5513 Commits

Author SHA1 Message Date
Anthony Minessale
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
Anthony Minessale
43733a6166 FS-6886 #comment addition of ignoring unhold as well 2014-10-02 15:48:29 -05:00
Spencer Thomason
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
Spencer Thomason
747322dcc6 Remove Contact header from BYE and CANCEL requests.
Per rfc3261 the Contact header is not applicable and MUST not appear in
the request.

FS-5868 #resolve
2014-10-02 12:24:46 -07:00
Anthony Minessale
9e9175321a FS-6886 #resolve 2014-10-02 11:30:13 -05:00
Flavio Grossi
5653551904 FS-5106 fire an event when a sip client doesn't respond to option-ping
When all-reg-options-ping is enabled, this adds a new custom event to mod_sofia
(sofia::sip_user_state), which is fired when a client stops responding to such
ping packets (or when it is reachable again).

Add two needed new columns to the sip_registrations table:
  - ping_status, which is "Reachable" or "Unreachable" depending on the client
    status;
  - ping_count, which tracks the number of ping responses received and is used
    to provide some kind of hysteresis to avoid firing the event in case of
    transitory network failures.

Then ping_count is checked against two threshold values, sip-user-ping-min
and sip-user-ping-max in a similar fashion as the ping-{max,min} options for
the gateways. These two values are configurable in the profile's xml
configuration file.

