[asterisk-commits] oej: trunk r99653 - in /trunk: ./ channels/chan_sip.c
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Tue Jan 22 15:09:37 CST 2008
Author: oej
Date: Tue Jan 22 15:09:37 2008
New Revision: 99653
URL: http://svn.digium.com/view/asterisk?view=rev&rev=99653
Log:
Merged revisions 99652 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
........
r99652 | oej | 2008-01-22 21:56:09 +0100 (Tis, 22 Jan 2008) | 4 lines
Thanks to Russell's education I realize that BUFSIZ has changed since I learned the C language
over 20 years ago... Resetting chan_sip to the size of BUFSIZ that I expected in my old
head to avoid too heavy memory allocations on some systems.
........
Modified:
trunk/ (props changed)
trunk/channels/chan_sip.c
Change Statistics:
0 files changed
Propchange: trunk/
------------------------------------------------------------------------------
Binary property 'branch-1.4-merged' - no diff available.
Modified: trunk/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/trunk/channels/chan_sip.c?view=diff&rev=99653&r1=99652&r2=99653
==============================================================================
--- trunk/channels/chan_sip.c (original)
+++ trunk/channels/chan_sip.c Tue Jan 22 15:09:37 2008
@@ -185,6 +185,8 @@
#define TRUE 1
#endif
+#define SIPBUFSIZE 512
+
#define XMIT_ERROR -2
/* #define VOCAL_DATA_HACK */
@@ -1103,9 +1105,9 @@
char referred_by[AST_MAX_EXTENSION]; /*!< Place to store REFERRED-BY extension */
char referred_by_name[AST_MAX_EXTENSION]; /*!< Place to store REFERRED-BY extension */
char refer_contact[AST_MAX_EXTENSION]; /*!< Place to store Contact info from a REFER extension */
- char replaces_callid[BUFSIZ]; /*!< Replace info: callid */
- char replaces_callid_totag[BUFSIZ/2]; /*!< Replace info: to-tag */
- char replaces_callid_fromtag[BUFSIZ/2]; /*!< Replace info: from-tag */
+ char replaces_callid[SIPBUFSIZE]; /*!< Replace info: callid */
+ char replaces_callid_totag[SIPBUFSIZE/2]; /*!< Replace info: to-tag */
+ char replaces_callid_fromtag[SIPBUFSIZE/2]; /*!< Replace info: from-tag */
struct sip_pvt *refer_call; /*!< Call we are referring. This is just a reference to a
* dialog owned by someone else, so we should not destroy
* it when the sip_refer object goes.
@@ -4037,7 +4039,7 @@
ast_set_flag(&p->flags[0], SIP_OUTGOING);
if (p->options->transfer) {
- char buf[BUFSIZ/2];
+ char buf[SIPBUFSIZE/2];
if (referer) {
if (sipdebug)
@@ -5029,7 +5031,7 @@
int text;
int needvideo = 0;
int needtext = 0;
- char buf[BUFSIZ];
+ char buf[SIPBUFSIZE];
char *decoded_exten;
{
@@ -5073,12 +5075,12 @@
/* Set the native formats for audio and merge in video */
tmp->nativeformats = ast_codec_choose(&i->prefs, what, 1) | video | text;
- ast_debug(3, "*** Our native formats are %s \n", ast_getformatname_multiple(buf, BUFSIZ, tmp->nativeformats));
- ast_debug(3, "*** Joint capabilities are %s \n", ast_getformatname_multiple(buf, BUFSIZ, i->jointcapability));
