.write = pjsip_acf_session_refresh_write,
};
+static char *app_pjsip_hangup = "PJSIPHangup";
+
/*!
* \brief Load the module
*
goto end;
}
+ if (ast_register_application_xml(app_pjsip_hangup, pjsip_app_hangup)) {
+ ast_log(LOG_WARNING, "Unable to register PJSIPHangup dialplan application\n");
+ goto end;
+ }
+ ast_manager_register_xml(app_pjsip_hangup, EVENT_FLAG_SYSTEM | EVENT_FLAG_CALL, pjsip_action_hangup);
+
+
ast_sip_register_service(&refer_callback_module);
ast_sip_session_register_supplement(&chan_pjsip_supplement);
ast_custom_function_unregister(&chan_pjsip_dial_contacts_function);
ast_custom_function_unregister(&chan_pjsip_parse_uri_function);
ast_custom_function_unregister(&session_refresh_function);
+ ast_unregister_application(app_pjsip_hangup);
+ ast_manager_unregister(app_pjsip_hangup);
+
ast_channel_unregister(&chan_pjsip_tech);
ast_rtp_glue_unregister(&chan_pjsip_rtp_glue);
ast_custom_function_unregister(&chan_pjsip_dial_contacts_function);
ast_custom_function_unregister(&chan_pjsip_parse_uri_function);
ast_custom_function_unregister(&session_refresh_function);
+ ast_unregister_application(app_pjsip_hangup);
+ ast_manager_unregister(app_pjsip_hangup);
ast_channel_unregister(&chan_pjsip_tech);
ao2_ref(chan_pjsip_tech.capabilities, -1);
<support_level>core</support_level>
***/
-/*** DOCUMENTATION
-<function name="PJSIP_DIAL_CONTACTS" language="en_US">
- <synopsis>
- Return a dial string for dialing all contacts on an AOR.
- </synopsis>
- <syntax>
- <parameter name="endpoint" required="true">
- <para>Name of the endpoint</para>
- </parameter>
- <parameter name="aor" required="false">
- <para>Name of an AOR to use, if not specified the configured AORs on the endpoint are used</para>
- </parameter>
- <parameter name="request_user" required="false">
- <para>Optional request user to use in the request URI</para>
- </parameter>
- </syntax>
- <description>
- <para>Returns a properly formatted dial string for dialing all contacts on an AOR.</para>
- </description>
-</function>
-<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>
- <syntax>
- <parameter name="media" required="true">
- <para>types of media offered</para>
- </parameter>
- </syntax>
- <description>
- <para>When read, returns the codecs offered based upon the media choice.</para>
- <para>When written, sets the codecs to offer when an outbound dial attempt is made,
- or when a session refresh is sent using <replaceable>PJSIP_SEND_SESSION_REFRESH</replaceable>.
- </para>
- </description>
- <see-also>
- <ref type="function">PJSIP_SEND_SESSION_REFRESH</ref>
- </see-also>
-</function>
-<function name="PJSIP_DTMF_MODE" language="en_US">
- <since>
- <version>13.18.0</version>
- <version>14.7.0</version>
- <version>15.1.0</version>
- <version>16.0.0</version>
- </since>
- <synopsis>
- Get or change the DTMF mode for a SIP call.
- </synopsis>
- <syntax>
- </syntax>
- <description>
- <para>When read, returns the current DTMF mode</para>
- <para>When written, sets the current DTMF mode</para>
- <para>This function uses the same DTMF mode naming as the dtmf_mode configuration option</para>
- </description>
-</function>
-<function name="PJSIP_MOH_PASSTHROUGH" language="en_US">
- <synopsis>
- Get or change the on-hold behavior for a SIP call.
- </synopsis>
- <syntax>
- </syntax>
- <description>
- <para>When read, returns the current moh passthrough mode</para>
- <para>When written, sets the current moh passthrough mode</para>
- <para>If <replaceable>yes</replaceable>, on-hold re-INVITEs are sent. If <replaceable>no</replaceable>, music on hold is generated.</para>
- <para>This function can be used to override the moh_passthrough configuration option</para>
- </description>
-</function>
-<function name="PJSIP_SEND_SESSION_REFRESH" language="en_US">
- <since>
- <version>13.12.0</version>
- <version>14.1.0</version>
- <version>15.0.0</version>
- </since>
- <synopsis>
- W/O: Initiate a session refresh via an UPDATE or re-INVITE on an established media session
- </synopsis>
- <syntax>
- <parameter name="update_type" required="false">
- <para>The type of update to send. Default is <literal>invite</literal>.</para>
- <enumlist>
- <enum name="invite">
- <para>Send the session refresh as a re-INVITE.</para>
- </enum>
- <enum name="update">
- <para>Send the session refresh as an UPDATE.</para>
- </enum>
- </enumlist>
- </parameter>
- </syntax>
- <description>
- <para>This function will cause the PJSIP stack to immediately refresh
- the media session for the channel. This will be done using either a
- re-INVITE (default) or an UPDATE request.
- </para>
- <para>This is most useful when combined with the <replaceable>PJSIP_MEDIA_OFFER</replaceable>
- dialplan function, as it allows the formats in use on a channel to be
- re-negotiated after call setup.</para>
- <warning>
- <para>The formats the endpoint supports are <emphasis>not</emphasis>
- checked or enforced by this function. Using this function to offer
- formats not supported by the endpoint <emphasis>may</emphasis> result
- in a loss of media.</para>
- </warning>
- <example title="Re-negotiate format to g722">
- ; Within some existing extension on an answered channel
- same => n,Set(PJSIP_MEDIA_OFFER(audio)=!all,g722)
- same => n,Set(PJSIP_SEND_SESSION_REFRESH()=invite)
- </example>
- </description>
- <see-also>
- <ref type="function">PJSIP_MEDIA_OFFER</ref>
- </see-also>
-</function>
-<function name="PJSIP_PARSE_URI" language="en_US">
- <since>
- <version>13.24.0</version>
- <version>16.1.0</version>
- <version>17.0.0</version>
- </since>
- <synopsis>
- Parse an uri and return a type part of the URI.
- </synopsis>
- <syntax>
- <parameter name="uri" required="true">
- <para>URI to parse</para>
- </parameter>
- <parameter name="type" required="true">
- <para>The <literal>type</literal> parameter specifies which URI part to read</para>
- <enumlist>
- <enum name="display">
- <para>Display name.</para>
- </enum>
- <enum name="scheme">
- <para>URI scheme.</para>
- </enum>
- <enum name="user">
- <para>User part.</para>
- </enum>
- <enum name="passwd">
- <para>Password part.</para>
- </enum>
- <enum name="host">
- <para>Host part.</para>
- </enum>
- <enum name="port">
- <para>Port number, or zero.</para>
- </enum>
- <enum name="user_param">
- <para>User parameter.</para>
- </enum>
- <enum name="method_param">
- <para>Method parameter.</para>
- </enum>
- <enum name="transport_param">
- <para>Transport parameter.</para>
- </enum>
- <enum name="ttl_param">
- <para>TTL param, or -1.</para>
- </enum>
- <enum name="lr_param">
- <para>Loose routing param, or zero.</para>
- </enum>
- <enum name="maddr_param">
- <para>Maddr param.</para>
- </enum>
- </enumlist>
- </parameter>
- </syntax>
- <description>
- <para>Parse an URI and return a specified part of the URI.</para>
- </description>
-</function>
-<info name="CHANNEL" language="en_US" tech="PJSIP">
- <enumlist>
- <enum name="rtp">
- <para>R/O Retrieve media related information.</para>
- <parameter name="type" required="true">
- <para>When <replaceable>rtp</replaceable> is specified, the
- <literal>type</literal> parameter must be provided. It specifies
- which RTP parameter to read.</para>
- <enumlist>
- <enum name="src">
- <para>Retrieve the local address for RTP.</para>
- </enum>
- <enum name="dest">
- <para>Retrieve the remote address for RTP.</para>
- </enum>
- <enum name="direct">
- <para>If direct media is enabled, this address is the remote address
- used for RTP.</para>
- </enum>
- <enum name="secure">
- <para>Whether or not the media stream is encrypted.</para>
- <enumlist>
- <enum name="0">
- <para>The media stream is not encrypted.</para>
- </enum>
- <enum name="1">
- <para>The media stream is encrypted.</para>
- </enum>
- </enumlist>
- </enum>
- <enum name="hold">
- <para>Whether or not the media stream is currently restricted
- due to a call hold.</para>
- <enumlist>
- <enum name="0">
- <para>The media stream is not held.</para>
- </enum>
- <enum name="1">
- <para>The media stream is held.</para>
- </enum>
- </enumlist>
- </enum>
- </enumlist>
- </parameter>
- <parameter name="media_type" required="false">
- <para>When <replaceable>rtp</replaceable> is specified, the
- <literal>media_type</literal> parameter may be provided. It specifies
- which media stream the chosen RTP parameter should be retrieved
- from.</para>
- <enumlist>
- <enum name="audio">
- <para>Retrieve information from the audio media stream.</para>
- <note><para>If not specified, <literal>audio</literal> is used
- by default.</para></note>
- </enum>
- <enum name="video">
- <para>Retrieve information from the video media stream.</para>
- </enum>
- </enumlist>
- </parameter>
- </enum>
- <enum name="rtcp">
- <para>R/O Retrieve RTCP statistics.</para>
- <parameter name="statistic" required="true">
- <para>When <replaceable>rtcp</replaceable> is specified, the
- <literal>statistic</literal> parameter must be provided. It specifies
