Merged revisions 81331 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ 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. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@81332 f38db490-d61c-443f-a65b-d21fe96a405b
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1 changed files with 4 additions and 7 deletions
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@ -16427,13 +16427,10 @@ static void check_rtp_timeout(struct sip_pvt *dialog, time_t t)
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usleep(1);
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sip_pvt_lock(dialog);
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}
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if (!(ast_rtp_get_bridged(dialog->rtp))) {
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ast_log(LOG_NOTICE, "Disconnecting call '%s' for lack of RTP activity in %ld seconds\n",
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dialog->owner->name, (long) (t - dialog->lastrtprx));
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/* Issue a softhangup */
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ast_softhangup_nolock(dialog->owner, AST_SOFTHANGUP_DEV);
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} else
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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));
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ast_log(LOG_NOTICE, "Disconnecting call '%s' for lack of RTP activity in %ld seconds\n",
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dialog->owner->name, (long) (t - dialog->lastrtprx));
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/* Issue a softhangup */
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ast_softhangup_nolock(dialog->owner, AST_SOFTHANGUP_DEV);
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ast_channel_unlock(dialog->owner);
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/* forget the timeouts for this call, since a hangup
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has already been requested and we don't want to
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