*
* \author Joshua Colp <jcolp@digium.com>
*
- * \brief Gulp SIP Channel Driver
+ * \brief PSJIP SIP Channel Driver
*
* \ingroup channel_drivers
*/
/*** MODULEINFO
<depend>pjproject</depend>
- <depend>res_sip</depend>
- <depend>res_sip_session</depend>
+ <depend>res_pjsip</depend>
+ <depend>res_pjsip_session</depend>
<support_level>core</support_level>
***/
#include "asterisk/stasis_channels.h"
#include "asterisk/indications.h"
-#include "asterisk/res_sip.h"
-#include "asterisk/res_sip_session.h"
+#include "asterisk/res_pjsip.h"
+#include "asterisk/res_pjsip_session.h"
/*** DOCUMENTATION
- <function name="GULP_DIAL_CONTACTS" language="en_US">
+ <function name="PJSIP_DIAL_CONTACTS" language="en_US">
<synopsis>
Return a dial string for dialing all contacts on an AOR.
</synopsis>
<para>Returns a properly formatted dial string for dialing all contacts on an AOR.</para>
</description>
</function>
- <function name="GULP_MEDIA_OFFER" language="en_US">
+ <function name="PJSIP_MEDIA_OFFER" language="en_US">
<synopsis>
Media and codec offerings to be set on an outbound SIP channel prior to dialing.
</synopsis>
</function>
***/
-static const char desc[] = "Gulp SIP Channel";
-static const char channel_type[] = "Gulp";
+static const char desc[] = "PJSIP Channel";
+static const char channel_type[] = "PJSIP";
static unsigned int chan_idx;
SIP_MEDIA_SIZE,
};
-struct gulp_pvt {
+struct chan_pjsip_pvt {
struct ast_sip_session_media *media[SIP_MEDIA_SIZE];
};
-static void gulp_pvt_dtor(void *obj)
+static void chan_pjsip_pvt_dtor(void *obj)
{
- struct gulp_pvt *pvt = obj;
+ struct chan_pjsip_pvt *pvt = obj;
int i;
for (i = 0; i < SIP_MEDIA_SIZE; ++i) {
}
/* \brief Asterisk core interaction functions */
-static struct ast_channel *gulp_request(const char *type, struct ast_format_cap *cap, const struct ast_channel *requestor, const char *data, int *cause);
-static int gulp_sendtext(struct ast_channel *ast, const char *text);
-static int gulp_digit_begin(struct ast_channel *ast, char digit);
-static int gulp_digit_end(struct ast_channel *ast, char digit, unsigned int duration);
-static int gulp_call(struct ast_channel *ast, const char *dest, int timeout);
-static int gulp_hangup(struct ast_channel *ast);
-static int gulp_answer(struct ast_channel *ast);
-static struct ast_frame *gulp_read(struct ast_channel *ast);
-static int gulp_write(struct ast_channel *ast, struct ast_frame *f);
-static int gulp_indicate(struct ast_channel *ast, int condition, const void *data, size_t datalen);
-static int gulp_transfer(struct ast_channel *ast, const char *target);
-static int gulp_fixup(struct ast_channel *oldchan, struct ast_channel *newchan);
-static int gulp_devicestate(const char *data);
-static int gulp_queryoption(struct ast_channel *ast, int option, void *data, int *datalen);
+static struct ast_channel *chan_pjsip_request(const char *type, struct ast_format_cap *cap, const struct ast_channel *requestor, const char *data, int *cause);
+static int chan_pjsip_sendtext(struct ast_channel *ast, const char *text);
+static int chan_pjsip_digit_begin(struct ast_channel *ast, char digit);
+static int chan_pjsip_digit_end(struct ast_channel *ast, char digit, unsigned int duration);
+static int chan_pjsip_call(struct ast_channel *ast, const char *dest, int timeout);
+static int chan_pjsip_hangup(struct ast_channel *ast);
+static int chan_pjsip_answer(struct ast_channel *ast);
+static struct ast_frame *chan_pjsip_read(struct ast_channel *ast);
+static int chan_pjsip_write(struct ast_channel *ast, struct ast_frame *f);
+static int chan_pjsip_indicate(struct ast_channel *ast, int condition, const void *data, size_t datalen);
+static int chan_pjsip_transfer(struct ast_channel *ast, const char *target);
+static int chan_pjsip_fixup(struct ast_channel *oldchan, struct ast_channel *newchan);
+static int chan_pjsip_devicestate(const char *data);
+static int chan_pjsip_queryoption(struct ast_channel *ast, int option, void *data, int *datalen);
/*! \brief PBX interface structure for channel registration */
-static struct ast_channel_tech gulp_tech = {
+static struct ast_channel_tech chan_pjsip_tech = {
.type = channel_type,
- .description = "Gulp SIP Channel Driver",
- .requester = gulp_request,
- .send_text = gulp_sendtext,
- .send_digit_begin = gulp_digit_begin,
- .send_digit_end = gulp_digit_end,
- .call = gulp_call,
- .hangup = gulp_hangup,
- .answer = gulp_answer,
- .read = gulp_read,
- .write = gulp_write,
- .write_video = gulp_write,
- .exception = gulp_read,
- .indicate = gulp_indicate,
- .transfer = gulp_transfer,
- .fixup = gulp_fixup,
- .devicestate = gulp_devicestate,
- .queryoption = gulp_queryoption,
+ .description = "PJSIP Channel Driver",
+ .requester = chan_pjsip_request,
+ .send_text = chan_pjsip_sendtext,
+ .send_digit_begin = chan_pjsip_digit_begin,
+ .send_digit_end = chan_pjsip_digit_end,
+ .call = chan_pjsip_call,
+ .hangup = chan_pjsip_hangup,
+ .answer = chan_pjsip_answer,
+ .read = chan_pjsip_read,
+ .write = chan_pjsip_write,
+ .write_video = chan_pjsip_write,
+ .exception = chan_pjsip_read,
+ .indicate = chan_pjsip_indicate,
+ .transfer = chan_pjsip_transfer,
+ .fixup = chan_pjsip_fixup,
+ .devicestate = chan_pjsip_devicestate,
+ .queryoption = chan_pjsip_queryoption,
.properties = AST_CHAN_TP_WANTSJITTER | AST_CHAN_TP_CREATESJITTER
};
/*! \brief SIP session interaction functions */
-static void gulp_session_begin(struct ast_sip_session *session);
-static void gulp_session_end(struct ast_sip_session *session);
-static int gulp_incoming_request(struct ast_sip_session *session, struct pjsip_rx_data *rdata);
-static void gulp_incoming_response(struct ast_sip_session *session, struct pjsip_rx_data *rdata);
+static void chan_pjsip_session_begin(struct ast_sip_session *session);
+static void chan_pjsip_session_end(struct ast_sip_session *session);
+static int chan_pjsip_incoming_request(struct ast_sip_session *session, struct pjsip_rx_data *rdata);
+static void chan_pjsip_incoming_response(struct ast_sip_session *session, struct pjsip_rx_data *rdata);
/*! \brief SIP session supplement structure */
-static struct ast_sip_session_supplement gulp_supplement = {
+static struct ast_sip_session_supplement chan_pjsip_supplement = {
.method = "INVITE",
.priority = AST_SIP_SESSION_SUPPLEMENT_PRIORITY_CHANNEL,
- .session_begin = gulp_session_begin,
- .session_end = gulp_session_end,
- .incoming_request = gulp_incoming_request,
- .incoming_response = gulp_incoming_response,
+ .session_begin = chan_pjsip_session_begin,
+ .session_end = chan_pjsip_session_end,
+ .incoming_request = chan_pjsip_incoming_request,
+ .incoming_response = chan_pjsip_incoming_response,
};
-static int gulp_incoming_ack(struct ast_sip_session *session, struct pjsip_rx_data *rdata);
+static int chan_pjsip_incoming_ack(struct ast_sip_session *session, struct pjsip_rx_data *rdata);
-static struct ast_sip_session_supplement gulp_ack_supplement = {
+static struct ast_sip_session_supplement chan_pjsip_ack_supplement = {
.method = "ACK",
.priority = AST_SIP_SESSION_SUPPLEMENT_PRIORITY_CHANNEL,
- .incoming_request = gulp_incoming_ack,
+ .incoming_request = chan_pjsip_incoming_ack,
};
/*! \brief Dialplan function for constructing a dial string for calling all contacts */
-static int gulp_dial_contacts(struct ast_channel *chan, const char *cmd, char *data, char *buf, size_t len)
+static int chan_pjsip_dial_contacts(struct ast_channel *chan, const char *cmd, char *data, char *buf, size_t len)
{
RAII_VAR(struct ast_sip_endpoint *, endpoint, NULL, ao2_cleanup);
RAII_VAR(struct ast_str *, dial, NULL, ast_free_ptr);
it_contacts = ao2_iterator_init(contacts, 0);
for (; (contact = ao2_iterator_next(&it_contacts)); ao2_ref(contact, -1)) {
- ast_str_append(&dial, -1, "Gulp/");
+ ast_str_append(&dial, -1, "PJSIP/");
if (!ast_strlen_zero(args.request_user)) {
ast_str_append(&dial, -1, "%s@", args.request_user);
return 0;
}
-static struct ast_custom_function gulp_dial_contacts_function = {
- .name = "GULP_DIAL_CONTACTS",
- .read = gulp_dial_contacts,
+static struct ast_custom_function chan_pjsip_dial_contacts_function = {
+ .name = "PJSIP_DIAL_CONTACTS",
+ .read = chan_pjsip_dial_contacts,
};
static int media_offer_read_av(struct ast_sip_session *session, char *buf,
name = ast_getformatname(&fmt);
if (ast_strlen_zero(name)) {
- ast_log(LOG_WARNING, "GULP_MEDIA_OFFER unrecognized format %s\n", name);
+ ast_log(LOG_WARNING, "PJSIP_MEDIA_OFFER unrecognized format %s\n", name);
continue;
}
}
static struct ast_custom_function media_offer_function = {
- .name = "GULP_MEDIA_OFFER",
+ .name = "PJSIP_MEDIA_OFFER",
.read = media_offer_read,
.write = media_offer_write
};
/*! \brief Function called by RTP engine to get local audio RTP peer */
-static enum ast_rtp_glue_result gulp_get_rtp_peer(struct ast_channel *chan, struct ast_rtp_instance **instance)
+static enum ast_rtp_glue_result chan_pjsip_get_rtp_peer(struct ast_channel *chan, struct ast_rtp_instance **instance)
{
struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(chan);
- struct gulp_pvt *pvt = channel->pvt;
+ struct chan_pjsip_pvt *pvt = channel->pvt;
struct ast_sip_endpoint *endpoint;
if (!pvt || !channel->session || !pvt->media[SIP_MEDIA_AUDIO]->rtp) {