- which RTCP statistic parameter to read.</para>
- <enumlist>
- <enum name="all">
- <para>Retrieve a summary of all RTCP statistics.</para>
- <para>The following data items are returned in a semi-colon
- delineated list:</para>
- <enumlist>
- <enum name="ssrc">
- <para>Our Synchronization Source identifier</para>
- </enum>
- <enum name="themssrc">
- <para>Their Synchronization Source identifier</para>
- </enum>
- <enum name="lp">
- <para>Our lost packet count</para>
- </enum>
- <enum name="rxjitter">
- <para>Received packet jitter</para>
- </enum>
- <enum name="rxcount">
- <para>Received packet count</para>
- </enum>
- <enum name="txjitter">
- <para>Transmitted packet jitter</para>
- </enum>
- <enum name="txcount">
- <para>Transmitted packet count</para>
- </enum>
- <enum name="rlp">
- <para>Remote lost packet count</para>
- </enum>
- <enum name="rtt">
- <para>Round trip time</para>
- </enum>
- <enum name="txmes">
- <para>Transmitted Media Experience Score</para>
- </enum>
- <enum name="rxmes">
- <para>Received Media Experience Score</para>
- </enum>
- </enumlist>
- </enum>
- <enum name="all_jitter">
- <para>Retrieve a summary of all RTCP Jitter statistics.</para>
- <para>The following data items are returned in a semi-colon
- delineated list:</para>
- <enumlist>
- <enum name="minrxjitter">
- <para>Our minimum jitter</para>
- </enum>
- <enum name="maxrxjitter">
- <para>Our max jitter</para>
- </enum>
- <enum name="avgrxjitter">
- <para>Our average jitter</para>
- </enum>
- <enum name="stdevrxjitter">
- <para>Our jitter standard deviation</para>
- </enum>
- <enum name="reported_minjitter">
- <para>Their minimum jitter</para>
- </enum>
- <enum name="reported_maxjitter">
- <para>Their max jitter</para>
- </enum>
- <enum name="reported_avgjitter">
- <para>Their average jitter</para>
- </enum>
- <enum name="reported_stdevjitter">
- <para>Their jitter standard deviation</para>
- </enum>
- </enumlist>
- </enum>
- <enum name="all_loss">
- <para>Retrieve a summary of all RTCP packet loss statistics.</para>
- <para>The following data items are returned in a semi-colon
- delineated list:</para>
- <enumlist>
- <enum name="minrxlost">
- <para>Our minimum lost packets</para>
- </enum>
- <enum name="maxrxlost">
- <para>Our max lost packets</para>
- </enum>
- <enum name="avgrxlost">
- <para>Our average lost packets</para>
- </enum>
- <enum name="stdevrxlost">
- <para>Our lost packets standard deviation</para>
- </enum>
- <enum name="reported_minlost">
- <para>Their minimum lost packets</para>
- </enum>
- <enum name="reported_maxlost">
- <para>Their max lost packets</para>
- </enum>
- <enum name="reported_avglost">
- <para>Their average lost packets</para>
- </enum>
- <enum name="reported_stdevlost">
- <para>Their lost packets standard deviation</para>
- </enum>
- </enumlist>
- </enum>
- <enum name="all_rtt">
- <para>Retrieve a summary of all RTCP round trip time information.</para>
- <para>The following data items are returned in a semi-colon
- delineated list:</para>
- <enumlist>
- <enum name="minrtt">
- <para>Minimum round trip time</para>
- </enum>
- <enum name="maxrtt">
- <para>Maximum round trip time</para>
- </enum>
- <enum name="avgrtt">
- <para>Average round trip time</para>
- </enum>
- <enum name="stdevrtt">
- <para>Standard deviation round trip time</para>
- </enum>
- </enumlist>
- </enum>
- <enum name="all_mes">
- <para>Retrieve a summary of all RTCP Media Experience Score information.</para>
- <para>The following data items are returned in a semi-colon
- delineated list:</para>
- <enumlist>
- <enum name="minmes">
- <para>Minimum MES based on us analysing received packets.</para>
- </enum>
- <enum name="maxmes">
- <para>Maximum MES based on us analysing received packets.</para>
- </enum>
- <enum name="avgmes">
- <para>Average MES based on us analysing received packets.</para>
- </enum>
- <enum name="stdevmes">
- <para>Standard deviation MES based on us analysing received packets.</para>
- </enum>
- <enum name="reported_minmes">
- <para>Minimum MES based on data we get in Sender and Receiver Reports sent by the remote end</para>
- </enum>
- <enum name="reported_maxmes">
- <para>Maximum MES based on data we get in Sender and Receiver Reports sent by the remote end</para>
- </enum>
- <enum name="reported_avgmes">
- <para>Average MES based on data we get in Sender and Receiver Reports sent by the remote end</para>
- </enum>
- <enum name="reported_stdevmes">
- <para>Standard deviation MES based on data we get in Sender and Receiver Reports sent by the remote end</para>
- </enum>
- </enumlist>
- </enum>
- <enum name="txcount"><para>Transmitted packet count</para></enum>
- <enum name="rxcount"><para>Received packet count</para></enum>
- <enum name="txjitter"><para>Transmitted packet jitter</para></enum>
- <enum name="rxjitter"><para>Received packet jitter</para></enum>
- <enum name="remote_maxjitter"><para>Their max jitter</para></enum>
- <enum name="remote_minjitter"><para>Their minimum jitter</para></enum>
- <enum name="remote_normdevjitter"><para>Their average jitter</para></enum>
- <enum name="remote_stdevjitter"><para>Their jitter standard deviation</para></enum>
- <enum name="local_maxjitter"><para>Our max jitter</para></enum>
- <enum name="local_minjitter"><para>Our minimum jitter</para></enum>
- <enum name="local_normdevjitter"><para>Our average jitter</para></enum>
- <enum name="local_stdevjitter"><para>Our jitter standard deviation</para></enum>
- <enum name="txploss"><para>Transmitted packet loss</para></enum>
- <enum name="rxploss"><para>Received packet loss</para></enum>
- <enum name="remote_maxrxploss"><para>Their max lost packets</para></enum>
- <enum name="remote_minrxploss"><para>Their minimum lost packets</para></enum>
- <enum name="remote_normdevrxploss"><para>Their average lost packets</para></enum>
- <enum name="remote_stdevrxploss"><para>Their lost packets standard deviation</para></enum>
- <enum name="local_maxrxploss"><para>Our max lost packets</para></enum>
- <enum name="local_minrxploss"><para>Our minimum lost packets</para></enum>
- <enum name="local_normdevrxploss"><para>Our average lost packets</para></enum>
- <enum name="local_stdevrxploss"><para>Our lost packets standard deviation</para></enum>
- <enum name="rtt"><para>Round trip time</para></enum>
- <enum name="maxrtt"><para>Maximum round trip time</para></enum>
- <enum name="minrtt"><para>Minimum round trip time</para></enum>
- <enum name="normdevrtt"><para>Average round trip time</para></enum>
- <enum name="stdevrtt"><para>Standard deviation round trip time</para></enum>
- <enum name="local_ssrc"><para>Our Synchronization Source identifier</para></enum>
- <enum name="remote_ssrc"><para>Their Synchronization Source identifier</para></enum>
- <enum name="txmes"><para>
- Current MES based on us analyzing rtt, jitter and loss
- in the actual received RTP stream received from the remote end.
- I.E. This is the MES for the incoming audio stream.
- </para></enum>
- <enum name="rxmes"><para>
- Current MES based on rtt and the jitter and loss values in
- RTCP sender and receiver reports we receive from the
- remote end. I.E. This is the MES for the outgoing audio stream.
- </para></enum>
- <enum name="remote_maxmes"><para>Max MES based on data we get in Sender and Receiver Reports sent by the remote end</para></enum>
- <enum name="remote_minmes"><para>Min MES based on data we get in Sender and Receiver Reports sent by the remote end</para></enum>
- <enum name="remote_normdevmes"><para>Average MES based on data we get in Sender and Receiver Reports sent by the remote end</para></enum>
- <enum name="remote_stdevmes"><para>Standard deviation MES based on data we get in Sender and Receiver Reports sent by the remote end</para></enum>
- <enum name="local_maxmes"><para>Max MES based on us analyzing the received RTP stream</para></enum>
- <enum name="local_minmes"><para>Min MES based on us analyzing the received RTP stream</para></enum>
- <enum name="local_normdevmes"><para>Average MES based on us analyzing the received RTP stream</para></enum>
- <enum name="local_stdevmes"><para>Standard deviation MES based on us analyzing the received RTP stream</para></enum>
- </enumlist>
- </parameter>
- <parameter name="media_type" required="false">
- <para>When <replaceable>rtcp</replaceable> is specified, the
- <literal>media_type</literal> parameter may be provided. It specifies
- which media stream the chosen RTCP parameter should be retrieved
- from.</para>
- <enumlist>
- <enum name="audio">
- <para>Retrieve information from the audio media stream.</para>
- <note><para>If not specified, <literal>audio</literal> is used
- by default.</para></note>
- </enum>
- <enum name="video">
- <para>Retrieve information from the video media stream.</para>
- </enum>
- </enumlist>
- </parameter>
- </enum>
- <enum name="endpoint">
- <para>R/O The name of the endpoint associated with this channel.
