diff options
author | eliel <eliel@f38db490-d61c-443f-a65b-d21fe96a405b> | 2008-11-03 12:35:05 +0000 |
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committer | eliel <eliel@f38db490-d61c-443f-a65b-d21fe96a405b> | 2008-11-03 12:35:05 +0000 |
commit | a58b020af32aa47f926e9651ba193b495b7690f4 (patch) | |
tree | 82d52c4dff424518af72316b47af8fd793ce16dc /channels | |
parent | da2d5b408c1975acd1d3dd39d5e0820343a9043b (diff) |
Add XML documentation for:
Applications
- SIPDtmfMode()
- SIPAddHeader()
Functions
- SIP_HEADER()
- SIPPEER()
- SIPCHANINFO()
- CHECKSIPDOMAIN()
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@153803 f38db490-d61c-443f-a65b-d21fe96a405b
Diffstat (limited to 'channels')
-rw-r--r-- | channels/chan_sip.c | 240 |
1 files changed, 178 insertions, 62 deletions
diff --git a/channels/chan_sip.c b/channels/chan_sip.c index d1cb709fd..0177fbbca 100644 --- a/channels/chan_sip.c +++ b/channels/chan_sip.c @@ -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); |