Also, if unregister-on-options-fail is enabled, the client is unregistered
based on the number of OPTIONS failure which is also checked against the
sip-user-ping-{min,max} values.
2014-10-02 12:34:47 +02:00
Anthony Minessale
8258180735 start jb at one frame since it now has better adaptation 2014-10-01 18:21:50 -05:00
Anthony Minessale
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
Brian West
8e408e9abe FS-6865 #resolve add XMPP priority to dingaling 2014-10-01 10:40:57 -05:00
Brian West
644b41f792 FS-6874 #resolve 2014-09-30 17:05:06 -05:00
Anthony Minessale
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
Anthony Minessale
0150c862a2 FS-6854 #comment try this patch 2014-09-30 20:35:19 +05:00
Dušan Dragić
a94fbe8079 mod_gsmopen: add tab completion for api commands 2014-09-29 13:25:30 +02:00
Giovanni Maruzzelli
4ce990504e Merge pull request #52 in FS/freeswitch from ~DDRAGIC/freeswitch:gsmopen_feature_additions to master
* commit 'a9b2e061dcd1d95322d27e169ac2f0016aa628a3':
  mod_gsmopen: clean up "gsm list" output a little
  mod_gsmopen: convert reported RSSI from AT+CSQ to dBm.
  mod_gsmopen: get device manufacturer, model and firmware version info.
  mod_gsmopen: add support for reading own number from ON phonebook using AT+CNUM
  mod_gsmopen: add AT+COPS support to get operator name.
2014-09-26 10:17:14 -05:00
Giovanni Maruzzelli
9e3a375c36 Merge pull request #54 in FS/freeswitch from ~DDRAGIC/freeswitch:bugfix/FS-6820-mod_gsmopen-executing-gsm-reload to master
* commit '9423953e028f8dd319a790ba1e5fdca37ff0cb2f':
  FS-6820 mod_gsmopen: fix total interfaces count when executing gsm reload
2014-09-26 10:14:46 -05:00
Giovanni Maruzzelli
0d538cd7b1 Merge pull request #42 in FS/freeswitch from ~DDRAGIC/freeswitch:FS-6799_fix_msg_index_check to master
* commit '9cf72b541e8184b2911b0bd78f9aee71cd6d44b4':
  FS-6799 fix reading sms in index 0
2014-09-26 10:13:44 -05:00
Brian West
7c89c21153 FS-6860 #resolve this was fixed once but was lost in the last sync 2014-09-26 09:00:09 -05:00
Anthony Minessale
f7de058acd FS-6854 #resolve 2014-09-25 21:44:02 +05:00
Anthony Minessale
9e72c8477f fix possible buffer overrun in websocket uri and sync the ws.c between sofia and verto (missing code from last commit) 2014-09-24 01:09:44 +05:00
Dušan Dragić
a9b2e061dc mod_gsmopen: clean up "gsm list" output a little
Replace tabs with spaces and add two columns, operator and imei.
2014-09-21 20:14:13 +02:00
Dušan Dragić
4aa7c98d5a mod_gsmopen: convert reported RSSI from AT+CSQ to dBm.
Add to gsmopen_dump and events.
2014-09-21 20:14:12 +02:00
Dušan Dragić
13a595a15e mod_gsmopen: get device manufacturer, model and firmware version info. 2014-09-21 20:14:05 +02:00
Dušan Dragić
79d962f38e mod_gsmopen: add support for reading own number from ON phonebook using AT+CNUM 2014-09-21 20:04:04 +02:00
Nathan Neulinger
1f5bb3470d mod_skinny: avoid truncation of non-null-terminated strings in protocol 2014-09-17 11:13:15 -05:00
Anthony Minessale
b2917e06db improve ssl errors 2014-09-17 02:14:43 +05:00
Seven Du
36addd5b61 bytes is signed 2014-09-16 19:15:12 +08:00
Nathan Neulinger
04269fdf19 mod_skinny: additional logging 2014-09-15 16:42:31 -05:00
Dušan Dragić
f262dbce94 FS-6821 mod_gsmopen: fix interface name in log
Fix interface name for logs emitted from mod_gsmopen.cpp during startup
2014-09-14 13:06:31 +02:00
Dušan Dragić
9423953e02 FS-6820 mod_gsmopen: fix total interfaces count when executing gsm reload 2014-09-14 12:24:19 +02:00
Anthony Minessale
efe0ebd318 FS-6818 #resolve 2014-09-12 18:49:58 +05:00
Dušan Dragić
d5f9de4fa3 mod_gsmopen: add AT+COPS support to get operator name.
For now expose the info in gsmopen_dump and events.
2014-09-11 22:33:28 +02:00
Travis Cross
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
Anthony Minessale
7144b25254 obey sip_copy_custom_headers on bye 2014-09-12 00:37:19 +05:00
Anthony Minessale
77c99b6306 FS-6806 #resolve #comment off by 1 error in last fix 2014-09-10 20:32:36 +05:00
Nathan Neulinger
574d19e56e mod_skinny: fix behavior of transfer when target extension falls through to voicemail - keep bridge from dropping out during that operation 2014-09-09 15:58:56 -05:00
Antonio
69d5cda6d6 resolve FS-6809 2014-09-09 15:33:19 +02:00
Nathan Neulinger
8973ffcc35 mod_skinny: improvements to error handling/detection 2014-09-09 08:30:46 -05:00
Nathan Neulinger
3c7e7c757a mod_skinny: more logging during transfer operations 2014-09-08 17:35:30 -05:00
Mike Jerris
98c8a9b508 Merge pull request #38 in FS/freeswitch from ~ALEXDG/freeswitch-event-for-gateway-ping:master to master
* 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
Anthony Minessale
a73583b5f3 FS-6806 #resolve 2014-09-09 00:09:31 +05:00
Dušan Dragić
9cf72b541e FS-6799 fix reading sms in index 0 2014-09-07 16:43:00 +02:00
Seven Du
a845755ea8 http 1.1 keepalive support 2014-09-07 12:21:42 +08:00
Seven Du
4e07845f2d fix incorrect string termination
if read multi times when waiting for a slow client, then bytes is much shorter than datalen
so it could incorrectly terminate the string and data could be lost
2014-09-07 11:45:12 +08:00
Seven Du
c02b2427e8 refactor http parsing and prevent read body more than content-length 2014-09-06 19:35:05 +08:00
Seven Du
7be60474ab respond to OPTIONS and only allows GET and HEAD on static resources 2014-09-06 17:21:58 +08:00
Seven Du
f3616557b6 parse x-www-form-urlencoded post body 2014-09-06 17:21:58 +08:00
Seven Du
a9b91550e9 add HTTP Basic auth 2014-09-06 17:21:57 +08:00
Seven Du
7f8cc54cfb add basic http virtual host support and fix some leaks 2014-09-06 17:21:57 +08:00
Alexander Haugg
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
Travis Cross
5c29d8d4fa Show gateway uptime in seconds
In `sofia status gateway ...` let's show the uptime in seconds rather
than in microseconds.  We'll output the uptime in microseconds in
`xmlstatus` and we'll label it as such.
2014-09-04 05:39:26 +00:00