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Allow Asterisk to compile under GCC 4.10

Browse Source 
This resolves a large number of compiler warnings from GCC 4.10 which
cause the build to fail under dev mode. The vast majority are
signed/unsigned mismatches in printf-style format strings.
........

Merged revisions 413586 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 413587 from http://svn.asterisk.org/svn/asterisk/branches/11
........

Merged revisions 413588 from http://svn.asterisk.org/svn/asterisk/branches/12


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@413589 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Kinsey Moore
2014-05-09 22:49:26 +00:00
parent f3b55da1b8
commit abd3e4040b
138 changed files with 726 additions and 717 deletions
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164
channels/chan_sip.c
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@@ -863,7 +863,7 @@ static int regobjs = 0; /*!< Registry objects */
/*! @} */
static struct ast_flags global_flags[3] = {{0}}; /*!< global SIP_ flags */
static int global_t38_maxdatagram; /*!< global T.38 FaxMaxDatagram override */
static unsigned int global_t38_maxdatagram; /*!< global T.38 FaxMaxDatagram override */
static struct stasis_subscription *network_change_sub; /*!< subscription id for network change events */
static struct stasis_subscription *acl_change_sub; /*!< subscription id for named ACL system change events */
@@ -3289,7 +3289,7 @@ static void *_sip_tcp_helper_thread(struct ast_tcptls_session_instance *tcptls_s
}
break;
default:
ast_log(LOG_ERROR, "Unknown tcptls thread alert '%d'\n", alert);
ast_log(LOG_ERROR, "Unknown tcptls thread alert '%u'\n", alert);
}
}
}
@@ -3539,7 +3539,7 @@ void dialog_unlink_all(struct sip_pvt *dialog)
void *registry_unref(struct sip_registry *reg, char *tag)
{
ast_debug(3, "SIP Registry %s: refcount now %d\n", reg->hostname, reg->refcount - 1);
ast_debug(3, "SIP Registry %s: refcount now %u\n", reg->hostname, reg->refcount - 1);
ASTOBJ_UNREF(reg, sip_registry_destroy);
return NULL;
}
@@ -3547,7 +3547,7 @@ void *registry_unref(struct sip_registry *reg, char *tag)
/*! \brief Add object reference to SIP registry */
static struct sip_registry *registry_addref(struct sip_registry *reg, char *tag)
{
ast_debug(3, "SIP Registry %s: refcount now %d\n", reg->hostname, reg->refcount + 1);
ast_debug(3, "SIP Registry %s: refcount now %u\n", reg->hostname, reg->refcount + 1);
return ASTOBJ_REF(reg); /* Add pointer to registry in packet */
}
@@ -3985,7 +3985,7 @@ static void build_via(struct sip_pvt *p)
snprintf(p->via, sizeof(p->via), "SIP/2.0/%s %s;branch=z9hG4bK%08x%s",
get_transport_pvt(p),
ast_sockaddr_stringify_remote(&p->ourip),
(int) p->branch, rport);
(unsigned)p->branch, rport);
}
/*! \brief NAT fix - decide which IP address to use for Asterisk server?
@@ -4343,7 +4343,7 @@ static enum sip_result __sip_reliable_xmit(struct sip_pvt *p, uint32_t seqno, in
struct sip_pkt *pkt = NULL;
int siptimer_a = DEFAULT_RETRANS;
int xmitres = 0;
int respid;
unsigned respid;
if (sipmethod == SIP_INVITE) {
/* Note this is a pending invite */
@@ -5063,7 +5063,7 @@ static int sip_sendhtml(struct ast_channel *chan, int subclass, const char *data
ast_string_field_build(p, url, "<%s>;mode=active", data);
if (sip_debug_test_pvt(p))
ast_debug(1, "Send URL %s, state = %d!\n", data, ast_channel_state(chan));
ast_debug(1, "Send URL %s, state = %u!\n", data, ast_channel_state(chan));
switch (ast_channel_state(chan)) {
case AST_STATE_RING:
@@ -5080,7 +5080,7 @@ static int sip_sendhtml(struct ast_channel *chan, int subclass, const char *data
}
break;
default:
ast_log(LOG_WARNING, "Don't know how to send URI when state is %d!\n", ast_channel_state(chan));
ast_log(LOG_WARNING, "Don't know how to send URI when state is %u!\n", ast_channel_state(chan));
}
return 0;
@@ -5855,7 +5855,7 @@ static void change_t38_state(struct sip_pvt *p, int state)
return;
p->t38.state = state;
ast_debug(2, "T38 state changed to %d on channel %s\n", p->t38.state, chan ? ast_channel_name(chan) : "<none>");
ast_debug(2, "T38 state changed to %u on channel %s\n", p->t38.state, chan ? ast_channel_name(chan) : "<none>");