- ast_debug(3, "*** Our capabilities are %s \n", ast_getformatname_multiple(buf, BUFSIZ, i->capability));
- ast_debug(3, "*** AST_CODEC_CHOOSE formats are %s \n", ast_getformatname_multiple(buf, BUFSIZ, ast_codec_choose(&i->prefs, what, 1)));
+ ast_debug(3, "*** Our native formats are %s \n", ast_getformatname_multiple(buf, SIPBUFSIZE, tmp->nativeformats));
+ ast_debug(3, "*** Joint capabilities are %s \n", ast_getformatname_multiple(buf, SIPBUFSIZE, i->jointcapability));
+ ast_debug(3, "*** Our capabilities are %s \n", ast_getformatname_multiple(buf, SIPBUFSIZE, i->capability));
+ ast_debug(3, "*** AST_CODEC_CHOOSE formats are %s \n", ast_getformatname_multiple(buf, SIPBUFSIZE, ast_codec_choose(&i->prefs, what, 1)));
if (i->prefcodec)
- ast_debug(3, "*** Our preferred formats from the incoming channel are %s \n", ast_getformatname_multiple(buf, BUFSIZ, i->prefcodec));
+ ast_debug(3, "*** Our preferred formats from the incoming channel are %s \n", ast_getformatname_multiple(buf, SIPBUFSIZE, i->prefcodec));
/* XXX Why are we choosing a codec from the native formats?? */
fmt = ast_best_codec(tmp->nativeformats);
@@ -6123,7 +6125,7 @@
int found_rtpmap_codecs[32];
int last_rtpmap_codec=0;
- char buf[BUFSIZ];
+ char buf[SIPBUFSIZE];
int rua_version;
if (!p->rtp) {
@@ -6625,19 +6627,19 @@
if (debug) {
/* shame on whoever coded this.... */
- char s1[BUFSIZ], s2[BUFSIZ], s3[BUFSIZ], s4[BUFSIZ], s5[BUFSIZ];
+ char s1[SIPBUFSIZE], s2[SIPBUFSIZE], s3[SIPBUFSIZE], s4[SIPBUFSIZE], s5[SIPBUFSIZE];
ast_verbose("Capabilities: us - %s, peer - audio=%s/video=%s/text=%s, combined - %s\n",
- ast_getformatname_multiple(s1, BUFSIZ, p->capability),
- ast_getformatname_multiple(s2, BUFSIZ, peercapability),
- ast_getformatname_multiple(s3, BUFSIZ, vpeercapability),
- ast_getformatname_multiple(s4, BUFSIZ, tpeercapability),
- ast_getformatname_multiple(s5, BUFSIZ, newjointcapability));
+ ast_getformatname_multiple(s1, SIPBUFSIZE, p->capability),
+ ast_getformatname_multiple(s2, SIPBUFSIZE, peercapability),
+ ast_getformatname_multiple(s3, SIPBUFSIZE, vpeercapability),
+ ast_getformatname_multiple(s4, SIPBUFSIZE, tpeercapability),
+ ast_getformatname_multiple(s5, SIPBUFSIZE, newjointcapability));
ast_verbose("Non-codec capabilities (dtmf): us - %s, peer - %s, combined - %s\n",
- ast_rtp_lookup_mime_multiple(s1, BUFSIZ, p->noncodeccapability, 0, 0),
- ast_rtp_lookup_mime_multiple(s2, BUFSIZ, peernoncodeccapability, 0, 0),
- ast_rtp_lookup_mime_multiple(s3, BUFSIZ, newnoncodeccapability, 0, 0));
+ ast_rtp_lookup_mime_multiple(s1, SIPBUFSIZE, p->noncodeccapability, 0, 0),
+ ast_rtp_lookup_mime_multiple(s2, SIPBUFSIZE, peernoncodeccapability, 0, 0),
+ ast_rtp_lookup_mime_multiple(s3, SIPBUFSIZE, newnoncodeccapability, 0, 0));
}
if (!newjointcapability) {
/* If T.38 was not negotiated either, totally bail out... */
@@ -6698,7 +6700,7 @@
}
/* Ok, we're going with this offer */
- ast_debug(2, "We're settling with these formats: %s\n", ast_getformatname_multiple(buf, BUFSIZ, p->jointcapability));
