From 768724f5e6aecda0f092a39728464508f738247e Mon Sep 17 00:00:00 2001
From: Mark Michelson <mmichelson@digium.com>
Date: Tue, 31 Jul 2007 15:22:32 +0000
Subject: [PATCH] Merged revisions 77824 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r77824 | mmichelson | 2007-07-31 10:21:22 -0500 (Tue, 31 Jul 2007) | 6 lines
This patch makes Asterisk send 100 Trying provisional responses upon receipt of re-invites. This makes it so that if there are two or more Asterisk
servers between endpoints, the Asterisk servers will not keep retransmitting the re-invites.
(closes issue #10274, reported by cstadlmann, patched by me with approval from file)
........
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@77825 65c4cc65-6c06-0410-ace0-fbb531ad65f3
---
channels/chan_sip.c | 2 ++
1 file changed, 2 insertions(+)
diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index af27a912e4..4770c9cee2 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -14720,6 +14720,8 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, int
case AST_STATE_UP:
ast_debug(2, "%s: This call is UP.... \n", c->name);
+ transmit_response(p, "100 Trying", req);
+
if (p->t38.state == T38_PEER_REINVITE) {
struct ast_channel *bridgepeer = NULL;
struct sip_pvt *bridgepvt = NULL;
--
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