/* If no channel was provided we can't send off a control frame */
if (!chan)
@@ -7508,7 +7508,7 @@ static int sip_write(struct ast_channel *ast, struct ast_frame *frame)
}
break;
default:
ast_log(LOG_WARNING, "Can't send %d type frames with SIP write\n", frame->frametype);
ast_log(LOG_WARNING, "Can't send %u type frames with SIP write\n", frame->frametype);
return 0;
}
@@ -8056,7 +8056,7 @@ static struct ast_channel *sip_new(struct sip_pvt *i, int state, const char *tit
sip_pvt_unlock(i);
/* Don't hold a sip pvt lock while we allocate a channel */
tmp = ast_channel_alloc(1, state, i->cid_num, i->cid_name, i->accountcode, i->exten, i->context, assignedids, requestor, i->amaflags, "SIP/%s-%08x", my_name, ast_atomic_fetchadd_int((int *)&chan_idx, +1));
tmp = ast_channel_alloc(1, state, i->cid_num, i->cid_name, i->accountcode, i->exten, i->context, assignedids, requestor, i->amaflags, "SIP/%s-%08x", my_name, (unsigned)ast_atomic_fetchadd_int((int *)&chan_idx, +1));
}
if (!tmp) {
ast_log(LOG_WARNING, "Unable to allocate AST channel structure for SIP channel\n");
@@ -8508,7 +8508,7 @@ static struct ast_frame *sip_rtp_read(struct ast_channel *ast, struct sip_pvt *p
}
ast_str_append(&out, 0, " -> ");
for (i = 0; i < f->datalen; i++) {
ast_str_append(&out, 0, "%02X ", arr[i]);
ast_str_append(&out, 0, "%02X ", (unsigned)arr[i]);
}
ast_verb(0, "%s\n", ast_str_buffer(out));
ast_free(out);
@@ -8632,7 +8632,7 @@ static char *generate_random_string(char *buf, size_t size)
for (x=0; x<4; x++)
val[x] = ast_random();
snprintf(buf, size, "%08lx%08lx%08lx%08lx", val[0], val[1], val[2], val[3]);
snprintf(buf, size, "%08lx%08lx%08lx%08lx", (unsigned long)val[0], (unsigned long)val[1], (unsigned long)val[2], (unsigned long)val[3]);
return buf;
}
@@ -8732,13 +8732,13 @@ static void build_callid_registry(struct sip_registry *reg, const struct ast_soc
/*! \brief Build SIP From tag value for REGISTER */
static void build_localtag_registry(struct sip_registry *reg)
{
ast_string_field_build(reg, localtag, "as%08lx", ast_random());
ast_string_field_build(reg, localtag, "as%08lx", (unsigned long)ast_random());
}
/*! \brief Make our SIP dialog tag */
static void make_our_tag(struct sip_pvt *pvt)
{
ast_string_field_build(pvt, tag, "as%08lx", ast_random());
ast_string_field_build(pvt, tag, "as%08lx", (unsigned long)ast_random());
}
/*! \brief Allocate Session-Timers struct w/in dialog */
@@ -9038,7 +9038,7 @@ static enum match_req_res match_req_to_dialog(struct sip_pvt *sip_pvt_ptr, struc
/* totag did not match what we had stored for them. */
char invite_branch[32] = { 0, };
if (sip_pvt_ptr->invite_branch) {
snprintf(invite_branch, sizeof(invite_branch), "z9hG4bK%08x", (int) sip_pvt_ptr->invite_branch);
snprintf(invite_branch, sizeof(invite_branch), "z9hG4bK%08x", (unsigned)sip_pvt_ptr->invite_branch);
}
/* Forked Request Detection
*
@@ -10046,7 +10046,7 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
int newnoncodeccapability;
const char *codecs;
int codec;
unsigned int codec;
/* SRTP */
int secure_audio = FALSE;
@@ -10054,7 +10054,7 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
/* Others */
int sendonly = -1;
int numberofports;
unsigned int numberofports;
int last_rtpmap_codec = 0;
int red_data_pt[10]; /* For T.140 RED */
int red_num_gen = 0; /* For T.140 RED */
@@ -10170,7 +10170,7 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
int text = FALSE;
int processed_crypto = FALSE;
char protocol[18] = {0,};
int x;
unsigned int x;
struct ast_rtp_engine_dtls *dtls;
numberofports = 0;
@@ -10215,7 +10215,7 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
/* Check number of ports offered for stream */
if (numberofports > 1) {
ast_log(LOG_WARNING, "%d ports offered for audio media, not supported by Asterisk. Will try anyway...\n", numberofports);
ast_log(LOG_WARNING, "%u ports offered for audio media, not supported by Asterisk. Will try anyway...\n", numberofports);
}
if ((!strcmp(protocol, "RTP/SAVPF") || !strcmp(protocol, "UDP/TLS/RTP/SAVPF")) && !ast_test_flag(&p->flags[2], SIP_PAGE3_USE_AVPF)) {
@@ -10287,7 +10287,7 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
goto process_sdp_cleanup;
}
if (debug) {