}
/*! \brief Function called by RTP engine to get local video RTP peer */
-static enum ast_rtp_glue_result gulp_get_vrtp_peer(struct ast_channel *chan, struct ast_rtp_instance **instance)
+static enum ast_rtp_glue_result chan_pjsip_get_vrtp_peer(struct ast_channel *chan, struct ast_rtp_instance **instance)
{
struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(chan);
- struct gulp_pvt *pvt = channel->pvt;
+ struct chan_pjsip_pvt *pvt = channel->pvt;
struct ast_sip_endpoint *endpoint;
if (!pvt || !channel->session || !pvt->media[SIP_MEDIA_VIDEO]->rtp) {
}
/*! \brief Function called by RTP engine to get peer capabilities */
-static void gulp_get_codec(struct ast_channel *chan, struct ast_format_cap *result)
+static void chan_pjsip_get_codec(struct ast_channel *chan, struct ast_format_cap *result)
{
struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(chan);
}
/*! \brief Function called by RTP engine to change where the remote party should send media */
-static int gulp_set_rtp_peer(struct ast_channel *chan,
+static int chan_pjsip_set_rtp_peer(struct ast_channel *chan,
struct ast_rtp_instance *rtp,
struct ast_rtp_instance *vrtp,
struct ast_rtp_instance *tpeer,
int nat_active)
{
struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(chan);
- struct gulp_pvt *pvt = channel->pvt;
+ struct chan_pjsip_pvt *pvt = channel->pvt;
struct ast_sip_session *session = channel->session;
int changed = 0;
struct ast_channel *bridge_peer;
}
/*! \brief Local glue for interacting with the RTP engine core */
-static struct ast_rtp_glue gulp_rtp_glue = {
- .type = "Gulp",
- .get_rtp_info = gulp_get_rtp_peer,
- .get_vrtp_info = gulp_get_vrtp_peer,
- .get_codec = gulp_get_codec,
- .update_peer = gulp_set_rtp_peer,
+static struct ast_rtp_glue chan_pjsip_rtp_glue = {
+ .type = "PJSIP",
+ .get_rtp_info = chan_pjsip_get_rtp_peer,
+ .get_vrtp_info = chan_pjsip_get_vrtp_peer,
+ .get_codec = chan_pjsip_get_codec,
+ .update_peer = chan_pjsip_set_rtp_peer,
};
-/*! \brief Function called to create a new Gulp Asterisk channel */
-static struct ast_channel *gulp_new(struct ast_sip_session *session, int state, const char *exten, const char *title, const char *linkedid, const char *cid_name)
+/*! \brief Function called to create a new PJSIP Asterisk channel */
+static struct ast_channel *chan_pjsip_new(struct ast_sip_session *session, int state, const char *exten, const char *title, const char *linkedid, const char *cid_name)
{
struct ast_channel *chan;
struct ast_format fmt;
- RAII_VAR(struct gulp_pvt *, pvt, NULL, ao2_cleanup);
+ RAII_VAR(struct chan_pjsip_pvt *, pvt, NULL, ao2_cleanup);
struct ast_sip_channel_pvt *channel;
- if (!(pvt = ao2_alloc(sizeof(*pvt), gulp_pvt_dtor))) {
+ if (!(pvt = ao2_alloc(sizeof(*pvt), chan_pjsip_pvt_dtor))) {
return NULL;
}
- if (!(chan = ast_channel_alloc(1, state, S_OR(session->id.number.str, ""), S_OR(session->id.name.str, ""), "", "", "", linkedid, 0, "Gulp/%s-%08x", ast_sorcery_object_get_id(session->endpoint),
+ if (!(chan = ast_channel_alloc(1, state, S_OR(session->id.number.str, ""), S_OR(session->id.name.str, ""), "", "", "", linkedid, 0, "PJSIP/%s-%08x", ast_sorcery_object_get_id(session->endpoint),
ast_atomic_fetchadd_int((int *)&chan_idx, +1)))) {
return NULL;
}
- ast_channel_tech_set(chan, &gulp_tech);
+ ast_channel_tech_set(chan, &chan_pjsip_tech);
if (!(channel = ast_sip_channel_pvt_alloc(pvt, session))) {
ast_hangup(chan);
return (status == PJ_SUCCESS) ? 0 : -1;
}
-/*! \brief Function called by core when we should answer a Gulp session */
-static int gulp_answer(struct ast_channel *ast)
+/*! \brief Function called by core when we should answer a PJSIP session */
+static int chan_pjsip_answer(struct ast_channel *ast)
{
struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(ast);
}
/*! \brief Internal helper function called when CNG tone is detected */
-static struct ast_frame *gulp_cng_tone_detected(struct ast_sip_session *session, struct ast_frame *f)
+static struct ast_frame *chan_pjsip_cng_tone_detected(struct ast_sip_session *session, struct ast_frame *f)
{
const char *target_context;
int exists;
}
/*! \brief Function called by core to read any waiting frames */
-static struct ast_frame *gulp_read(struct ast_channel *ast)
+static struct ast_frame *chan_pjsip_read(struct ast_channel *ast)
{
struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(ast);
- struct gulp_pvt *pvt = channel->pvt;
+ struct chan_pjsip_pvt *pvt = channel->pvt;
struct ast_frame *f;
struct ast_sip_session_media *media = NULL;
int rtcp = 0;
if (f && (f->frametype == AST_FRAME_DTMF)) {
if (f->subclass.integer == 'f') {
ast_debug(3, "Fax CNG detected on %s\n", ast_channel_name(ast));
- f = gulp_cng_tone_detected(channel->session, f);
+ f = chan_pjsip_cng_tone_detected(channel->session, f);
} else {
ast_debug(3, "* Detected inband DTMF '%c' on '%s'\n", f->subclass.integer,
ast_channel_name(ast));
}
/*! \brief Function called by core to write frames */
-static int gulp_write(struct ast_channel *ast, struct ast_frame *frame)
+static int chan_pjsip_write(struct ast_channel *ast, struct ast_frame *frame)
{
struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(ast);
- struct gulp_pvt *pvt = channel->pvt;
+ struct chan_pjsip_pvt *pvt = channel->pvt;
struct ast_sip_session_media *media;
int res = 0;
case AST_FRAME_MODEM:
break;
default:
- ast_log(LOG_WARNING, "Can't send %d type frames with Gulp\n", frame->frametype);
+ ast_log(LOG_WARNING, "Can't send %d type frames with PJSIP\n", frame->frametype);
break;
}
{
struct fixup_data *fix_data = data;
struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(fix_data->chan);
- struct gulp_pvt *pvt = channel->pvt;
+ struct chan_pjsip_pvt *pvt = channel->pvt;
channel->session->channel = fix_data->chan;
if (pvt->media[SIP_MEDIA_AUDIO] && pvt->media[SIP_MEDIA_AUDIO]->rtp) {
}
/*! \brief Function called by core to change the underlying owner channel */
-static int gulp_fixup(struct ast_channel *oldchan, struct ast_channel *newchan)
+static int chan_pjsip_fixup(struct ast_channel *oldchan, struct ast_channel *newchan)
{
struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(newchan);
struct fixup_data fix_data;
}
/*! \brief Function called to get the device state of an endpoint */
-static int gulp_devicestate(const char *data)
+static int chan_pjsip_devicestate(const char *data)
{
RAII_VAR(struct ast_sip_endpoint *, endpoint, ast_sorcery_retrieve_by_id(ast_sip_get_sorcery(), "endpoint", data), ao2_cleanup);
enum ast_device_state state = AST_DEVICE_UNKNOWN;
}
/*! \brief Function called to query options on a channel */
-static int gulp_queryoption(struct ast_channel *ast, int option, void *data, int *datalen)
+static int chan_pjsip_queryoption(struct ast_channel *ast, int option, void *data, int *datalen)
{
struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(ast);
struct ast_sip_session *session = channel->session;
}
/*! \brief Function called by core to ask the channel to indicate some sort of condition */
-static int gulp_indicate(struct ast_channel *ast, int condition, const void *data, size_t datalen)
+static int chan_pjsip_indicate(struct ast_channel *ast, int condition, const void *data, size_t datalen)
{
struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(ast);
- struct gulp_pvt *pvt = channel->pvt;
+ struct chan_pjsip_pvt *pvt = channel->pvt;
struct ast_sip_session_media *media;
int response_code = 0;
int res = 0;
} else {
res = -1;
}
- ast_devstate_changed(AST_DEVICE_UNKNOWN, AST_DEVSTATE_CACHABLE, "Gulp/%s", ast_sorcery_object_get_id(channel->session->endpoint));
+ ast_devstate_changed(AST_DEVICE_UNKNOWN, AST_DEVSTATE_CACHABLE, "PJSIP/%s", ast_sorcery_object_get_id(channel->session->endpoint));
break;
case AST_CONTROL_BUSY:
if (ast_channel_state(ast) != AST_STATE_UP) {
}
/*! \brief Function called by core for Asterisk initiated transfer */
-static int gulp_transfer(struct ast_channel *chan, const char *target)
+static int chan_pjsip_transfer(struct ast_channel *chan, const char *target)
{
struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(chan);
struct transfer_data *trnf_data = transfer_data_alloc(channel->session, target);
}
/*! \brief Function called by core to start a DTMF digit */
-static int gulp_digit_begin(struct ast_channel *chan, char digit)
+static int chan_pjsip_digit_begin(struct ast_channel *chan, char digit)
{
struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(chan);
- struct gulp_pvt *pvt = channel->pvt;
+ struct chan_pjsip_pvt *pvt = channel->pvt;
struct ast_sip_session_media *media = pvt->media[SIP_MEDIA_AUDIO];
int res = 0;
}
/*! \brief Function called by core to stop a DTMF digit */
-static int gulp_digit_end(struct ast_channel *ast, char digit, unsigned int duration)
+static int chan_pjsip_digit_end(struct ast_channel *ast, char digit, unsigned int duration)
{
struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(ast);
- struct gulp_pvt *pvt = channel->pvt;
+ struct chan_pjsip_pvt *pvt = channel->pvt;
struct ast_sip_session_media *media = pvt->media[SIP_MEDIA_AUDIO];
int res = 0;
}
/*! \brief Function called by core to actually start calling a remote party */
