[asterisk-commits] file: branch group/vldtmf r40491 - in /team/group/vldtmf: ./ configs/

[asterisk-commits at lists.digium.com]

Author: file
Date: Fri Aug 18 22:59:04 2006
New Revision: 40491

URL: http://svn.digium.com/view/asterisk?rev=40491&view=rev
Log:
Add crazy documentation.

Modified:
    team/group/vldtmf/UPGRADE.txt
    team/group/vldtmf/configs/sip.conf.sample

Modified: team/group/vldtmf/UPGRADE.txt
URL: http://svn.digium.com/view/asterisk/team/group/vldtmf/UPGRADE.txt?rev=40491&r1=40490&r2=40491&view=diff
==============================================================================
--- team/group/vldtmf/UPGRADE.txt (original)
+++ team/group/vldtmf/UPGRADE.txt Fri Aug 18 22:59:04 2006
@@ -288,6 +288,12 @@
   option in sip.conf is removed to osp.conf as authpolicy. allowguest option
   in sip.conf cannot be set as osp anymore. 
 
+* The Asterisk RTP stack has been changed in regards to RFC2833 reception
+  and transmission. Packets will now be sent with proper duration instead of all
+  at once. If you are receiving calls from a pre-1.4 Asterisk installation you
+  will want to turn on the rfc2833compensate option. Without this option your
+  DTMF reception may act poorly.
+
 The Zap channel:
 
 * Support for MFC/R2 has been removed, as it has not been functional for some

Modified: team/group/vldtmf/configs/sip.conf.sample
URL: http://svn.digium.com/view/asterisk/team/group/vldtmf/configs/sip.conf.sample?rev=40491&r1=40490&r2=40491&view=diff
==============================================================================
--- team/group/vldtmf/configs/sip.conf.sample (original)
+++ team/group/vldtmf/configs/sip.conf.sample Fri Aug 18 22:59:04 2006
@@ -417,7 +417,7 @@
 ; subscribecontext	      subscribecontext
 ; videosupport		      videosupport
 ; maxcallbitrate	      maxcallbitrate
-;                             mailbox
+; rfc2833compensate           mailbox
 ;                             username
 ;                             template
 ;                             fromdomain
@@ -431,6 +431,7 @@
 ;                             rtpholdtimeout
 ;                             sendrpid
 ;                             outboundproxy
+;                             rfc2833compensate
 
 ;[sip_proxy]
 ; For incoming calls only. Example: FWD (Free World Dialup)
@@ -587,3 +588,9 @@
 				; Normally you do NOT need to set this parameter
 ;setvar=CUSTID=5678		; Channel variable to be set for all calls from this device
 
+;[pre14-asterisk]
+;type=friend
+;secret=digium
+;host=dynamic
+;rfc2833compensate=yes		; Compensate for pre-1.4 DTMF transmission from another Asterisk machine.
+				; You must have this turned on or DTMF reception will work improperly.