ast_verbose("Found RTP audio format %d\n", codec);
ast_verbose("Found RTP audio format %u\n", codec);
}
ast_rtp_codecs_payloads_set_m_type(&newaudiortp, NULL, codec);
@@ -10320,7 +10320,7 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
/* Check number of ports offered for stream */
if (numberofports > 1) {
ast_log(LOG_WARNING, "%d ports offered for video stream, not supported by Asterisk. Will try anyway...\n", numberofports);
ast_log(LOG_WARNING, "%u ports offered for video stream, not supported by Asterisk. Will try anyway...\n", numberofports);
}
if (has_media_stream(p, SDP_VIDEO)) {
@@ -10367,7 +10367,7 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
goto process_sdp_cleanup;
}
if (debug) {
ast_verbose("Found RTP video format %d\n", codec);
ast_verbose("Found RTP video format %u\n", codec);
}
ast_rtp_codecs_payloads_set_m_type(&newvideortp, NULL, codec);
}
@@ -10399,7 +10399,7 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
/* Check number of ports offered for stream */
if (numberofports > 1) {
ast_log(LOG_WARNING, "%d ports offered for text stream, not supported by Asterisk. Will try anyway...\n", numberofports);
ast_log(LOG_WARNING, "%u ports offered for text stream, not supported by Asterisk. Will try anyway...\n", numberofports);
}
if (!strcmp(protocol, "RTP/AVPF") && !ast_test_flag(&p->flags[2], SIP_PAGE3_USE_AVPF)) {
@@ -10431,7 +10431,7 @@ static int process_sdp(struct sip_pvt *p, struct sip_request *req, int t38action
goto process_sdp_cleanup;
}
if (debug) {
ast_verbose("Found RTP text format %d\n", codec);
ast_verbose("Found RTP text format %u\n", codec);
}
ast_rtp_codecs_payloads_set_m_type(&newtextrtp, NULL, codec);
}
@@ -11065,7 +11065,7 @@ static int process_sdp_a_ice(const char *a, struct sip_pvt *p, struct ast_rtp_in
int found = FALSE;
char ufrag[256], pwd[256], foundation[32], transport[4], address[46], cand_type[6], relay_address[46] = "";
struct ast_rtp_engine_ice_candidate candidate = { 0, };
int port, relay_port = 0;
unsigned int port, relay_port = 0;
if (!instance || !(ice = ast_rtp_instance_get_ice(instance))) {
return found;
@@ -11077,7 +11077,7 @@ static int process_sdp_a_ice(const char *a, struct sip_pvt *p, struct ast_rtp_in
} else if (sscanf(a, "ice-pwd: %255s", pwd) == 1) {
ice->set_authentication(instance, NULL, pwd);
found = TRUE;
} else if (sscanf(a, "candidate: %31s %30u %3s %30u %23s %30u typ %5s %*s %23s %*s %30u", foundation, &candidate.id, transport, &candidate.priority,
} else if (sscanf(a, "candidate: %31s %30u %3s %30u %23s %30u typ %5s %*s %23s %*s %30u", foundation, &candidate.id, transport, (unsigned *)&candidate.priority,
address, &port, cand_type, relay_address, &relay_port) >= 7) {
candidate.foundation = foundation;
candidate.transport = transport;
@@ -11167,7 +11167,7 @@ static int process_sdp_a_dtls(const char *a, struct sip_pvt *p, struct ast_rtp_i
static int process_sdp_a_audio(const char *a, struct sip_pvt *p, struct ast_rtp_codecs *newaudiortp, int *last_rtpmap_codec)
{
int found = FALSE;
int codec;
unsigned int codec;
char mimeSubtype[128];
char fmtp_string[256];
unsigned int sample_rate;
@@ -11203,18 +11203,18 @@ static int process_sdp_a_audio(const char *a, struct sip_pvt *p, struct ast_rtp_
if (!(ast_rtp_codecs_payloads_set_rtpmap_type_rate(newaudiortp, NULL, codec, "audio", mimeSubtype,
ast_test_flag(&p->flags[0], SIP_G726_NONSTANDARD) ? AST_RTP_OPT_G726_NONSTANDARD : 0, sample_rate))) {
if (debug)
ast_verbose("Found audio description format %s for ID %d\n", mimeSubtype, codec);
ast_verbose("Found audio description format %s for ID %u\n", mimeSubtype, codec);
//found_rtpmap_codecs[last_rtpmap_codec] = codec;
(*last_rtpmap_codec)++;
found = TRUE;
} else {
ast_rtp_codecs_payloads_unset(newaudiortp, NULL, codec);
if (debug)
ast_verbose("Found unknown media description format %s for ID %d\n", mimeSubtype, codec);
ast_verbose("Found unknown media description format %s for ID %u\n", mimeSubtype, codec);
}
} else {
if (debug)
ast_verbose("Discarded description format %s for ID %d\n", mimeSubtype, codec);
ast_verbose("Discarded description format %s for ID %u\n", mimeSubtype, codec);
}
} else if (sscanf(a, "fmtp: %30u %255[^\t\n]", &codec, fmtp_string) == 2) {
