Merged revisions 77490 via svnmerge from
authorMark Michelson <mmichelson@digium.com>
Fri, 27 Jul 2007 14:31:35 +0000 (14:31 +0000)
committerMark Michelson <mmichelson@digium.com>
Fri, 27 Jul 2007 14:31:35 +0000 (14:31 +0000)
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r77490 | mmichelson | 2007-07-27 09:30:43 -0500 (Fri, 27 Jul 2007) | 3 lines

"re-invite" was misspelled

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@77491 65c4cc65-6c06-0410-ace0-fbb531ad65f3

channels/chan_sip.c

index bbc6ea5..659690a 100644 (file)
@@ -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);