- Use the <replaceable>PJSIP_ENDPOINT</replaceable> function to obtain
- further endpoint related information.</para>
- </enum>
- <enum name="contact">
- <para>R/O The name of the contact associated with this channel.
- Use the <replaceable>PJSIP_CONTACT</replaceable> function to obtain
- further contact related information. Note this may not be present and if so
- is only available on outgoing legs.</para>
- </enum>
- <enum name="aor">
- <para>R/O The name of the AOR associated with this channel.
- Use the <replaceable>PJSIP_AOR</replaceable> function to obtain
- further AOR related information. Note this may not be present and if so
- is only available on outgoing legs.</para>
- </enum>
- <enum name="pjsip">
- <para>R/O Obtain information about the current PJSIP channel and its
- session.</para>
- <parameter name="type" required="true">
- <para>When <replaceable>pjsip</replaceable> is specified, the
- <literal>type</literal> parameter must be provided. It specifies
- which signalling parameter to read.</para>
- <enumlist>
- <enum name="call-id">
- <para>The SIP call-id.</para>
- </enum>
- <enum name="secure">
- <para>Whether or not the signalling uses a secure transport.</para>
- <enumlist>
- <enum name="0"><para>The signalling uses a non-secure transport.</para></enum>
- <enum name="1"><para>The signalling uses a secure transport.</para></enum>
- </enumlist>
- </enum>
- <enum name="target_uri">
- <para>The contact URI where requests are sent.</para>
- </enum>
- <enum name="local_uri">
- <para>The local URI.</para>
- </enum>
- <enum name="local_tag">
- <para>Tag in From header</para>
- </enum>
- <enum name="remote_uri">
- <para>The remote URI.</para>
- </enum>
- <enum name="remote_tag">
- <para>Tag in To header</para>
- </enum>
- <enum name="request_uri">
- <para>The request URI of the incoming <literal>INVITE</literal>
- associated with the creation of this channel.</para>
- </enum>
- <enum name="t38state">
- <para>The current state of any T.38 fax on this channel.</para>
- <enumlist>
- <enum name="DISABLED"><para>T.38 faxing is disabled on this channel.</para></enum>
- <enum name="LOCAL_REINVITE"><para>Asterisk has sent a <literal>re-INVITE</literal> to the remote end to initiate a T.38 fax.</para></enum>
- <enum name="REMOTE_REINVITE"><para>The remote end has sent a <literal>re-INVITE</literal> to Asterisk to initiate a T.38 fax.</para></enum>
- <enum name="ENABLED"><para>A T.38 fax session has been enabled.</para></enum>
- <enum name="REJECTED"><para>A T.38 fax session was attempted but was rejected.</para></enum>
- </enumlist>
- </enum>
- <enum name="local_addr">
- <para>On inbound calls, the full IP address and port number that
- the <literal>INVITE</literal> request was received on. On outbound
- calls, the full IP address and port number that the <literal>INVITE</literal>
- request was transmitted from.</para>
- </enum>
- <enum name="remote_addr">
- <para>On inbound calls, the full IP address and port number that
- the <literal>INVITE</literal> request was received from. On outbound
- calls, the full IP address and port number that the <literal>INVITE</literal>
- request was transmitted to.</para>
- </enum>
- </enumlist>
- </parameter>
- </enum>
- </enumlist>
-</info>
-<info name="CHANNEL_EXAMPLES" language="en_US" tech="PJSIP">
- <example title="PJSIP specific CHANNEL examples">
- ; Log the current Call-ID
- same => n,Log(NOTICE, ${CHANNEL(pjsip,call-id)})
-
- ; Log the destination address of the audio stream
- same => n,Log(NOTICE, ${CHANNEL(rtp,dest)})
-
- ; Store the round-trip time associated with a
- ; video stream in the CDR field video-rtt
- same => n,Set(CDR(video-rtt)=${CHANNEL(rtcp,rtt,video)})
- </example>
-</info>
-***/
-
#include "asterisk.h"
#include <pjsip.h>
#include "asterisk/module.h"
#include "asterisk/acl.h"
#include "asterisk/app.h"
+#include "asterisk/conversions.h"
#include "asterisk/channel.h"
#include "asterisk/stream.h"
#include "asterisk/format.h"
return ast_sip_push_task_wait_serializer(channel->session->serializer, refresh_write_cb, &rdata);
}
+
+struct hangup_data {
+ struct ast_sip_session *session;
+ int response_code;
+};
+
+/*!
+ * \brief Serializer task to hangup channel
+ */
+static int pjsip_hangup(void *obj)
+{
+ struct hangup_data *hdata = obj;
+ pjsip_tx_data *packet = NULL;
+
+ if ((hdata->session->inv_session->state != PJSIP_INV_STATE_DISCONNECTED) &&
+ (pjsip_inv_answer(hdata->session->inv_session, hdata->response_code, NULL, NULL, &packet) == PJ_SUCCESS)) {
+ ast_sip_session_send_response(hdata->session, packet);
+ }
+
+ return 0;
+}
+
+/*!
+ * \brief Callback that validates the response code
+ */
+static int response_code_validator(const char *channel_name,
+ const char *response) {
+ int response_code;
+
+ int rc = ast_str_to_int(response, &response_code);
+ if (rc != 0) {
+ response_code = ast_sip_str2rc(response);
+ if (response_code < 0) {
+ ast_log(LOG_WARNING, "%s: Unrecognized response code parameter '%s'."
+ " Defaulting to 603 DECLINE\n",
+ channel_name, response);
+ return PJSIP_SC_DECLINE;
+ }
+ }
+
+ if (response_code < 400 || response_code > 699) {
+ ast_log(LOG_WARNING, "%s: Response code %d is out of range 400 -> 699."
+ " Defaulting to 603 DECLINE\n",
+ channel_name, response_code);
+ return PJSIP_SC_DECLINE;
+ }
+ return response_code;
+}
+
+/*!
+ * \brief Called by pjsip_app_hangup and pjsip_action_hangup
+ * to actually perform the hangup
+ */
+static void pjsip_app_hangup_handler(struct ast_channel *chan, int response_code)
+{
+ struct ast_sip_channel_pvt *channel;
+ struct hangup_data hdata = { NULL, -1 };
+ const char *tag = ast_channel_name(chan);
+
+ hdata.response_code = response_code;
+
+ ast_channel_lock(chan);
+ if (strcmp(ast_channel_tech(chan)->type, "PJSIP")) {
+ ast_log(LOG_WARNING, "%s: Not a PJSIP channel\n", tag);
+ ast_channel_unlock(chan);
+ return;
+ }
+
+ channel = ast_channel_tech_pvt(chan);
+ hdata.session = channel->session;
+
+ if (hdata.session->inv_session->role != PJSIP_ROLE_UAS || (
+ hdata.session->inv_session->state != PJSIP_INV_STATE_INCOMING &&
+ hdata.session->inv_session->state != PJSIP_INV_STATE_EARLY)) {
+ ast_log(LOG_WARNING, "%s: Not an incoming channel or invalid state '%s'\n",
+ tag, pjsip_inv_state_name(hdata.session->inv_session->state));
+ ast_channel_unlock(chan);
+ return;
+ }
+
+ ast_channel_unlock(chan);
+
+ if (ast_sip_push_task_wait_serializer(channel->session->serializer,
+ pjsip_hangup, &hdata) != 0) {
+ ast_log(LOG_WARNING, "%s: failed to push hangup task to serializer\n", tag);
+ }
+
+ return;
+}
+
+/*!
+ * \brief PJSIPHangup Dialplan App
+ */
+int pjsip_app_hangup(struct ast_channel *chan, const char *data)
+{
+ int response_code;
+ const char *tag = ast_channel_name(chan);
+
+ if (ast_strlen_zero(data)) {
+ ast_log(LOG_WARNING, "%s: Missing response code parameter\n", tag);
+ return -1;
+ }
+
+ response_code = response_code_validator(tag, data);
+
+ pjsip_app_hangup_handler(chan, response_code);
+
+ return -1;
+}
+
+/*!
