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authoroej <oej@f38db490-d61c-443f-a65b-d21fe96a405b>2006-04-10 13:19:36 +0000
committeroej <oej@f38db490-d61c-443f-a65b-d21fe96a405b>2006-04-10 13:19:36 +0000
commitf61aeffd402993cbbf0ef5027ed15f1acf054eba (patch)
treed6996ee2777069066e92ac4ec1f10fe14da922a9 /channels
parent3eca9bec9d8b4315f28cffb2b7dfd49b15ae7c7f (diff)
Small fix
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@18797 f38db490-d61c-443f-a65b-d21fe96a405b
Diffstat (limited to 'channels')
-rw-r--r--channels/chan_sip.c5
1 files changed, 4 insertions, 1 deletions
diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 52aa62884..6a1c6c159 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -4730,7 +4730,7 @@ static int transmit_reinvite_with_sdp(struct sip_pvt *p)
if (sipdebug)
add_header(&req, "X-asterisk-info", "SIP re-invite (RTP bridge)");
if (recordhistory)
- append_history(p, "%s", "Re-invite sent");
+ append_history(p, "ReInv", "Re-invite sent");
add_sdp(&req, p);
/* Use this as the basis */
copy_request(&p->initreq, &req);
@@ -10151,6 +10151,9 @@ static void handle_response(struct sip_pvt *p, int resp, char *rest, struct sip_
} else {
/* Responses to OUTGOING SIP requests on INCOMING calls
get handled here. As well as out-of-call message responses */
+ if (ast_test_flag(req, SIP_PKT_DEBUG))
+ ast_verbose("SIP Response message for INCOMING dialog %s arrived\n", msg);
+
if (resp == 200) {
/* Tags in early session is replaced by the tag in 200 OK, which is
the final reply to our INVITE */