[asterisk-commits] eliel: trunk r153803 - /trunk/channels/chan_sip.c
SVN commits to the Asterisk project
asterisk-commits at lists.digium.com
Mon Nov 3 06:35:08 CST 2008
Author: eliel
Date: Mon Nov 3 06:35:05 2008
New Revision: 153803
URL: http://svn.digium.com/view/asterisk?view=rev&rev=153803
Log:
Add XML documentation for:
Applications
- SIPDtmfMode()
- SIPAddHeader()
Functions
- SIP_HEADER()
- SIPPEER()
- SIPCHANINFO()
- CHECKSIPDOMAIN()
Modified:
trunk/channels/chan_sip.c
Modified: trunk/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/trunk/channels/chan_sip.c?view=diff&rev=153803&r1=153802&r2=153803
==============================================================================
--- trunk/channels/chan_sip.c (original)
+++ trunk/channels/chan_sip.c Mon Nov 3 06:35:05 2008
@@ -273,6 +273,182 @@
#include "asterisk/ast_version.h"
#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
@@ -15473,11 +15649,6 @@
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 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 @@
/*! \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 @@
/*! \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 @@
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 @@
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);
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