here.
FS-6101 --resolve
Thanks-to: Emmanuel Schmidbauer <e.schmidbauer@gmail.com>
2014-02-20 16:07:53 +00:00
b3c02a547e
FS-6230 --resolve There was no NAT detection enabled on the calls, this patch will activate it explicitly when used with force-rport to avoid a similar problem with other users
2014-02-14 23:50:17 +05:00
c46bc9d565
FS-6223 --resolve
2014-02-13 22:11:34 +05:00
a0d6ebd80d
FS-6222 --resolve
2014-02-13 20:46:05 +05:00
7ed4ed65db
0a63cdc4b2 this might help
2014-02-13 05:50:39 +05:00
0a63cdc4b2
FS-6222 try this patch
2014-02-13 04:08:24 +05:00
35f2bcccf7
FS-6221 --resolve
2014-02-13 03:26:09 +05:00
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
d3b9aaba60
Add mod_sofia support for setting socket TCP keepalive
...
Some OSes like linux provide a mechanism for sending TCP keepalive
pings. Sofia provides its own mechanism for doing this in userspace.
We shouldn't conflate these two mechanisms.
FS-6104
2014-02-11 06:27:29 +00:00
c52ac4817c
FS-6128 FS-6200 --resolve allocating the sofia_private on the nua_handle seems to lead to memory corruption, changing it back to malloc as done in the version before the regression
2014-02-11 04:05:40 +05:00
386a95f47d
FS-6208 --resolve that patch is correct but this other code will protect the seg in either case
2014-02-10 21:44:19 +05:00
900db14f1d
FS-6203 --resolve
2014-02-07 22:34:34 +05:00
0539cc166b
FS-6202 --resolve
2014-02-07 22:13:23 +05:00
4c068c637d
FS-6197 --resolve the line numbers in the trace do not match the version you reported make sure you know what version you are using
2014-02-06 21:47:24 +05:00
57abca11a8
Support setting TLS cipher suites in mod_sofia
...
This adds a parameter tls-ciphers in which we can specify the TLS
cipher suite specification for OpenSSL.
2014-02-06 06:04:02 +00:00
d4a13742ba
Default to TLSv1 through TLSv1.2 if unspecified
2014-02-06 02:47:40 +00:00
8c2c464bec
Support setting enabled TLS versions in mod_sofia
...
Previously if tls-version was set to tlsv1 we supported only TLSv1,
but if it was set to sslv23 we supported all versions of TLS. This
was a weird incorrectly documented behavior that we hope no one was
relying on.
Now we can pass a comma-separated list of TLS/SSL versions that we
would like to support in tls-version.
FS-5839 --resolve
2014-02-06 02:20:19 +00:00
a121665b14
FS-6190 --resolve slightly modified, please test
2014-02-06 02:31:56 +05:00
8fa385b0d7
Make bypass-media-after-hold a media-option
...
FS-6186 --resolve
Thanks-to: Matteo Brancaleoni <mbrancaleoni@voismart.it>
2014-02-05 19:36:15 +00:00
5e8fe326c7
Move documentation to correct place
2014-02-05 17:59:08 +00:00
3ecb504fda
FS-6186 --resolve
2014-02-05 07:15:08 +05:00
b65d2a9a78
FS-5396 --resolve add gethost function to call gethostbyname as desired
2014-02-05 01:08:31 +05:00
4021b5ce14
try to avoid compiler optimization snafu
2014-02-05 00:31:10 +05:00
a21cbf8142
FS-6165 --resolve
2014-01-30 00:20:21 +05:00
8dccd213fb
FS-6085 --resolve
2014-01-28 21:56:02 +05:00
c537fc6f61
Fixing leak in presense where the contact header doesn't include the fs_path option during a presense subscription
2014-01-22 15:28:24 -08:00
5a820fdc78
Disabled some compiler warnings on Windows build (for Visual Studio 2012), also disabled code analysis one some projects. This should make the Windows build more clean in the future.
2014-01-22 22:47:11 +01:00
084e245085
Lots of trivial tweaks to minimize compiler warnings, especially on 64-bit platform: Added missing type casts, changed types, added missing NULL checks.