struct ast_format *format;
@@ -11232,7 +11232,7 @@ static int process_sdp_a_audio(const char *a, struct sip_pvt *p, struct ast_rtp_
case AST_FORMAT_SIREN7:
if (sscanf(fmtp_string, "bitrate=%30u", &bit_rate) == 1) {
if (bit_rate != 32000) {
ast_log(LOG_WARNING, "Got Siren7 offer at %d bps, but only 32000 bps supported; ignoring.\n", bit_rate);
ast_log(LOG_WARNING, "Got Siren7 offer at %u bps, but only 32000 bps supported; ignoring.\n", bit_rate);
ast_rtp_codecs_payloads_unset(newaudiortp, NULL, codec);
} else {
found = TRUE;
@@ -11242,7 +11242,7 @@ static int process_sdp_a_audio(const char *a, struct sip_pvt *p, struct ast_rtp_
case AST_FORMAT_SIREN14:
if (sscanf(fmtp_string, "bitrate=%30u", &bit_rate) == 1) {
if (bit_rate != 48000) {
ast_log(LOG_WARNING, "Got Siren14 offer at %d bps, but only 48000 bps supported; ignoring.\n", bit_rate);
ast_log(LOG_WARNING, "Got Siren14 offer at %u bps, but only 48000 bps supported; ignoring.\n", bit_rate);
ast_rtp_codecs_payloads_unset(newaudiortp, NULL, codec);
} else {
found = TRUE;
@@ -11252,7 +11252,7 @@ static int process_sdp_a_audio(const char *a, struct sip_pvt *p, struct ast_rtp_
case AST_FORMAT_G719:
if (sscanf(fmtp_string, "bitrate=%30u", &bit_rate) == 1) {
if (bit_rate != 64000) {
ast_log(LOG_WARNING, "Got G.719 offer at %d bps, but only 64000 bps supported; ignoring.\n", bit_rate);
ast_log(LOG_WARNING, "Got G.719 offer at %u bps, but only 64000 bps supported; ignoring.\n", bit_rate);
ast_rtp_codecs_payloads_unset(newaudiortp, NULL, codec);
} else {
found = TRUE;
@@ -11269,7 +11269,7 @@ static int process_sdp_a_audio(const char *a, struct sip_pvt *p, struct ast_rtp_
static int process_sdp_a_video(const char *a, struct sip_pvt *p, struct ast_rtp_codecs *newvideortp, int *last_rtpmap_codec)
{
int found = FALSE;
int codec;
unsigned int codec;
char mimeSubtype[128];
unsigned int sample_rate;
int debug = sip_debug_test_pvt(p);
@@ -11283,19 +11283,19 @@ static int process_sdp_a_video(const char *a, struct sip_pvt *p, struct ast_rtp_
|| !strncasecmp(mimeSubtype, "VP8", 3)) {
if (!(ast_rtp_codecs_payloads_set_rtpmap_type_rate(newvideortp, NULL, codec, "video", mimeSubtype, 0, sample_rate))) {
if (debug)
ast_verbose("Found video description format %s for ID %d\n", mimeSubtype, codec);
ast_verbose("Found video description format %s for ID %u\n", mimeSubtype, codec);
//found_rtpmap_codecs[last_rtpmap_codec] = codec;
(*last_rtpmap_codec)++;
found = TRUE;
} else {
ast_rtp_codecs_payloads_unset(newvideortp, NULL, codec);
if (debug)
ast_verbose("Found unknown media description format %s for ID %d\n", mimeSubtype, codec);
ast_verbose("Found unknown media description format %s for ID %u\n", mimeSubtype, codec);
}
}
} else {
if (debug)
ast_verbose("Discarded description format %s for ID %d\n", mimeSubtype, codec);
ast_verbose("Discarded description format %s for ID %u\n", mimeSubtype, codec);
}
} else if (sscanf(a, "fmtp: %30u %255s", &codec, fmtp_string) == 2) {
struct ast_format *format;
@@ -11315,7 +11315,7 @@ static int process_sdp_a_video(const char *a, struct sip_pvt *p, struct ast_rtp_
static int process_sdp_a_text(const char *a, struct sip_pvt *p, struct ast_rtp_codecs *newtextrtp, char *red_fmtp, int *red_num_gen, int *red_data_pt, int *last_rtpmap_codec)
{
int found = FALSE;
int codec;
unsigned int codec;
char mimeSubtype[128];
unsigned int sample_rate;
char *red_cp;
@@ -11333,25 +11333,25 @@ static int process_sdp_a_text(const char *a, struct sip_pvt *p, struct ast_rtp_c
} else if (!strncasecmp(mimeSubtype, "RED", 3)) { /* Text with Redudancy */
if (p->trtp) {
ast_rtp_codecs_payloads_set_rtpmap_type_rate(newtextrtp, NULL, codec, "text", mimeSubtype, 0, sample_rate);
sprintf(red_fmtp, "fmtp:%d ", codec);
sprintf(red_fmtp, "fmtp:%u ", codec);
if (debug)
ast_verbose("RED submimetype has payload type: %d\n", codec);
ast_verbose("RED submimetype has payload type: %u\n", codec);
found = TRUE;
}
}
} else {
if (debug)
ast_verbose("Discarded description format %s for ID %d\n", mimeSubtype, codec);
ast_verbose("Discarded description format %s for ID %u\n", mimeSubtype, codec);
}
} else if (!strncmp(a, red_fmtp, strlen(red_fmtp))) {