+ ast_debug(2, "We're settling with these formats: %s\n", ast_getformatname_multiple(buf, SIPBUFSIZE, p->jointcapability));
if (!p->owner) /* There's no open channel owning us so we can return here. For a re-invite or so, we proceed */
return 0;
@@ -6707,10 +6709,10 @@
if (!(p->owner->nativeformats & p->jointcapability) && (p->jointcapability & AST_FORMAT_AUDIO_MASK)) {
if (debug) {
- char s1[BUFSIZ], s2[BUFSIZ];
+ char s1[SIPBUFSIZE], s2[SIPBUFSIZE];
ast_debug(1, "Oooh, we need to change our audio formats since our peer supports only %s and not %s\n",
- ast_getformatname_multiple(s1, BUFSIZ, p->jointcapability),
- ast_getformatname_multiple(s2, BUFSIZ, p->owner->nativeformats));
+ ast_getformatname_multiple(s1, SIPBUFSIZE, p->jointcapability),
+ ast_getformatname_multiple(s2, SIPBUFSIZE, p->owner->nativeformats));
}
p->owner->nativeformats = ast_codec_choose(&p->prefs, p->jointcapability, 1) | (p->capability & vpeercapability) | (p->capability & tpeercapability);
ast_set_read_format(p->owner, p->owner->readformat);
@@ -6934,7 +6936,7 @@
/*! \brief Add route header into request per learned route */
static void add_route(struct sip_request *req, struct sip_route *route)
{
- char r[BUFSIZ*2], *p;
+ char r[SIPBUFSIZE*2], *p;
int n, rem = sizeof(r);
if (!route)
@@ -7098,7 +7100,7 @@
snprintf(tmp, sizeof(tmp), "%d", p->expiry);
add_header(resp, "Expires", tmp);
if (p->expiry) { /* Only add contact if we have an expiry time */
- char contact[BUFSIZ];
+ char contact[SIPBUFSIZE];
snprintf(contact, sizeof(contact), "%s;expires=%d", p->our_contact, p->expiry);
add_header(resp, "Contact", contact); /* Not when we unregister */
}
@@ -7790,8 +7792,8 @@
int min_video_packet_size = 0;
int min_text_packet_size = 0;
- char codecbuf[BUFSIZ];
- char buf[BUFSIZ];
+ char codecbuf[SIPBUFSIZE];
+ char buf[SIPBUFSIZE];
/* Set the SDP session name */
snprintf(subject, sizeof(subject), "s=%s\r\n", ast_strlen_zero(global_sdpsession) ? "-" : global_sdpsession);
@@ -8026,7 +8028,7 @@
/* Update lastrtprx when we send our SDP */
p->lastrtprx = p->lastrtptx = time(NULL); /* XXX why both ? */
- ast_debug(3, "Done building SDP. Settling with this capability: %s\n", ast_getformatname_multiple(buf, BUFSIZ, capability));
+ ast_debug(3, "Done building SDP. Settling with this capability: %s\n", ast_getformatname_multiple(buf, SIPBUFSIZE, capability));
return AST_SUCCESS;
}
@@ -8195,7 +8197,7 @@
/*! \brief Check Contact: URI of SIP message */
static void extract_uri(struct sip_pvt *p, struct sip_request *req)
{
- char stripped[BUFSIZ];
+ char stripped[SIPBUFSIZE];
char *c;
ast_copy_string(stripped, get_header(req, "Contact"), sizeof(stripped));
@@ -8305,8 +8307,8 @@
struct ast_str *invite = ast_str_alloca(256);
char from[256];
char to[256];
- char tmp_n[BUFSIZ/2]; /* build a local copy of 'n' if needed */
- char tmp_l[BUFSIZ/2]; /* build a local copy of 'l' if needed */
+ char tmp_n[SIPBUFSIZE/2]; /* build a local copy of 'n' if needed */