+ * \brief PJSIPHangup Manager Action
+ */
+int pjsip_action_hangup(struct mansession *s, const struct message *m)
+{
+ return ast_manager_hangup_helper(s, m,
+ pjsip_app_hangup_handler, response_code_validator);
+}
--- /dev/null
+<?xml version="1.0" encoding="UTF-8"?>
+<!DOCTYPE docs SYSTEM "appdocsxml.dtd">
+<docs>
+ <application name="PJSIPHangup" language="en_US">
+ <synopsis>
+ Hangup an incoming PJSIP channel with a SIP response code
+ </synopsis>
+ <syntax>
+ <parameter name="Cause" required="true">
+ <para>May be one of...</para>
+ <enumlist>
+ <enum name="Response code"><para>A numeric response code in the range 400 ->699</para></enum>
+ <enum name="Response code name"><para>A response code name from
+ <literal>third-party/pjproject/source/pjsip/include/pjsip/sip_msg.h</literal>
+ such as <literal>USE_IDENTITY_HEADER</literal> or
+ <literal>PJSIP_SC_USE_IDENTITY_HEADER</literal></para></enum>
+ </enumlist>
+ </parameter>
+ </syntax>
+ <description>
+ <para>
+ Hangs up an incoming PJSIP channel and returns the
+ specified SIP response code in the final response to the caller.
+ </para>
+ <para>
+ </para>
+ <warning><para>
+ This function must be called BEFORE anything that
+ might cause any other final (non 1XX) response to be sent.
+ For example calling <literal>Answer()</literal> or
+ <literal>Playback</literal> without the
+ <literal>noanswer</literal> option will cause the call
+ to be answered and a final 200 response to be sent.
+ </para></warning>
+ <para>
+ </para>
+ <para>As with the <literal>Hangup</literal> application,
+ the dialplan will terminate after calling this function.</para>
+ <para>
+ </para>
+ <para>The cause code set on the channel will be translated to
+ a standard ISDN cause code using the table defined in
+ ast_sip_hangup_sip2cause() in res_pjsip.c</para>
+ <para>
+ </para>
+ <example title="Terminate call with 437 response code">
+ same = n,PJSIPHangup(437)
+ </example>
+ <example title="Terminate call with 437 response code using the response code name">
+ same = n,PJSIPHangup(UNSUPPORTED_CERTIFICATE)
+ </example>
+ <example title="Terminate call with 437 response code based on condition">
+ same = n,ExecIf($[${SOMEVALUE} = ${SOME_BAD_VALUE}]?PJSIPHangup(437))
+ </example>
+ </description>
+ </application>
+
+ <manager name="PJSIPHangup" language="en_US">
+ <synopsis>
+ Hangup an incoming PJSIP channel with a SIP response code
+ </synopsis>
+ <syntax>
+ <xi:include xpointer="xpointer(/docs/manager[@name='Login']/syntax/parameter[@name='ActionID'])" />
+ <xi:include xpointer="xpointer(/docs/manager[@name='Hangup']/syntax/parameter[@name='Channel'])" />
+ <xi:include xpointer="xpointer(/docs/application[@name='PJSIPHangup']/syntax/parameter[@name='Cause'])" />
+ </syntax>
+ <description>
+ <para>
+ Hangs up an incoming PJSIP channel and returns the
+ specified SIP response code in the final response to the caller.
+ </para>
+ <para>
+ </para>
+ <warning><para>
+ This function must be called BEFORE anything that
+ might cause any other final (non 1XX) response to be sent.
+ For example calling <literal>Answer()</literal> or
+ <literal>Playback</literal> without the
+ <literal>noanswer</literal> option will cause the call
+ to be answered and a final 200 response to be sent.
+ </para></warning>
+ <para>
+ </para>
+ <para>The cause code set on the channel will be translated to
+ a standard ISDN cause code using the table defined in
+ ast_sip_hangup_sip2cause() in res_pjsip.c</para>
+ <para>
+ </para>
+ <example title="Terminate call with 437 response code">
+ Action: PJSIPHangup
+ ActionID: 12345678
+ Channel: PJSIP/alice-00000002
+ Cause: 437
+ </example>
+ <example title="Terminate call with 437 response code using the response code name">
+ Action: PJSIPHangup
+ ActionID: 12345678
+ Channel: PJSIP/alice-00000002
+ Cause: UNSUPPORTED_CERTIFICATE
+ </example>
+ </description>
+ </manager>
+
+ <function name="PJSIP_DIAL_CONTACTS" language="en_US">
+ <synopsis>
+ Return a dial string for dialing all contacts on an AOR.
+ </synopsis>
+ <syntax>
+ <parameter name="endpoint" required="true">
+ <para>Name of the endpoint</para>
+ </parameter>
+ <parameter name="aor" required="false">
+ <para>Name of an AOR to use, if not specified the configured AORs on the endpoint are used</para>
+ </parameter>
+ <parameter name="request_user" required="false">
+ <para>Optional request user to use in the request URI</para>
+ </parameter>
+ </syntax>
+ <description>
+ <para>Returns a properly formatted dial string for dialing all contacts on an AOR.</para>
+ </description>
+ </function>
+ <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>
+ <syntax>
+ <parameter name="media" required="true">
+ <para>types of media offered</para>
+ </parameter>
+ </syntax>
+ <description>
+ <para>When read, returns the codecs offered based upon the media choice.</para>
+ <para>When written, sets the codecs to offer when an outbound dial attempt is made,
+ or when a session refresh is sent using <replaceable>PJSIP_SEND_SESSION_REFRESH</replaceable>.
+ </para>
+ </description>
+ <see-also>
+ <ref type="function">PJSIP_SEND_SESSION_REFRESH</ref>
+ </see-also>
+ </function>
+ <function name="PJSIP_DTMF_MODE" language="en_US">
+ <since>
+ <version>13.18.0</version>
+ <version>14.7.0</version>
+ <version>15.1.0</version>
+ <version>16.0.0</version>
+ </since>
+ <synopsis>
+ Get or change the DTMF mode for a SIP call.
+ </synopsis>
+ <syntax>
+ </syntax>
+ <description>
+ <para>When read, returns the current DTMF mode</para>
+ <para>When written, sets the current DTMF mode</para>
+ <para>This function uses the same DTMF mode naming as the dtmf_mode configuration option</para>
+ </description>
+ </function>
+ <function name="PJSIP_MOH_PASSTHROUGH" language="en_US">
+ <synopsis>
+ Get or change the on-hold behavior for a SIP call.
+ </synopsis>
+ <syntax>
+ </syntax>
+ <description>
+ <para>When read, returns the current moh passthrough mode</para>
+ <para>When written, sets the current moh passthrough mode</para>
+ <para>If <replaceable>yes</replaceable>, on-hold re-INVITEs are sent. If <replaceable>no</replaceable>, music on hold is generated.</para>
+ <para>This function can be used to override the moh_passthrough configuration option</para>
+ </description>
+ </function>
+ <function name="PJSIP_SEND_SESSION_REFRESH" language="en_US">
+ <since>
+ <version>13.12.0</version>
+ <version>14.1.0</version>
+ <version>15.0.0</version>
+ </since>
+ <synopsis>
+ W/O: Initiate a session refresh via an UPDATE or re-INVITE on an established media session
+ </synopsis>
+ <syntax>
+ <parameter name="update_type" required="false">
+ <para>The type of update to send. Default is <literal>invite</literal>.</para>
+ <enumlist>
+ <enum name="invite">
+ <para>Send the session refresh as a re-INVITE.</para>
+ </enum>
+ <enum name="update">
+ <para>Send the session refresh as an UPDATE.</para>
+ </enum>
+ </enumlist>
+ </parameter>
+ </syntax>
+ <description>
+ <para>This function will cause the PJSIP stack to immediately refresh
+ the media session for the channel. This will be done using either a
+ re-INVITE (default) or an UPDATE request.
+ </para>
+ <para>This is most useful when combined with the <replaceable>PJSIP_MEDIA_OFFER</replaceable>
+ dialplan function, as it allows the formats in use on a channel to be
+ re-negotiated after call setup.</para>
+ <warning>
+ <para>The formats the endpoint supports are <emphasis>not</emphasis>
+ checked or enforced by this function. Using this function to offer
+ formats not supported by the endpoint <emphasis>may</emphasis> result
+ in a loss of media.</para>
+ </warning>
+ <example title="Re-negotiate format to g722">
+ ; Within some existing extension on an answered channel
+ same => n,Set(PJSIP_MEDIA_OFFER(audio)=!all,g722)
+ same => n,Set(PJSIP_SEND_SESSION_REFRESH()=invite)
+ </example>
+ </description>
+ <see-also>
+ <ref type="function">PJSIP_MEDIA_OFFER</ref>
+ </see-also>
+ </function>
+ <function name="PJSIP_PARSE_URI" language="en_US">
+ <since>
+ <version>13.24.0</version>
+ <version>16.1.0</version>
+ <version>17.0.0</version>
+ </since>
+ <synopsis>
+ Parse an uri and return a type part of the URI.
