[asterisk-commits] file: trunk r104533 - in /trunk: channels/chan_sip.c main/rtp.c
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Wed Feb 27 09:31:10 CST 2008
Author: file
Date: Wed Feb 27 09:31:09 2008
New Revision: 104533
URL: http://svn.digium.com/view/asterisk?view=rev&rev=104533
Log:
Fix T38 passthrough regression introduced by state changes.
(closes issue #12078)
Reported by: dimas
Patches:
v1-12078.patch uploaded by dimas (license 88)
(closes issue #12074)
Reported by: Ivan
Modified:
trunk/channels/chan_sip.c
trunk/main/rtp.c
Modified: trunk/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/trunk/channels/chan_sip.c?view=diff&rev=104533&r1=104532&r2=104533
==============================================================================
--- trunk/channels/chan_sip.c (original)
+++ trunk/channels/chan_sip.c Wed Feb 27 09:31:09 2008
@@ -14521,12 +14521,12 @@
if (IS_SIP_TECH(bridgepeer->tech)) {
bridgepvt = (struct sip_pvt*)(bridgepeer->tech_pvt);
if (bridgepvt->udptl) {
- if (p->t38.state == T38_PEER_REINVITE) {
+ if (p->t38.state == T38_ENABLED && bridgepvt->t38.state == T38_PEER_REINVITE) {
sip_handle_t38_reinvite(bridgepeer, p, 0);
ast_rtp_set_rtptimers_onhold(p->rtp);
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)) {
+ } else if (p->t38.state == T38_DISABLED && bridgepvt->t38.state == T38_ENABLED) {
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??? */
@@ -14544,10 +14544,6 @@
ast_debug(2, "Strange... The other side of the bridge is not a SIP channel\n");
change_t38_state(p, T38_DISABLED);
}
- }
- if ((p->t38.state == T38_LOCAL_REINVITE) || (p->t38.state == T38_LOCAL_DIRECT)) {
- /* If there was T38 reinvite and we are supposed to answer with 200 OK than this should set us to T38 negotiated mode */
- change_t38_state(p, T38_ENABLED);
}
if (!req->ignore && p->owner) {
@@ -20795,6 +20791,7 @@
ast_debug(3, "Sending reinvite on SIP '%s' - It's UDPTL soon redirected to IP %s:%d\n", p->callid, ast_inet_ntoa(p->udptlredirip.sin_addr), ntohs(p->udptlredirip.sin_port));
else
ast_debug(3, "Sending reinvite on SIP '%s' - It's UDPTL soon redirected to us (IP %s)\n", p->callid, ast_inet_ntoa(p->ourip.sin_addr));
+ change_t38_state(p, T38_LOCAL_REINVITE);
transmit_reinvite_with_sdp(p, TRUE, FALSE);
} else if (!ast_test_flag(&p->flags[0], SIP_PENDINGBYE)) {
if (flag)
Modified: trunk/main/rtp.c
URL: http://svn.digium.com/view/asterisk/trunk/main/rtp.c?view=diff&rev=104533&r1=104532&r2=104533
==============================================================================
--- trunk/main/rtp.c (original)
+++ trunk/main/rtp.c Wed Feb 27 09:31:09 2008
@@ -3359,7 +3359,8 @@
} else if ((fr->frametype == AST_FRAME_CONTROL) && !(flags & AST_BRIDGE_IGNORE_SIGS)) {
if ((fr->subclass == AST_CONTROL_HOLD) ||
(fr->subclass == AST_CONTROL_UNHOLD) ||
- (fr->subclass == AST_CONTROL_VIDUPDATE)) {
+ (fr->subclass == AST_CONTROL_VIDUPDATE) ||
+ (fr->subclass == AST_CONTROL_T38)) {
if (fr->subclass == AST_CONTROL_HOLD) {
/* If we someone went on hold we want the other side to reinvite back to us */
if (who == c0)
@@ -3597,7 +3598,8 @@
} else if ((fr->frametype == AST_FRAME_CONTROL) && !(flags & AST_BRIDGE_IGNORE_SIGS)) {
if ((fr->subclass == AST_CONTROL_HOLD) ||
(fr->subclass == AST_CONTROL_UNHOLD) ||
- (fr->subclass == AST_CONTROL_VIDUPDATE)) {
+ (fr->subclass == AST_CONTROL_VIDUPDATE) ||
+ (fr->subclass == AST_CONTROL_T38)) {
/* If we are going on hold, then break callback mode and P2P bridging */
if (fr->subclass == AST_CONTROL_HOLD) {
if (p0_callback)
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