2014-01-22 22:30:23 +01:00
643c88f49b
FS-6140 --resolve
2014-01-23 02:13:10 +05:00
bfeed7514b
add version to log
2014-01-20 23:34:32 +05:00
8dfbd91a3d
fold version macros into runtime functions to avoid cascade rebuilds when modifying the core
2014-01-17 01:54:47 +05:00
3eb645a336
FS-6093 --resolve
2014-01-09 14:30:41 -05:00
ef730b75aa
FS-5987 pushing the patch now since no matter what its better than before
2013-12-11 00:04:26 +05:00
30325093d4
FS-6009 --resolve
2013-11-27 23:55:00 -06:00
2747660269
FS-5959 FS-5958
2013-11-25 12:08:17 -06:00
8015a3a0a6
fix typo that caused a seg in presesnce
2013-11-23 02:32:02 +05:00
4129adc9ac
update more fields on re-subscribe
2013-11-22 22:21:32 +05:00
ccaa3ae732
FS-5959
2013-11-21 01:38:21 +05:00
4fb80bf301
FS-5977 --resolve
2013-11-15 21:33:05 +05:00
e603c17de8
try to improve performance of subscribe sip handler
2013-11-13 21:41:37 +05:00
0d1b647197
remainder of previous patch to pass through dynamic payload numbers used
2013-11-12 00:42:32 +05:00
f0335d0f51
FS-5948 --resolve
2013-11-08 03:53:12 +05:00
bb9adff511
rework media code
2013-11-08 03:48:00 +05:00
ecedd2d21c
FS-5844
2013-10-31 10:35:53 -05:00
75ad91e185
FS-5844 --resolve
2013-10-30 20:26:01 -05:00
9ce66748af
Minor code duplicatation removal.
2013-10-28 16:56:50 -07:00
556e243361
FS-5910 --resolve
2013-10-24 14:29:29 -05:00
37774268c0
FS-5882 --resolve
2013-10-17 22:40:07 +05:00
25e7be1c60
FS-2731 apply patch
2013-10-16 10:56:01 -04:00
00c46450ff
In a case where calls to phones with presence enabled as well as presence disabled,
...
there were cases where the presence enabled calls weren't turning off lights
due to the fact that there were calls still up on the phone(though those calls had presence disabled).
2013-10-15 19:09:39 -07:00
4a172402d2
some refactoring and plumbing for 1.4
2013-10-16 02:24:32 +05:00
87831d1672
Document purpose of sofia profile aliasing
2013-10-09 20:09:14 +00:00
8ceb262b9e
FS-5806 please test HEAD
2013-09-20 10:20:01 -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
aa222c7aea
While sofia does garbage collection, in other locations we still call su_free after calls to sip_header_as_string. So adding it here too since we've finished using the full_contact variable.
2013-09-15 16:05:11 -07:00
5197c726be
add via_protocol to request so you can be informed if you wish to force expires due to tcp keepalives being on.