+ </synopsis>
+ <syntax>
+ <parameter name="uri" required="true">
+ <para>URI to parse</para>
+ </parameter>
+ <parameter name="type" required="true">
+ <para>The <literal>type</literal> parameter specifies which URI part to read</para>
+ <enumlist>
+ <enum name="display">
+ <para>Display name.</para>
+ </enum>
+ <enum name="scheme">
+ <para>URI scheme.</para>
+ </enum>
+ <enum name="user">
+ <para>User part.</para>
+ </enum>
+ <enum name="passwd">
+ <para>Password part.</para>
+ </enum>
+ <enum name="host">
+ <para>Host part.</para>
+ </enum>
+ <enum name="port">
+ <para>Port number, or zero.</para>
+ </enum>
+ <enum name="user_param">
+ <para>User parameter.</para>
+ </enum>
+ <enum name="method_param">
+ <para>Method parameter.</para>
+ </enum>
+ <enum name="transport_param">
+ <para>Transport parameter.</para>
+ </enum>
+ <enum name="ttl_param">
+ <para>TTL param, or -1.</para>
+ </enum>
+ <enum name="lr_param">
+ <para>Loose routing param, or zero.</para>
+ </enum>
+ <enum name="maddr_param">
+ <para>Maddr param.</para>
+ </enum>
+ </enumlist>
+ </parameter>
+ </syntax>
+ <description>
+ <para>Parse an URI and return a specified part of the URI.</para>
+ </description>
+ </function>
+
+ <info name="CHANNEL" language="en_US" tech="PJSIP">
+ <enumlist>
+ <enum name="rtp">
+ <para>R/O Retrieve media related information.</para>
+ <parameter name="type" required="true">
+ <para>When <replaceable>rtp</replaceable> is specified, the
+ <literal>type</literal> parameter must be provided. It specifies
+ which RTP parameter to read.</para>
+ <enumlist>
+ <enum name="src">
+ <para>Retrieve the local address for RTP.</para>
+ </enum>
+ <enum name="dest">
+ <para>Retrieve the remote address for RTP.</para>
+ </enum>
+ <enum name="direct">
+ <para>If direct media is enabled, this address is the remote address
+ used for RTP.</para>
+ </enum>
+ <enum name="secure">
+ <para>Whether or not the media stream is encrypted.</para>
+ <enumlist>
+ <enum name="0">
+ <para>The media stream is not encrypted.</para>
+ </enum>
+ <enum name="1">
+ <para>The media stream is encrypted.</para>
+ </enum>
+ </enumlist>
+ </enum>
+ <enum name="hold">
+ <para>Whether or not the media stream is currently restricted
+ due to a call hold.</para>
+ <enumlist>
+ <enum name="0">
+ <para>The media stream is not held.</para>
+ </enum>
+ <enum name="1">
+ <para>The media stream is held.</para>
+ </enum>
+ </enumlist>
+ </enum>
+ </enumlist>
+ </parameter>
+ <parameter name="media_type" required="false">
+ <para>When <replaceable>rtp</replaceable> is specified, the
+ <literal>media_type</literal> parameter may be provided. It specifies
+ which media stream the chosen RTP parameter should be retrieved
+ from.</para>
+ <enumlist>
+ <enum name="audio">
+ <para>Retrieve information from the audio media stream.</para>
+ <note><para>If not specified, <literal>audio</literal> is used
+ by default.</para></note>
+ </enum>
+ <enum name="video">
+ <para>Retrieve information from the video media stream.</para>
+ </enum>
+ </enumlist>
+ </parameter>
+ </enum>
+ <enum name="rtcp">
+ <para>R/O Retrieve RTCP statistics.</para>
+ <parameter name="statistic" required="true">
+ <para>When <replaceable>rtcp</replaceable> is specified, the
+ <literal>statistic</literal> parameter must be provided. It specifies
+ which RTCP statistic parameter to read.</para>
+ <enumlist>
+ <enum name="all">
+ <para>Retrieve a summary of all RTCP statistics.</para>
+ <para>The following data items are returned in a semi-colon
+ delineated list:</para>
+ <enumlist>
+ <enum name="ssrc">
+ <para>Our Synchronization Source identifier</para>
+ </enum>
+ <enum name="themssrc">
+ <para>Their Synchronization Source identifier</para>
+ </enum>
+ <enum name="lp">
+ <para>Our lost packet count</para>
+ </enum>
+ <enum name="rxjitter">
+ <para>Received packet jitter</para>
+ </enum>
+ <enum name="rxcount">
+ <para>Received packet count</para>
+ </enum>
+ <enum name="txjitter">
+ <para>Transmitted packet jitter</para>
+ </enum>
+ <enum name="txcount">
+ <para>Transmitted packet count</para>
+ </enum>
+ <enum name="rlp">
+ <para>Remote lost packet count</para>
+ </enum>
+ <enum name="rtt">
+ <para>Round trip time</para>
+ </enum>
+ <enum name="txmes">
+ <para>Transmitted Media Experience Score</para>
+ </enum>
+ <enum name="rxmes">
+ <para>Received Media Experience Score</para>
+ </enum>
+ </enumlist>
+ </enum>
+ <enum name="all_jitter">
+ <para>Retrieve a summary of all RTCP Jitter statistics.</para>
+ <para>The following data items are returned in a semi-colon
+ delineated list:</para>
+ <enumlist>
+ <enum name="minrxjitter">
+ <para>Our minimum jitter</para>
+ </enum>
+ <enum name="maxrxjitter">
+ <para>Our max jitter</para>
+ </enum>
+ <enum name="avgrxjitter">
+ <para>Our average jitter</para>
+ </enum>
+ <enum name="stdevrxjitter">
+ <para>Our jitter standard deviation</para>
+ </enum>
+ <enum name="reported_minjitter">
+ <para>Their minimum jitter</para>
+ </enum>
+ <enum name="reported_maxjitter">
+ <para>Their max jitter</para>
+ </enum>
+ <enum name="reported_avgjitter">
+ <para>Their average jitter</para>
+ </enum>
+ <enum name="reported_stdevjitter">
+ <para>Their jitter standard deviation</para>
+ </enum>
+ </enumlist>
+ </enum>
+ <enum name="all_loss">
+ <para>Retrieve a summary of all RTCP packet loss statistics.</para>
+ <para>The following data items are returned in a semi-colon
+ delineated list:</para>
+ <enumlist>
+ <enum name="minrxlost">
+ <para>Our minimum lost packets</para>
+ </enum>
+ <enum name="maxrxlost">
+ <para>Our max lost packets</para>
+ </enum>
+ <enum name="avgrxlost">
+ <para>Our average lost packets</para>
+ </enum>
+ <enum name="stdevrxlost">
+ <para>Our lost packets standard deviation</para>
+ </enum>
+ <enum name="reported_minlost">
+ <para>Their minimum lost packets</para>
+ </enum>
+ <enum name="reported_maxlost">
+ <para>Their max lost packets</para>
+ </enum>
+ <enum name="reported_avglost">
+ <para>Their average lost packets</para>
+ </enum>
+ <enum name="reported_stdevlost">
+ <para>Their lost packets standard deviation</para>
+ </enum>
+ </enumlist>
+ </enum>
+ <enum name="all_rtt">
+ <para>Retrieve a summary of all RTCP round trip time information.</para>
+ <para>The following data items are returned in a semi-colon
+ delineated list:</para>
+ <enumlist>
+ <enum name="minrtt">
+ <para>Minimum round trip time</para>
+ </enum>
+ <enum name="maxrtt">
+ <para>Maximum round trip time</para>
+ </enum>
+ <enum name="avgrtt">
+ <para>Average round trip time</para>
+ </enum>
+ <enum name="stdevrtt">
+ <para>Standard deviation round trip time</para>
+ </enum>
+ </enumlist>
+ </enum>
+ <enum name="all_mes">
+ <para>Retrieve a summary of all RTCP Media Experience Score information.</para>
+ <para>The following data items are returned in a semi-colon
+ delineated list:</para>
+ <enumlist>
+ <enum name="minmes">
+ <para>Minimum MES based on us analysing received packets.</para>
+ </enum>
+ <enum name="maxmes">
+ <para>Maximum MES based on us analysing received packets.</para>
+ </enum>
+ <enum name="avgmes">
+ <para>Average MES based on us analysing received packets.</para>
+ </enum>
+ <enum name="stdevmes">
+ <para>Standard deviation MES based on us analysing received packets.</para>
+ </enum>
+ <enum name="reported_minmes">
+ <para>Minimum MES based on data we get in Sender and Receiver Reports sent by the remote end</para>
+ </enum>
+ <enum name="reported_maxmes">
+ <para>Maximum MES based on data we get in Sender and Receiver Reports sent by the remote end</para>
+ </enum>
+ <enum name="reported_avgmes">
+ <para>Average MES based on data we get in Sender and Receiver Reports sent by the remote end</para>
+ </enum>
+ <enum name="reported_stdevmes">
+ <para>Standard deviation MES based on data we get in Sender and Receiver Reports sent by the remote end</para>
+ </enum>
+ </enumlist>
+ </enum>
+ <enum name="txcount"><para>Transmitted packet count</para></enum>
+ <enum name="rxcount"><para>Received packet count</para></enum>
+ <enum name="txjitter"><para>Transmitted packet jitter</para></enum>
+ <enum name="rxjitter"><para>Received packet jitter</para></enum>
+ <enum name="remote_maxjitter"><para>Their max jitter</para></enum>
+ <enum name="remote_minjitter"><para>Their minimum jitter</para></enum>
+ <enum name="remote_normdevjitter"><para>Their average jitter</para></enum>
+ <enum name="remote_stdevjitter"><para>Their jitter standard deviation</para></enum>
+ <enum name="local_maxjitter"><para>Our max jitter</para></enum>
+ <enum name="local_minjitter"><para>Our minimum jitter</para></enum>
+ <enum name="local_normdevjitter"><para>Our average jitter</para></enum>
+ <enum name="local_stdevjitter"><para>Our jitter standard deviation</para></enum>
+ <enum name="txploss"><para>Transmitted packet loss</para></enum>
+ <enum name="rxploss"><para>Received packet loss</para></enum>
+ <enum name="remote_maxrxploss"><para>Their max lost packets</para></enum>
+ <enum name="remote_minrxploss"><para>Their minimum lost packets</para></enum>
+ <enum name="remote_normdevrxploss"><para>Their average lost packets</para></enum>
+ <enum name="remote_stdevrxploss"><para>Their lost packets standard deviation</para></enum>
+ <enum name="local_maxrxploss"><para>Our max lost packets</para></enum>
+ <enum name="local_minrxploss"><para>Our minimum lost packets</para></enum>
+ <enum name="local_normdevrxploss"><para>Our average lost packets</para></enum>
+ <enum name="local_stdevrxploss"><para>Our lost packets standard deviation</para></enum>
+ <enum name="rtt"><para>Round trip time</para></enum>
+ <enum name="maxrtt"><para>Maximum round trip time</para></enum>
+ <enum name="minrtt"><para>Minimum round trip time</para></enum>
+ <enum name="normdevrtt"><para>Average round trip time</para></enum>
+ <enum name="stdevrtt"><para>Standard deviation round trip time</para></enum>
+ <enum name="local_ssrc"><para>Our Synchronization Source identifier</para></enum>
+ <enum name="remote_ssrc"><para>Their Synchronization Source identifier</para></enum>
+ <enum name="txmes"><para>
+ Current MES based on us analyzing rtt, jitter and loss
+ in the actual received RTP stream received from the remote end.