-static int gulp_call(struct ast_channel *ast, const char *dest, int timeout)
+static int chan_pjsip_call(struct ast_channel *ast, const char *dest, int timeout)
{
struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(ast);
}
/*! \brief Clear a channel from a session along with its PVT */
-static void clear_session_and_channel(struct ast_sip_session *session, struct ast_channel *ast, struct gulp_pvt *pvt)
+static void clear_session_and_channel(struct ast_sip_session *session, struct ast_channel *ast, struct chan_pjsip_pvt *pvt)
{
session->channel = NULL;
if (pvt->media[SIP_MEDIA_AUDIO] && pvt->media[SIP_MEDIA_AUDIO]->rtp) {
struct hangup_data *h_data = data;
struct ast_channel *ast = h_data->chan;
struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(ast);
- struct gulp_pvt *pvt = channel->pvt;
+ struct chan_pjsip_pvt *pvt = channel->pvt;
struct ast_sip_session *session = channel->session;
int cause = h_data->cause;
return 0;
}
-/*! \brief Function called by core to hang up a Gulp session */
-static int gulp_hangup(struct ast_channel *ast)
+/*! \brief Function called by core to hang up a PJSIP session */
+static int chan_pjsip_hangup(struct ast_channel *ast)
{
struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(ast);
- struct gulp_pvt *pvt = channel->pvt;
+ struct chan_pjsip_pvt *pvt = channel->pvt;
int cause = hangup_cause2sip(ast_channel_hangupcause(channel->session->channel));
struct hangup_data *h_data = hangup_data_alloc(cause, ast);
);
if (ast_strlen_zero(tmp)) {
- ast_log(LOG_ERROR, "Unable to create Gulp channel with empty destination\n");
+ ast_log(LOG_ERROR, "Unable to create PJSIP channel with empty destination\n");
req_data->cause = AST_CAUSE_CHANNEL_UNACCEPTABLE;
return -1;
}
}
if (ast_strlen_zero(endpoint_name)) {
- ast_log(LOG_ERROR, "Unable to create Gulp channel with empty endpoint name\n");
+ ast_log(LOG_ERROR, "Unable to create PJSIP channel with empty endpoint name\n");
req_data->cause = AST_CAUSE_CHANNEL_UNACCEPTABLE;
} else if (!(endpoint = ast_sorcery_retrieve_by_id(ast_sip_get_sorcery(), "endpoint", endpoint_name))) {
- ast_log(LOG_ERROR, "Unable to create Gulp channel - endpoint '%s' was not found\n", endpoint_name);
+ ast_log(LOG_ERROR, "Unable to create PJSIP channel - endpoint '%s' was not found\n", endpoint_name);
req_data->cause = AST_CAUSE_NO_ROUTE_DESTINATION;
return -1;
}
return 0;
}
-/*! \brief Function called by core to create a new outgoing Gulp session */
-static struct ast_channel *gulp_request(const char *type, struct ast_format_cap *cap, const struct ast_channel *requestor, const char *data, int *cause)
+/*! \brief Function called by core to create a new outgoing PJSIP session */
+static struct ast_channel *chan_pjsip_request(const char *type, struct ast_format_cap *cap, const struct ast_channel *requestor, const char *data, int *cause)
{
struct request_data req_data;
RAII_VAR(struct ast_sip_session *, session, NULL, ao2_cleanup);
session = req_data.session;
- if (!(session->channel = gulp_new(session, AST_STATE_DOWN, NULL, NULL, requestor ? ast_channel_linkedid(requestor) : NULL, NULL))) {
+ if (!(session->channel = chan_pjsip_new(session, AST_STATE_DOWN, NULL, NULL, requestor ? ast_channel_linkedid(requestor) : NULL, NULL))) {
/* Session needs to be terminated prematurely */
return NULL;
}
return 0;
}
-/*! \brief Function called by core to send text on Gulp session */
-static int gulp_sendtext(struct ast_channel *ast, const char *text)
+/*! \brief Function called by core to send text on PJSIP session */
+static int chan_pjsip_sendtext(struct ast_channel *ast, const char *text)
{
struct ast_sip_channel_pvt *channel = ast_channel_tech_pvt(ast);
struct sendtext_data *data = sendtext_data_create(channel->session, text);
return 0;
}
-static void gulp_session_begin(struct ast_sip_session *session)
+static void chan_pjsip_session_begin(struct ast_sip_session *session)
{
RAII_VAR(struct ast_datastore *, datastore, NULL, ao2_cleanup);
}
/*! \brief Function called when the session ends */
-static void gulp_session_end(struct ast_sip_session *session)
+static void chan_pjsip_session_end(struct ast_sip_session *session)
{
if (!session->channel) {
return;
}
/*! \brief Function called when a request is received on the session */
-static int gulp_incoming_request(struct ast_sip_session *session, struct pjsip_rx_data *rdata)
+static int chan_pjsip_incoming_request(struct ast_sip_session *session, struct pjsip_rx_data *rdata)
{
pjsip_tx_data *packet = NULL;
return 0;
}
- if (!(session->channel = gulp_new(session, AST_STATE_RING, session->exten, NULL, NULL, NULL))) {
+ if (!(session->channel = chan_pjsip_new(session, AST_STATE_RING, session->exten, NULL, NULL, NULL))) {
if (pjsip_inv_end_session(session->inv_session, 503, NULL, &packet) == PJ_SUCCESS) {
ast_sip_session_send_response(session, packet);
}
- ast_log(LOG_ERROR, "Failed to allocate new GULP channel on incoming SIP INVITE\n");
+ ast_log(LOG_ERROR, "Failed to allocate new PJSIP channel on incoming SIP INVITE\n");
return -1;
}
/* channel gets created on incoming request, but we wait to call start
break;
}
- ast_debug(3, "Started PBX on new GULP channel %s\n", ast_channel_name(session->channel));
+ ast_debug(3, "Started PBX on new PJSIP channel %s\n", ast_channel_name(session->channel));
return (res == AST_PBX_SUCCESS) ? 0 : -1;
}
};
/*! \brief Function called when a response is received on the session */
-static void gulp_incoming_response(struct ast_sip_session *session, struct pjsip_rx_data *rdata)
+static void chan_pjsip_incoming_response(struct ast_sip_session *session, struct pjsip_rx_data *rdata)
{
struct pjsip_status_line status = rdata->msg_info.msg->line.status;
}
}
-static int gulp_incoming_ack(struct ast_sip_session *session, struct pjsip_rx_data *rdata)
+static int chan_pjsip_incoming_ack(struct ast_sip_session *session, struct pjsip_rx_data *rdata)
{
if (rdata->msg_info.msg->line.req.method.id == PJSIP_ACK_METHOD) {
if (session->endpoint->media.direct_media.enabled) {
*/
static int load_module(void)
{
- if (!(gulp_tech.capabilities = ast_format_cap_alloc())) {
+ if (!(chan_pjsip_tech.capabilities = ast_format_cap_alloc())) {
return AST_MODULE_LOAD_DECLINE;
}
- ast_format_cap_add_all_by_type(gulp_tech.capabilities, AST_FORMAT_TYPE_AUDIO);
+ ast_format_cap_add_all_by_type(chan_pjsip_tech.capabilities, AST_FORMAT_TYPE_AUDIO);
- ast_rtp_glue_register(&gulp_rtp_glue);
+ ast_rtp_glue_register(&chan_pjsip_rtp_glue);
- if (ast_channel_register(&gulp_tech)) {
+ if (ast_channel_register(&chan_pjsip_tech)) {
ast_log(LOG_ERROR, "Unable to register channel class %s\n", channel_type);
goto end;
}
- if (ast_custom_function_register(&gulp_dial_contacts_function)) {
- ast_log(LOG_ERROR, "Unable to register GULP_DIAL_CONTACTS dialplan function\n");
+ if (ast_custom_function_register(&chan_pjsip_dial_contacts_function)) {
+ ast_log(LOG_ERROR, "Unable to register PJSIP_DIAL_CONTACTS dialplan function\n");
goto end;
}
if (ast_custom_function_register(&media_offer_function)) {
- ast_log(LOG_WARNING, "Unable to register GULP_MEDIA_OFFER dialplan function\n");
+ ast_log(LOG_WARNING, "Unable to register PJSIP_MEDIA_OFFER dialplan function\n");
}
- if (ast_sip_session_register_supplement(&gulp_supplement)) {
- ast_log(LOG_ERROR, "Unable to register Gulp supplement\n");
+ if (ast_sip_session_register_supplement(&chan_pjsip_supplement)) {
+ ast_log(LOG_ERROR, "Unable to register PJSIP supplement\n");
goto end;
}
if (ast_sip_session_register_supplement(&pbx_start_supplement)) {
- ast_log(LOG_ERROR, "Unable to register Gulp pbx start supplement\n");
- ast_sip_session_unregister_supplement(&gulp_supplement);
+ ast_log(LOG_ERROR, "Unable to register PJSIP pbx start supplement\n");
+ ast_sip_session_unregister_supplement(&chan_pjsip_supplement);
goto end;
}
- if (ast_sip_session_register_supplement(&gulp_ack_supplement)) {
- ast_log(LOG_ERROR, "Unable to register Gulp ACK supplement\n");
+ if (ast_sip_session_register_supplement(&chan_pjsip_ack_supplement)) {
+ ast_log(LOG_ERROR, "Unable to register PJSIP ACK supplement\n");
ast_sip_session_unregister_supplement(&pbx_start_supplement);
- ast_sip_session_unregister_supplement(&gulp_supplement);
+ ast_sip_session_unregister_supplement(&chan_pjsip_supplement);
goto end;
}
end:
ast_custom_function_unregister(&media_offer_function);
- ast_custom_function_unregister(&gulp_dial_contacts_function);
- ast_channel_unregister(&gulp_tech);
- ast_rtp_glue_unregister(&gulp_rtp_glue);
+ ast_custom_function_unregister(&chan_pjsip_dial_contacts_function);
+ ast_channel_unregister(&chan_pjsip_tech);
+ ast_rtp_glue_unregister(&chan_pjsip_rtp_glue);
return AST_MODULE_LOAD_FAILURE;
}
return -1;
}
-/*! \brief Unload the Gulp channel from Asterisk */
+/*! \brief Unload the PJSIP channel from Asterisk */
static int unload_module(void)
{
ast_custom_function_unregister(&media_offer_function);
- ast_sip_session_unregister_supplement(&gulp_supplement);
+ ast_sip_session_unregister_supplement(&chan_pjsip_supplement);
ast_sip_session_unregister_supplement(&pbx_start_supplement);
- ast_custom_function_unregister(&gulp_dial_contacts_function);
- ast_channel_unregister(&gulp_tech);