2013-09-12 15:24:19 -05:00
6b6bceb730
mitigate zoiper endless loop bug by not sending notify with expires: 0 when zoiper is detected
2013-09-12 20:21:41 +05:00
ce4cfc97c4
FS-5772 improve odds a little more but the phone is broken here
2013-09-11 02:55:31 +05:00
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
149acf0c11
oops
2013-09-09 22:53:54 +05:00
7b03cb511c
FS-5717 --resolve
2013-09-09 20:08:48 +05:00
c4add0663c
FS-5762 FS-5765 --resolve
2013-09-06 23:28:46 +05:00
af85dcc5c0
FS-5757
2013-09-06 04:48:56 +05:00
21f1743b30
FS-5757 take 200
2013-09-06 01:52:11 +05:00
2967c97d11
FS-5757 --resolve
2013-09-05 20:35:31 +05:00
9337413715
FS-5759 --resolve regression from e45774e764 re: FS-5620
2013-09-05 05:39:03 +05:00
ae171ba53a
FS-5757 revert
2013-09-04 22:15:54 +05:00
5bf27df0a4
FS-5752 --resolve
2013-09-04 21:05:13 +05:00
56e617324b
FS-5757 --resolve
2013-09-04 20:30:25 +05:00
787b7a73e2
FS-5748 --resolve regression from FS-5665 511efc5cf0
2013-09-04 01:08:00 +05:00
8f0dbcaaf8
FS-4757 this should fix it but I am not happy with the way this is done but since polycom is one of the only phones that does this right I have no other choice
2013-08-29 22:23:08 +05:00
2e2e4caa45
FS-5737 revert
2013-08-29 22:20:00 +05:00
0d26dbb318
FS-5737 --resolve
2013-08-29 21:03:01 +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
b4792e7a4e
recommit
2013-08-21 22:26:04 +05:00
0624f61d48
revert
2013-08-21 22:24:20 +05:00
e052d98a28
add isfocus to outbound invites
2013-08-21 22:21:26 +05:00
655fe8408d
fix rfc4579 method=bye handling
2013-08-21 22:00:47 +05:00
f284746703
fix rtp flags
2013-08-19 20:52:34 +08:00
8e45f23d02
FS-5709 --resolve
2013-08-17 02:16:28 +05:00
b76c39ade1
FS-5700 --resolve
2013-08-15 21:19:57 +05:00
129a509cfb
FS-5648 --resolve
2013-08-14 09:48:05 -05:00
8859df6789
FS-5683 --resolve
2013-08-14 02:24:53 +05:00
ae64753112
FS-5698 --resolve
2013-08-14 02:08:54 +05:00
3ec15cce68
doh blonde moment
2013-08-12 17:27:31 -05:00
56d1d07d61
url_decode the callee_name
2013-08-12 15:19:36 -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
4eee5aee8c
FS-5682 --resolve new syntax <action application="set" data="sip_multipart=application/pidf+xml:1234"/> use push instead of set to add more to the same channel
2013-08-08 22:55:13 +05:00
1b01e9f01a
FS-5682 revert
2013-08-08 22:55:13 +05:00
eead164cc2
FS-5682
2013-08-08 05:15:10 +05:00
11d6c0877a
FS-5665 --resolve this should push the extra ones into the variable and convert it to an array
2013-08-07 20:59:08 +05:00
511efc5cf0
FS-5652 --resolve
2013-07-29 16:55:16 -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
569dd9c540
FS-5634 --resolve this patch should fix it but comment it out instead in the old version to get the same effect
2013-07-24 11:23:54 -05:00
3ddbad4dd3
let sofia recover work in some tcp situations
2013-07-23 13:52:49 -05:00
f08c3309f6
FS-5621
2013-07-20 14:07:01 -05:00
e45774e764
FS-5620 --resolve
2013-07-19 21:51:18 -05:00
d88b2ed793
FS-5498 --close remove debug line
2013-07-20 02:17:59 +08:00
a6bffd38cb
FS-5498
2013-07-19 12:03:51 -05:00
663c180cd9
FS-5615 --resolve
2013-07-17 17:08:03 -05:00
c98b01a205
make display in contact compat with broken aastra
2013-07-17 16:35:24 -05:00
9f52531630
FS-5613 --resolve only alphanumeric plus - _ . are now allowed
2013-07-17 09:21:28 -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
90021b2953
mod_sofia: Add missing format string to switch_event_add_body() call
...
Fixes GCC format string warning/error.
Signed-off-by: Stefan Knoblich <stkn@openisdn.net>
2013-07-11 01:15:16 +02:00
be98eb34ab
FS-5559 --resolve
2013-07-10 11:49:58 -05:00
fd1893197e
FS-5586 --resolve
2013-07-08 07:55:50 -05:00
51fd5a6cee
--resolve FS-5426
2013-07-03 14:13:05 -05:00
bf89fbd81d
--resolve FS-5432
2013-07-03 14:13:05 -05:00
dac93d7936
FS-5527 --resolve
2013-06-28 10:42:06 -05:00
5324063f44
Added support for blind transfer notification of negative responses
2013-06-26 23:17:49 -04:00
bf5fa172e5
FS-5547 --resolve
2013-06-26 10:47:40 -05:00
32adc789f6
make noexpandtab explicit in all vim modelines other than xml files
2013-06-25 11:50:17 -05:00
f35ea43b38
FS-5513 --resolve
2013-06-20 18:33:40 -05:00
4baf0276d6
FS-5513 --resolve
2013-06-18 21:16:56 -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
a55d70ed62
add kirk wireless servers to safe as they do not do rport
2013-05-31 11:54:05 -05:00
9ace496fe3
full_contact must be freed by sofia. There isn't a chance of this having blown anything up, but Coverity is right that this should be initialized with NULL to be safe against future refactors.