/* count numbers of generations in fmtp */
red_cp = &red_fmtp[strlen(red_fmtp)];
strncpy(red_fmtp, a, 100);
sscanf(red_cp, "%30u", &red_data_pt[*red_num_gen]);
sscanf(red_cp, "%30u", (unsigned *)&red_data_pt[*red_num_gen]);
red_cp = strtok(red_cp, "/");
while (red_cp && (*red_num_gen)++ < AST_RED_MAX_GENERATION) {
sscanf(red_cp, "%30u", &red_data_pt[*red_num_gen]);
sscanf(red_cp, "%30u", (unsigned *)&red_data_pt[*red_num_gen]);
red_cp = strtok(NULL, "/");
}
red_cp = red_fmtp;
@@ -11383,10 +11383,10 @@ static int process_sdp_a_image(const char *a, struct sip_pvt *p)
}
if ((sscanf(attrib, "t38faxmaxbuffer:%30u", &x) == 1)) {
ast_debug(3, "MaxBufferSize:%d\n", x);
ast_debug(3, "MaxBufferSize:%u\n", x);
found = TRUE;
} else if ((sscanf(attrib, "t38maxbitrate:%30u", &x) == 1) || (sscanf(attrib, "t38faxmaxrate:%30u", &x) == 1)) {
ast_debug(3, "T38MaxBitRate: %d\n", x);
ast_debug(3, "T38MaxBitRate: %u\n", x);
switch (x) {
case 14400:
p->t38.their_parms.rate = AST_T38_RATE_14400;
@@ -11415,7 +11415,7 @@ static int process_sdp_a_image(const char *a, struct sip_pvt *p)
} else if ((sscanf(attrib, "t38faxmaxdatagram:%30u", &x) == 1) || (sscanf(attrib, "t38maxdatagram:%30u", &x) == 1)) {
/* override the supplied value if the configuration requests it */
if (((signed int) p->t38_maxdatagram >= 0) && ((unsigned int) p->t38_maxdatagram > x)) {
ast_debug(1, "Overriding T38FaxMaxDatagram '%d' with '%d'\n", x, p->t38_maxdatagram);
ast_debug(1, "Overriding T38FaxMaxDatagram '%u' with '%d'\n", x, p->t38_maxdatagram);
x = p->t38_maxdatagram;
}
ast_debug(3, "FaxMaxDatagram: %u\n", x);
@@ -11423,7 +11423,7 @@ static int process_sdp_a_image(const char *a, struct sip_pvt *p)
found = TRUE;
} else if ((strncmp(attrib, "t38faxfillbitremoval", sizeof("t38faxfillbitremoval") - 1) == 0)) {
if (sscanf(attrib, "t38faxfillbitremoval:%30u", &x) == 1) {
ast_debug(3, "FillBitRemoval: %d\n", x);
ast_debug(3, "FillBitRemoval: %u\n", x);
if (x == 1) {
p->t38.their_parms.fill_bit_removal = TRUE;
}
@@ -11434,7 +11434,7 @@ static int process_sdp_a_image(const char *a, struct sip_pvt *p)
found = TRUE;
} else if ((strncmp(attrib, "t38faxtranscodingmmr", sizeof("t38faxtranscodingmmr") - 1) == 0)) {
if (sscanf(attrib, "t38faxtranscodingmmr:%30u", &x) == 1) {
ast_debug(3, "Transcoding MMR: %d\n", x);
ast_debug(3, "Transcoding MMR: %u\n", x);
if (x == 1) {
p->t38.their_parms.transcoding_mmr = TRUE;
}
@@ -11445,7 +11445,7 @@ static int process_sdp_a_image(const char *a, struct sip_pvt *p)
found = TRUE;
} else if ((strncmp(attrib, "t38faxtranscodingjbig", sizeof("t38faxtranscodingjbig") - 1) == 0)) {
if (sscanf(attrib, "t38faxtranscodingjbig:%30u", &x) == 1) {
ast_debug(3, "Transcoding JBIG: %d\n", x);
ast_debug(3, "Transcoding JBIG: %u\n", x);
if (x == 1) {
p->t38.their_parms.transcoding_jbig = TRUE;
}
@@ -11540,7 +11540,7 @@ static int finalize_content(struct sip_request *req)
return -1;
}
snprintf(clen, sizeof(clen), "%zd", ast_str_strlen(req->content));
snprintf(clen, sizeof(clen), "%zu", ast_str_strlen(req->content));
add_header(req, "Content-Length", clen);
if (ast_str_strlen(req->content)) {
@@ -12768,7 +12768,7 @@ static void add_ice_to_sdp(struct ast_rtp_instance *instance, struct ast_str **a
i = ao2_iterator_init(candidates, 0);
while ((candidate = ao2_iterator_next(&i))) {
ast_str_append(a_buf, 0, "a=candidate:%s %d %s %d ", candidate->foundation, candidate->id, candidate->transport, candidate->priority);
ast_str_append(a_buf, 0, "a=candidate:%s %u %s %d ", candidate->foundation, candidate->id, candidate->transport, candidate->priority);
ast_str_append(a_buf, 0, "%s ", ast_sockaddr_stringify_host(&candidate->address));
if (candidate->type == AST_RTP_ICE_CANDIDATE_TYPE_SRFLX
@@ -12870,7 +12870,7 @@ static void add_codec_to_sdp(const struct sip_pvt *p,
unsigned int rate;
if (debug)
ast_verbose("Adding codec %d (%s) to SDP\n", format->id, ast_getformatname(format));
ast_verbose("Adding codec %u (%s) to SDP\n", format->id, ast_getformatname(format));
if (((rtp_code = ast_rtp_codecs_payload_code(ast_rtp_instance_get_codecs(p->rtp), 1, format, 0)) == -1) ||
!(mime = ast_rtp_lookup_mime_subtype2(1, format, 0, ast_test_flag(&p->flags[0], SIP_G726_NONSTANDARD) ? AST_RTP_OPT_G726_NONSTANDARD : 0)) ||