- ast_rtp_glue_unregister(&gulp_rtp_glue);
+ ast_custom_function_unregister(&chan_pjsip_dial_contacts_function);
+ ast_channel_unregister(&chan_pjsip_tech);
+ ast_rtp_glue_unregister(&chan_pjsip_rtp_glue);
return 0;
}
-AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_LOAD_ORDER, "Gulp SIP Channel Driver",
+AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_LOAD_ORDER, "PJSIP Channel Driver",
.load = load_module,
.unload = unload_module,
.reload = reload,
$(if $(filter res_ael_share,$(EMBEDDED_MODS)),modules.link,res_ael_share.so): ael/ael_lex.o ael/ael.tab.o ael/pval.o
ael/ael_lex.o ael/ael.tab.o ael/pval.o: _ASTCFLAGS+=$(call MOD_ASTCFLAGS,res_ael_share)
-$(if $(filter res_sip,$(EMBEDDED_MODS)),modules.link,res_sip.so): $(subst .c,.o,$(wildcard res_sip/*.c))
-$(subst .c,.o,$(wildcard res_sip/*.c)): _ASTCFLAGS+=$(call MOD_ASTCFLAGS,res_sip)
+$(if $(filter res_pjsip,$(EMBEDDED_MODS)),modules.link,res_pjsip.so): $(subst .c,.o,$(wildcard res_pjsip/*.c))
+$(subst .c,.o,$(wildcard res_pjsip/*.c)): _ASTCFLAGS+=$(call MOD_ASTCFLAGS,res_pjsip)
$(if $(filter res_stasis,$(EMBEDDED_MODS)),modules.link,res_stasis.so): $(subst .c,.o,$(wildcard stasis/*.c))
$(subst .c,.o,$(wildcard stasis/*.c)): _ASTCFLAGS+=$(call MOD_ASTCFLAGS,res_stasis)
clean::
rm -f snmp/*.[oi] ael/*.[oi] ais/*.[oi] ari/*.[oi]
- rm -f res_sip/*.[oi] stasis/*.[oi]
+ rm -f res_pjsip/*.[oi] stasis/*.[oi]
rm -f parking/*.o parking/*.i
$(if $(filter res_parking,$(EMBEDDED_MODS)),modules.link,res_parking.so): $(subst .c,.o,$(wildcard parking/*.c))
#include <pjsip_simple.h>
#include <pjlib.h>
-#include "asterisk/res_sip.h"
-#include "res_sip/include/res_sip_private.h"
+#include "asterisk/res_pjsip.h"
+#include "res_pjsip/include/res_pjsip_private.h"
#include "asterisk/linkedlists.h"
#include "asterisk/logger.h"
#include "asterisk/lock.h"
***/
/*** DOCUMENTATION
- <configInfo name="res_sip" language="en_US">
+ <configInfo name="res_pjsip" language="en_US">
<synopsis>SIP Resource using PJProject</synopsis>
- <configFile name="res_sip.conf">
+ <configFile name="pjsip.conf">
<configObject name="endpoint">
<synopsis>Endpoint</synopsis>
<description><para>
accomplished via Addresses of Record (AoRs) which have one or more
contacts assicated with them. Endpoints <emphasis>NOT</emphasis> configured to
use a <literal>transport</literal> will default to first transport found
- in <filename>res_sip.conf</filename> that matches its type.
+ in <filename>pjsip.conf</filename> that matches its type.
</para>
<para>Example: An Endpoint has been configured with no transport.
When it comes time to call an AoR, PJSIP will find the
<synopsis>Authentication Object(s) associated with the endpoint</synopsis>
<description><para>
This is a comma-delimited list of <replaceable>auth</replaceable> sections defined
- in <filename>res_sip.conf</filename> to be used to verify inbound connection attempts.
+ in <filename>pjsip.conf</filename> to be used to verify inbound connection attempts.
</para><para>
Endpoints without an <literal>authentication</literal> object
configured will allow connections without vertification.
This will set the desired transport configuration to send SIP data through.
</para>
<warning><para>Not specifying a transport will <emphasis>DEFAULT</emphasis>
- to the first configured transport in <filename>res_sip.conf</filename> which is
+ to the first configured transport in <filename>pjsip.conf</filename> which is
valid for the URI we are trying to contact.
</para></warning>
</description>
identification from the endpoint from headers such as P-Asserted-Identity
or Remote-Party-ID header. This option applies both to calls originating from the
endpoint and calls originating from Asterisk. If <literal>no</literal>, the
- configured Caller-ID from res_sip.conf will always be used as the identity for
+ configured Caller-ID from pjsip.conf will always be used as the identity for
the endpoint.</para></description>
</configOption>
<configOption name="trust_id_outbound" default="no">
<synopsis>Send private identification details to the endpoint.</synopsis>
- <description><para>This option determines whether res_sip will send private
+ <description><para>This option determines whether res_pjsip will send private
identification information to the endpoint. If <literal>no</literal>,
private Caller-ID information will not be forwarded to the endpoint.
"Private" in this case refers to any method of restricting identification.
<synopsis>Use Endpoint's requested packetisation interval</synopsis>
</configOption>
<configOption name="use_avpf" default="no">
- <synopsis>Determines whether res_sip will use and enforce usage of AVPF for this
+ <synopsis>Determines whether res_pjsip will use and enforce usage of AVPF for this
endpoint.</synopsis>
<description><para>
- If set to <literal>yes</literal>, res_sip will use use the AVPF or SAVPF RTP
+ If set to <literal>yes</literal>, res_pjsip will use use the AVPF or SAVPF RTP
profile for all media offers on outbound calls and media updates and will
decline media offers not using the AVPF or SAVPF profile.
</para><para>
- If set to <literal>no</literal>, res_sip will use use the AVP or SAVP RTP
+ If set to <literal>no</literal>, res_pjsip will use use the AVP or SAVP RTP
profile for all media offers on outbound calls and media updates and will
decline media offers not using the AVP or SAVP profile.
</para></description>
</configOption>
<configOption name="media_encryption" default="no">
- <synopsis>Determines whether res_sip will use and enforce usage of media encryption
+ <synopsis>Determines whether res_pjsip will use and enforce usage of media encryption
for this endpoint.</synopsis>
<description>
<enumlist>
<enum name="no"><para>
- res_sip will offer no encryption and allow no encryption to be setup.
+ res_pjsip will offer no encryption and allow no encryption to be setup.
</para></enum>
<enum name="sdes"><para>
- res_sip will offer standard SRTP setup via in-SDP keys. Encrypted SIP
+ res_pjsip will offer standard SRTP setup via in-SDP keys. Encrypted SIP
transport should be used in conjunction with this option to prevent
exposure of media encryption keys.
</para></enum>
<enum name="dtls"><para>
- res_sip will offer DTLS-SRTP setup.
+ res_pjsip will offer DTLS-SRTP setup.
</para></enum>
</enumlist>
</description>
</configOption>
<configOption name="inband_progress" default="no">
- <synopsis>Determines whether chan_gulp will indicate ringing using inband
+ <synopsis>Determines whether chan_pjsip will indicate ringing using inband
progress.</synopsis>
<description><para>
- If set to <literal>yes</literal>, chan_gulp will send a 183 Session Progress
+ If set to <literal>yes</literal>, chan_pjsip will send a 183 Session Progress
when told to indicate ringing and will immediately start sending ringing
as audio.
</para><para>
- If set to <literal>no</literal>, chan_gulp will send a 180 Ringing when told
+ If set to <literal>no</literal>, chan_pjsip will send a 180 Ringing when told
to indicate ringing and will NOT send it as audio.
</para></description>
</configOption>
<synopsis>The number of in-use channels which will cause busy to be returned as device state</synopsis>
<description><para>
When the number of in-use channels for the endpoint matches the devicestate_busy_at setting the
- Gulp channel driver will return busy as the device state instead of in use.
+ PJSIP channel driver will return busy as the device state instead of in use.
</para></description>
</configOption>
<configOption name="t38udptl" default="no">
</para>
<enumlist>
<enum name="active"><para>
- res_sip will make a connection to the peer.
+ res_pjsip will make a connection to the peer.
</para></enum>
<enum name="passive"><para>
- res_sip will accept connections from the peer.
+ res_pjsip will accept connections from the peer.
</para></enum>
<enum name="actpass"><para>
- res_sip will offer and accept connections from the peer.
+ res_pjsip will offer and accept connections from the peer.
</para></enum>
</enumlist>
</description>
<emphasis>Transports</emphasis>
</para>
<para>There are different transports and protocol derivatives
- supported by <literal>res_sip</literal>. They are in order of
+ supported by <literal>res_pjsip</literal>. They are in order of
preference: UDP, TCP, and WebSocket (WS).</para>
<warning><para>
Multiple endpoints using the same connection is <emphasis>NOT</emphasis>
<configObject name="aor">
<synopsis>The configuration for a location of an endpoint</synopsis>
<description><para>
- An AoR is what allows Asterisk to contact an endpoint via res_sip. If no
+ An AoR is what allows Asterisk to contact an endpoint via res_pjsip. If no
AoRs are specified, an endpoint will not be reachable by Asterisk.
Beyond that, an AoR has other uses within Asterisk.
</para><para>
The settings in this section are global. In addition to being global, the values will
not be re-evaluated when a reload is performed. This is because the values must be set
before the SIP stack is initialized. The only way to reset these values is to either
- restart Asterisk, or unload res_sip.so and then load it again.
+ restart Asterisk, or unload res_pjsip.so and then load it again.
