[asterisk-commits] trunk r18787 - /trunk/channels/chan_sip.c
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asterisk-commits at lists.digium.com
Mon Apr 10 03:30:40 MST 2006
Author: rizzo
Date: Mon Apr 10 05:30:38 2006
New Revision: 18787
URL: http://svn.digium.com/view/asterisk?rev=18787&view=rev
Log:
struct sip_request cleanup:
- remove a debug field that was read but never set, so it was basically
unused as well as the code testing it (also removed);
- make scalar fields contiguous so any array overflow will be
less harmful;
Modified:
trunk/channels/chan_sip.c
Modified: trunk/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/trunk/channels/chan_sip.c?rev=18787&r1=18786&r2=18787&view=diff
==============================================================================
--- trunk/channels/chan_sip.c (original)
+++ trunk/channels/chan_sip.c Mon Apr 10 05:30:38 2006
@@ -482,12 +482,11 @@
int len; /*!< Length */
int headers; /*!< # of SIP Headers */
int method; /*!< Method of this request */
+ int lines; /*!< SDP Content */
+ unsigned int flags; /*!< SIP_PKT Flags for this packet */
char *header[SIP_MAX_HEADERS];
- int lines; /*!< SDP Content */
char *line[SIP_MAX_LINES];
char data[SIP_MAX_PACKET];
- int debug; /*!< Debug flag for this packet */
- unsigned int flags; /*!< SIP_PKT Flags for this packet */
};
/*! \brief structure used in transfers */
@@ -10156,8 +10155,6 @@
} else {
/* Responses to OUTGOING SIP requests on INCOMING calls
get handled here. As well as out-of-call message responses */
- if (req->debug)
- ast_verbose("SIP Response message for INCOMING dialog %s arrived\n", msg);
if (resp == 200) {
/* Tags in early session is replaced by the tag in 200 OK, which is
the final reply to our INVITE */
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