@@ -12886,9 +12886,9 @@ static void add_codec_to_sdp(const struct sip_pvt *p,
ast_str_append(m_buf, 0, " %d", rtp_code);
/* Opus mandates 2 channels in rtpmap */
if ((int)format->id == AST_FORMAT_OPUS) {
ast_str_append(a_buf, 0, "a=rtpmap:%d %s/%d/2\r\n", rtp_code, mime, rate);
ast_str_append(a_buf, 0, "a=rtpmap:%d %s/%u/2\r\n", rtp_code, mime, rate);
} else {
ast_str_append(a_buf, 0, "a=rtpmap:%d %s/%d\r\n", rtp_code, mime, rate);
ast_str_append(a_buf, 0, "a=rtpmap:%d %s/%u\r\n", rtp_code, mime, rate);
}
ast_format_sdp_generate(format, rtp_code, a_buf);
@@ -12952,7 +12952,7 @@ static void add_vcodec_to_sdp(const struct sip_pvt *p, struct ast_format *format
return;
if (debug)
ast_verbose("Adding video codec %d (%s) to SDP\n", format->id, ast_getformatname(format));
ast_verbose("Adding video codec %u (%s) to SDP\n", format->id, ast_getformatname(format));
if (((rtp_code = ast_rtp_codecs_payload_code(ast_rtp_instance_get_codecs(p->vrtp), 1, format, 0)) == -1) ||
!(subtype = ast_rtp_lookup_mime_subtype2(1, format, 0, 0)) ||
@@ -12961,7 +12961,7 @@ static void add_vcodec_to_sdp(const struct sip_pvt *p, struct ast_format *format
}
ast_str_append(m_buf, 0, " %d", rtp_code);
ast_str_append(a_buf, 0, "a=rtpmap:%d %s/%d\r\n", rtp_code, subtype, rate);
ast_str_append(a_buf, 0, "a=rtpmap:%d %s/%u\r\n", rtp_code, subtype, rate);
/* VP8: add RTCP FIR support */
if ((int)format->id == AST_FORMAT_VP8) {
ast_str_append(a_buf, 0, "a=rtcp-fb:* ccm fir\r\n");
@@ -12981,13 +12981,13 @@ static void add_tcodec_to_sdp(const struct sip_pvt *p, struct ast_format *format
return;
if (debug)
ast_verbose("Adding text codec %d (%s) to SDP\n", format->id, ast_getformatname(format));
ast_verbose("Adding text codec %u (%s) to SDP\n", format->id, ast_getformatname(format));
if ((rtp_code = ast_rtp_codecs_payload_code(ast_rtp_instance_get_codecs(p->trtp), 1, format, 0)) == -1)
return;
ast_str_append(m_buf, 0, " %d", rtp_code);
ast_str_append(a_buf, 0, "a=rtpmap:%d %s/%d\r\n", rtp_code,
ast_str_append(a_buf, 0, "a=rtpmap:%d %s/%u\r\n", rtp_code,
ast_rtp_lookup_mime_subtype2(1, format, 0, 0),
ast_rtp_lookup_sample_rate2(1, format, 0));
/* Add fmtp code here */
@@ -13033,12 +13033,12 @@ static void add_noncodec_to_sdp(const struct sip_pvt *p, int format,
int rtp_code;
if (debug)
ast_verbose("Adding non-codec 0x%x (%s) to SDP\n", format, ast_rtp_lookup_mime_subtype2(0, NULL, format, 0));
ast_verbose("Adding non-codec 0x%x (%s) to SDP\n", (unsigned)format, ast_rtp_lookup_mime_subtype2(0, NULL, format, 0));
if ((rtp_code = ast_rtp_codecs_payload_code(ast_rtp_instance_get_codecs(p->rtp), 0, NULL, format)) == -1)
return;
ast_str_append(m_buf, 0, " %d", rtp_code);
ast_str_append(a_buf, 0, "a=rtpmap:%d %s/%d\r\n", rtp_code,
ast_str_append(a_buf, 0, "a=rtpmap:%d %s/%u\r\n", rtp_code,
ast_rtp_lookup_mime_subtype2(0, NULL, format, 0),
ast_rtp_lookup_sample_rate2(0, NULL, format));
if (format == AST_RTP_DTMF) /* Indicate we support DTMF and FLASH... */
@@ -13503,8 +13503,8 @@ static enum sip_result add_sdp(struct sip_request *resp, struct sip_pvt *p, int
"IP6" : "IP4", ast_sockaddr_stringify_addr_remote(&udptldest));
}
ast_str_append(&a_modem, 0, "a=T38FaxVersion:%d\r\n", p->t38.our_parms.version);
ast_str_append(&a_modem, 0, "a=T38MaxBitRate:%d\r\n", t38_get_rate(p->t38.our_parms.rate));
ast_str_append(&a_modem, 0, "a=T38FaxVersion:%u\r\n", p->t38.our_parms.version);
ast_str_append(&a_modem, 0, "a=T38MaxBitRate:%u\r\n", t38_get_rate(p->t38.our_parms.rate));
if (p->t38.our_parms.fill_bit_removal) {
ast_str_append(&a_modem, 0, "a=T38FaxFillBitRemoval\r\n");
}
@@ -14355,7 +14355,7 @@ static int transmit_invite(struct sip_pvt *p, int sipmethod, int sdp, int init,
if (sdp) {
offered_media_list_destroy(p);
if (p->udptl && p->t38.state == T38_LOCAL_REINVITE) {
ast_debug(1, "T38 is in state %d on channel %s\n", p->t38.state, p->owner ? ast_channel_name(p->owner) : "<none>");
ast_debug(1, "T38 is in state %u on channel %s\n", p->t38.state, p->owner ? ast_channel_name(p->owner) : "<none>");
add_sdp(&req, p, FALSE, FALSE, TRUE);
} else if (p->rtp) {
try_suggested_sip_codec(p);