</para></description>
<configOption name="timert1" default="500">
<synopsis>Set transaction timer T1 value (milliseconds).</synopsis>
ast_sip_initialize_global_headers();
- if (ast_res_sip_initialize_configuration()) {
+ if (ast_res_pjsip_initialize_configuration()) {
ast_log(LOG_ERROR, "Failed to initialize SIP configuration. Aborting load\n");
goto error;
}
goto error;
}
- ast_res_sip_init_options_handling(0);
+ ast_res_pjsip_init_options_handling(0);
- ast_res_sip_init_contact_transports();
+ ast_res_pjsip_init_contact_transports();
return AST_MODULE_LOAD_SUCCESS;
error:
ast_sip_destroy_distributor();
- ast_res_sip_destroy_configuration();
+ ast_res_pjsip_destroy_configuration();
ast_sip_destroy_global_headers();
if (monitor_thread) {
stop_monitor_thread();
static int reload_module(void)
{
- if (ast_res_sip_reload_configuration()) {
+ if (ast_res_pjsip_reload_configuration()) {
return AST_MODULE_LOAD_DECLINE;
}
- ast_res_sip_init_options_handling(1);
+ ast_res_pjsip_init_options_handling(1);
return 0;
}
static int unload_module(void)
{
ast_sip_destroy_distributor();
- ast_res_sip_destroy_configuration();
+ ast_res_pjsip_destroy_configuration();
ast_sip_destroy_global_headers();
if (monitor_thread) {
stop_monitor_thread();
#include <pjsip.h>
#include <pjlib.h>
-#include "asterisk/res_sip.h"
+#include "asterisk/res_pjsip.h"
#include "asterisk/logger.h"
#include "asterisk/sorcery.h"
/*! \brief Initialize sorcery with auth support */
int ast_sip_initialize_sorcery_auth(struct ast_sorcery *sorcery)
{
- ast_sorcery_apply_default(sorcery, SIP_SORCERY_AUTH_TYPE, "config", "res_sip.conf,criteria=type=auth");
+ ast_sorcery_apply_default(sorcery, SIP_SORCERY_AUTH_TYPE, "config", "pjsip.conf,criteria=type=auth");
if (ast_sorcery_object_register(sorcery, SIP_SORCERY_AUTH_TYPE, auth_alloc, NULL, auth_apply)) {
return -1;
#include "pjsip.h"
#include "pjlib.h"
-#include "asterisk/res_sip.h"
+#include "asterisk/res_pjsip.h"
#include "asterisk/logger.h"
#include "asterisk/sorcery.h"
/*! \brief Initialize sorcery with domain alias support */
int ast_sip_initialize_sorcery_domain_alias(struct ast_sorcery *sorcery)
{
- ast_sorcery_apply_default(sorcery, SIP_SORCERY_DOMAIN_ALIAS_TYPE, "config", "res_sip.conf,criteria=type=domain_alias");
+ ast_sorcery_apply_default(sorcery, SIP_SORCERY_DOMAIN_ALIAS_TYPE, "config", "pjsip.conf,criteria=type=domain_alias");
if (ast_sorcery_object_register(sorcery, SIP_SORCERY_DOMAIN_ALIAS_TYPE, domain_alias_alloc, NULL, NULL)) {
return -1;
#include <pjsip.h>
#include <pjlib.h>
-#include "asterisk/res_sip.h"
+#include "asterisk/res_pjsip.h"
#include "asterisk/sorcery.h"
#include "asterisk/ast_version.h"
{
snprintf(default_useragent, sizeof(default_useragent), "%s %s", DEFAULT_USERAGENT_PREFIX, ast_get_version());
- ast_sorcery_apply_default(sorcery, "global", "config", "res_sip.conf,criteria=type=global");
+ ast_sorcery_apply_default(sorcery, "global", "config", "pjsip.conf,criteria=type=global");
if (ast_sorcery_object_register(sorcery, "global", global_alloc, NULL, global_apply)) {
return -1;
/*** MODULEINFO
<depend>pjproject</depend>
- <depend>res_sip</depend>
+ <depend>res_pjsip</depend>
<support_level>core</support_level>
***/
#include "asterisk.h"
#include <pjsip.h>
-#include "asterisk/res_sip.h"
+#include "asterisk/res_pjsip.h"
#include "asterisk/logger.h"
#include "asterisk/sorcery.h"
#include "asterisk/acl.h"
int ast_sip_initialize_sorcery_security(struct ast_sorcery *sorcery)
{
ast_sorcery_apply_default(sorcery, SIP_SORCERY_SECURITY_TYPE,
- "config", "res_sip.conf,criteria=type=security");
+ "config", "pjsip.conf,criteria=type=security");
if (ast_sorcery_object_register(sorcery, SIP_SORCERY_SECURITY_TYPE,
security_alloc, NULL, NULL)) {
#include <pjsip.h>
#include <pjlib.h>
-#include "asterisk/res_sip.h"
+#include "asterisk/res_pjsip.h"
#include "asterisk/sorcery.h"
-#include "include/res_sip_private.h"
+#include "include/res_pjsip_private.h"
#define TIMER_T1_MIN 100
#define DEFAULT_TIMER_T1 500
return -1;
}
- ast_sorcery_apply_config(system_sorcery, "res_sip");
+ ast_sorcery_apply_config(system_sorcery, "res_pjsip");
- ast_sorcery_apply_default(system_sorcery, "system", "config", "res_sip.conf,criteria=type=system");
+ ast_sorcery_apply_default(system_sorcery, "system", "config", "pjsip.conf,criteria=type=system");
if (ast_sorcery_object_register(system_sorcery, "system", system_alloc, NULL, system_apply)) {
ast_sorcery_unref(system_sorcery);
#include <pjsip.h>
#include <pjlib.h>
-#include "asterisk/res_sip.h"
+#include "asterisk/res_pjsip.h"
#include "asterisk/logger.h"
#include "asterisk/astobj2.h"
#include "asterisk/sorcery.h"
/*! \brief Initialize sorcery with transport support */
int ast_sip_initialize_sorcery_transport(struct ast_sorcery *sorcery)
{
- ast_sorcery_apply_default(sorcery, "transport", "config", "res_sip.conf,criteria=type=transport");
+ ast_sorcery_apply_default(sorcery, "transport", "config", "pjsip.conf,criteria=type=transport");
if (ast_sorcery_object_register(sorcery, "transport", transport_alloc, NULL, transport_apply)) {
return -1;
/*
- * res_sip.h
+ * res_pjsip.h
*
* Created on: Jan 25, 2013
* Author: mjordan
struct ao2_container;
/*!
- * \brief Initialize the configuration for res_sip
+ * \brief Initialize the configuration for res_pjsip
*/
-int ast_res_sip_initialize_configuration(void);
+int ast_res_pjsip_initialize_configuration(void);
/*!
* \brief Annihilate the configuration objects
*/
-void ast_res_sip_destroy_configuration(void);
+void ast_res_pjsip_destroy_configuration(void);
/*!
* \brief Reload the configuration
*/
-int ast_res_sip_reload_configuration(void);
+int ast_res_pjsip_reload_configuration(void);
/*!
* \brief Initialize OPTIONS request handling.
* \retval 0 on success
* \retval other on failure
*/
-int ast_res_sip_init_options_handling(int reload);
+int ast_res_pjsip_init_options_handling(int reload);
/*!
* \brief Initialize transport storage for contacts.
* \retval 0 on success
* \retval other on failure
*/
-int ast_res_sip_init_contact_transports(void);
+int ast_res_pjsip_init_contact_transports(void);
/*!
* \brief Initialize outbound authentication support
#include "pjsip.h"
#include "pjlib.h"
-#include "asterisk/res_sip.h"
+#include "asterisk/res_pjsip.h"
#include "asterisk/logger.h"
#include "asterisk/astobj2.h"
#include "asterisk/sorcery.h"
-#include "include/res_sip_private.h"
+#include "include/res_pjsip_private.h"
#define CONTACT_TRANSPORTS_BUCKETS 7
static struct ao2_container *contact_transports;
int ast_sip_initialize_sorcery_location(struct ast_sorcery *sorcery)
{
ast_sorcery_apply_default(sorcery, "contact", "astdb", "registrar");
- ast_sorcery_apply_default(sorcery, "aor", "config", "res_sip.conf,criteria=type=aor");
+ ast_sorcery_apply_default(sorcery, "aor", "config", "pjsip.conf,criteria=type=aor");
if (ast_sorcery_object_register(sorcery, "contact", contact_alloc, NULL, NULL) ||
ast_sorcery_object_register(sorcery, "aor", aor_alloc, NULL, NULL)) {
return 0;
}
-int ast_res_sip_init_contact_transports(void)
+int ast_res_pjsip_init_contact_transports(void)
{
if (contact_transports) {
ao2_t_ref(contact_transports, -1, "Remove old contact transports");
#include <pjsip.h>
#include <pjsip_ua.h>
-#include "asterisk/res_sip.h"
-#include "include/res_sip_private.h"
+#include "asterisk/res_pjsip.h"
+#include "include/res_pjsip_private.h"
#include "asterisk/cli.h"
#include "asterisk/astobj2.h"
#include "asterisk/utils.h"
ast_endpoint_blob_publish(persistent->endpoint, ast_endpoint_state_type(), blob);
- ast_devstate_changed(AST_DEVICE_UNKNOWN, AST_DEVSTATE_CACHABLE, "Gulp/%s", ast_endpoint_get_resource(persistent->endpoint));
+ ast_devstate_changed(AST_DEVICE_UNKNOWN, AST_DEVSTATE_CACHABLE, "PJSIP/%s", ast_endpoint_get_resource(persistent->endpoint));
return 0;
}
switch (cmd) {
case CLI_INIT:
- e->command = "sip show endpoints";
+ e->command = "pjsip show endpoints";
e->usage =
- "Usage: sip show endpoints\n"
- " Show the registered SIP endpoints\n";
+ "Usage: pjsip show endpoints\n"
+ " Show the registered PJSIP endpoints\n";
return NULL;
case CLI_GENERATE:
return NULL;
switch (cmd) {
case CLI_INIT:
- e->command = "sip show endpoint";
+ e->command = "pjsip show endpoint";
e->usage =
- "Usage: sip show endpoint <endpoint>\n"
- " Show the given SIP endpoint.\n";
+ "Usage: pjsip show endpoint <endpoint>\n"
+ " Show the given PJSIP endpoint.\n";
return NULL;
case CLI_GENERATE:
return NULL;
}
static struct ast_cli_entry cli_commands[] = {
- AST_CLI_DEFINE(handle_cli_show_endpoints, "Show SIP Endpoints"),
- AST_CLI_DEFINE(cli_show_endpoint, "Show SIP Endpoint")
+ AST_CLI_DEFINE(handle_cli_show_endpoints, "Show PJSIP Endpoints"),
+ AST_CLI_DEFINE(cli_show_endpoint, "Show PJSIP Endpoint")
};
static int dtmf_handler(const struct aco_option *opt, struct ast_variable *var, void *obj)
return NULL;
}
- if (!(persistent->endpoint = ast_endpoint_create("Gulp", ast_sorcery_object_get_id(endpoint)))) {
+ if (!(persistent->endpoint = ast_endpoint_create("PJSIP", ast_sorcery_object_get_id(endpoint)))) {
return NULL;
}
return 0;
}
-int ast_res_sip_initialize_configuration(void)
