mirror of
https://github.com/asterisk/asterisk.git
synced 2025-09-03 11:25:35 +00:00
channels/pjsip/dialplan_functions: Add an option for extracting the SIP call-id
This patch adds a new option to the CHANNEL function that allows for the extraction of the SIP call-id. It is used in conjunction with the 'pjsip' option, and will return the Call-ID of the INVITE request that established the PJSIP channel. ASTERISK-25352 Change-Id: I278d1f8bcfe3a53c5aa1dadebc14e92b0abd476a
This commit is contained in:
@@ -318,6 +318,9 @@
|
||||
<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>
|
||||
@@ -594,6 +597,8 @@ static int channel_read_pjsip(struct ast_channel *chan, const char *type, const
|
||||
if (ast_strlen_zero(type)) {
|
||||
ast_log(LOG_WARNING, "You must supply a type field for 'pjsip' information\n");
|
||||
return -1;
|
||||
} else if (!strcmp(type, "call-id")) {
|
||||
snprintf(buf, buflen, "%.*s", (int) pj_strlen(&dlg->call_id->id), pj_strbuf(&dlg->call_id->id));
|
||||
} else if (!strcmp(type, "secure")) {
|
||||
#ifdef HAVE_PJSIP_GET_DEST_INFO
|
||||
pjsip_host_info dest;
|
||||
|
Reference in New Issue
Block a user