[asterisk-commits] rmudgett: branch group/issue8824 r145527 - /team/group/issue8824/configs/

SVN commits to the Asterisk project asterisk-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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