+int ast_res_pjsip_initialize_configuration(void)
{
if (ast_cli_register_multiple(cli_commands, ARRAY_LEN(cli_commands))) {
return -1;
return -1;
}
- ast_sorcery_apply_config(sip_sorcery, "res_sip");
+ ast_sorcery_apply_config(sip_sorcery, "res_pjsip");
if (ast_sip_initialize_sorcery_auth(sip_sorcery)) {
ast_log(LOG_ERROR, "Failed to register SIP authentication support\n");
return -1;
}
- ast_sorcery_apply_default(sip_sorcery, "endpoint", "config", "res_sip.conf,criteria=type=endpoint");
+ ast_sorcery_apply_default(sip_sorcery, "endpoint", "config", "pjsip.conf,criteria=type=endpoint");
ast_sorcery_apply_default(sip_sorcery, "nat_hook", "memory", NULL);
return 0;
}
-void ast_res_sip_destroy_configuration(void)
+void ast_res_pjsip_destroy_configuration(void)
{
ast_cli_unregister_multiple(cli_commands, ARRAY_LEN(cli_commands));
ast_sorcery_unref(sip_sorcery);
}
-int ast_res_sip_reload_configuration(void)
+int ast_res_pjsip_reload_configuration(void)
{
if (sip_sorcery) {
ast_sorcery_reload(sip_sorcery);
#include <pjsip.h>
-#include "asterisk/res_sip.h"
+#include "asterisk/res_pjsip.h"
static int distribute(void *data);
static pj_bool_t distributor(pjsip_rx_data *rdata);
#include <pjsip.h>
#include <pjlib.h>
-#include "asterisk/res_sip.h"
+#include "asterisk/res_pjsip.h"
#include "asterisk/linkedlists.h"
static pj_status_t add_request_headers(pjsip_tx_data *tdata);
#include <pjsip_ua.h>
#include <pjlib.h>
-#include "asterisk/res_sip.h"
+#include "asterisk/res_pjsip.h"
#include "asterisk/channel.h"
#include "asterisk/pbx.h"
#include "asterisk/astobj2.h"
#include "asterisk/cli.h"
#include "asterisk/time.h"
-#include "include/res_sip_private.h"
+#include "include/res_pjsip_private.h"
#define DEFAULT_LANGUAGE "en"
#define DEFAULT_ENCODING "text/plain"
qualify_and_schedule_permanent_cb, NULL);
}
-int ast_res_sip_init_options_handling(int reload)
+int ast_res_pjsip_init_options_handling(int reload)
{
const pj_str_t STR_OPTIONS = { "OPTIONS", 7 };
#define bzero bzero
#include "pjsip.h"
-#include "asterisk/res_sip.h"
+#include "asterisk/res_pjsip.h"
#include "asterisk/module.h"
-#include "include/res_sip_private.h"
+#include "include/res_pjsip_private.h"
static pj_bool_t outbound_auth(pjsip_rx_data *rdata);
#include <pjsip.h>
-#include "asterisk/res_sip.h"
+#include "asterisk/res_pjsip.h"
#include "asterisk/security_events.h"
static int find_transport_in_use(void *obj, void *arg, int flags)
/*** MODULEINFO
<depend>pjproject</depend>
- <depend>res_sip</depend>
+ <depend>res_pjsip</depend>
<support_level>core</support_level>
***/
#include <pjsip.h>
-#include "asterisk/res_sip.h"
+#include "asterisk/res_pjsip.h"
#include "asterisk/module.h"
#include "asterisk/logger.h"
#include "asterisk/sorcery.h"
#include "asterisk/acl.h"
/*** DOCUMENTATION
- <configInfo name="res_sip_acl" language="en_US">
+ <configInfo name="res_pjsip_acl" language="en_US">
<synopsis>SIP ACL module</synopsis>
<description><para>
<emphasis>ACL</emphasis>
</para>
- <para>The ACL module used by <literal>res_sip</literal>. This module is
+ <para>The ACL module used by <literal>res_pjsip</literal>. This module is
independent of <literal>endpoints</literal> and operates on all inbound
- SIP communication using res_sip.
+ SIP communication using res_pjsip.
</para><para>
It should be noted that this module can also reference ACLs from
<filename>acl.conf</filename>.
and <literal>Contact Header</literal>. It is possible to create a combined ACL using
both IP and Contact.
</para></description>
- <configFile name="res_sip.conf">
+ <configFile name="pjsip.conf">
<configObject name="acl">
<synopsis>Access Control List</synopsis>
<configOption name="acl">
return 0;
}
-AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_LOAD_ORDER, "SIP ACL Resource",
+AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_LOAD_ORDER, "PJSIP ACL Resource",
.load = load_module,
.unload = unload_module,
.load_pri = AST_MODPRI_APP_DEPEND,
#include <pjsip.h>
-#include "asterisk/res_sip.h"
+#include "asterisk/res_pjsip.h"
#include "asterisk/logger.h"
#include "asterisk/module.h"
#include "asterisk/strings.h"
/*** MODULEINFO
<depend>pjproject</depend>
- <depend>res_sip</depend>
+ <depend>res_pjsip</depend>
<support_level>core</support_level>
***/
return 0;
}
-AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_LOAD_ORDER, "SIP authentication resource",
+AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_LOAD_ORDER, "PJSIP authentication resource",
.load = load_module,
.unload = unload_module,
.reload = reload_module,
/*** MODULEINFO
<depend>pjproject</depend>
- <depend>res_sip</depend>
- <depend>res_sip_session</depend>
+ <depend>res_pjsip</depend>
+ <depend>res_pjsip_session</depend>
<support_level>core</support_level>
***/
#include <pjsip.h>
#include <pjsip_ua.h>
-#include "asterisk/res_sip.h"
-#include "asterisk/res_sip_session.h"
+#include "asterisk/res_pjsip.h"
+#include "asterisk/res_pjsip_session.h"
#include "asterisk/channel.h"
#include "asterisk/module.h"
#include "asterisk/callerid.h"
return 0;
}
-AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_LOAD_ORDER, "SIP Caller ID Support",
+AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_LOAD_ORDER, "PJSIP Caller ID Support",
.load = load_module,
.unload = unload_module,
.load_pri = AST_MODPRI_APP_DEPEND,
/*** MODULEINFO
<depend>pjproject</depend>
- <depend>res_sip</depend>
- <depend>res_sip_session</depend>
+ <depend>res_pjsip</depend>
+ <depend>res_pjsip_session</depend>
<support_level>core</support_level>
***/
#include <pjsip.h>
#include <pjsip_ua.h>
-#include "asterisk/res_sip.h"
-#include "asterisk/res_sip_session.h"
+#include "asterisk/res_pjsip.h"
+#include "asterisk/res_pjsip_session.h"
#include "asterisk/callerid.h"
#include "asterisk/channel.h"
#include "asterisk/module.h"
return 0;
}
-AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_LOAD_ORDER, "SIP Add Diversion Header Support",
+AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_LOAD_ORDER, "PJSIP Add Diversion Header Support",
.load = load_module,
.unload = unload_module,
.load_pri = AST_MODPRI_APP_DEPEND,
/*** MODULEINFO
<depend>pjproject</depend>
- <depend>res_sip</depend>
- <depend>res_sip_session</depend>
+ <depend>res_pjsip</depend>
+ <depend>res_pjsip_session</depend>
<support_level>core</support_level>
***/
#include <pjsip.h>
#include <pjsip_ua.h>
-#include "asterisk/res_sip.h"
-#include "asterisk/res_sip_session.h"
+#include "asterisk/res_pjsip.h"
+#include "asterisk/res_pjsip_session.h"
#include "asterisk/module.h"
static int dtmf_info_incoming_request(struct ast_sip_session *session, struct pjsip_rx_data *rdata)
return 0;
}
-AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_LOAD_ORDER, "SIP DTMF INFO Support",
+AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_LOAD_ORDER, "PJSIP DTMF INFO Support",
.load = load_module,
.unload = unload_module,
.load_pri = AST_MODPRI_APP_DEPEND,
/*** MODULEINFO
<depend>pjproject</depend>
- <depend>res_sip</depend>
+ <depend>res_pjsip</depend>
<support_level>core</support_level>
***/
#include <pjsip.h>
-#include "asterisk/res_sip.h"
+#include "asterisk/res_pjsip.h"
#include "asterisk/module.h"
static int get_endpoint_details(pjsip_rx_data *rdata, char *domain, size_t domain_size)
return 0;
}
-AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_LOAD_ORDER, "SIP Anonymous endpoint identifier",
+AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_LOAD_ORDER, "PJSIP Anonymous endpoint identifier",
.load = load_module,
.unload = unload_module,
.load_pri = AST_MODPRI_DEFAULT,
/*** MODULEINFO
<depend>pjproject</depend>
- <depend>res_sip</depend>
+ <depend>res_pjsip</depend>
<defaultenabled>no</defaultenabled>
<support_level>core</support_level>
***/
#include <pjsip.h>
-#include "asterisk/res_sip.h"
+#include "asterisk/res_pjsip.h"
#include "asterisk/module.h"
static struct ast_sip_endpoint *constant_identify(pjsip_rx_data *rdata)
return 0;
}
-AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_LOAD_ORDER, "SIP Constant Endpoint Identifier",
+AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_LOAD_ORDER, "PJSIP Constant Endpoint Identifier",
.load = load_module,
.unload = unload_module,
.load_pri = AST_MODPRI_APP_DEPEND,
/*** MODULEINFO
<depend>pjproject</depend>
- <depend>res_sip</depend>
+ <depend>res_pjsip</depend>
<support_level>core</support_level>
***/
#include <pjsip.h>
-#include "asterisk/res_sip.h"
+#include "asterisk/res_pjsip.h"
#include "asterisk/module.h"
#include "asterisk/acl.h"
/*** DOCUMENTATION
- <configInfo name="res_sip_endpoint_identifier_ip" language="en_US">
+ <configInfo name="res_pjsip_endpoint_identifier_ip" language="en_US">
<synopsis>Module that identifies endpoints via source IP address</synopsis>
- <configFile name="res_sip.conf">
+ <configFile name="res_pjsip.conf">
<configObject name="identify">
<configOption name="endpoint">
<synopsis>Name of Endpoint</synopsis>
static int load_module(void)
{
- ast_sorcery_apply_default(ast_sip_get_sorcery(), "identify", "config", "res_sip.conf,criteria=type=identify");
+ ast_sorcery_apply_default(ast_sip_get_sorcery(), "identify", "config", "res_pjsip.conf,criteria=type=identify");
