[svn-commits] rmudgett: branch group/issue8824 r145527 - /team/group/issue8824/configs/
SVN commits to the Digium repositories
svn-commits at lists.digium.com
Wed Oct 1 13:24:06 CDT 2008
Author: rmudgett
Date: Wed Oct 1 13:24:05 2008
New Revision: 145527
URL: http://svn.digium.com/view/asterisk?view=rev&rev=145527
Log:
Restored JITTER BUFFER CONFIGURATION to mISDN sample config file and removed trailing white space.
Modified:
team/group/issue8824/configs/misdn.conf.sample
Modified: team/group/issue8824/configs/misdn.conf.sample
URL: http://svn.digium.com/view/asterisk/team/group/issue8824/configs/misdn.conf.sample?view=diff&rev=145527&r1=145526&r2=145527
==============================================================================
--- team/group/issue8824/configs/misdn.conf.sample (original)
+++ team/group/issue8824/configs/misdn.conf.sample Wed Oct 1 13:24:05 2008
@@ -7,13 +7,13 @@
; for debugging and general setup, things that are not bound to port groups
;
-[general]
+[general]
;
; Sets the Path to the misdn-init.conf (for nt_ptp mode checking)
;
misdn_init=/etc/misdn-init.conf
-; set debugging flag:
+; set debugging flag:
; 0 - No Debug
; 1 - mISDN Messages and * - Messages, and * - State changes
; 2 - Messages + Message specific Informations (e.g. bearer capability)
@@ -26,8 +26,8 @@
-; set debugging file and flags for mISDNuser (NT-Stack)
-;
+; set debugging file and flags for mISDNuser (NT-Stack)
+;
; flags can be or'ed with the following values:
;
; DBGM_NET 0x00000001
@@ -57,7 +57,7 @@
ntdebugfile=/var/log/misdn-nt.log
-; some pbx systems do cut the L1 for some milliseconds, to avoid
+; some pbx systems do cut the L1 for some milliseconds, to avoid
; dropping running calls, we can set this flag to yes and tell
; mISDNuser not to drop the calls on L2_RELEASE
ntkeepcalls=no
@@ -82,7 +82,7 @@
;
stop_tone_after_first_digit=yes
-; whether to append overlapdialed Digits to Extension or not
+; whether to append overlapdialed Digits to Extension or not
;
; default value: yes
;
@@ -109,14 +109,40 @@
;
crypt_keys=test,muh
+;------------------------------ JITTER BUFFER CONFIGURATION --------------------------
+; jbenable = yes ; Enables the use of a jitterbuffer on the receiving side of a
+ ; SIP channel. Defaults to "no". An enabled jitterbuffer will
+ ; be used only if the sending side can create and the receiving
+ ; side can not accept jitter. The SIP channel can accept jitter,
+ ; thus a jitterbuffer on the receive SIP side will be used only
+ ; if it is forced and enabled.
+
+; jbforce = no ; Forces the use of a jitterbuffer on the receive side of a SIP
+ ; channel. Defaults to "no".
+
+; jbmaxsize = 200 ; Max length of the jitterbuffer in milliseconds.
+
+; jbresyncthreshold = 1000 ; Jump in the frame timestamps over which the jitterbuffer is
+ ; resynchronized. Useful to improve the quality of the voice, with
+ ; big jumps in/broken timestamps, usually sent from exotic devices
+ ; and programs. Defaults to 1000.
+
+; jbimpl = fixed ; Jitterbuffer implementation, used on the receiving side of a SIP
+ ; channel. Two implementations are currently available - "fixed"
+ ; (with size always equals to jbmaxsize) and "adaptive" (with
+ ; variable size, actually the new jb of IAX2). Defaults to fixed.
+
+; jblog = no ; Enables jitterbuffer frame logging. Defaults to "no".
+;-----------------------------------------------------------------------------------
+
; users sections:
-;
+;
; name your sections as you wish but not "general" or "default" !
; the sections are Groups, you can dial out in extensions.conf
-; with Dial(mISDN/g:extern/101) where extern is a section name,
-; chan_misdn tries every port in this section to find a
+; with Dial(mISDN/g:extern/101) where extern is a section name,
+; chan_misdn tries every port in this section to find a
; new free channel
-;
+;
; The default section is not a group section, it just contains config elements
; which are inherited by group sections.
;
@@ -141,7 +167,7 @@
;
; Either if we should produce DTMF Tones ourselves
-;
+;
senddtmf=yes
;
@@ -164,8 +190,8 @@
;
allowed_bearers=all
-; Prefixes for national and international, those are put before the
-; oad if an according dialplan is set by the other end.
+; Prefixes for national and international, those are put before the
+; oad if an according dialplan is set by the other end.
