[asterisk-commits] branch oej/02-labarea r22199 - in /team/oej/02-labarea: ./ channels/ configs/

[asterisk-commits at lists.digium.com]

Author: oej
Date: Sun Apr 23 08:20:19 2006
New Revision: 22199

URL: http://svn.digium.com/view/asterisk?rev=22199&view=rev
Log:
Reset, resolve, go

Modified:
    team/oej/02-labarea/   (props changed)
    team/oej/02-labarea/channels/chan_sip.c
    team/oej/02-labarea/configs/sip.conf.sample

Propchange: team/oej/02-labarea/
------------------------------------------------------------------------------
    automerge = http://edvina.net/training/

Propchange: team/oej/02-labarea/
------------------------------------------------------------------------------
--- svnmerge-integrated (original)
+++ svnmerge-integrated Sun Apr 23 08:20:19 2006
@@ -1,1 +1,1 @@
-/trunk:1-21291
+/trunk:1-22195

Modified: team/oej/02-labarea/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/team/oej/02-labarea/channels/chan_sip.c?rev=22199&r1=22198&r2=22199&view=diff
==============================================================================
--- team/oej/02-labarea/channels/chan_sip.c (original)
+++ team/oej/02-labarea/channels/chan_sip.c Sun Apr 23 08:20:19 2006
@@ -61,6 +61,8 @@
  * if it's a response to an outbound request, it's sent to handle_response().
  * If it is a request, handle_request sends it to one of a list of functions
  * depending on the request type - INVITE, OPTIONS, REFER, BYE, CANCEL etc
+ * sipsock_read locks the ast_channel if it exists (an active call) and
+ * unlocks it after we have processed the SIP message.
  *
  * A new INVITE is sent to handle_request_invite(), that will end up
  * starting a new channel in the PBX, the new channel after that executing
@@ -12035,6 +12037,7 @@
 }
 
 /*! \brief Read data from SIP socket
+\note sipsock_read locks the owner channel while we are processing the SIP message
 \return 1 on error, 0 on success
 \note Successful messages is connected to SIP call and forwarded to handle_request() 
 */
@@ -12092,6 +12095,8 @@
 	/* Process request, with netlock held */
 retrylock:
 	ast_mutex_lock(&netlock);
+
+	/* Find the active SIP dialog or create a new one */
 	p = find_call(&req, &sin, req.method);	/* returns p locked */
 	if (p) {
 		/* Go ahead and lock the owner if it has one -- we may need it */

Modified: team/oej/02-labarea/configs/sip.conf.sample
URL: http://svn.digium.com/view/asterisk/team/oej/02-labarea/configs/sip.conf.sample?rev=22199&r1=22198&r2=22199&view=diff
==============================================================================
--- team/oej/02-labarea/configs/sip.conf.sample (original)
+++ team/oej/02-labarea/configs/sip.conf.sample Sun Apr 23 08:20:19 2006
@@ -56,7 +56,8 @@
 				; Default is yes
 ;autodomain=yes			; Turn this on to have Asterisk add local host
 				; name and local IP to domain list.
-;pedantic=yes			; Enable slow, pedantic checking for Pingtel
+;pedantic=yes			; Enable checking of tags in headers, 
+				; international character conversions in URIs
 				; and multiline formatted headers for strict
 				; SIP compatibility (defaults to "no")