+ I.E. This is the MES for the incoming audio stream.
+ </para></enum>
+ <enum name="rxmes"><para>
+ Current MES based on rtt and the jitter and loss values in
+ RTCP sender and receiver reports we receive from the
+ remote end. I.E. This is the MES for the outgoing audio stream.
+ </para></enum>
+ <enum name="remote_maxmes"><para>Max MES based on data we get in Sender and Receiver Reports sent by the remote end</para></enum>
+ <enum name="remote_minmes"><para>Min MES based on data we get in Sender and Receiver Reports sent by the remote end</para></enum>
+ <enum name="remote_normdevmes"><para>Average MES based on data we get in Sender and Receiver Reports sent by the remote end</para></enum>
+ <enum name="remote_stdevmes"><para>Standard deviation MES based on data we get in Sender and Receiver Reports sent by the remote end</para></enum>
+ <enum name="local_maxmes"><para>Max MES based on us analyzing the received RTP stream</para></enum>
+ <enum name="local_minmes"><para>Min MES based on us analyzing the received RTP stream</para></enum>
+ <enum name="local_normdevmes"><para>Average MES based on us analyzing the received RTP stream</para></enum>
+ <enum name="local_stdevmes"><para>Standard deviation MES based on us analyzing the received RTP stream</para></enum>
+ </enumlist>
+ </parameter>
+ <parameter name="media_type" required="false">
+ <para>When <replaceable>rtcp</replaceable> is specified, the
+ <literal>media_type</literal> parameter may be provided. It specifies
+ which media stream the chosen RTCP parameter should be retrieved
+ from.</para>
+ <enumlist>
+ <enum name="audio">
+ <para>Retrieve information from the audio media stream.</para>
+ <note><para>If not specified, <literal>audio</literal> is used
+ by default.</para></note>
+ </enum>
+ <enum name="video">
+ <para>Retrieve information from the video media stream.</para>
+ </enum>
+ </enumlist>
+ </parameter>
+ </enum>
+ <enum name="endpoint">
+ <para>R/O The name of the endpoint associated with this channel.
+ Use the <replaceable>PJSIP_ENDPOINT</replaceable> function to obtain
+ further endpoint related information.</para>
+ </enum>
+ <enum name="contact">
+ <para>R/O The name of the contact associated with this channel.
+ Use the <replaceable>PJSIP_CONTACT</replaceable> function to obtain
+ further contact related information. Note this may not be present and if so
+ is only available on outgoing legs.</para>
+ </enum>
+ <enum name="aor">
+ <para>R/O The name of the AOR associated with this channel.
+ Use the <replaceable>PJSIP_AOR</replaceable> function to obtain
+ further AOR related information. Note this may not be present and if so
+ is only available on outgoing legs.</para>
+ </enum>
+ <enum name="pjsip">
+ <para>R/O Obtain information about the current PJSIP channel and its
+ session.</para>
+ <parameter name="type" required="true">
+ <para>When <replaceable>pjsip</replaceable> is specified, the
+ <literal>type</literal> parameter must be provided. It specifies
+ which signalling parameter to read.</para>
+ <enumlist>
+ <enum name="call-id">
+ <para>The SIP call-id.</para>
+ </enum>
+ <enum name="secure">
+ <para>Whether or not the signalling uses a secure transport.</para>
+ <enumlist>
+ <enum name="0"><para>The signalling uses a non-secure transport.</para></enum>
+ <enum name="1"><para>The signalling uses a secure transport.</para></enum>
+ </enumlist>
+ </enum>
+ <enum name="target_uri">
+ <para>The contact URI where requests are sent.</para>
+ </enum>
+ <enum name="local_uri">
+ <para>The local URI.</para>
+ </enum>
+ <enum name="local_tag">
+ <para>Tag in From header</para>
+ </enum>
+ <enum name="remote_uri">
+ <para>The remote URI.</para>
+ </enum>
+ <enum name="remote_tag">
+ <para>Tag in To header</para>
+ </enum>
+ <enum name="request_uri">
+ <para>The request URI of the incoming <literal>INVITE</literal>
+ associated with the creation of this channel.</para>
+ </enum>
+ <enum name="t38state">
+ <para>The current state of any T.38 fax on this channel.</para>
+ <enumlist>
+ <enum name="DISABLED"><para>T.38 faxing is disabled on this channel.</para></enum>
+ <enum name="LOCAL_REINVITE"><para>Asterisk has sent a <literal>re-INVITE</literal> to the remote end to initiate a T.38 fax.</para></enum>
+ <enum name="REMOTE_REINVITE"><para>The remote end has sent a <literal>re-INVITE</literal> to Asterisk to initiate a T.38 fax.</para></enum>
+ <enum name="ENABLED"><para>A T.38 fax session has been enabled.</para></enum>
+ <enum name="REJECTED"><para>A T.38 fax session was attempted but was rejected.</para></enum>
+ </enumlist>
+ </enum>
+ <enum name="local_addr">
+ <para>On inbound calls, the full IP address and port number that
+ the <literal>INVITE</literal> request was received on. On outbound
+ calls, the full IP address and port number that the <literal>INVITE</literal>
+ request was transmitted from.</para>
+ </enum>
+ <enum name="remote_addr">
+ <para>On inbound calls, the full IP address and port number that
+ the <literal>INVITE</literal> request was received from. On outbound
+ calls, the full IP address and port number that the <literal>INVITE</literal>
+ request was transmitted to.</para>
+ </enum>
+ </enumlist>
+ </parameter>
+ </enum>
+ </enumlist>
+ </info>
+ <info name="CHANNEL_EXAMPLES" language="en_US" tech="PJSIP">
+ <example title="PJSIP specific CHANNEL examples">
+ ; Log the current Call-ID
+ same => n,Log(NOTICE, ${CHANNEL(pjsip,call-id)})
+
+ ; Log the destination address of the audio stream
+ same => n,Log(NOTICE, ${CHANNEL(rtp,dest)})
+
+ ; Store the round-trip time associated with a
+ ; video stream in the CDR field video-rtt
+ same => n,Set(CDR(video-rtt)=${CHANNEL(rtcp,rtt,video)})
+ </example>
+ </info>
+</docs>
\ No newline at end of file
*/
int pjsip_acf_parse_uri_read(struct ast_channel *chan, const char *cmd, char *data, char *buf, size_t len);
-#endif /* _PJSIP_DIALPLAN_FUNCTIONS */
\ No newline at end of file
+/*!
+ * \brief Hang up an incoming PJSIP channel with a SIP response code
+ * \param chan The channel the function is called on
+ * \param data SIP response code or name
+ *
+ * \retval 0 on success
+ * \retval -1 on failure
+ */
+int pjsip_app_hangup(struct ast_channel *chan, const char *data);
+
+/*!
