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automerge commit
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.2-netsec@25055 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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@@ -1233,7 +1233,6 @@ void setup_rtp_connection(unsigned call_reference, const char *remoteIp, int rem
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*/
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void connection_made(unsigned call_reference, const char *token)
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{
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struct ast_channel *c = NULL;
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struct oh323_pvt *pvt;
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if (h323debug)
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@@ -1435,7 +1434,6 @@ int setup_outgoing_call(call_details_t *cd)
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*/
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void chan_ringing(unsigned call_reference, const char *token)
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{
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struct ast_channel *c = NULL;
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struct oh323_pvt *pvt;
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if (h323debug)
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@@ -2281,7 +2279,7 @@ static char *convertcap(int cap)
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}
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}
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static int oh323_set_rtp_peer(struct ast_channel *chan, struct ast_rtp *rtp, struct ast_rtp *vrtp, int codecs)
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static int oh323_set_rtp_peer(struct ast_channel *chan, struct ast_rtp *rtp, struct ast_rtp *vrtp, int codecs, int nat_active)
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{
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/* XXX Deal with Video */
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struct oh323_pvt *pvt;
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@@ -2310,7 +2308,7 @@ static struct ast_rtp_protocol oh323_rtp = {
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.type = type,
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.get_rtp_info = oh323_get_rtp_peer,
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.get_vrtp_info = oh323_get_vrtp_peer,
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.set_rtp_peer= oh323_set_rtp_peer,
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.set_rtp_peer = oh323_set_rtp_peer,
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};
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int load_module()
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