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ari/pjsip: Make it possible to control transfers through ARI

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Introduce a ChannelTransfer event and the ability to notify progress to
ARI. Implement emitting this event from the PJSIP channel instead of
handling the transfer in Asterisk when configured.

Introduce a dialplan function to the PJSIP channel to switch between the
"core" and "ari-only" behavior.

UserNote: Call transfers on the PJSIP channel can now be controlled by
ARI. This can be enabled by using the PJSIP_TRANSFER_HANDLING(ari-only)
dialplan function.
This commit is contained in:
Holger Hans Peter Freyther
2024-06-15 16:01:58 +08:00
committed by github-actions[bot]
parent 1f1fa06f69
commit 9fc631d028
18 changed files with 1462 additions and 6 deletions
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33
channels/pjsip/dialplan_functions.c
View File

@@ -1368,3 +1368,36 @@ 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);
}
int pjsip_transfer_handling_write(struct ast_channel *chan, const char *cmd, char *data, const char *value)
{
struct ast_sip_channel_pvt *channel;
int ret = 0;
if (!chan) {
ast_log(LOG_WARNING, "No channel was provided to %s function.\n", cmd);
return -1;
}
ast_channel_lock(chan);
if (strcmp(ast_channel_tech(chan)->type, "PJSIP")) {
ast_log(LOG_WARNING, "Cannot call %s on a non-PJSIP channel %s\n", cmd, ast_channel_name(chan));
ast_channel_unlock(chan);
return -1;
}
channel = ast_channel_tech_pvt(chan);
if (ast_strlen_zero(value) || !strcmp(value, "core")) {
channel->session->transferhandling_ari = 0;
} else if (!strcmp(value, "ari-only")) {
channel->session->transferhandling_ari = 1;
} else {
ast_log(AST_LOG_WARNING, "Cannot set unknown transfer handling '%s' on channel '%s', transfer handling will remain unchanged.",
value, ast_channel_name(chan));
ret = -1;
}
ast_channel_unlock(chan);
return ret;
}

26
channels/pjsip/dialplan_functions_doc.xml
View File

@@ -405,6 +405,32 @@
<para>Parse the contents of the provided variable as a URI and return a specified part of the URI.</para>
</description>
</function>
<function name="PJSIP_TRANSFER_HANDLING" language="en_US">
<since>
<version>22.3.0</version>
<version>21.8.0</version>
<version>20.13.0</version>
</since>
<synopsis>
Set how transfers are handled for a PJSIP channel.
</synopsis>
<syntax>
<parameter name="mode" required="false">
<para>How transfers are handled for a PJSIP channel. Default is <literal>core</literal>.</para>
<enumlist>
<enum name="core">
<para>Asterisk will handle attended and blind transfers.</para>
</enum>
<enum name="ari-only">
<para>Asterisk will generate ARI events on incoming SIP REFER.</para>
</enum>
</enumlist>
</parameter>
</syntax>
<description>
<para>When written, sets the transferhandling behavior</para>
</description>
</function>
<info name="CHANNEL" language="en_US" tech="PJSIP">
<enumlist>

12
channels/pjsip/include/dialplan_functions.h
View File

@@ -168,4 +168,16 @@ int pjsip_app_hangup(struct ast_channel *chan, const char *data);
*/
int pjsip_action_hangup(struct mansession *s, const struct message *m);
/*!
* \brief PJSIP_TRANSFER_HANDLING function write callback
* \param chan The channel the function is called on
* \param cmd the Name of the function
* \param data Arguments passed to the function
* \param value Value to be set by the function
*
* \retval 0 on success
* \retval -1 on failure
*/
int pjsip_transfer_handling_write(struct ast_channel *chan, const char *cmd, char *data, const char *value);
#endif /* _PJSIP_DIALPLAN_FUNCTIONS */