[asterisk-commits] rmudgett: branch group/issue8824 r144304 - in /team/group/issue8824: channels...
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Wed Sep 24 15:21:51 CDT 2008
Author: rmudgett
Date: Wed Sep 24 15:21:51 2008
New Revision: 144304
URL: http://svn.digium.com/view/asterisk?view=rev&rev=144304
Log:
chan_misdn documentation/description changes
Modified:
team/group/issue8824/channels/chan_misdn.c
team/group/issue8824/channels/misdn/isdn_lib.c
team/group/issue8824/channels/misdn_config.c
team/group/issue8824/configs/misdn.conf.sample
team/group/issue8824/doc/tex/misdn.tex
Modified: team/group/issue8824/channels/chan_misdn.c
URL: http://svn.digium.com/view/asterisk/team/group/issue8824/channels/chan_misdn.c?view=diff&rev=144304&r1=144303&r2=144304
==============================================================================
--- team/group/issue8824/channels/chan_misdn.c (original)
+++ team/group/issue8824/channels/chan_misdn.c Wed Sep 24 15:21:51 2008
@@ -5920,7 +5920,8 @@
" jb - Set jitter buffer length, optarg is length\n"
" jt - Set jitter buffer upper threshold, optarg is threshold\n"
" jn - Disable jitter buffer\n"
- " n - disable DSP on channel, disables: Echocancel, DTMF Detection and Volume Control.\n"
+ " n - Disable mISDN DSP on channel.\n"
+ " Disables: echo cancel, DTMF detection, and volume control.\n"
" p - Caller ID presentation,\n"
" optarg is either 'allowed' or 'restricted'\n"
" s - Send Non-inband DTMF as inband\n"
Modified: team/group/issue8824/channels/misdn/isdn_lib.c
URL: http://svn.digium.com/view/asterisk/team/group/issue8824/channels/misdn/isdn_lib.c?view=diff&rev=144304&r1=144303&r2=144304
==============================================================================
--- team/group/issue8824/channels/misdn/isdn_lib.c (original)
+++ team/group/issue8824/channels/misdn/isdn_lib.c Wed Sep 24 15:21:51 2008
@@ -163,19 +163,24 @@
void get_show_stack_details(int port, char *buf)
{
- struct misdn_stack *stack=get_misdn_stack();
-
- for ( ; stack; stack=stack->next) {
- if (stack->port == port) break;
+ struct misdn_stack *stack = get_misdn_stack();
+
+ for (; stack; stack = stack->next) {
+ if (stack->port == port) {
+ break;
+ }
}
if (stack) {
- sprintf(buf, "* Port %d Type %s Prot. %s L2Link %s L1Link:%s Blocked:%d",
- stack->port, stack->nt ? "NT" : "TE", stack->ptp ? "PTP" : "PMP",
- stack->l2link ? "UP" : "DOWN", stack->l1link ? "UP" : "DOWN",
+ sprintf(buf, "* Port %2d Type %s Prot. %s L2Link %s L1Link:%s Blocked:%d",
+ stack->port,
+ stack->nt ? "NT" : "TE",
+ stack->ptp ? "PTP" : "PMP",
+ stack->l2link ? "UP " : "DOWN",
+ stack->l1link ? "UP " : "DOWN",
stack->blocked);
} else {
- buf[0]=0;
+ buf[0] = 0;
}
}
Modified: team/group/issue8824/channels/misdn_config.c
URL: http://svn.digium.com/view/asterisk/team/group/issue8824/channels/misdn_config.c?view=diff&rev=144304&r1=144303&r2=144304
==============================================================================
--- team/group/issue8824/channels/misdn_config.c (original)
+++ team/group/issue8824/channels/misdn_config.c Wed Sep 24 15:21:51 2008
@@ -318,13 +318,13 @@
{ "faxdetect_context", MISDN_CFG_FAXDETECT_CONTEXT, MISDN_CTYPE_STR, NO_DEFAULT, NONE,
"Context to jump into if we detect a fax. Don't set this if you want to stay in the current context." },
{ "l1watcher_timeout", MISDN_CFG_L1_TIMEOUT, MISDN_CTYPE_BOOLINT, "0", 4,
- "Watches the layer 1. If the layer 1 is down, it tries to\n"
- "\tget it up. The timeout is given in seconds. with 0 as value it\n"
- "\tdoes not watch the l1 at all\n"
- "\n"
