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Merge "chan_sip: bigger buffers for headers, better failure mode"

This commit is contained in:
zuul 2016-06-16 17:59:32 -05:00 committed by Gerrit Code Review
parent 3c80f84cd0 80ff7912a1
commit 947f76a971
1 changed files with 46 additions and 31 deletions
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77
channels/chan_sip.c
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@ -14154,9 +14154,10 @@ static void build_contact(struct sip_pvt *p, struct sip_request *req, int incomi
/*! \brief Initiate new SIP request to peer/user */
static void initreqprep(struct sip_request *req, struct sip_pvt *p, int sipmethod, const char * const explicit_uri)
{
struct ast_str *invite = ast_str_alloca(256);
char from[256];
char to[256];
#define SIPHEADER 256
struct ast_str *invite = ast_str_create(SIPHEADER);
struct ast_str *from = ast_str_create(SIPHEADER);
struct ast_str *to = ast_str_create(SIPHEADER);
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 ? */
@ -14258,34 +14259,40 @@ static void initreqprep(struct sip_request *req, struct sip_pvt *p, int sipmetho
ourport = (p->fromdomainport && (p->fromdomainport != STANDARD_SIP_PORT)) ? p->fromdomainport : ast_sockaddr_port(&p->ourip);
if (!sip_standard_port(p->socket.type, ourport)) {
ret = snprintf(from, sizeof(from), "<sip:%s@%s:%d>;tag=%s", tmp_l, d, ourport, p->tag);
ret = ast_str_set(&from, 0, "<sip:%s@%s:%d>;tag=%s", tmp_l, d, ourport, p->tag);
} else {
ret = snprintf(from, sizeof(from), "<sip:%s@%s>;tag=%s", tmp_l, d, p->tag);
ret = ast_str_set(&from, 0, "<sip:%s@%s>;tag=%s", tmp_l, d, p->tag);
}
if (ret < 0 || ret >= sizeof(from)) { /* a return value of size or more means that the output was truncated */
if (ret == AST_DYNSTR_BUILD_FAILED) {
/* We don't have an escape path from here... */
ast_log(LOG_ERROR, "The From header was truncated in call '%s'. This call setup will fail.\n", p->callid);
/* Make sure that the field contains something non-broken.
See https://issues.asterisk.org/jira/browse/ASTERISK-26069
*/
ast_str_set(&from, 3, "<>");
}
/* If a caller id name was specified, prefix a display name, if there is enough room. */
if (cid_has_name || !cid_has_num) {
size_t written = strlen(from);
ssize_t left = sizeof(from) - written - 4; /* '"" \0' */
if (left > 0) {
size_t name_len;
if (sip_cfg.pedanticsipchecking) {
ast_escape_quoted(n, tmp_n, MIN(left + 1, sizeof(tmp_n)));
n = tmp_n;
}
name_len = strlen(n);
if (left < name_len) {
name_len = left;
}
memmove(from + name_len + 3, from, written + 1);
from[0] = '"';
memcpy(from + 1, n, name_len);
from[name_len + 1] = '"';
from[name_len + 2] = ' ';
size_t written = ast_str_strlen(from);
size_t name_len;
if (sip_cfg.pedanticsipchecking) {
ast_escape_quoted(n, tmp_n, sizeof(tmp_n));
n = tmp_n;
}
name_len = strlen(n);
ret = ast_str_make_space(&from, name_len + written + 4);
if (ret == 0) {
/* needed again, as ast_str_make_space coud've changed the pointer */
char *from_buf = ast_str_buffer(from);
memmove(from_buf + name_len + 3, from_buf, written + 1);
from_buf[0] = '"';
memcpy(from_buf + 1, n, name_len);
from_buf[name_len + 1] = '"';
from_buf[name_len + 2] = ' ';
}
}
@ -14328,24 +14335,28 @@ static void initreqprep(struct sip_request *req, struct sip_pvt *p, int sipmetho
/*! \todo Need to add back the VXML URL here at some point, possibly use build_string for all this junk */
if (!strchr(p->todnid, '@')) {
/* We have no domain in the dnid */
ret = snprintf(to, sizeof(to), "<sip:%s@%s>%s%s", p->todnid, p->tohost, ast_strlen_zero(p->theirtag) ? "" : ";tag=", p->theirtag);
ret = ast_str_set(&to, 0, "<sip:%s@%s>%s%s", p->todnid, p->tohost, ast_strlen_zero(p->theirtag) ? "" : ";tag=", p->theirtag);
} else {
ret = snprintf(to, sizeof(to), "<sip:%s>%s%s", p->todnid, ast_strlen_zero(p->theirtag) ? "" : ";tag=", p->theirtag);
ret = ast_str_set(&to, 0, "<sip:%s>%s%s", p->todnid, ast_strlen_zero(p->theirtag) ? "" : ";tag=", p->theirtag);
}
} else {
if (sipmethod == SIP_NOTIFY && !ast_strlen_zero(p->theirtag)) {
/* If this is a NOTIFY, use the From: tag in the subscribe (RFC 3265) */
ret = snprintf(to, sizeof(to), "<%s%s>;tag=%s", (strncasecmp(p->uri, "sip:", 4) ? "sip:" : ""), p->uri, p->theirtag);
ret = ast_str_set(&to, 0, "<%s%s>;tag=%s", (strncasecmp(p->uri, "sip:", 4) ? "sip:" : ""), p->uri, p->theirtag);
} else if (p->options && p->options->vxml_url) {
/* If there is a VXML URL append it to the SIP URL */
ret = snprintf(to, sizeof(to), "<%s>;%s", p->uri, p->options->vxml_url);
ret = ast_str_set(&to, 0, "<%s>;%s", p->uri, p->options->vxml_url);
} else {
ret = snprintf(to, sizeof(to), "<%s>", p->uri);
ret = ast_str_set(&to, 0, "<%s>", p->uri);
}
}
if (ret < 0 || ret >= sizeof(to)) { /* a return value of size or more means that the output was truncated */
if (ret == AST_DYNSTR_BUILD_FAILED) {
/* We don't have an escape path from here... */
ast_log(LOG_ERROR, "The To header was truncated in call '%s'. This call setup will fail.\n", p->callid);
/* Make sure that the field contains something non-broken.
See https://issues.asterisk.org/jira/browse/ASTERISK-26069
*/
ast_str_set(&to, 3, "<>");
}
init_req(req, sipmethod, p->uri);
@ -14360,8 +14371,8 @@ static void initreqprep(struct sip_request *req, struct sip_pvt *p, int sipmetho
*/
add_route(req, &p->route, 0);
add_header(req, "From", from);
add_header(req, "To", to);
add_header(req, "From", ast_str_buffer(from));
add_header(req, "To", ast_str_buffer(to));
ast_string_field_set(p, exten, l);
build_contact(p, req, 0);
add_header(req, "Contact", p->our_contact);
@ -14370,6 +14381,10 @@ static void initreqprep(struct sip_request *req, struct sip_pvt *p, int sipmetho
if (!ast_strlen_zero(global_useragent)) {
add_header(req, "User-Agent", global_useragent);
}
ast_free(from);
ast_free(to);
ast_free(invite);
}
/*! \brief Add "Diversion" header to outgoing message