ffb93653ab
Merge pull request #859 in FS/freeswitch from ~SJTHOMASON/freeswitch:bugfix/FS-9183-call-with-t.38-reinvite-fails-after to master
...
* commit '5b71dffb8004c6e7045afeaf42c6a785c7496975':
FS-9183: [mod_sofia] Handle 415 Unsupported Media Type as 488
2016-08-17 10:59:17 -05:00
c278c9f63f
FS-9375 #resolve [DTMF not working on OPUS after Call Transfer ] #comment Can we try to reproduce with this version on all 3 boxes (not just the patch but the whole rev as-is)
2016-08-15 17:10:42 -05:00
2e3227b50f
FS-9422 #resolve [Freeswitch Exit/Crash on SDP Negotiation] #comment renegotiate-codec-on-hold renegotiate-codec-on-reinvite are both removed in this commit
2016-08-12 14:10:23 -05:00
5b71dffb80
FS-9183: [mod_sofia] Handle 415 Unsupported Media Type as 488
...
Handle 415 Unsupported Media Type the same as 488 Not Acceptable Here
after t.38 ReINVITE. This resolves an issue where the call would fail and
translates the response code to the more standard 488 Not Acceptable Here
allowing the call to continue in audio mode.
FS-9183 #resolve
2016-08-11 15:13:48 -07:00
589a0e682e
FS-9276 new feature proxy in-dialog calls sip notify and info similar to proxy hold
2016-08-01 11:44:12 +01:00
f34dac749e
revert FS-9368
2016-07-28 14:13:37 -05:00
e42997ef08
FS-9376 #comment please try this
2016-07-22 22:43:55 -05:00
c556e5269b
FS-9368
2016-07-21 14:32:04 -05:00
f43522877c
FS-9356 #resolve [DTMF not recognized when coming from a Cisco SIP trunk]
2016-07-18 11:11:32 -05:00
addf7555bf
FS-9277: sip info with record: on and off doesn't start and stop call recording sessions
2016-07-11 14:27:49 +01:00
3a3f456e88
FS-9316 #resolve [INVITE with empty SDP from Cisco VCS cannot setup video]
2016-07-06 11:36:54 -05:00
78c2ed0526
FS-9241 Use tls_public_url instead of tls_url in INVITE Contact when NAT is detected
2016-07-05 17:15:04 +02:00
5c9f98b016
FS-9297: [mod_sofia] fix multiple crashes from passing invalid null values in sofia.conf
2016-06-24 16:45:29 -05:00
05cc770818
FS-9192-proxy-hold: proxy hold when proxy media and proxy mode are disabled; its similiar to proxy-refer
2016-06-15 10:10:47 +01:00
18dbefa6da
FS-9214 regression
2016-06-13 12:46:42 -04:00
b9d31dc9b7
FS-9214: fix 3pcc behavior
...
Fix callflow issues with 3pcc=true and 3pcc=proxy and interactions with sip_wait_for_aleg_ack
removes passthrough of 183 on 3pcc=proxy (that was previously not functioning)
2016-06-07 11:43:08 -05:00
2766eaf49e
FS-9225: [mod_sofia] Allow to force SIP REGISTER Expires: to be within configured range.
...
Allow to force SIP REGISTER Expires: to be within MIN and MAX values.
Not just to force it to one specific value using "sip-force-expires".
