[asterisk-dev] [asterisk-commits] may: trunk r369092 - in /trunk: ./ addons/chan_ooh323.c
Matthew Jordan
mjordan at digium.com
Tue Jun 19 20:47:35 CDT 2012
It appears as if this patch broke the build on trunk:
http://bamboo.asterisk.org/browse/TESTING-ASTERISKTRUNK-387/
chan_ooh323.c: In function ?do_monitor?:
chan_ooh323.c:3752: error: dereferencing pointer to incomplete type
make[2]: *** [chan_ooh323.o] Error 1
make[1]: *** [addons] Error 2
gmake: *** [_cleantest_all] Error 2
chan_ooh323.c: In function ?do_monitor?:
chan_ooh323.c:3752: error: dereferencing pointer to incomplete type
make[2]: *** [chan_ooh323.o] Error 1
make[1]: *** [addons] Error 2
gmake: *** [_cleantest_all] Error 2
--
Matthew Jordan
Digium, Inc. | Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com & http://asterisk.org
----- Original Message -----
> From: "SVN commits to the Asterisk project" <asterisk-commits at lists.digium.com>
> To: asterisk-commits at lists.digium.com, svn-commits at lists.digium.com
> Sent: Tuesday, June 19, 2012 6:36:47 PM
> Subject: [asterisk-commits] may: trunk r369092 - in /trunk: ./ addons/chan_ooh323.c
>
> Author: may
> Date: Tue Jun 19 18:36:43 2012
> New Revision: 369092
>
> URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=369092
> Log:
> check rtptimeouts in ooh323 channels as per config file
> (rtp voice, video, udptl except rtcp)
>
> (closes issue ASTERISK-19179)
> Reported by: TSAREGORODTSEV Yury
> Patches:
> 19179-ooh323-ast10.patch
> ........
>
> Merged revisions 369091 from
> http://svn.asterisk.org/svn/asterisk/branches/10
>
> Modified:
> trunk/ (props changed)
> trunk/addons/chan_ooh323.c
>
> Propchange: trunk/
> ------------------------------------------------------------------------------
> Binary property 'branch-10-merged' - no diff available.
>
> Modified: trunk/addons/chan_ooh323.c
> URL:
> http://svnview.digium.com/svn/asterisk/trunk/addons/chan_ooh323.c?view=diff&rev=369092&r1=369091&r2=369092
> ==============================================================================
> --- trunk/addons/chan_ooh323.c (original)
> +++ trunk/addons/chan_ooh323.c Tue Jun 19 18:36:43 2012
> @@ -1151,6 +1151,8 @@
>
> if (p) {
> ast_mutex_lock(&p->lock);
> +
> + p->lastrtptx = time(NULL);
>
> if (f->frametype == AST_FRAME_MODEM) {
> ast_debug(1, "Send UDPTL %d/%d len %d for %s\n",
> @@ -3735,6 +3737,24 @@
> h323_next = h323->next;
>
> /* TODO: Need to add rtptimeout keepalive support */
> +
> + if (h323->rtp && h323->rtptimeout && h323->lastrtptx &&
> + h323->lastrtptx + h323->rtptimeout < t) {
> + ast_rtp_instance_sendcng(h323->rtp, 0);
> + h323->lastrtptx = time(NULL);
> + }
> +
> + if (h323->rtp && h323->owner && h323->rtptimeout &&
> + h323->lastrtprx &&
> + h323->lastrtprx + h323->rtptimeout < t) {
> + if (!ast_channel_trylock(h323->owner)) {
> + ast_softhangup_nolock(h323->owner, AST_SOFTHANGUP_DEV);
> + ast_log(LOG_NOTICE, "Disconnecting call '%s' for lack of RTP
> activity in %ld seconds\n", h323->owner->name, (long) (t -
> h323->lastrtprx));
> + ast_channel_unlock(h323->owner);
> + }
> +
> + }
> +
> if (ast_test_flag(h323, H323_NEEDDESTROY)) {
> ooh323_destroy (h323);
> } /* else if (ast_test_flag(h323, H323_NEEDSTART) &&
> h323->owner) {
> @@ -4610,12 +4630,14 @@
> switch (ast_channel_fdno(ast)) {
> case 0:
> f = ast_rtp_instance_read(p->rtp, 0); /* RTP Audio */
> + p->lastrtprx = time(NULL);
> break;
> case 1:
> f = ast_rtp_instance_read(p->rtp, 1); /* RTCP Control Channel */
> break;
> case 2:
> f = ast_rtp_instance_read(p->vrtp, 0); /* RTP Video */
> + p->lastrtprx = time(NULL);
> break;
> case 3:
> f = ast_rtp_instance_read(p->vrtp, 1); /* RTCP Control Channel for
> video */
> @@ -4626,6 +4648,7 @@
> ast_debug(1, "Got UDPTL %d/%d len %d for %s\n",
> f->frametype, f->subclass.integer, f->datalen,
> ast_channel_name(ast));
> }
> + p->lastrtprx = time(NULL);
> break;
>
> default:
>
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-commits mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-commits
>
More information about the asterisk-dev
mailing list