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Documentation update
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@20900 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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@@ -801,10 +801,10 @@ static struct sip_pvt {
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int rtpholdtimeout; /*!< RTP timeout when on hold */
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int rtpkeepalive; /*!< Send RTP packets for keepalive */
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enum transfermodes allowtransfer; /*! SIP Refer restriction scheme */
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enum subscriptiontype subscribed; /*!< Is this dialog a subscription? */
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int stateid;
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int laststate; /*!< Last known extension state */
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int dialogver;
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enum subscriptiontype subscribed; /*!< SUBSCRIBE: Is this dialog a subscription? */
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int stateid; /*!< SUBSCRIBE: ID for devicestate subscriptions */
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int laststate; /*!< SUBSCRIBE: Last known extension state */
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int dialogver; /*!< SUBSCRIBE: Version for subscription dialog-info */
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struct ast_dsp *vad; /*!< Voice Activation Detection dsp */
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@@ -815,7 +815,7 @@ static struct sip_pvt {
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struct ast_rtp *vrtp; /*!< Video RTP session */
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struct sip_pkt *packets; /*!< Packets scheduled for re-transmission */
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struct sip_history_head *history; /*!< History of this SIP dialog */
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struct ast_variable *chanvars; /*!< Channel variables to set for call */
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struct ast_variable *chanvars; /*!< Channel variables to set for inbound call */
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struct sip_pvt *next; /*!< Next dialog in chain */
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struct sip_invite_param *options; /*!< Options for INVITE */
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} *iflist = NULL;
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