New profile parameters:
"sip-force-expires-min"
"sip-force-expires-max"
2016-06-03 05:30:01 -07:00
a838c29e41
Merge pull request #862 in FS/freeswitch from ~LAZEDO/freeswitch:feature/FS-9188 to master
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* commit '08199210c1f548a1aa9a302c32eb89fabbf90301':
[mod_sofia] [FS-9188] added channel var to suppress auto-answer notify
2016-05-26 16:28:07 -05:00
d8164a95f2
Merge pull request #633 in FS/freeswitch from ~LAZEDO/freeswitch:FS-8652 to master
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* commit 'dc7753f51667e235ec7171a697f14989c9759e6f':
FS-8652 handle early-only param in replaces header
2016-05-23 14:27:27 -05:00
dc7753f516
FS-8652 handle early-only param in replaces header
2016-05-23 18:59:42 +01:00
08199210c1
[mod_sofia] [FS-9188] added channel var to suppress auto-answer notify
2016-05-20 10:49:40 +01:00
2b10500231
FS-9160 #resolve tweak sip_invite_failure_* chan vars for properly reporting last outbound call failure when there are multiple bridge attempts on a single call
2016-05-18 15:29:59 -05:00
465a0b6f1d
FS-9034 revert commit in sofia.c that prevents register in new thread
2016-05-17 18:40:27 +02:00
0a8e115b98
FS-9060: [mod_sofia] correct issues with hold and broken soa negotiations after performing a bypass media reinvite
2016-04-22 12:26:13 -05:00
252f0e2618
FS-9083 [mod-sofia] Pass On SIP headers from leg A to B
2016-04-19 10:28:12 +02:00
93d45b3bce
FS-9049
2016-04-10 11:39:28 -05:00
647cc257fd
FS-8913 #resolve [Problem with transfer when using bypass_media + SRTP + Inbound late negotiation]
2016-03-31 16:45:34 -05:00
9ec3253563
Merge pull request #771 in FS/freeswitch from ~SAFAROV/freeswitch2:FS-7125 to master
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* commit 'ff1c384e13f07afaac98334532667ebf8093a1f1':
FS-7125 Added sofia event "wrong_calls_state". This is for fail2ban logging
2016-03-29 14:06:54 -05:00
ff1c384e13
FS-7125 Added sofia event "wrong_calls_state". This is for fail2ban logging
2016-03-29 21:21:38 +03:00
0335cc3291
FS-8918 #resolve [10 Second timeout after Notify during Proxy refer.]
2016-03-28 13:14:38 -05:00
8423f28d43
FS-8841 #resolve [Debian - FS Hang ]
2016-03-17 18:48:59 -05:00
f96e016135
FS-6544
2016-02-05 11:56:44 -06:00
dd2ad0e300
FS-6833
2016-02-04 15:25:33 -06:00
93f0da09d3
Merge pull request #674 in FS/freeswitch from ~LAZEDO/freeswitch:feature/FS-8763 to master
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* commit 'f80d6cc5517c6886dc15dc446c0519811c55d04e':
FS-8763 [mod_sofia] set is_auth on switch_ivr_set_user result
2016-01-21 12:00:14 -06:00
cffb310f3f
FS-8759 #resolve [segfault on 1.7] #comment please update to master
2016-01-20 11:56:18 -06:00
f80d6cc551
FS-8763 [mod_sofia] set is_auth on switch_ivr_set_user result
2016-01-20 16:09:17 +00:00
e39605dc2c
FS-8715: [mod_sofia] make the oubound_proxy on the profile consistent with how we do the same thing on the gateway
2016-01-08 11:06:12 -06:00
6c17410c25
add parsing for userLocation header
2015-12-22 16:57:41 -06:00
0a4e394cc8
add parsing for userLocation header
2015-12-22 16:54:56 -06:00
e23aa12a2d
FS-8643: fix mod_sofia memory leaks
2015-12-10 22:42:55 -08:00
1868859638
FS-8573 #resolve [unidirectional audio after resuming from hold in bypass media mode]
2015-11-25 13:56:25 -06:00
de7a123c8b
FS-8536 #resolve [Send Keyframe when getting SIP INFO with picture_fast_update]
2015-11-13 14:15:39 -06:00
6b5b1a3d86
FS-8389: [build] Fix msvc 2015 build warnings
2015-10-26 13:45:48 -04:00
4e1ec89009
FS-6833 FS-6834 fix regression
2015-10-13 12:17:29 -05:00
aa7bc32375
FS-7834 #resolve [MOH doesn't work with inbound-bypass-media and resume-media-on-hold]
2015-10-13 11:51:51 -05:00
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
6837062563
FS-6833 add content-type header to ack with sdp
2015-10-09 10:28:02 -05:00
43ef01fbbe
correct version of proposed patch
2015-09-23 11:58:57 -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
331cde3c92
FS-8053: handle a=sendonly, a=sendrecv, a=recvonly to change who is sending video during a call
2015-09-09 18:10:52 -05:00
7e2082a55a
FS-8140 Fix user_name typo in sofia_handle_sip_i_invite
2015-09-09 10:33:23 +02:00
2feae3fc69
FS-6833 #comment please test this branch
2015-09-01 16:31:23 -05:00
7efc1f3bfb
FS-7135: [mod_sofia] fix response to re-invite with duplicate sdp (such as we get from seesion refresh) when soa is disabled to include an sdp. Fixes t.38 fax failure on session refresh
2015-08-27 11:42:40 -05:00
f929a9ccbf
FS-7966: we still need to do declarations first until we fully switch to 2015 and change the flags on other builds to not error on code before declaration
2015-08-21 18:47:58 -04:00
5e456fe9a6
FS-7966. First pass at moving to Visual Studio 2015.