@@ -14861,7 +14861,7 @@ static int transmit_cc_notify(struct ast_cc_agent *agent, struct sip_pvt *subscr
char subscription_state_hdr[64];
if (state < CC_QUEUED || state > CC_READY) {
ast_log(LOG_WARNING, "Invalid state provided for transmit_cc_notify (%d)\n", state);
ast_log(LOG_WARNING, "Invalid state provided for transmit_cc_notify (%u)\n", state);
return -1;
}
@@ -15808,7 +15808,7 @@ void sip_auth_headers(enum sip_auth_type code, char **header, char **respheader)
*header = "Proxy-Authenticate";
*respheader = "Proxy-Authorization";
} else {
ast_verbose("-- wrong response code %d\n", code);
ast_verbose("-- wrong response code %u\n", code);
*header = *respheader = "Invalid";
}
}
@@ -16498,7 +16498,7 @@ static int build_path(struct sip_pvt *p, struct sip_peer *peer, struct sip_reque
static void build_nonce(struct sip_pvt *p, int forceupdate)
{
if (p->stalenonce || forceupdate || ast_strlen_zero(p->nonce)) {
ast_string_field_build(p, nonce, "%08lx", ast_random()); /* Create nonce for challenge */
ast_string_field_build(p, nonce, "%08lx", (unsigned long)ast_random()); /* Create nonce for challenge */
p->stalenonce = 0;
}
}
@@ -20334,7 +20334,7 @@ static char *_sip_show_peer(int type, int fd, struct mansession *s, const struct
ast_cli(fd, " DirectMedACL : %s\n", AST_CLI_YESNO(ast_acl_list_is_empty(peer->directmediaacl) == 0));
ast_cli(fd, " T.38 support : %s\n", AST_CLI_YESNO(ast_test_flag(&peer->flags[1], SIP_PAGE2_T38SUPPORT)));
ast_cli(fd, " T.38 EC mode : %s\n", faxec2str(ast_test_flag(&peer->flags[1], SIP_PAGE2_T38SUPPORT)));
ast_cli(fd, " T.38 MaxDtgrm: %d\n", peer->t38_maxdatagram);
ast_cli(fd, " T.38 MaxDtgrm: %u\n", peer->t38_maxdatagram);
ast_cli(fd, " DirectMedia : %s\n", AST_CLI_YESNO(ast_test_flag(&peer->flags[0], SIP_DIRECT_MEDIA)));
ast_cli(fd, " PromiscRedir : %s\n", AST_CLI_YESNO(ast_test_flag(&peer->flags[0], SIP_PROMISCREDIR)));
ast_cli(fd, " User=Phone : %s\n", AST_CLI_YESNO(ast_test_flag(&peer->flags[0], SIP_USEREQPHONE)));
@@ -20473,7 +20473,7 @@ static char *_sip_show_peer(int type, int fd, struct mansession *s, const struct
astman_append(s, "SIP-TextSupport: %s\r\n", (ast_test_flag(&peer->flags[1], SIP_PAGE2_TEXTSUPPORT)?"Y":"N"));
astman_append(s, "SIP-T.38Support: %s\r\n", (ast_test_flag(&peer->flags[1], SIP_PAGE2_T38SUPPORT)?"Y":"N"));
astman_append(s, "SIP-T.38EC: %s\r\n", faxec2str(ast_test_flag(&peer->flags[1], SIP_PAGE2_T38SUPPORT)));
astman_append(s, "SIP-T.38MaxDtgrm: %d\r\n", peer->t38_maxdatagram);
astman_append(s, "SIP-T.38MaxDtgrm: %u\r\n", peer->t38_maxdatagram);
astman_append(s, "SIP-Sess-Timers: %s\r\n", stmode2str(peer->stimer.st_mode_oper));
astman_append(s, "SIP-Sess-Refresh: %s\r\n", strefresherparam2str(peer->stimer.st_ref));
astman_append(s, "SIP-Sess-Expires: %d\r\n", peer->stimer.st_max_se);
@@ -20978,7 +20978,7 @@ static char *sip_show_settings(struct ast_cli_entry *e, int cmd, struct ast_cli_
ast_cli(a->fd, " T.38 support: %s\n", AST_CLI_YESNO(ast_test_flag(&global_flags[1], SIP_PAGE2_T38SUPPORT)));
ast_cli(a->fd, " T.38 EC mode: %s\n", faxec2str(ast_test_flag(&global_flags[1], SIP_PAGE2_T38SUPPORT)));
ast_cli(a->fd, " T.38 MaxDtgrm: %d\n", global_t38_maxdatagram);
ast_cli(a->fd, " T.38 MaxDtgrm: %u\n", global_t38_maxdatagram);
if (!realtimepeers && !realtimeregs)
ast_cli(a->fd, " SIP realtime: Disabled\n" );
else
@@ -20992,10 +20992,10 @@ static char *sip_show_settings(struct ast_cli_entry *e, int cmd, struct ast_cli_
ast_cli(a->fd, " IP ToS RTP audio: %s\n", ast_tos2str(global_tos_audio));
ast_cli(a->fd, " IP ToS RTP video: %s\n", ast_tos2str(global_tos_video));
ast_cli(a->fd, " IP ToS RTP text: %s\n", ast_tos2str(global_tos_text));
ast_cli(a->fd, " 802.1p CoS SIP: %d\n", global_cos_sip);
ast_cli(a->fd, " 802.1p CoS RTP audio: %d\n", global_cos_audio);
ast_cli(a->fd, " 802.1p CoS RTP video: %d\n", global_cos_video);
ast_cli(a->fd, " 802.1p CoS RTP text: %d\n", global_cos_text);
ast_cli(a->fd, " 802.1p CoS SIP: %u\n", global_cos_sip);
ast_cli(a->fd, " 802.1p CoS RTP audio: %u\n", global_cos_audio);
ast_cli(a->fd, " 802.1p CoS RTP video: %u\n", global_cos_video);
ast_cli(a->fd, " 802.1p CoS RTP text: %u\n", global_cos_text);
ast_cli(a->fd, " Jitterbuffer enabled: %s\n", AST_CLI_YESNO(ast_test_flag(&global_jbconf, AST_JB_ENABLED)));
