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Add XML documentation for: 

Applications
        - SIPDtmfMode()
        - SIPAddHeader()
     Functions
        - SIP_HEADER()
        - SIPPEER()
        - SIPCHANINFO()
        - CHECKSIPDOMAIN()



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@153803 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Eliel C. Sardanons 2008-11-03 12:35:05 +00:00
parent 191ffe5bce
commit 105a020b6f
1 changed files with 178 additions and 62 deletions
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240
channels/chan_sip.c
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@ -274,6 +274,182 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
#include "asterisk/event.h"
#include "asterisk/tcptls.h"
/*** DOCUMENTATION
<application name="SIPDtmfMode" language="en_US">
<synopsis>
Change the dtmfmode for a SIP call.
</synopsis>
<syntax>
<parameter name="mode" required="true">
<enumlist>
<enum name="inband" />
<enum name="info" />
<enum name="rfc2833" />
</enumlist>
</parameter>
</syntax>
<description>
<para>Changes the dtmfmode for a SIP call.</para>
</description>
</application>
<application name="SIPAddHeader" language="en_US">
<synopsis>
Add a SIP header to the outbound call.
</synopsis>
<syntax argsep=":">
<parameter name="Header" required="true" />
<parameter name="Content" required="true" />
</syntax>
<description>
<para>Adds a header to a SIP call placed with DIAL.</para>
<para>Remember to use the X-header if you are adding non-standard SIP
headers, like <literal>X-Asterisk-Accountcode:</literal>. Use this with care.
Adding the wrong headers may jeopardize the SIP dialog.</para>
<para>Always returns <literal>0</literal>.</para>
</description>
</application>
<function name="SIP_HEADER" language="en_US">
<synopsis>
Gets the specified SIP header.
</synopsis>
<syntax>
<parameter name="name" required="true" />
<parameter name="number">
<para>If not specified, defaults to <literal>1</literal>.</para>
</parameter>
</syntax>
<description>
<para>Since there are several headers (such as Via) which can occur multiple
times, SIP_HEADER takes an optional second argument to specify which header with
that name to retrieve. Headers start at offset <literal>1</literal>.</para>
</description>
</function>
<function name="SIPPEER" language="en_US">
<synopsis>
Gets SIP peer information.
</synopsis>
<syntax>
<parameter name="peername" required="true" />
<parameter name="item">
<enumlist>
<enum name="ip">
<para>(default) The ip address.</para>
</enum>
<enum name="port">
<para>The port number.</para>
</enum>
<enum name="mailbox">
<para>The configured mailbox.</para>
</enum>
<enum name="context">
<para>The configured context.</para>
</enum>
<enum name="expire">
<para>The epoch time of the next expire.</para>
</enum>
<enum name="dynamic">
<para>Is it dynamic? (yes/no).</para>
</enum>
<enum name="callerid_name">
<para>The configured Caller ID name.</para>
</enum>
<enum name="callerid_num">
<para>The configured Caller ID number.</para>
</enum>
<enum name="callgroup">
<para>The configured Callgroup.</para>
</enum>
<enum name="pickupgroup">
<para>The configured Pickupgroup.</para>
</enum>
<enum name="codecs">
<para>The configured codecs.</para>
</enum>
<enum name="status">
<para>Status (if qualify=yes).</para>
</enum>
<enum name="regexten">
<para>Registration extension.</para>
</enum>
<enum name="limit">
<para>Call limit (call-limit).</para>
</enum>
<enum name="busylevel">
<para>Configured call level for signalling busy.</para>
</enum>
<enum name="curcalls">
<para>Current amount of calls. Only available if call-limit is set.</para>
</enum>
<enum name="language">
<para>Default language for peer.</para>
</enum>
<enum name="accountcode">
<para>Account code for this peer.</para>
</enum>
<enum name="useragent">
<para>Current user agent id for peer.</para>
</enum>
<enum name="chanvar[name]">
<para>A channel variable configured with setvar for this peer.</para>
</enum>
<enum name="codec[x]">
<para>Preferred codec index number <replaceable>x</replaceable> (beginning with zero).</para>
</enum>
</enumlist>
</parameter>
</syntax>
<description />
</function>
<function name="SIPCHANINFO" language="en_US">
<synopsis>
Gets the specified SIP parameter from the current channel.
</synopsis>
<syntax>
<parameter name="item" required="true">
<enumlist>
<enum name="peerip">
<para>The IP address of the peer.</para>
</enum>
<enum name="recvip">
<para>The source IP address of the peer.</para>
</enum>
<enum name="from">
<para>The URI from the <literal>From:</literal> header.</para>
</enum>
<enum name="uri">
<para>The URI from the <literal>Contact:</literal> header.</para>
</enum>
<enum name="useragent">
<para>The useragent.</para>
</enum>
<enum name="peername">
<para>The name of the peer.</para>
</enum>
<enum name="t38passthrough">
<para><literal>1</literal> if T38 is offered or enabled in this channel,
otherwise <literal>0</literal>.</para>
</enum>
</enumlist>
</parameter>
</syntax>
<description />
</function>
<function name="CHECKSIPDOMAIN" language="en_US">
<synopsis>
Checks if domain is a local domain.
</synopsis>
<syntax>
<parameter name="domain" required="true" />
</syntax>
<description>
<para>This function checks if the <replaceable>domain</replaceable> in the argument is configured
as a local SIP domain that this Asterisk server is configured to handle.
Returns the domain name if it is locally handled, otherwise an empty string.