if (ast_sorcery_object_register(ast_sip_get_sorcery(), "identify", ip_identify_alloc, NULL, NULL)) {
return AST_MODULE_LOAD_DECLINE;
return 0;
}
-AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_LOAD_ORDER, "SIP IP endpoint identifier",
+AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_LOAD_ORDER, "PJSIP IP endpoint identifier",
.load = load_module,
.reload = reload_module,
.unload = unload_module,
/*** MODULEINFO
<depend>pjproject</depend>
- <depend>res_sip</depend>
+ <depend>res_pjsip</depend>
<support_level>core</support_level>
***/
#include <pjsip.h>
-#include "asterisk/res_sip.h"
+#include "asterisk/res_pjsip.h"
#include "asterisk/module.h"
static int get_endpoint_details(pjsip_rx_data *rdata, char *endpoint, size_t endpoint_size, char *domain, size_t domain_size)
return 0;
}
-AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_LOAD_ORDER, "SIP username endpoint identifier",
+AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_LOAD_ORDER, "PJSIP username endpoint identifier",
.load = load_module,
.unload = unload_module,
.load_pri = AST_MODPRI_APP_DEPEND,
/*** MODULEINFO
<depend>pjproject</depend>
- <depend>res_sip</depend>
- <depend>res_sip_pubsub</depend>
+ <depend>res_pjsip</depend>
+ <depend>res_pjsip_pubsub</depend>
<support_level>core</support_level>
***/
#include <pjsip_simple.h>
#include <pjlib.h>
-#include "asterisk/res_sip.h"
-#include "asterisk/res_sip_pubsub.h"
-#include "asterisk/res_sip_exten_state.h"
+#include "asterisk/res_pjsip.h"
+#include "asterisk/res_pjsip_pubsub.h"
+#include "asterisk/res_pjsip_exten_state.h"
#include "asterisk/module.h"
#include "asterisk/logger.h"
#include "asterisk/astobj2.h"
return 0;
}
-AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_GLOBAL_SYMBOLS | AST_MODFLAG_LOAD_ORDER, "SIP Extension State Notifications",
+AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_GLOBAL_SYMBOLS | AST_MODFLAG_LOAD_ORDER, "PJSIP Extension State Notifications",
.load = load_module,
.unload = unload_module,
.load_pri = AST_MODPRI_CHANNEL_DEPEND,
/*** MODULEINFO
<depend>pjproject</depend>
- <depend>res_sip</depend>
+ <depend>res_pjsip</depend>
<support_level>core</support_level>
***/
#include <pjsip.h>
-#include "asterisk/res_sip.h"
+#include "asterisk/res_pjsip.h"
#include "asterisk/module.h"
#include "asterisk/logger.h"
return 0;
}
-AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_LOAD_ORDER, "SIP Packet Logger",
+AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_LOAD_ORDER, "PJSIP Packet Logger",
.load = load_module,
.unload = unload_module,
.load_pri = AST_MODPRI_APP_DEPEND,
/*** MODULEINFO
<depend>pjproject</depend>
- <depend>res_sip</depend>
- <depend>res_sip_session</depend>
+ <depend>res_pjsip</depend>
+ <depend>res_pjsip_session</depend>
<support_level>core</support_level>
***/
#include "asterisk/message.h"
#include "asterisk/module.h"
#include "asterisk/pbx.h"
-#include "asterisk/res_sip.h"
-#include "asterisk/res_sip_session.h"
+#include "asterisk/res_pjsip.h"
+#include "asterisk/res_pjsip_session.h"
const pjsip_method pjsip_message_method = {PJSIP_OTHER_METHOD, {"MESSAGE", 7} };
return 0;
}
-AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_LOAD_ORDER, "SIP Messaging Support",
+AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_LOAD_ORDER, "PJSIP Messaging Support",
.load = load_module,
.unload = unload_module,
.load_pri = AST_MODPRI_APP_DEPEND,
/*** MODULEINFO
<depend>pjproject</depend>
- <depend>res_sip</depend>
+ <depend>res_pjsip</depend>
<support_level>core</support_level>
***/
#include <pjsip_simple.h>
#include <pjlib.h>
-#include "asterisk/res_sip.h"
-#include "asterisk/res_sip_pubsub.h"
+#include "asterisk/res_pjsip.h"
+#include "asterisk/res_pjsip_pubsub.h"
#include "asterisk/module.h"
#include "asterisk/logger.h"
#include "asterisk/astobj2.h"
/*** MODULEINFO
<depend>pjproject</depend>
- <depend>res_sip</depend>
- <depend>res_sip_pubsub</depend>
+ <depend>res_pjsip</depend>
+ <depend>res_pjsip_pubsub</depend>
<support_level>core</support_level>
***/
return 0;
}
-AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_LOAD_ORDER, "SIP MWI resource",
+AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_LOAD_ORDER, "PJSIP MWI resource",
.load = load_module,
.unload = unload_module,
.reload = reload,
/*** MODULEINFO
<depend>pjproject</depend>
- <depend>res_sip</depend>
+ <depend>res_pjsip</depend>
<support_level>core</support_level>
***/
#include <pjsip.h>
#include <pjsip_ua.h>
-#include "asterisk/res_sip.h"
+#include "asterisk/res_pjsip.h"
#include "asterisk/module.h"
#include "asterisk/acl.h"
return 0;
}
-AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_LOAD_ORDER, "SIP NAT Support",
+AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_LOAD_ORDER, "PJSIP NAT Support",
.load = load_module,
.unload = unload_module,
.load_pri = AST_MODPRI_APP_DEPEND,
/*** MODULEINFO
<depend>pjproject</depend>
- <depend>res_sip</depend>
+ <depend>res_pjsip</depend>
<support_level>core</support_level>
***/
#include "asterisk/manager.h"
#include "asterisk/module.h"
#include "asterisk/pbx.h"
-#include "asterisk/res_sip.h"
+#include "asterisk/res_pjsip.h"
#include "asterisk/sorcery.h"
#define CONTENT_TYPE_SIZE 64
switch (cmd) {
case CLI_INIT:
- e->command = "gulp notify";
+ e->command = "pjsip notify";
e->usage =
- "Usage:
gulp notify <type> <peer> [<peer>...]\n"
+ "Usage:
pjsip notify <type> <peer> [<peer>...]\n"
" Send a NOTIFY request to an endpoint\n"
" Message types are defined in sip_notify.conf\n";
return NULL;
struct ast_variable *vars = astman_get_variables(m);
if (ast_strlen_zero(endpoint_name)) {
- astman_send_error(s, m, "GulpNotify requires a channel name");
+ astman_send_error(s, m, "PJSIPNotify requires a channel name");
return 0;
}
}
ast_cli_register_multiple(cli_options, ARRAY_LEN(cli_options));
- ast_manager_register_xml("GulpNotify", EVENT_FLAG_SYSTEM, manager_notify);
+ ast_manager_register_xml("PJSIPNotify", EVENT_FLAG_SYSTEM, manager_notify);
return AST_MODULE_LOAD_SUCCESS;
}
static int unload_module(void)
{
- ast_manager_unregister("GulpNotify");
+ ast_manager_unregister("PJSIPNotify");
aco_info_destroy(¬ify_cfg);
return 0;
}
-AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_LOAD_ORDER, "CLI/AGI SIP Notify Support",
+AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_LOAD_ORDER, "CLI/AMI PJSIP Notify Support",
.load = load_module,
.reload = reload_module,
.unload = unload_module,
/*** MODULEINFO
<depend>pjproject</depend>
- <depend>res_sip</depend>
- <depend>res_sip_session</depend>
+ <depend>res_pjsip</depend>
+ <depend>res_pjsip_session</depend>
<support_level>core</support_level>
***/
#include <pjsip_ua.h>
#include "asterisk/features.h"
-#include "asterisk/res_sip.h"
-#include "asterisk/res_sip_session.h"
+#include "asterisk/res_pjsip.h"
+#include "asterisk/res_pjsip_session.h"
#include "asterisk/module.h"
#include "asterisk/features_config.h"
return 0;
}
-AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_LOAD_ORDER, "SIP INFO One Touch Recording Support",
+AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_LOAD_ORDER, "PJSIP INFO One Touch Recording Support",
.load = load_module,
.unload = unload_module,
.load_pri = AST_MODPRI_APP_DEPEND,
/*** MODULEINFO
<depend>pjproject</depend>
- <depend>res_sip</depend>
+ <depend>res_pjsip</depend>
<support_level>core</support_level>
***/
#include <pjsip.h>
-#include "asterisk/res_sip.h"
+#include "asterisk/res_pjsip.h"
#include "asterisk/logger.h"
#include "asterisk/module.h"
#include "asterisk/strings.h"
return 0;
}
-AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_LOAD_ORDER, "SIP authentication resource",
+AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_LOAD_ORDER, "PJSIP authentication resource",
.load = load_module,
.unload = unload_module,
.load_pri = AST_MODPRI_CHANNEL_DEPEND,
/*** MODULEINFO
<depend>pjproject</depend>
- <depend>res_sip</depend>
+ <depend>res_pjsip</depend>
<support_level>core</support_level>
***/
#include <pjsip.h>
#include <pjsip_ua.h>
-#include "asterisk/res_sip.h"
+#include "asterisk/res_pjsip.h"
#include "asterisk/module.h"
#include "asterisk/taskprocessor.h"
/*** DOCUMENTATION
- <configInfo name="res_sip_outbound_registration" language="en_US">
+ <configInfo name="res_pjsip_outbound_registration" language="en_US">
<synopsis>SIP resource for outbound registrations</synopsis>
<description><para>
<emphasis>Outbound Registration</emphasis>
</para>
- <para>This module allows <literal>res_sip</literal> to register to other SIP servers.</para>
+ <para>This module allows <literal>res_pjsip</literal> to register to other SIP servers.</para>
</description>
- <configFile name="res_sip.conf">
+ <configFile name="pjsip.conf">
<configObject name="registration">
<synopsis>The configuration for outbound registration</synopsis>
<description><para>
Registration is <emphasis>COMPLETELY</emphasis> separate from the rest of
- <literal>res_sip.conf</literal>. A minimal configuration consists of
+ <literal>pjsip.conf</literal>. A minimal configuration consists of
setting a <literal>server_uri</literal> and a <literal>client_uri</literal>.