+ * \brief Manager action to hang up an incoming PJSIP channel with a SIP response code
+ * \param s session
+ * \param m message
+ *
+ * \retval 0 on success
+ * \retval -1 on failure
+ */
+int pjsip_action_hangup(struct mansession *s, const struct message *m);
+
+#endif /* _PJSIP_DIALPLAN_FUNCTIONS */
*/
struct stasis_message_router *ast_manager_get_message_router(void);
+/*!
+ * \brief Callback used by ast_manager_hangup_helper
+ * that will actually hangup a channel
+ *
+ * \param chan The channel to hang up
+ * \param causecode Cause code to set on the channel
+ */
+typedef void (*manager_hangup_handler_t)(struct ast_channel *chan, int causecode);
+
+/*!
+ * \brief Callback used by ast_manager_hangup_helper
+ * that will validate the cause code.
+
+ * \param channel_name Mostly for displaying log messages
+ * \param cause Cause code string
+ *
+ * \returns integer cause code
+ */
+typedef int (*manager_hangup_cause_validator_t)(const char *channel_name,
+ const char *cause);
+
+/*!
+ * \brief A manager helper function that hangs up a channel using a supplied
+ * channel type specific hangup function and cause code validator
+ *
+ * This function handles the lookup of channel(s) and the AMI interaction
+ * but uses the supplied callbacks to actually perform the hangup. It can be
+ * used to implement a custom AMI 'Hangup' action without having to duplicate
+ * all the code in the standard Hangup action.
+ *
+ * \param s Session
+ * \param m Message
+ * \param handler Function that actually performs the hangup
+ * \param cause_validator Function that validates the cause code
+ *
+ * \retval 0 on success.
+ * \retval non-zero on error.
+ */
+int ast_manager_hangup_helper(struct mansession *s, const struct message *m,
+ manager_hangup_handler_t handler, manager_hangup_cause_validator_t cause_validator);
+
#endif /* _ASTERISK_MANAGER_H */
*/
const int ast_sip_hangup_sip2cause(int cause);
+/*!
+ * \brief Convert name to SIP response code
+ *
+ * \param name SIP response code name matching one of the
+ * enum names defined in "enum pjsip_status_code"
+ * defined in sip_msg.h. May be specified with or
+ * without the PJSIP_SC_ prefix.
+ *
+ * \retval SIP response code
+ * \retval -1 if matching code not found
+ */
+int ast_sip_str2rc(const char *name);
#endif /* _RES_PJSIP_H */
return 0;
}
-static int action_hangup(struct mansession *s, const struct message *m)
+int ast_manager_hangup_helper(struct mansession *s,
+ const struct message *m, manager_hangup_handler_t hangup_handler,
+ manager_hangup_cause_validator_t cause_validator)
{
struct ast_channel *c = NULL;
int causecode = 0; /* all values <= 0 mean 'do not set hangupcause in channel' */
idText[0] = '\0';
}
- if (!ast_strlen_zero(cause)) {
+ if (cause_validator) {
+ causecode = cause_validator(name_or_regex, cause);
+ } else if (!ast_strlen_zero(cause)) {
char *endptr;
causecode = strtol(cause, &endptr, 10);
if (causecode < 0 || causecode > 127 || *endptr != '\0') {
ast_sockaddr_stringify_addr(&s->session->addr),
ast_channel_name(c));
- ast_channel_softhangup_withcause_locked(c, causecode);
+ hangup_handler(c, causecode);
c = ast_channel_unref(c);
astman_send_ack(s, m, "Channel Hungup");
ast_sockaddr_stringify_addr(&s->session->addr),
ast_channel_name(c));
- ast_channel_softhangup_withcause_locked(c, causecode);
+ hangup_handler(c, causecode);
channels_matched++;
astman_append(s,
return 0;
}
+static int action_hangup(struct mansession *s, const struct message *m)
+{
+ return ast_manager_hangup_helper(s, m,
+ ast_channel_softhangup_withcause_locked, NULL);
+}
+
static int action_setvar(struct mansession *s, const struct message *m)
{
struct ast_channel *c = NULL;
return 0;
}
+struct response_code_map {
+ int code;
+ const char *long_name;
+ const char *short_name;
+};
+
+/*
+ * This map was generated from sip_msg.h with
+ *
+ * sed -n -r -e 's/^\s+(PJSIP_SC_([^ =]+))\s*=\s*[0-9]+,/{ \1, "\1", "\2" },/gp' \
+ * third-party/pjproject/source/pjsip/include/pjsip/sip_msg.h
+ *
+ */
+static const struct response_code_map rc_map[] = {
+ { PJSIP_SC_NULL, "PJSIP_SC_NULL", "NULL" },
+ { PJSIP_SC_TRYING, "PJSIP_SC_TRYING", "TRYING" },
+ { PJSIP_SC_RINGING, "PJSIP_SC_RINGING", "RINGING" },
+ { PJSIP_SC_CALL_BEING_FORWARDED, "PJSIP_SC_CALL_BEING_FORWARDED", "CALL_BEING_FORWARDED" },
+ { PJSIP_SC_QUEUED, "PJSIP_SC_QUEUED", "QUEUED" },
+ { PJSIP_SC_PROGRESS, "PJSIP_SC_PROGRESS", "PROGRESS" },
+ { PJSIP_SC_EARLY_DIALOG_TERMINATED, "PJSIP_SC_EARLY_DIALOG_TERMINATED", "EARLY_DIALOG_TERMINATED" },
+ { PJSIP_SC_OK, "PJSIP_SC_OK", "OK" },
+ { PJSIP_SC_ACCEPTED, "PJSIP_SC_ACCEPTED", "ACCEPTED" },
+ { PJSIP_SC_NO_NOTIFICATION, "PJSIP_SC_NO_NOTIFICATION", "NO_NOTIFICATION" },
+ { PJSIP_SC_MULTIPLE_CHOICES, "PJSIP_SC_MULTIPLE_CHOICES", "MULTIPLE_CHOICES" },
+ { PJSIP_SC_MOVED_PERMANENTLY, "PJSIP_SC_MOVED_PERMANENTLY", "MOVED_PERMANENTLY" },
+ { PJSIP_SC_MOVED_TEMPORARILY, "PJSIP_SC_MOVED_TEMPORARILY", "MOVED_TEMPORARILY" },
+ { PJSIP_SC_USE_PROXY, "PJSIP_SC_USE_PROXY", "USE_PROXY" },
+ { PJSIP_SC_ALTERNATIVE_SERVICE, "PJSIP_SC_ALTERNATIVE_SERVICE", "ALTERNATIVE_SERVICE" },
+ { PJSIP_SC_BAD_REQUEST, "PJSIP_SC_BAD_REQUEST", "BAD_REQUEST" },
+ { PJSIP_SC_UNAUTHORIZED, "PJSIP_SC_UNAUTHORIZED", "UNAUTHORIZED" },
+ { PJSIP_SC_PAYMENT_REQUIRED, "PJSIP_SC_PAYMENT_REQUIRED", "PAYMENT_REQUIRED" },
+ { PJSIP_SC_FORBIDDEN, "PJSIP_SC_FORBIDDEN", "FORBIDDEN" },
+ { PJSIP_SC_NOT_FOUND, "PJSIP_SC_NOT_FOUND", "NOT_FOUND" },