if (ast_test_flag(&global_jbconf, AST_JB_ENABLED)) {
ast_cli(a->fd, " Jitterbuffer forced: %s\n", AST_CLI_YESNO(ast_test_flag(&global_jbconf, AST_JB_FORCED)));
@@ -22166,7 +22166,7 @@ static int build_reply_digest(struct sip_pvt *p, int method, char* digest, int d
else
snprintf(uri, sizeof(uri), "%s:%s@%s", p->socket.type == AST_TRANSPORT_TLS ? "sips" : "sip", p->username, ast_sockaddr_stringify_host_remote(&p->sa));
snprintf(cnonce, sizeof(cnonce), "%08lx", ast_random());
snprintf(cnonce, sizeof(cnonce), "%08lx", (unsigned long)ast_random());
/* Check if we have peer credentials */
ao2_lock(p);
@@ -22224,7 +22224,7 @@ static int build_reply_digest(struct sip_pvt *p, int method, char* digest, int d
p->noncecount++;
if (!ast_strlen_zero(p->qop))
snprintf(resp, sizeof(resp), "%s:%s:%08x:%s:%s:%s", a1_hash, p->nonce, p->noncecount, cnonce, "auth", a2_hash);
snprintf(resp, sizeof(resp), "%s:%s:%08x:%s:%s:%s", a1_hash, p->nonce, (unsigned)p->noncecount, cnonce, "auth", a2_hash);
else
snprintf(resp, sizeof(resp), "%s:%s:%s", a1_hash, p->nonce, a2_hash);
ast_md5_hash(resp_hash, resp);
@@ -22236,7 +22236,7 @@ static int build_reply_digest(struct sip_pvt *p, int method, char* digest, int d
/* XXX We hard code our qop to "auth" for now. XXX */
if (!ast_strlen_zero(p->qop))
snprintf(digest, digest_len, "Digest username=\"%s\", realm=\"%s\", algorithm=MD5, uri=\"%s\", nonce=\"%s\", response=\"%s\"%s, qop=auth, cnonce=\"%s\", nc=%08x", username, p->realm, uri, p->nonce, resp_hash, opaque, cnonce, p->noncecount);
snprintf(digest, digest_len, "Digest username=\"%s\", realm=\"%s\", algorithm=MD5, uri=\"%s\", nonce=\"%s\", response=\"%s\"%s, qop=auth, cnonce=\"%s\", nc=%08x", username, p->realm, uri, p->nonce, resp_hash, opaque, cnonce, (unsigned)p->noncecount);
else
snprintf(digest, digest_len, "Digest username=\"%s\", realm=\"%s\", algorithm=MD5, uri=\"%s\", nonce=\"%s\", response=\"%s\"%s", username, p->realm, uri, p->nonce, resp_hash, opaque);
@@ -22405,7 +22405,7 @@ static int function_sippeer(struct ast_channel *chan, const char *cmd, char *dat
} else if (!strcasecmp(colname, "codecs")) {
ast_getformatname_multiple(buf, len -1, peer->caps);
} else if (!strcasecmp(colname, "encryption")) {
snprintf(buf, len, "%d", ast_test_flag(&peer->flags[1], SIP_PAGE2_USE_SRTP));
snprintf(buf, len, "%u", ast_test_flag(&peer->flags[1], SIP_PAGE2_USE_SRTP));
} else if (!strncasecmp(colname, "chanvar[", 8)) {
char *chanvar=colname + 8;
struct ast_variable *v;
@@ -25130,7 +25130,7 @@ static int handle_request_invite_st(struct sip_pvt *p, struct sip_request *req,
break;
default:
ast_log(LOG_ERROR, "Internal Error %d at %s:%d\n", st_get_mode(p, 1), __FILE__, __LINE__);
ast_log(LOG_ERROR, "Internal Error %u at %s:%d\n", st_get_mode(p, 1), __FILE__, __LINE__);
break;
}
} else {
@@ -25884,7 +25884,7 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, str
p->invitestate = INV_TERMINATED;
break;
default:
ast_log(LOG_WARNING, "Don't know how to handle INVITE in state %d\n", ast_channel_state(c));
ast_log(LOG_WARNING, "Don't know how to handle INVITE in state %u\n", ast_channel_state(c));
transmit_response(p, "100 Trying", req);
break;
}
@@ -27998,7 +27998,7 @@ static int handle_incoming(struct sip_pvt *p, struct sip_request *req, struct as
(could be new request in existing SIP dialog as well...)
*/
p->method = req->method; /* Find out which SIP method they are using */
ast_debug(4, "**** Received %s (%d) - Command in SIP %s\n", sip_methods[p->method].text, sip_methods[p->method].id, cmd);
ast_debug(4, "**** Received %s (%u) - Command in SIP %s\n", sip_methods[p->method].text, sip_methods[p->method].id, cmd);
if (p->icseq && (p->icseq > seqno) ) {
if (p->pendinginvite && seqno == p->pendinginvite && (req->method == SIP_ACK || req->method == SIP_CANCEL)) {
@@ -29833,7 +29833,7 @@ static void set_insecure_flags (struct ast_flags *flags, const char *value, int
\returns non-zero if any config options were handled, zero otherwise
*/
static int handle_t38_options(struct ast_flags *flags, struct ast_flags *mask, struct ast_variable *v,
int *maxdatagram)
unsigned int *maxdatagram)
{
int res = 1;