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authormmichelson <mmichelson@f38db490-d61c-443f-a65b-d21fe96a405b>2007-07-31 15:21:22 +0000
committermmichelson <mmichelson@f38db490-d61c-443f-a65b-d21fe96a405b>2007-07-31 15:21:22 +0000
commitf38be48a2f20583eee445c135693ef20855849cb (patch)
treee0eb0aec112fadfc42239f527f28352f21a4459b /channels
parent1b46005ab3e28acc2f38463545facab752e9eca6 (diff)
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: http://svn.digium.com/svn/asterisk/branches/1.4@77824 f38db490-d61c-443f-a65b-d21fe96a405b
Diffstat (limited to 'channels')
-rw-r--r--channels/chan_sip.c2
1 files changed, 2 insertions, 0 deletions
diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 763d0fc9b..a70d5e58a 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -13697,6 +13697,8 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, int
if (option_debug > 1)
ast_log(LOG_DEBUG, "%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;