From 5873462c2e0bf74ecfd316db3ece34424870acdf Mon Sep 17 00:00:00 2001
From: Olle Johansson <oej@edvina.net>
Date: Sun, 23 Apr 2006 06:22:29 +0000
Subject: [PATCH] - Add doxygen documentation for sipsock_read locking -
 Improve documentation of pedantic   (related to issue #7016)

  From the air above Russia...


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@22163 65c4cc65-6c06-0410-ace0-fbb531ad65f3
---
 channels/chan_sip.c     | 5 +++++
 configs/sip.conf.sample | 3 ++-
 2 files changed, 7 insertions(+), 1 deletion(-)

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index cfbcb16086..276577dd16 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -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
@@ -11867,6 +11869,7 @@ static int handle_request(struct sip_pvt *p, struct sip_request *req, struct soc
 }
 
 /*! \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() 
 */
@@ -11924,6 +11927,8 @@ static int sipsock_read(int *id, int fd, short events, void *ignore)
 	/* 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 */
diff --git a/configs/sip.conf.sample b/configs/sip.conf.sample
index 3133feee0a..d391ec7c68 100644
--- a/configs/sip.conf.sample
+++ b/configs/sip.conf.sample
@@ -56,7 +56,8 @@ srvlookup=yes			; Enable DNS SRV lookups on outbound calls
 				; 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")
 
-- 
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