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authorMark Michelson <mmichelson@digium.com>2007-07-27 14:31:35 +0000
committerMark Michelson <mmichelson@digium.com>2007-07-27 14:31:35 +0000
commitd17d9c542b8ed74d7deff1ecdafd2a3c34fa1ad5 (patch)
tree24c77a32df88c8a053938c98f16ddcc18c1de2ad /channels
parent7bd9b905e4e532ac1164162c10de0f56c327794c (diff)
Merged revisions 77490 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r77490 | mmichelson | 2007-07-27 09:30:43 -0500 (Fri, 27 Jul 2007) | 3 lines "re-invite" was misspelled ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@77491 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Diffstat (limited to 'channels')
-rw-r--r--channels/chan_sip.c2
1 files changed, 1 insertions, 1 deletions
diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index bbc6ea5f3..659690ac7 100644
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -12872,7 +12872,7 @@ static void handle_response_invite(struct sip_pvt *p, int resp, char *rest, stru
if (p->vrtp)
ast_rtp_set_rtptimers_onhold(p->vrtp); /* Turn off RTP timers while we send fax */
} else if (p->t38.state == T38_DISABLED && bridgepeer && (bridgepvt->t38.state == T38_ENABLED)) {
- ast_log(LOG_WARNING, "RTP re-inivte after T38 session not handled yet !\n");
+ ast_log(LOG_WARNING, "RTP re-invite after T38 session not handled yet !\n");
/* Insted of this we should somehow re-invite the other side of the bridge to RTP */
/* XXXX Should we really destroy this session here, without any response at all??? */
sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);