+ { PJSIP_SC_METHOD_NOT_ALLOWED, "PJSIP_SC_METHOD_NOT_ALLOWED", "METHOD_NOT_ALLOWED" },
+ { PJSIP_SC_NOT_ACCEPTABLE, "PJSIP_SC_NOT_ACCEPTABLE", "NOT_ACCEPTABLE" },
+ { PJSIP_SC_PROXY_AUTHENTICATION_REQUIRED, "PJSIP_SC_PROXY_AUTHENTICATION_REQUIRED", "PROXY_AUTHENTICATION_REQUIRED" },
+ { PJSIP_SC_REQUEST_TIMEOUT, "PJSIP_SC_REQUEST_TIMEOUT", "REQUEST_TIMEOUT" },
+ { PJSIP_SC_CONFLICT, "PJSIP_SC_CONFLICT", "CONFLICT" },
+ { PJSIP_SC_GONE, "PJSIP_SC_GONE", "GONE" },
+ { PJSIP_SC_LENGTH_REQUIRED, "PJSIP_SC_LENGTH_REQUIRED", "LENGTH_REQUIRED" },
+ { PJSIP_SC_CONDITIONAL_REQUEST_FAILED, "PJSIP_SC_CONDITIONAL_REQUEST_FAILED", "CONDITIONAL_REQUEST_FAILED" },
+ { PJSIP_SC_REQUEST_ENTITY_TOO_LARGE, "PJSIP_SC_REQUEST_ENTITY_TOO_LARGE", "REQUEST_ENTITY_TOO_LARGE" },
+ { PJSIP_SC_REQUEST_URI_TOO_LONG, "PJSIP_SC_REQUEST_URI_TOO_LONG", "REQUEST_URI_TOO_LONG" },
+ { PJSIP_SC_UNSUPPORTED_MEDIA_TYPE, "PJSIP_SC_UNSUPPORTED_MEDIA_TYPE", "UNSUPPORTED_MEDIA_TYPE" },
+ { PJSIP_SC_UNSUPPORTED_URI_SCHEME, "PJSIP_SC_UNSUPPORTED_URI_SCHEME", "UNSUPPORTED_URI_SCHEME" },
+ { PJSIP_SC_UNKNOWN_RESOURCE_PRIORITY, "PJSIP_SC_UNKNOWN_RESOURCE_PRIORITY", "UNKNOWN_RESOURCE_PRIORITY" },
+ { PJSIP_SC_BAD_EXTENSION, "PJSIP_SC_BAD_EXTENSION", "BAD_EXTENSION" },
+ { PJSIP_SC_EXTENSION_REQUIRED, "PJSIP_SC_EXTENSION_REQUIRED", "EXTENSION_REQUIRED" },
+ { PJSIP_SC_SESSION_TIMER_TOO_SMALL, "PJSIP_SC_SESSION_TIMER_TOO_SMALL", "SESSION_TIMER_TOO_SMALL" },
+ { PJSIP_SC_INTERVAL_TOO_BRIEF, "PJSIP_SC_INTERVAL_TOO_BRIEF", "INTERVAL_TOO_BRIEF" },
+ { PJSIP_SC_BAD_LOCATION_INFORMATION, "PJSIP_SC_BAD_LOCATION_INFORMATION", "BAD_LOCATION_INFORMATION" },
+ { PJSIP_SC_USE_IDENTITY_HEADER, "PJSIP_SC_USE_IDENTITY_HEADER", "USE_IDENTITY_HEADER" },
+ { PJSIP_SC_PROVIDE_REFERRER_HEADER, "PJSIP_SC_PROVIDE_REFERRER_HEADER", "PROVIDE_REFERRER_HEADER" },
+ { PJSIP_SC_FLOW_FAILED, "PJSIP_SC_FLOW_FAILED", "FLOW_FAILED" },
+ { PJSIP_SC_ANONIMITY_DISALLOWED, "PJSIP_SC_ANONIMITY_DISALLOWED", "ANONIMITY_DISALLOWED" },
+ { PJSIP_SC_BAD_IDENTITY_INFO, "PJSIP_SC_BAD_IDENTITY_INFO", "BAD_IDENTITY_INFO" },
+ { PJSIP_SC_UNSUPPORTED_CERTIFICATE, "PJSIP_SC_UNSUPPORTED_CERTIFICATE", "UNSUPPORTED_CERTIFICATE" },
+ { PJSIP_SC_INVALID_IDENTITY_HEADER, "PJSIP_SC_INVALID_IDENTITY_HEADER", "INVALID_IDENTITY_HEADER" },
+ { PJSIP_SC_FIRST_HOP_LACKS_OUTBOUND_SUPPORT, "PJSIP_SC_FIRST_HOP_LACKS_OUTBOUND_SUPPORT", "FIRST_HOP_LACKS_OUTBOUND_SUPPORT" },
+ { PJSIP_SC_MAX_BREADTH_EXCEEDED, "PJSIP_SC_MAX_BREADTH_EXCEEDED", "MAX_BREADTH_EXCEEDED" },
+ { PJSIP_SC_BAD_INFO_PACKAGE, "PJSIP_SC_BAD_INFO_PACKAGE", "BAD_INFO_PACKAGE" },
+ { PJSIP_SC_CONSENT_NEEDED, "PJSIP_SC_CONSENT_NEEDED", "CONSENT_NEEDED" },
+ { PJSIP_SC_TEMPORARILY_UNAVAILABLE, "PJSIP_SC_TEMPORARILY_UNAVAILABLE", "TEMPORARILY_UNAVAILABLE" },
+ { PJSIP_SC_CALL_TSX_DOES_NOT_EXIST, "PJSIP_SC_CALL_TSX_DOES_NOT_EXIST", "CALL_TSX_DOES_NOT_EXIST" },
+ { PJSIP_SC_LOOP_DETECTED, "PJSIP_SC_LOOP_DETECTED", "LOOP_DETECTED" },
+ { PJSIP_SC_TOO_MANY_HOPS, "PJSIP_SC_TOO_MANY_HOPS", "TOO_MANY_HOPS" },
+ { PJSIP_SC_ADDRESS_INCOMPLETE, "PJSIP_SC_ADDRESS_INCOMPLETE", "ADDRESS_INCOMPLETE" },
+ { PJSIP_SC_BUSY_HERE, "PJSIP_SC_BUSY_HERE", "BUSY_HERE" },
+ { PJSIP_SC_REQUEST_TERMINATED, "PJSIP_SC_REQUEST_TERMINATED", "REQUEST_TERMINATED" },
+ { PJSIP_SC_NOT_ACCEPTABLE_HERE, "PJSIP_SC_NOT_ACCEPTABLE_HERE", "NOT_ACCEPTABLE_HERE" },
+ { PJSIP_SC_BAD_EVENT, "PJSIP_SC_BAD_EVENT", "BAD_EVENT" },
+ { PJSIP_SC_REQUEST_UPDATED, "PJSIP_SC_REQUEST_UPDATED", "REQUEST_UPDATED" },
+ { PJSIP_SC_REQUEST_PENDING, "PJSIP_SC_REQUEST_PENDING", "REQUEST_PENDING" },
+ { PJSIP_SC_UNDECIPHERABLE, "PJSIP_SC_UNDECIPHERABLE", "UNDECIPHERABLE" },
+ { PJSIP_SC_SECURITY_AGREEMENT_NEEDED, "PJSIP_SC_SECURITY_AGREEMENT_NEEDED", "SECURITY_AGREEMENT_NEEDED" },
+ { PJSIP_SC_INTERNAL_SERVER_ERROR, "PJSIP_SC_INTERNAL_SERVER_ERROR", "INTERNAL_SERVER_ERROR" },
+ { PJSIP_SC_NOT_IMPLEMENTED, "PJSIP_SC_NOT_IMPLEMENTED", "NOT_IMPLEMENTED" },
+ { PJSIP_SC_BAD_GATEWAY, "PJSIP_SC_BAD_GATEWAY", "BAD_GATEWAY" },
+ { PJSIP_SC_SERVICE_UNAVAILABLE, "PJSIP_SC_SERVICE_UNAVAILABLE", "SERVICE_UNAVAILABLE" },
+ { PJSIP_SC_SERVER_TIMEOUT, "PJSIP_SC_SERVER_TIMEOUT", "SERVER_TIMEOUT" },
+ { PJSIP_SC_VERSION_NOT_SUPPORTED, "PJSIP_SC_VERSION_NOT_SUPPORTED", "VERSION_NOT_SUPPORTED" },
+ { PJSIP_SC_MESSAGE_TOO_LARGE, "PJSIP_SC_MESSAGE_TOO_LARGE", "MESSAGE_TOO_LARGE" },
+ { PJSIP_SC_PUSH_NOTIFICATION_SERVICE_NOT_SUPPORTED, "PJSIP_SC_PUSH_NOTIFICATION_SERVICE_NOT_SUPPORTED", "PUSH_NOTIFICATION_SERVICE_NOT_SUPPORTED" },
+ { PJSIP_SC_PRECONDITION_FAILURE, "PJSIP_SC_PRECONDITION_FAILURE", "PRECONDITION_FAILURE" },
+ { PJSIP_SC_BUSY_EVERYWHERE, "PJSIP_SC_BUSY_EVERYWHERE", "BUSY_EVERYWHERE" },
+ { PJSIP_SC_DECLINE, "PJSIP_SC_DECLINE", "DECLINE" },
+ { PJSIP_SC_DOES_NOT_EXIST_ANYWHERE, "PJSIP_SC_DOES_NOT_EXIST_ANYWHERE", "DOES_NOT_EXIST_ANYWHERE" },
+ { PJSIP_SC_NOT_ACCEPTABLE_ANYWHERE, "PJSIP_SC_NOT_ACCEPTABLE_ANYWHERE", "NOT_ACCEPTABLE_ANYWHERE" },
+ { PJSIP_SC_UNWANTED, "PJSIP_SC_UNWANTED", "UNWANTED" },
+ { PJSIP_SC_REJECTED, "PJSIP_SC_REJECTED", "REJECTED" },
+};
+
+int ast_sip_str2rc(const char *name)
+{
+ int i;
+
+ if (ast_strlen_zero(name)) {
+ return -1;
+ }
+
+ for (i = 0; i < ARRAY_LEN(rc_map); i++) {
+ if (strcasecmp(rc_map[i].short_name, name) == 0 ||
+ strcasecmp(rc_map[i].long_name, name) == 0) {
+ return rc_map[i].code;
+ }
+ }
+
+ return -1;
+}
+
+
#ifdef TEST_FRAMEWORK
AST_TEST_DEFINE(xml_sanitization_end_null)
{
projects / thirdparty / asterisk.git / commitdiff
