Asterisk
/
asterisk
mirror of https://github.com/asterisk/asterisk
1
0
Fork 0
Code Issues Releases Wiki Activity

Thanks to Russell's education I realize that BUFSIZ has changed since I learned the C language

Browse Source 
over 20 years ago... Resetting chan_sip to the size of BUFSIZ that I expected in my old 
head to avoid to heavy memory allocations on some systems.


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@99652 65c4cc65-6c06-0410-ace0-fbb531ad65f3
1.4
Olle Johansson 18 years ago
parent fa20a738d7
commit f9cf98d4bf
1 changed files with 43 additions and 41 deletions
Show Stats Download Patch File Download Diff File

84
channels/chan_sip.c
Unescape View File

@ -159,6 +159,8 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#define TRUE 1
#endif
#define SIPBUFSIZE 512
#define XMIT_ERROR -2
#define VIDEO_CODEC_MASK 0x1fc0000 /*!< Video codecs from H.261 thru AST_FORMAT_MAX_VIDEO */
@ -889,9 +891,9 @@ struct sip_refer {
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 */
int attendedtransfer; /*!< Attended or blind transfer? */
int localtransfer; /*!< Transfer to local domain? */
@ -2984,7 +2986,7 @@ static int sip_call(struct ast_channel *ast, char *dest, int timeout)
ast_set_flag(&p->flags[0], SIP_OUTGOING);
if (p->options->transfer) {
char buf[BUFSIZ/2];
char buf[SIPBUFSIZE/2];
if (referer) {
if (sipdebug && option_debug > 2)
@ -3996,13 +3998,13 @@ static struct ast_channel *sip_new(struct sip_pvt *i, int state, const char *tit
/* Set the native formats for audio and merge in video */
tmp->nativeformats = ast_codec_choose(&i->prefs, what, 1) | video;
if (option_debug > 2) {
char buf[BUFSIZ];
ast_log(LOG_DEBUG, "*** Our native formats are %s \n", ast_getformatname_multiple(buf, BUFSIZ, tmp->nativeformats));
ast_log(LOG_DEBUG, "*** Joint capabilities are %s \n", ast_getformatname_multiple(buf, BUFSIZ, i->jointcapability));
ast_log(LOG_DEBUG, "*** Our capabilities are %s \n", ast_getformatname_multiple(buf, BUFSIZ, i->capability));
ast_log(LOG_DEBUG, "*** AST_CODEC_CHOOSE formats are %s \n", ast_getformatname_multiple(buf, BUFSIZ, ast_codec_choose(&i->prefs, what, 1)));
char buf[SIPBUFSIZE];
ast_log(LOG_DEBUG, "*** Our native formats are %s \n", ast_getformatname_multiple(buf, SIPBUFSIZE, tmp->nativeformats));
ast_log(LOG_DEBUG, "*** Joint capabilities are %s \n", ast_getformatname_multiple(buf, SIPBUFSIZE, i->jointcapability));
ast_log(LOG_DEBUG, "*** Our capabilities are %s \n", ast_getformatname_multiple(buf, SIPBUFSIZE, i->capability));
ast_log(LOG_DEBUG, "*** AST_CODEC_CHOOSE formats are %s \n", ast_getformatname_multiple(buf, SIPBUFSIZE, ast_codec_choose(&i->prefs, what, 1)));
if (i->prefcodec)
ast_log(LOG_DEBUG, "*** Our preferred formats from the incoming channel are %s \n", ast_getformatname_multiple(buf, BUFSIZ, i->prefcodec));
ast_log(LOG_DEBUG, "*** 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?? */
@ -5409,18 +5411,18 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req)
if (debug) {
/* shame on whoever coded this.... */
char s1[BUFSIZ], s2[BUFSIZ], s3[BUFSIZ], s4[BUFSIZ];
char s1[SIPBUFSIZE], s2[SIPBUFSIZE], s3[SIPBUFSIZE], s4[SIPBUFSIZE];
ast_verbose("Capabilities: us - %s, peer - audio=%s/video=%s, combined - %s\n",
ast_getformatname_multiple(s1, BUFSIZ, p->capability),
ast_getformatname_multiple(s2, BUFSIZ, newpeercapability),
ast_getformatname_multiple(s3, BUFSIZ, vpeercapability),
ast_getformatname_multiple(s4, BUFSIZ, newjointcapability));
ast_getformatname_multiple(s1, SIPBUFSIZE, p->capability),
ast_getformatname_multiple(s2, SIPBUFSIZE, newpeercapability),
ast_getformatname_multiple(s3, SIPBUFSIZE, vpeercapability),
ast_getformatname_multiple(s4, 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... */
@ -5474,8 +5476,8 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req)
/* Ok, we're going with this offer */
if (option_debug > 1) {
char buf[BUFSIZ];
ast_log(LOG_DEBUG, "We're settling with these formats: %s\n", ast_getformatname_multiple(buf, BUFSIZ, p->jointcapability));
char buf[SIPBUFSIZE];
ast_log(LOG_DEBUG, "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 */
