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Merged revisions 81331 via svnmerge from

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https://origsvn.digium.com/svn/asterisk/branches/1.4

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r81331 | file | 2007-08-29 11:13:55 -0300 (Wed, 29 Aug 2007) | 4 lines

(closes issue #9690)
Reported by: mattv
Make rtp timeouts work even if two RTP streams are directly bridged in the RTP stack.

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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@81332 f38db490-d61c-443f-a65b-d21fe96a405b
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file 2007-08-29 14:16:07 +00:00
parent 800fbaee68
commit 16a860ddf9
1 changed files with 4 additions and 7 deletions
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11
channels/chan_sip.c
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@ -16427,13 +16427,10 @@ static void check_rtp_timeout(struct sip_pvt *dialog, time_t t)
usleep(1);
sip_pvt_lock(dialog);
}
if (!(ast_rtp_get_bridged(dialog->rtp))) {
ast_log(LOG_NOTICE, "Disconnecting call '%s' for lack of RTP activity in %ld seconds\n",
dialog->owner->name, (long) (t - dialog->lastrtprx));
/* Issue a softhangup */
ast_softhangup_nolock(dialog->owner, AST_SOFTHANGUP_DEV);
} else
ast_log(LOG_NOTICE, "'%s' will not be disconnected in %ld seconds because it is directly bridged to another RTP stream\n", dialog->owner->name, (long) (t - dialog->lastrtprx));
ast_log(LOG_NOTICE, "Disconnecting call '%s' for lack of RTP activity in %ld seconds\n",
dialog->owner->name, (long) (t - dialog->lastrtprx));
/* Issue a softhangup */
ast_softhangup_nolock(dialog->owner, AST_SOFTHANGUP_DEV);
ast_channel_unlock(dialog->owner);
/* forget the timeouts for this call, since a hangup
has already been requested and we don't want to