- "\tThis option is only read at loading time of chan_misdn, which\n"
- "\tmeans you need to unload and load chan_misdn to change the value,\n"
- "\tan Asterisk restart should do the trick." },
+ "Monitors L1 of the port. If L1 is down it tries\n"
+ "\tto bring it up. The polling timeout is given in seconds.\n"
+ "\tSetting the value to 0 disables monitoring L1 of the port.\n"
+ "\n"
+ "\tThis option is only read at chan_misdn loading time.\n"
+ "\tYou need to unload and load chan_misdn to change the\n"
+ "\tvalue. An asterisk restart will also do the trick.\n" },
{ "overlapdial", MISDN_CFG_OVERLAP_DIAL, MISDN_CTYPE_BOOLINT, "0", 4,
"Enables overlap dial for the given amount of seconds.\n"
"\tPossible values are positive integers or:\n"
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=144304&r1=144303&r2=144304
==============================================================================
--- team/group/issue8824/configs/misdn.conf.sample (original)
+++ team/group/issue8824/configs/misdn.conf.sample Wed Sep 24 15:21:51 2008
@@ -76,19 +76,6 @@
bridging=no
-;
-; watches the L1s of every port. If one l1 is down it tries to
-; get it up. The timeout is given in seconds. with 0 as value it
-; does not watch the l1 at all
-;
-; default value: 0
-;
-; this option is only read at loading time of chan_misdn,
-; which means you need to unload and load chan_misdn to change the
-; value, an asterisk restart should do the trick
-;
-l1watcher_timeout=0
-
; stops dialtone after getting first digit on nt Port
;
; default value: yes
@@ -124,43 +111,15 @@
; users sections:
;
-; name your sections as you which but not "general" !
+; 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
; new free channel
;
-
; The default section is not a group section, it just contains config elements
; which are inherited by group sections.
;
-
-;------------------------------ 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".
-;-----------------------------------------------------------------------------------
-
[default]
; define your default context here
@@ -232,6 +191,19 @@
;
+; Monitors L1 of the port. If L1 is down it tries
+; to bring it up. The polling timeout is given in seconds.
+; Setting the value to 0 disables monitoring L1 of the port.
+;
+; default value: 0
+;
+; This option is only read at chan_misdn loading time.
+; You need to unload and load chan_misdn to change the
+; value. An asterisk restart will also do the trick.
+;
+l1watcher_timeout=0
+
+;
; 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.
@@ -399,15 +371,6 @@
; default value: no
;
;echocancel=no
-
-; Set this to no to disable echotraining. You can enter a number > 10
-; the value is a multiple of 0.125 ms.
-;
-; default value: no
-; yes = 2000
-; no = 0
-;
-echotraining=no
;
; chan_misdns jitterbuffer, default 4000
Modified: team/group/issue8824/doc/tex/misdn.tex
URL: http://svn.digium.com/view/asterisk/team/group/issue8824/doc/tex/misdn.tex?view=diff&rev=144304&r1=144303&r2=144304
==============================================================================
--- team/group/issue8824/doc/tex/misdn.tex (original)
+++ team/group/issue8824/doc/tex/misdn.tex Wed Sep 24 15:21:51 2008
@@ -220,7 +220,7 @@
You can only use "misdn send display" when an Asterisk channel is created and
isdn is in the correct state. "correct state" means that you have established a
-call to another phone (mustnot be isdn though).
+call to another phone (must not be isdn though).
Then you use it like this:
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