2013-05-18 19:13:18 -07:00
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
f718a0572f
Alice meet rabit hole. There is a leak here on every invite that comes back from a status 180 when the other side is not FS, but you have presence enabled. Thanks to Coverity scan for location this condition. This 'might' not have been leaking if sofia was automatically freeing this when the sofia object was destroyed, but all other locations where sip_header_as_string where called they were all followed by a su_free, so I assume not.
2013-05-18 16:35:03 -07:00
f519ad195a
If the mod_sofia global presense hash is null, then make sure to free the matches variable before going to done.
2013-05-18 16:05:40 -07:00
ea881b191f
media/srtp refactoring pass
2013-05-17 15:39:30 -05:00
8d7aa3ff65
FS-5441: --resolve
2013-05-17 12:39:17 -07:00
1eae2a1941
FS-5410 --resolve
2013-05-13 13:15:33 -05:00
bba4540fe5
add context
2013-05-10 16:31:12 -04:00
bb6b02f2d6
FS-5403 --resolve
2013-05-10 09:59:02 -05:00
16690e491d
FS-5399 --resolve
2013-05-09 14:05:09 -05:00
01a748869d
regression parsing rtp-timeout-sec from the profile
2013-05-07 08:50:02 -05:00
283e5f5f6a
FS-5365 --resolve
2013-05-03 13:14:01 -05:00
44840fc97a
fix regression in sofia profile param for t38-passthru
2013-05-02 22:36:39 -05:00
9ef0323499
FS-5355 --resolve
2013-04-29 10:09:14 -05:00
c4e7c302f9
FS-5351 --resolve
2013-04-27 08:19:18 -05:00
5b9318c109
FS-4990 change uuid_debug_audio to uuid_debug_media and add video params
2013-04-27 06:47:06 +08:00
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
d5ec949142
Add application layer keepalive options to sofia and expose config options
2013-04-25 13:34:07 -05:00
eeeb4fa445
FS-5340 --resolve
2013-04-24 16:35:58 -05:00
e1c325dcb5
FS-5341 --resolve
2013-04-23 14:49:07 -05:00
ec43377df7
FS-5339 --resolve
2013-04-23 10:26:16 -05:00
b0cdad5d5d
add geolocation header parser
2013-04-18 16:44:40 -05:00
c8a6c99f84
FS-5322 --resolve
2013-04-18 16:27:45 -05:00
54d47599e9
FS-5318 update
2013-04-17 17:51:11 -05:00
cb9d41dd0e
FS-5318 --resolve
2013-04-17 15:59:48 -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
0fe47df2d7
don't send presence for hangup if you never sent it for early in presence_disable_early mode, it causes issues in yealink
2013-04-09 22:32:44 -05:00
c9a49b0ff3
debug
2013-04-09 21:42:32 -05:00
a67505ba74
FS-4934
2013-04-09 21:38:11 -05:00
cbdf2ceaac
FS-5265
2013-04-08 16:30:00 -05:00
965047615c
FS-5279 --resolve some mods made to the patch
2013-04-08 11:48:20 -05:00
1e251568cc
FS-5265
2013-04-06 16:32:01 -05:00
e245e81c3b
sla barg cid regression
2013-04-05 16:41:51 -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
52c1d3c76b
FS-5265 --resolve
2013-04-04 12:35:22 -05:00
e35b087f51
remove depricated vars causing issues
2013-04-02 21:12:14 -05:00
69c3c7d8e2
FS-5257 --resolve you seem to be setting sip_secure_media on an inbound leg, this is not correct. This patch should auto-correct that but also beware that the variable has changed to rtp_secure_media after this commit
2013-04-02 20:05:46 -05:00
Michael Jerris