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-rw-r--r--configs/basic-pbx/modules.conf1
-rw-r--r--configs/samples/cdr.conf.sample2
-rw-r--r--configs/samples/confbridge.conf.sample6
-rw-r--r--configs/samples/config_test.conf.sample8
-rw-r--r--configs/samples/musiconhold.conf.sample23
-rw-r--r--configs/samples/pjsip.conf.sample28
-rw-r--r--configs/samples/voicemail.conf.sample3
7 files changed, 67 insertions, 4 deletions
diff --git a/configs/basic-pbx/modules.conf b/configs/basic-pbx/modules.conf
index 8de58bf2d..9711745ed 100644
--- a/configs/basic-pbx/modules.conf
+++ b/configs/basic-pbx/modules.conf
@@ -77,7 +77,6 @@ load = res_pjsip_exten_state.so
load = res_pjsip_header_funcs.so
load = res_pjsip_logger.so
load = res_pjsip_messaging.so
-load = res_pjsip_multihomed.so
load = res_pjsip_mwi_body_generator.so
load = res_pjsip_mwi.so
load = res_pjsip_nat.so
diff --git a/configs/samples/cdr.conf.sample b/configs/samples/cdr.conf.sample
index e175a2a76..1d0af7864 100644
--- a/configs/samples/cdr.conf.sample
+++ b/configs/samples/cdr.conf.sample
@@ -17,7 +17,7 @@
; party. Setting this to "yes" will make calls to extensions that don't answer
; and don't set a B side channel (such as by using the Dial application)
; receive CDR log entries. If this option is set to "no", then those log
-; entries will not be created. Unasnwered Calls which get offered to an
+; entries will not be created. Unanswered Calls which get offered to an
; outgoing line will always receive log entries regardless of this option, and
; that is the intended behaviour.
;unanswered = no
diff --git a/configs/samples/confbridge.conf.sample b/configs/samples/confbridge.conf.sample
index 0e07f6b16..265b95342 100644
--- a/configs/samples/confbridge.conf.sample
+++ b/configs/samples/confbridge.conf.sample
@@ -218,6 +218,12 @@ type=bridge
; Default is en (English).
;regcontext=conferences ; The name of the context into which to register conference names as extensions.
+;video_update_discard=2000 ; Amount of time (in milliseconds) to discard video update requests after sending a video
+ ; update request. Default is 2000. A video update request is a request for a full video
+ ; intra-frame. Clients can request this if they require a full frame in order to decode
+ ; the video stream. Since a full frame can be large limiting how often they occur can
+ ; reduce bandwidth usage at the cost of increasing how long it may take a newly joined
+ ; channel to receive the video stream.
; All sounds in the conference are customizable using the bridge profile options below.
; Simply state the option followed by the filename or full path of the filename after
diff --git a/configs/samples/config_test.conf.sample b/configs/samples/config_test.conf.sample
index 2fff45ece..b7cb21292 100644
--- a/configs/samples/config_test.conf.sample
+++ b/configs/samples/config_test.conf.sample
@@ -6,6 +6,10 @@
[global]
intopt=-1
uintopt=1
+timelenopt1=1ms
+timelenopt2=1s
+timelenopt3=1m
+timelenopt4=1h
doubleopt=0.1
sockaddropt=1.2.3.4:1234
boolopt=true
@@ -23,6 +27,10 @@ customopt=yes
[item]
intopt=-1
uintopt=1
+timelenopt1=1
+timelenopt2=1
+timelenopt3=1
+timelenopt4=1
doubleopt=0.1
sockaddropt=1.2.3.4:1234
boolopt=true
diff --git a/configs/samples/musiconhold.conf.sample b/configs/samples/musiconhold.conf.sample
index b2980fc1d..741bde603 100644
--- a/configs/samples/musiconhold.conf.sample
+++ b/configs/samples/musiconhold.conf.sample
@@ -97,3 +97,26 @@ directory=moh
;mode=custom
;directory=/var/lib/asterisk/mohmp3
;application=/site/sw/bin/madplay -Q -o raw:- --mono -R 8000 -a -12
+
+; By default, when res_musiconhold reloads or unloads, it sends a HUP signal
+; to custom applications (and all descendants), waits 100ms, then sends a
+; TERM signal, waits 100ms, then finally sends a KILL signal. An application
+; which is interacting with an external device and/or spawns children of its
+; own may not be able to exit cleanly in the default times, expecially if sent
+; a KILL signal, or if it's children are getting signals directly from
+; res_musiconhoild. To allow extra time, the 'kill_escalation_delay'
+; class option can be used to set the number of milliseconds res_musiconhold
+; waits before escalating kill signals, with the default being the current
+; 100ms. To control to whom the signals are sent, the "kill_method"
+; class option can be set to "process_group" (the default, existing behavior),
+; which sends signals to the application and its descendants directly, or
+; "process" which sends signals only to the application itself.
