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Add support for accepting and sending T.38 in the initial INVITE.

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(closes issue #9402)
Reported by: thdei


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@91439 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Joshua Colp
2007-12-06 16:14:26 +00:00
parent 95dce3bbba
commit 4edfc25a1f
1 changed files with 18 additions and 0 deletions
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18
channels/chan_sip.c
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@@ -4052,6 +4052,10 @@ static struct ast_channel *sip_new(struct sip_pvt *i, int state, const char *tit
if (i->rtp)
ast_jb_configure(tmp, &global_jbconf);
/* If the INVITE contains T.38 SDP information set the proper channel variable so a created outgoing call will also have T.38 */
if (i->udptl && i->t38.state == T38_PEER_DIRECT)
pbx_builtin_setvar_helper(tmp, "_T38CALL", "1");
/* Set channel variables for this call from configuration */
for (v = i->chanvars ; v ; v = v->next)
pbx_builtin_setvar_helper(tmp, v->name, v->value);
@@ -12175,6 +12179,20 @@ static void handle_response_invite(struct sip_pvt *p, int resp, char *rest, stru
if (p->owner && !ast_test_flag(req, SIP_PKT_IGNORE))
ast_queue_control(p->owner, AST_CONTROL_CONGESTION);
ast_set_flag(&p->flags[0], SIP_NEEDDESTROY);
} else if (p->udptl && p->t38.state == T38_LOCAL_DIRECT) {
/* We tried to send T.38 out in an initial INVITE and the remote side rejected it,
right now we can't fall back to audio so totally abort.
*/
p->t38.state = T38_DISABLED;
/* Try to reset RTP timers */
ast_rtp_set_rtptimers_onhold(p->rtp);
ast_log(LOG_ERROR, "Got error on T.38 initial invite. Bailing out.\n");
/* The dialog is now terminated */
if (p->owner && !ast_test_flag(req, SIP_PKT_IGNORE))
ast_queue_control(p->owner, AST_CONTROL_CONGESTION);
ast_set_flag(&p->flags[0], SIP_NEEDDESTROY);
sip_alreadygone(p);
} else {
/* We can't set up this call, so give up */
if (p->owner && !ast_test_flag(req, SIP_PKT_IGNORE))