@ -5486,10 +5488,10 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req)
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_log(LOG_DEBUG, "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);
ast_set_read_format(p->owner, p->owner->readformat);
@ -5685,7 +5687,7 @@ static int copy_via_headers(struct sip_pvt *p, struct sip_request *req, const st
/*! \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)
@ -5839,7 +5841,7 @@ static int respprep(struct sip_request *resp, struct sip_pvt *p, const char *msg
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 */
}
@ -6427,7 +6429,7 @@ static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p)
if (option_debug > 1) {
char codecbuf[BUFSIZ];
char codecbuf[SIPBUFSIZE];
ast_log(LOG_DEBUG, "** Our capability: %s Video flag: %s\n", ast_getformatname_multiple(codecbuf, sizeof(codecbuf), capability), ast_test_flag(&p->flags[0], SIP_NOVIDEO) ? "True" : "False");
ast_log(LOG_DEBUG, "** Our prefcodec: %s \n", ast_getformatname_multiple(codecbuf, sizeof(codecbuf), p->prefcodec));
}
@ -6604,8 +6606,8 @@ static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p)
p->lastrtprx = p->lastrtptx = time(NULL); /* XXX why both ? */
if (option_debug > 2) {
char buf[BUFSIZ];
ast_log(LOG_DEBUG, "Done building SDP. Settling with this capability: %s\n", ast_getformatname_multiple(buf, BUFSIZ, capability));
char buf[SIPBUFSIZE];
ast_log(LOG_DEBUG, "Done building SDP. Settling with this capability: %s\n", ast_getformatname_multiple(buf, SIPBUFSIZE, capability));
}
return AST_SUCCESS;
@ -6769,7 +6771,7 @@ static int transmit_reinvite_with_t38_sdp(struct sip_pvt *p)
/*! \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));
@ -6876,8 +6878,8 @@ static void initreqprep(struct sip_request *req, struct sip_pvt *p, int sipmetho
size_t invite_max = sizeof(invite_buf);
char from[256];
char to[256];
char tmp[BUFSIZ/2];
char tmp2[BUFSIZ/2];
char tmp[SIPBUFSIZE/2];
char tmp2[SIPBUFSIZE/2];
const char *l = NULL, *n = NULL;
const char *urioptions = "";
@ -7024,7 +7026,7 @@ static int transmit_invite(struct sip_pvt *p, int sipmethod, int sdp, int init)
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)) {
@ -7337,7 +7339,7 @@ static int transmit_sip_request(struct sip_pvt *p, struct sip_request *req)
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);
@ -7924,7 +7926,7 @@ static void reg_source_db(struct sip_peer *peer)
/*! \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 */
@ -8008,8 +8010,8 @@ static int set_address_from_contact(struct sip_pvt *pvt)
/*! \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, *n, *pt;
@ -10581,7 +10583,7 @@ static int sip_show_settings(int fd, int argc, char *argv[])
{
int realtimepeers;
int realtimeusers;
char codec_buf[BUFSIZ];
char codec_buf[SIPBUFSIZE];
realtimepeers = ast_check_realtime("sippeers");
realtimeusers = ast_check_realtime("sipusers");
@ -10745,7 +10747,7 @@ static int __sip_show_channels(int fd, int argc, char *argv[], int subscriptions
referstatus = referstatus2str(cur->refer->status);
}
if (cur->subscribed == NONE && !subscriptions) {
char formatbuf[BUFSIZ/2];
char formatbuf[SIPBUFSIZE/2];
ast_cli(fd, FORMAT, ast_inet_ntoa(cur->sa.sin_addr),
S_OR(cur->username, S_OR(cur->cid_num, "(None)")),
cur->callid,
@ -10925,7 +10927,7 @@ static int sip_show_channel(int fd, int argc, char *argv[])
ast_mutex_lock(&iflock);
for (cur = iflist; cur; cur = cur->next) {
if (!strncasecmp(cur->callid, argv[3], len)) {
char formatbuf[BUFSIZ/2];
char formatbuf[SIPBUFSIZE/2];
ast_cli(fd,"\n");
if (cur->subscribed != NONE)
ast_cli(fd, " * Subscription (type: %s)\n", subscription_type2str(cur->subscribed));
@ -11881,7 +11883,7 @@ static struct ast_custom_function sipchaninfo_function = {
/*! \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, *uri, *t;
char *domain;
@ -14482,7 +14484,7 @@ static int handle_request_refer(struct sip_pvt *p, struct sip_request *req, int
}
/* Generate a Replaces string to be used in the INVITE during attended transfer */
if (p->refer->replaces_callid && !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,

Write Preview
Loading…
Cancel
Save