;
; default values: nationalprefix : 0
; internationalprefix : 00
@@ -181,7 +207,7 @@
rxgain=0
txgain=0
-; some telcos especially in NL seem to need this set to yes, also in
+; some telcos especially in NL seem to need this set to yes, also in
; switzerland this seems to be important
;
; default value: no
@@ -204,7 +230,7 @@
l1watcher_timeout=0
;
-; This option defines, if chan_misdn should check the L1 on a PMP
+; This option defines, if chan_misdn should check the L1 on a PMP
; before making a group call on it. The L1 may go down for PMP Ports
; so we might need this.
; But be aware! a broken or plugged off cable might be used for a group call
@@ -217,19 +243,19 @@
;
-; in PMP this option defines which cause should be sent out to
+; in PMP this option defines which cause should be sent out to
; the 3. caller. chan_misdn does not support callwaiting on TE
-; PMP side. This allows to modify the RELEASE_COMPLETE cause
+; PMP side. This allows to modify the RELEASE_COMPLETE cause
; at least.
;
reject_cause=16
;
-; Send Setup_Acknowledge on incoming calls anyway (instead of PROCEEDING),
-; this requests additional Infos, so we can waitfordigits
+; Send Setup_Acknowledge on incoming calls anyway (instead of PROCEEDING),
+; this requests additional Infos, so we can waitfordigits
; without much issues. This works only for PTP Ports
-;
+;
; default value: no
;
need_more_infos=no
@@ -251,10 +277,10 @@
method=standard
-; specify if chan_misdn should collect digits before going into the
+; specify if chan_misdn should collect digits before going into the
; dialplan, you can choose yes=4 Seconds, no, or specify the amount
; of seconds you need;
-;
+;
overlapdial=yes
;
@@ -266,7 +292,7 @@
; localdialplan -> callerid
; cpndialplan -> connected party number
;
-; dialplan options:
+; dialplan options:
;
; 0 - unknown
; 1 - International
@@ -284,7 +310,7 @@
;
-; turn this to no if you don't mind correct handling of Progress Indicators
+; turn this to no if you don't mind correct handling of Progress Indicators
;
early_bconnect=yes
@@ -292,16 +318,16 @@
;
; turn this on if you like to send Tone Indications to a Incoming
; isdn channel on a TE Port. Rarely used, only if the Telco allows
-; you to send indications by yourself, normally the Telco sends the
+; you to send indications by yourself, normally the Telco sends the
; indications to the remote party.
-;
+;
; default: no
;
incoming_early_audio=no
; uncomment the following to get into s extension at extension conf
; there you can use DigitTimeout if you can't or don't want to use
-; isdn overlap dial.
+; isdn overlap dial.
; note: This will jump into the s exten for every exten!
;
; default value: no
@@ -309,7 +335,7 @@
;always_immediate=no
;
-; set this to yes if you want to generate your own dialtone
+; set this to yes if you want to generate your own dialtone
; with always_immediate=yes, else chan_misdn generates the dialtone
;
; default value: no
@@ -317,9 +343,9 @@
nodialtone=no
-; uncomment the following if you want callers which called exactly the
+; uncomment the following if you want callers which called exactly the
; base number (so no extension is set) jump to the s extension.
-; if the user dials something more it jumps to the correct extension
+; if the user dials something more it jumps to the correct extension
; instead
;
; default value: no
@@ -347,7 +373,7 @@
; from asterisks CALLERPRES function.
; s=0, p=0 -> callerid presented
; s=1, p=1 -> callerid restricted (the remote end does not see it!)
-;
+;
; default values s=-1, p=-1
presentation=-1
screen=-1
@@ -374,7 +400,7 @@
;
; chan_misdns jitterbuffer, default 4000
-;
+;
jitterbuffer=4000
;
@@ -384,7 +410,7 @@
;
-; change this to yes, if you want to bridge a mISDN data channel to
+; change this to yes, if you want to bridge a mISDN data channel to
; another channel type or to an application.
;
hdlc=no
@@ -392,8 +418,8 @@
;
; defines the maximum amount of incoming calls per port for
-; this group. Calls which exceed the maximum will be marked with
-; the channel variable MAX_OVERFLOW. It will contain the amount of
+; this group. Calls which exceed the maximum will be marked with
+; the channel variable MAX_OVERFLOW. It will contain the amount of
; overflowed calls
;
max_incoming=-1
@@ -405,7 +431,7 @@
max_outgoing=-1
[intern]
-; define your ports, e.g. 1,2 (depends on mISDN-driver loading order)
+; define your ports, e.g. 1,2 (depends on mISDN-driver loading order)
ports=1,2
; context where to go to when incoming Call on one of the above ports
context=Intern
@@ -417,16 +443,16 @@
; configs. For backwards compatibility you can still set ptp here.
;
ports=3
-
+
[first_extern]
; again port defs
ports=4
; again a context for incoming calls
context=Extern1
-; msns for te ports, listen on those numbers on the above ports, and
+; msns for te ports, listen on those numbers on the above ports, and
; indicate the incoming calls to asterisk
-; here you can give a comma separated list or simply an '*' for
-; any msn.
+; here you can give a comma separated list or simply an '*' for
+; any msn.
msns=*
; here an example with given msns
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