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authorDavid Vossel <dvossel@digium.com>2011-07-06 17:39:36 +0000
committerDavid Vossel <dvossel@digium.com>2011-07-06 17:39:36 +0000
commita7c6f0445e7bbd1f7aea1687eae52c3ca7cd5756 (patch)
tree5b543ca42fd46fc57a7bcabb3d848d7349163333
parent6e62aa2c7d3d08413989e2797f861eb379b27b52 (diff)
Fixes newlines from being stripped from out of dialog sip MESSAGES.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@326544 65c4cc65-6c06-0410-ace0-fbb531ad65f3
-rw-r--r--channels/chan_sip.c6
1 files changed, 5 insertions, 1 deletions
diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 18eba2371..1f2c72379 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -16079,7 +16079,11 @@ static void receive_message(struct sip_pvt *p, struct sip_request *req, struct a
return;
}
- if (get_msg_text2(&buf, req, FALSE)) {
+ /* If this is an out of dialog msg, add back newlines, otherwise strip the new lines.
+ * In dialog msg's newlines are stripped to preserve the behavior of how Asterisk has worked
+ * in the past. If it is found later that new lines can be added into in dialog msgs as well,
+ * then change this. */
+ if (get_msg_text2(&buf, req, p->owner ? FALSE : TRUE)) {
ast_log(LOG_WARNING, "Unable to retrieve text from %s\n", p->callid);
transmit_response(p, "202 Accepted", req);
if (!p->owner)