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
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.
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.
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>
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.
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.
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
Jon Bergli Heier