</para></description>
<configOption name="auth_rejection_permanent" default="yes">
<synopsis>Transport used for outbound authentication</synopsis>
<description>
<note><para>A <replaceable>transport</replaceable> configured in
- <literal>res_sip.conf</literal>. As with other <literal>res_sip</literal> modules, this will use the first available transport of the appropriate type if unconfigured.</para></note>
+ <literal>pjsip.conf</literal>. As with other <literal>res_pjsip</literal> modules, this will use the first available transport of the appropriate type if unconfigured.</para></note>
</description>
</configOption>
<configOption name="type">
static int load_module(void)
{
- ast_sorcery_apply_default(ast_sip_get_sorcery(), "registration", "config", "res_sip.conf,criteria=type=registration");
+ ast_sorcery_apply_default(ast_sip_get_sorcery(), "registration", "config", "pjsip.conf,criteria=type=registration");
if (ast_sorcery_object_register(ast_sip_get_sorcery(), "registration", sip_outbound_registration_alloc, NULL, sip_outbound_registration_apply)) {
return AST_MODULE_LOAD_DECLINE;
return 0;
}
-AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_LOAD_ORDER, "SIP Outbound Registration Support",
+AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_LOAD_ORDER, "PJSIP Outbound Registration Support",
.load = load_module,
.reload = reload_module,
.unload = unload_module,
/*** MODULEINFO
<depend>pjproject</depend>
- <depend>res_sip</depend>
- <depend>res_sip_pubsub</depend>
- <depend>res_sip_exten_state</depend>
+ <depend>res_pjsip</depend>
+ <depend>res_pjsip_pubsub</depend>
+ <depend>res_pjsip_exten_state</depend>
<support_level>core</support_level>
***/
#include <pjlib.h>
#include "asterisk/module.h"
-#include "asterisk/res_sip.h"
-#include "asterisk/res_sip_exten_state.h"
+#include "asterisk/res_pjsip.h"
+#include "asterisk/res_pjsip_exten_state.h"
enum state {
NOTIFY_OPEN,
return 0;
}
-AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_LOAD_ORDER, "SIP Extension State PIDF Provider",
+AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_LOAD_ORDER, "PJSIP Extension State PIDF Provider",
.load = load_module,
.unload = unload_module,
.load_pri = AST_MODPRI_CHANNEL_DEPEND,
/*** MODULEINFO
<depend>pjproject</depend>
- <depend>res_sip</depend>
+ <depend>res_pjsip</depend>
<support_level>core</support_level>
***/
#include <pjsip_simple.h>
#include <pjlib.h>
-#include "asterisk/res_sip_pubsub.h"
+#include "asterisk/res_pjsip_pubsub.h"
#include "asterisk/module.h"
#include "asterisk/linkedlists.h"
#include "asterisk/astobj2.h"
#include "asterisk/uuid.h"
#include "asterisk/taskprocessor.h"
#include "asterisk/sched.h"
-#include "asterisk/res_sip.h"
+#include "asterisk/res_pjsip.h"
static pj_bool_t pubsub_on_rx_request(pjsip_rx_data *rdata);
return 0;
}
-AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_GLOBAL_SYMBOLS | AST_MODFLAG_LOAD_ORDER, "SIP event resource",
+AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_GLOBAL_SYMBOLS | AST_MODFLAG_LOAD_ORDER, "PJSIP event resource",
.load = load_module,
.unload = unload_module,
.load_pri = AST_MODPRI_CHANNEL_DEPEND,
/*** MODULEINFO
<depend>pjproject</depend>
- <depend>res_sip</depend>
- <depend>res_sip_session</depend>
- <depend>res_sip_pubsub</depend>
+ <depend>res_pjsip</depend>
+ <depend>res_pjsip_session</depend>
+ <depend>res_pjsip_pubsub</depend>
<support_level>core</support_level>
***/
#include <pjsip.h>
#include <pjsip_ua.h>
-#include "asterisk/res_sip.h"
-#include "asterisk/res_sip_session.h"
+#include "asterisk/res_pjsip.h"
+#include "asterisk/res_pjsip_session.h"
#include "asterisk/module.h"
#include "asterisk/pbx.h"
#include "asterisk/taskprocessor.h"
return 0;
}
-AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_LOAD_ORDER, "SIP Blind and Attended Transfer Support",
+AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_LOAD_ORDER, "PJSIP Blind and Attended Transfer Support",
.load = load_module,
.unload = unload_module,
.load_pri = AST_MODPRI_APP_DEPEND,
/*** MODULEINFO
<depend>pjproject</depend>
- <depend>res_sip</depend>
+ <depend>res_pjsip</depend>
<support_level>core</support_level>
***/
#include <pjsip.h>
#include <pjsip_ua.h>
-#include "asterisk/res_sip.h"
+#include "asterisk/res_pjsip.h"
#include "asterisk/module.h"
/*! \brief Internal function which returns the expiration time for a contact */
return 0;
}
-AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_LOAD_ORDER, "SIP Registrar Support",
+AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_LOAD_ORDER, "PJSIP Registrar Support",
.load = load_module,
.unload = unload_module,
.load_pri = AST_MODPRI_APP_DEPEND,
/*** MODULEINFO
<depend>pjproject</depend>
- <depend>res_sip</depend>
+ <depend>res_pjsip</depend>
<support_level>core</support_level>
***/
#include <pjsip.h>
-#include "asterisk/res_sip.h"
+#include "asterisk/res_pjsip.h"
#include "asterisk/module.h"
#include "asterisk/sched.h"
return 0;
}
-AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_LOAD_ORDER, "SIP Contact Auto-Expiration",
+AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_LOAD_ORDER, "PJSIP Contact Auto-Expiration",
.load = load_module,
.unload = unload_module,
.load_pri = AST_MODPRI_APP_DEPEND,
/*** MODULEINFO
<depend>pjproject</depend>
- <depend>res_sip</depend>
- <depend>res_sip_session</depend>
+ <depend>res_pjsip</depend>
+ <depend>res_pjsip_session</depend>
<support_level>core</support_level>
***/
#include <pjsip.h>
#include <pjsip_ua.h>
-#include "asterisk/res_sip.h"
-#include "asterisk/res_sip_session.h"
+#include "asterisk/res_pjsip.h"
+#include "asterisk/res_pjsip_session.h"
#include "asterisk/module.h"
#include "asterisk/causes.h"
return 0;
}
-AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_LOAD_ORDER, "SIP RFC3326 Support",
+AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_LOAD_ORDER, "PJSIP RFC3326 Support",
.load = load_module,
.unload = unload_module,
.load_pri = AST_MODPRI_APP_DEPEND,
/*** MODULEINFO
<depend>pjproject</depend>
- <depend>res_sip</depend>
- <depend>res_sip_session</depend>
+ <depend>res_pjsip</depend>
+ <depend>res_pjsip_session</depend>
<support_level>core</support_level>
***/
#include "asterisk/acl.h"
#include "asterisk/sdp_srtp.h"
-#include "asterisk/res_sip.h"
-#include "asterisk/res_sip_session.h"
+#include "asterisk/res_pjsip.h"
+#include "asterisk/res_pjsip_session.h"
/*! \brief Scheduler for RTCP purposes */
static struct ast_sched_context *sched;
return AST_MODULE_LOAD_FAILURE;
}
-AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_LOAD_ORDER, "SIP SDP RTP/AVP stream handler",
+AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_LOAD_ORDER, "PJSIP SDP RTP/AVP stream handler",
.load = load_module,
.unload = unload_module,
.load_pri = AST_MODPRI_CHANNEL_DRIVER,
/*** MODULEINFO
<depend>pjproject</depend>
- <depend>res_sip</depend>
+ <depend>res_pjsip</depend>
<support_level>core</support_level>
***/
#include <pjsip_ua.h>
#include <pjlib.h>
-#include "asterisk/res_sip.h"
-#include "asterisk/res_sip_session.h"
+#include "asterisk/res_pjsip.h"
+#include "asterisk/res_pjsip_session.h"
#include "asterisk/datastore.h"
#include "asterisk/module.h"
#include "asterisk/logger.h"
-#include "asterisk/res_sip.h"
+#include "asterisk/res_pjsip.h"
#include "asterisk/astobj2.h"
#include "asterisk/lock.h"
#include "asterisk/uuid.h"
return 0;
}
-AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_GLOBAL_SYMBOLS | AST_MODFLAG_LOAD_ORDER, "SIP Session resource",
+AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_GLOBAL_SYMBOLS | AST_MODFLAG_LOAD_ORDER, "PJSIP Session resource",
.load = load_module,
.unload = unload_module,
.load_pri = AST_MODPRI_APP_DEPEND,
/*** MODULEINFO
<depend>pjproject</depend>
- <depend>res_sip</depend>
- <depend>res_sip_session</depend>
+ <depend>res_pjsip</depend>
+ <depend>res_pjsip_session</depend>
<support_level>core</support_level>
***/
#include "asterisk/channel.h"
#include "asterisk/acl.h"
-#include "asterisk/res_sip.h"
-#include "asterisk/res_sip_session.h"
+#include "asterisk/res_pjsip.h"
+#include "asterisk/res_pjsip_session.h"
/*! \brief The number of seconds after receiving a T.38 re-invite before automatically rejecting it */
#define T38_AUTOMATIC_REJECTION_SECONDS 5
return AST_MODULE_LOAD_FAILURE;
}
-AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_LOAD_ORDER, "SIP T.38 UDPTL Support",
+AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_LOAD_ORDER, "PJSIP T.38 UDPTL Support",
.load = load_module,
.unload = unload_module,
.load_pri = AST_MODPRI_CHANNEL_DRIVER,
/*** MODULEINFO
<depend>pjproject</depend>
- <depend>res_sip</depend>
+ <depend>res_pjsip</depend>
<depend>res_http_websocket</depend>
<support_level>core</support_level>
***/
#include "asterisk/module.h"
#include "asterisk/http_websocket.h"
-#include "asterisk/res_sip.h"
-#include "asterisk/res_sip_session.h"
+#include "asterisk/res_pjsip.h"
+#include "asterisk/res_pjsip_session.h"
#include "asterisk/taskprocessor.h"
static int transport_type_ws;
return 0;
}
-AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_LOAD_ORDER, "SIP WebSocket Transport Support",
+AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_LOAD_ORDER, "PJSIP WebSocket Transport Support",
.load = load_module,
.unload = unload_module,
.load_pri = AST_MODPRI_APP_DEPEND,
projects / thirdparty / asterisk.git / commitdiff