...
The new C compiler breaks a lot of things. snprintf and timespec now exist, and redefining causes an error.
Many more things are warnings, so warnings-as-errors will fail - remove it from some projects for now.
V8: don't pass VS version to build batch file.
mod_sofia: Config has too-long if/elseif chain. Break this up to avoid "parser stack overflow; program too complex".
Add mod_conference.h to project and dir to includes.
2015-08-21 18:28:32 -04:00
7644b6834f
FS-7760 add some more params
2015-07-29 15:53:29 -05:00
2751039784
add dial-prefix and absolute-dial-string to the nightmare xml
2015-07-15 16:57:05 -05:00
e910e86bfb
FS-7760: revise channel fetch on nightmare transfer
2015-07-15 16:28:29 -05:00
58043483cd
FS-7773 [mod_sofia] transfer events #resolve
2015-07-10 19:46:10 +01:00
f0417cdf57
FS-7760 test_pflag not test_flag
2015-07-03 16:35:52 -05:00
b2b9555d34
FS-7760 #resolve
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added xml fetch for channels to externally support nightmare transfer
depends on channel-xml-fetch-on-nightmare-transfer profile param (default is disabled)
2015-07-03 19:28:17 +01:00
20be305b1c
Merge pull request #263 in FS/freeswitch from ~LAZEDO/freeswitch-fs-7463:FS-7463 to master
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* commit 'e12608bd11c62de715c1d6eecf05a8fe8b8becb0':
FS-7463 #resolve
2015-07-03 12:28:16 -05:00
9d40e1397d
FS-7641
2015-06-12 15:59:02 -05:00
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
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FS-6801 #resolve
2015-06-02 00:47:18 -04:00
e12608bd11
FS-7463 #resolve
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allow intercept of replaced call-id when processing the replaces header by providing a a-leg parameter in replaces header.
without this change one leg calls could not be picked because there is no bridge_uuid
2015-06-01 20:11:40 +01:00
bbad395866
Merge pull request #174 in FS/freeswitch from ~FLAVIO/freeswitch:origin/bugfix/FS-7196 to master
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* commit 'f135232ef5ab3f53f76cfa8ec2a1225f7700de90':
FS-7196: add recording_follow_transfer support for semi attended transfers
2015-06-01 13:52:05 -05:00
bd02432d03
Merge pull request #232 in FS/freeswitch from ~ITALOROSSI/freeswitch:bugfix/FS-7242-delayed-bye-after-REFER to master
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* commit 'beeea0172411f6274effdac8b90f970fa317b159':
FS-7242 Waiting for state CS_PARK when receiving REFER.
2015-06-01 13:10:43 -05:00
0c4f85111f
FS-7337: Add support for Remote-Party-ID header in UPDATE request.
2015-06-01 12:25:13 -05:00
200199e05f
CID:1023990: remove unused code
2015-05-28 12:47:23 -05:00
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
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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
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* 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
b33563138b
conditionally allow intercept of replaced call-id when processing replaces header
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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
beeea01724
FS-7242 Waiting for state CS_PARK when receiving REFER.
2015-04-07 22:11:27 -03:00
ed0a434b95
Fail the call with 502 bad gateway if the outbound leg returns a redirect with an invalid URI.
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FS-7385 #resolve
2015-03-25 12:46:59 -07:00
d562bc601a
Merge pull request #201 in FS/freeswitch from ~SJTHOMASON/freeswitch:bugfix/FS-7308-cannot-disable-sofia-sla-sql-query to master
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* 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
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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
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
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
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
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
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
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
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
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
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
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
7144b25254
obey sip_copy_custom_headers on bye
2014-09-12 00:37:19 +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
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
1dc44067cd
FS-6770 #resolve
2014-08-27 13:28:15 -05:00
64060c7dbd
Add sofia gateway parameter "destination-prefix"
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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
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
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
c58e64bf6d
add some more headers to sip messages
2014-07-16 00:55:46 +05:00
28e90addf2
FS-5501 fire event on profile start
2014-07-09 08:53:14 -05:00
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
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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
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
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
388e119e55
CID: 1214221,1214222
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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
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
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
0ba59aa6c2
Expanding on 052638186b to support boolean tel url params in mod_sofia
2014-04-25 13:49:59 -07:00
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
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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.
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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
Brian West