Check the <literal>domain=</literal> configuration in <filename>sip.conf</filename>.</para>
</description>
</function>
***/
#ifndef FALSE
#define FALSE 0
#endif
@ -15473,11 +15649,6 @@ static int func_header_read(struct ast_channel *chan, const char *function, char
static struct ast_custom_function sip_header_function = {
.name = "SIP_HEADER",
.synopsis = "Gets the specified SIP header",
.syntax = "SIP_HEADER(<name>[,<number>])",
.desc = "Since there are several headers (such as Via) which can occur multiple\n"
"times, SIP_HEADER takes an optional second argument to specify which header with\n"
"that name to retrieve. Headers start at offset 1.\n",
.read = func_header_read,
};
@ -15497,13 +15668,7 @@ static int func_check_sipdomain(struct ast_channel *chan, const char *cmd, char
static struct ast_custom_function checksipdomain_function = {
.name = "CHECKSIPDOMAIN",
.synopsis = "Checks if domain is a local domain",
.syntax = "CHECKSIPDOMAIN(<domain|IP>)",
.read = func_check_sipdomain,
.desc = "This function checks if the domain in the argument is configured\n"
"as a local SIP domain that this Asterisk server is configured to handle.\n"
"Returns the domain name if it is locally handled, otherwise an empty string.\n"
"Check the domain= configuration in sip.conf\n",
};
/*! \brief ${SIPPEER()} Dialplan function - reads peer data */
@ -15596,33 +15761,7 @@ static int function_sippeer(struct ast_channel *chan, const char *cmd, char *dat
/*! \brief Structure to declare a dialplan function: SIPPEER */
static struct ast_custom_function sippeer_function = {
.name = "SIPPEER",
.synopsis = "Gets SIP peer information",
.syntax = "SIPPEER(<peername>[,item])",
.read = function_sippeer,
.desc = "Valid items are:\n"
"- ip (default) The IP address.\n"
"- port The port number\n"
"- mailbox The configured mailbox.\n"
"- context The configured context.\n"
"- expire The epoch time of the next expire.\n"
"- dynamic Is it dynamic? (yes/no).\n"
"- callerid_name The configured Caller ID name.\n"
"- callerid_num The configured Caller ID number.\n"
"- callgroup The configured Callgroup.\n"
"- pickupgroup The configured Pickupgroup.\n"
"- codecs The configured codecs.\n"
"- status Status (if qualify=yes).\n"
"- regexten Registration extension\n"
"- limit Call limit (call-limit)\n"
"- busylevel Configured call level for signalling busy\n"
"- curcalls Current amount of calls \n"
" Only available if call-limit is set\n"
"- language Default language for peer\n"
"- accountcode Account code for this peer\n"
"- useragent Current user agent id for peer\n"
"- chanvar[name] A channel variable configured with setvar for this peer.\n"
"- codec[x] Preferred codec index number 'x' (beginning with zero).\n"
"\n"
};
/*! \brief ${SIPCHANINFO()} Dialplan function - reads sip channel data */
@ -15687,17 +15826,7 @@ static int function_sipchaninfo_read(struct ast_channel *chan, const char *cmd,
/*! \brief Structure to declare a dialplan function: SIPCHANINFO */
static struct ast_custom_function sipchaninfo_function = {
.name = "SIPCHANINFO",
.synopsis = "Gets the specified SIP parameter from the current channel",
.syntax = "SIPCHANINFO(item)",
.read = function_sipchaninfo_read,
.desc = "Valid items are:\n"
"- peerip The IP address of the peer.\n"
"- recvip The source IP address of the peer.\n"
"- from The URI from the From: header.\n"
"- uri The URI from the Contact: header.\n"
"- useragent The useragent.\n"
"- peername The name of the peer.\n"
"- t38passthrough 1 if T38 is offered or enabled in this channel, otherwise 0\n"
};
/*! \brief Parse 302 Moved temporalily response
@ -23248,21 +23377,8 @@ static int sip_set_rtp_peer(struct ast_channel *chan, struct ast_rtp *rtp, struc
return 0;
}
static char *synopsis_dtmfmode = "Change the dtmfmode for a SIP call";
static char *descrip_dtmfmode = " SIPDtmfMode(inband|info|rfc2833): Changes the dtmfmode for a SIP call\n";
static char *app_dtmfmode = "SIPDtmfMode";
static char *app_sipaddheader = "SIPAddHeader";
static char *synopsis_sipaddheader = "Add a SIP header to the outbound call";
static char *descrip_sipaddheader = ""
" SIPAddHeader(Header: Content):\n"
"Adds a header to a SIP call placed with DIAL.\n"
"Remember to user the X-header if you are adding non-standard SIP\n"
"headers, like \"X-Asterisk-Accountcode:\". Use this with care.\n"
"Adding the wrong headers may jeopardize the SIP dialog.\n"
"Always returns 0\n";
/*! \brief Set the DTMFmode for an outbound SIP call (application) */
static int sip_dtmfmode(struct ast_channel *chan, void *data)
@ -23633,8 +23749,8 @@ static int load_module(void)
ast_udptl_proto_register(&sip_udptl);
/* Register dialplan applications */
ast_register_application(app_dtmfmode, sip_dtmfmode, synopsis_dtmfmode, descrip_dtmfmode);
ast_register_application(app_sipaddheader, sip_addheader, synopsis_sipaddheader, descrip_sipaddheader);
ast_register_application_xml(app_dtmfmode, sip_dtmfmode);
ast_register_application_xml(app_sipaddheader, sip_addheader);
/* Register dialplan functions */
ast_custom_function_register(&sip_header_function);