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Merged revisions 77490 via svnmerge from

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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: http://svn.digium.com/svn/asterisk/trunk@77491 f38db490-d61c-443f-a65b-d21fe96a405b
This commit is contained in:
mmichelson 2007-07-27 14:31:35 +00:00
parent 08f2930c10
commit 04a8d2dc08
1 changed files with 1 additions and 1 deletions
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2
channels/chan_sip.c
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@ -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);