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
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.
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"/>
This patch allows conference flags to be set dynamically from the
dial plan by either passing them to the conference application in
the +flags{ } string or by setting the "conference_flags" dial plan
variable.
The +flags{ } string is currently used to set *user* flags only.
This patch changes this by allowing the +flags{ } string to contain
conference related flags as well (for example wait_mod). It shouldn't
be a problem to pass both types of flags via +flags{ } as long as
the user and conference flag names are kept unique.
FS-5099 #resolve
Freeswitch tries to fix timing issues (wrong ptime) on re-invite the same way
it does for the initial invite. This results in small audio glitches, while it
sends a couple of packets with different ptime, before the timing detection
logic figures out the remote (broken) endpoint true ptime.
In order to avoid unnecessary timing changes, this patch overwrites the
advertised ptime from known broken endpoints with the ptime, which was detected
by freeswitch. It does this by checking if the sip_h_X-Broken-PTIME (1.2.x) or
rtp_h_X-Broken-PTIME (master) variables are set.
FS-6644 #resolve
* 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.
