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Small fix 

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@18797 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Olle Johansson 2006-04-10 13:19:36 +00:00
parent c3b3feab76
commit f74538df11
1 changed files with 4 additions and 1 deletions
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5
channels/chan_sip.c
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@ -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 */