[asterisk-dev] [asterisk-commits] may: branch 1.8 r369090 - /branches/1.8/addons/chan_ooh323.c

[Matthew Jordan]

Similarly on 1.8:

http://bamboo.asterisk.org/browse/TESTING-ASTERISK18BRANCH-263/

cc1: warnings being treated as errors
chan_ooh323.c: In function ?do_monitor?:
chan_ooh323.c:3468: error: implicit declaration of function ?ast_channel_name?
chan_ooh323.c:3468: error: format ?%s? expects type ?char *?, but argument 6 has type ?int?
make[1]: *** [chan_ooh323.o] Error 1
gmake: *** [addons] Error 2
cc1: warnings being treated as errors
chan_ooh323.c: In function ?do_monitor?:
chan_ooh323.c:3468: error: implicit declaration of function ?ast_channel_name?
chan_ooh323.c:3468: error: format ?%s? expects type ?char *?, but argument 6 has type ?int?
make[1]: *** [chan_ooh323.o] Error 1
gmake: *** [addons] Error 2

Interestingly, Asterisk 10 compiled just fine.

--
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 6:28:13 PM
> Subject: [asterisk-commits] may: branch 1.8 r369090 -        /branches/1.8/addons/chan_ooh323.c
> 
> Author: may
> Date: Tue Jun 19 18:28:09 2012
> New Revision: 369090
> 
> URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=369090
> 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-ooh323-2.patch
> 
> 
> Modified:
>     branches/1.8/addons/chan_ooh323.c
> 
> Modified: branches/1.8/addons/chan_ooh323.c
> URL:
> http://svnview.digium.com/svn/asterisk/branches/1.8/addons/chan_ooh323.c?view=diff&rev=369090&r1=369089&r2=369090
> ==============================================================================
> --- branches/1.8/addons/chan_ooh323.c (original)
> +++ branches/1.8/addons/chan_ooh323.c Tue Jun 19 18:28:09 2012
> @@ -1154,6 +1154,8 @@
>  
>          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",
> @@ -3451,6 +3453,24 @@
>                          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", ast_channel_name(h323->owner), (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) {
> @@ -4277,12 +4297,14 @@
>          switch (ast->fdno) {
>          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 */
> @@ -4291,6 +4313,7 @@
>                  f = ast_udptl_read(p->udptl);                /* UDPTL t.38 data */
>                  if (gH323Debug) ast_debug(1, "Got UDPTL %d/%d len %d for %s\n",
>                                  f->frametype, f->subclass.integer, f->datalen, ast->name);
> +                p->lastrtprx = time(NULL);
>                  break;
>  
>          default:
> 
> 
> --
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