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"/>
* 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.
when video floor is locked by a member, changing audio floor on del_member
will cause the video floor lock cleared unexpectedly, this commit fixes that.
When we specifically release all limits on a channel we destroy the
hash table stored in the "limit_hash" private channel data but we
don't destroy the private data as it will be reclaimed as part of the
session. If limit increment is called after the limit release we can
reuse that channel private, but we need to check whether the hash
table is null first. Fortunately this makes the code look better
anyway.
FS-6775 #resolve
FS-6783 #resolve
