Documentation update

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@20900 65c4cc65-6c06-0410-ace0-fbb531ad65f3

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 1e0fbab48b2fcdf8821a37b7cf32aa0261537c5d..039712e4ff35e02f0d2639b933b69661f02659cf 100644 (file)
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -801,10 +801,10 @@ static struct sip_pvt {
        int rtpholdtimeout;                     /*!< RTP timeout when on hold */
        int rtpkeepalive;                       /*!< Send RTP packets for keepalive */
        enum transfermodes allowtransfer;       /*! SIP Refer restriction scheme */
-       enum subscriptiontype subscribed;       /*!< Is this dialog a subscription?  */
-       int stateid;
-       int laststate;                          /*!< Last known extension state */
-       int dialogver;
+       enum subscriptiontype subscribed;       /*!< SUBSCRIBE: Is this dialog a subscription?  */
+       int stateid;                            /*!< SUBSCRIBE: ID for devicestate subscriptions */
+       int laststate;                          /*!< SUBSCRIBE: Last known extension state */
+       int dialogver;                          /*!< SUBSCRIBE: Version for subscription dialog-info */
        
        struct ast_dsp *vad;                    /*!< Voice Activation Detection dsp */
        
@@ -815,7 +815,7 @@ static struct sip_pvt {
        struct ast_rtp *vrtp;                   /*!< Video RTP session */
        struct sip_pkt *packets;                /*!< Packets scheduled for re-transmission */
        struct sip_history_head *history;       /*!< History of this SIP dialog */
-       struct ast_variable *chanvars;          /*!< Channel variables to set for call */
+       struct ast_variable *chanvars;          /*!< Channel variables to set for inbound call */
        struct sip_pvt *next;                   /*!< Next dialog in chain */
        struct sip_invite_param *options;       /*!< Options for INVITE */
 } *iflist = NULL;