+
+;[sox_from_device]
+;mode=custom
+;directory=/var/lib/asterisk/mohmp3
+;application=/usr/bin/sox -q -t alsa -c 2 -r 48000 hw:1 -c 1 -r 8000 -t raw -s -
+; Wait 500ms before escalating kill signals
+;kill_escalation_delay=500
+; Send signals to just the child process instead of all descendants
+;kill_method=process
diff --git a/configs/samples/pjsip.conf.sample b/configs/samples/pjsip.conf.sample
index a992ff8b6..3c3e52a05 100644
--- a/configs/samples/pjsip.conf.sample
+++ b/configs/samples/pjsip.conf.sample
@@ -608,8 +608,15 @@
;direct_media_glare_mitigation=none ; Mitigation of direct media re INVITE
; glare (default: "none")
;direct_media_method=invite ; Direct Media method type (default: "invite")
-;connected_line_method=invite ; Connected line method type (default:
- ; "invite")
+;connected_line_method=invite ; Connected line method type.
+ ; When set to "invite", check the remote's
+ ; Allow header and if UPDATE is allowed, send
+ ; UPDATE instead of INVITE to avoid SDP
+ ; renegotiation. If UPDATE is not Allowed,
+ ; send INVITE.
+ ; If set to "update", send UPDATE regardless
+ ; of what the remote Allows.
+ ; (default: "invite")
;direct_media=yes ; Determines whether media may flow directly between
; endpoints (default: "yes")
;disable_direct_media_on_nat=no ; Disable direct media session refreshes when
@@ -774,6 +781,23 @@
; transfer (default: "yes"). The value "no" is useful
; for some SIP phones (Mitel/Aastra, Snom) which expect
; a sip/frag "200 OK" after REFER has been accepted.
+;notify_early_inuse_ringing = ; Whether to notifies dialog-info 'early'
+ ; on INUSE && RINGING state (default: "no").
+ ; The value "yes" is useful for some SIP phones
+ ; (Cisco SPA) to be able to indicate and pick up
+ ; ringing devices.
+;max_audio_streams= ; The maximum number of allowed negotiated audio streams
+ ; (default: 1)
+;max_video_streams= ; The maximum number of allowed negotiated video streams
+ ; (default: 1)
+;webrtc= ; When set to "yes" this also enables the following values that are needed
+ ; for webrtc: rtcp_mux, use_avpf, ice_support, and use_received_transport.
+ ; The following configuration settings also get defaulted as follows:
+ ; media_encryption=dtls
+ ; dtls_verify=fingerprint
+ ; dtls_setup=actpass
+ ; A dtls_cert_file and a dtls_ca_file still need to be specified.
+ ; Default for this option is "no"
;==========================AUTH SECTION OPTIONS=========================
;[auth]
diff --git a/configs/samples/voicemail.conf.sample b/configs/samples/voicemail.conf.sample
index f8221eebe..84e83a344 100644
--- a/configs/samples/voicemail.conf.sample
+++ b/configs/samples/voicemail.conf.sample
@@ -227,6 +227,9 @@ pagerdateformat=%A, %B %d, %Y at %r
;imapclosetimeout=60 ; The TCP close timeout (in seconds)
;imapreadtimeout=60 ; The TCP read timeout (in seconds)
;imapwritetimeout=60 ; The TCP write timeout (in seconds)
+;imap_poll_logout=no ; If pollmailboxes=yes, then specify whether need to
+ ; disconnect from the IMAP server after polling.
+ ; Default: no
; -----------------------------------------------------------------------------
;