+ char tmp_l[SIPBUFSIZE/2]; /* build a local copy of 'l' if needed */
const char *l = NULL; /* XXX what is this, exactly ? */
const char *n = NULL; /* XXX what is this, exactly ? */
const char *urioptions = "";
@@ -8473,7 +8475,7 @@
append_date(&req);
if (sipmethod == SIP_REFER) { /* Call transfer */
if (p->refer) {
- char buf[BUFSIZ];
+ char buf[SIPBUFSIZE];
if (!ast_strlen_zero(p->refer->refer_to))
add_header(&req, "Refer-To", p->refer->refer_to);
if (!ast_strlen_zero(p->refer->referred_by)) {
@@ -8794,7 +8796,7 @@
static int transmit_notify_with_sipfrag(struct sip_pvt *p, int cseq, char *message, int terminate)
{
struct sip_request req;
- char tmp[BUFSIZ/2];
+ char tmp[SIPBUFSIZE/2];
reqprep(&req, p, SIP_NOTIFY, 0, 1);
snprintf(tmp, sizeof(tmp), "refer;id=%d", cseq);
@@ -9422,7 +9424,7 @@
/*! \brief Save contact header for 200 OK on INVITE */
static int parse_ok_contact(struct sip_pvt *pvt, struct sip_request *req)
{
- char contact[BUFSIZ];
+ char contact[SIPBUFSIZE];
char *c;
/* Look for brackets */
@@ -9495,8 +9497,8 @@
/*! \brief Parse contact header and save registration (peer registration) */
static enum parse_register_result parse_register_contact(struct sip_pvt *pvt, struct sip_peer *peer, struct sip_request *req)
{
- char contact[BUFSIZ];
- char data[BUFSIZ];
+ char contact[SIPBUFSIZE];
+ char data[SIPBUFSIZE];
const char *expires = get_header(req, "Expires");
int expiry = atoi(expires);
char *curi, *host, *pt, *curi2;
@@ -12636,7 +12638,7 @@
int realtimepeers;
int realtimeusers;
int realtimeregs;
- char codec_buf[BUFSIZ];
+ char codec_buf[SIPBUFSIZE];
const char *msg; /* temporary msg pointer */
switch (cmd) {
@@ -12866,7 +12868,7 @@
if (cur->subscribed == NONE && !arg->subscriptions) {
/* set if SIP transfer in progress */
const char *referstatus = cur->refer ? referstatus2str(cur->refer->status) : "";
- char formatbuf[BUFSIZ/2];
+ char formatbuf[SIPBUFSIZE/2];
ast_cli(arg->fd, FORMAT, ast_inet_ntoa(dst->sin_addr),
S_OR(cur->username, S_OR(cur->cid_num, "(None)")),
@@ -13112,7 +13114,7 @@
dialoglist_lock();
for (cur = dialoglist; cur; cur = cur->next) {
if (!strncasecmp(cur->callid, a->argv[3], len)) {
- char formatbuf[BUFSIZ/2];
+ char formatbuf[SIPBUFSIZE/2];
ast_cli(a->fd,"\n");
if (cur->subscribed != NONE)
ast_cli(a->fd, " * Subscription (type: %s)\n", subscription_type2str(cur->subscribed));
@@ -14126,7 +14128,7 @@
/*! \brief Parse 302 Moved temporalily response */
static void parse_moved_contact(struct sip_pvt *p, struct sip_request *req)
{
- char tmp[BUFSIZ];
+ char tmp[SIPBUFSIZE];
char *s, *e, *t;
char *domain;
@@ -16882,7 +16884,7 @@
}
/* Generate a Replaces string to be used in the INVITE during attended transfer */
if (!ast_strlen_zero(p->refer->replaces_callid)) {
- char tempheader[BUFSIZ];
+ char tempheader[SIPBUFSIZE];
snprintf(tempheader, sizeof(tempheader), "%s%s%s%s%s", p->refer->replaces_callid,
p->refer->replaces_callid_totag ? ";to-tag=" : "",
p->refer->replaces_callid_totag,
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