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Add capability to remove added SIP headers *before* INVITE is generated.

Browse Source 
(closes issue #14246)
Reported by: klaus3000
Patches: 
      2patch_chan_sip_SIPRemoveHeader_trunk.txt uploaded by klaus3000 (license 65)



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@168638 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Olle Johansson
2009-01-15 13:35:50 +00:00
parent 526cc089a9
commit 27c1d88021
1 changed files with 61 additions and 0 deletions
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channels/chan_sip.c
+61
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@@ -301,6 +301,32 @@ ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
<para>Always returns <literal>0</literal>.</para>
</description>
</application>
<application name="SIPRemoveHeader" language="en_US">
<synopsis>
Remove SIP headers previously added with SIPAddHeader
</synopsis>
<syntax>
<parameter name="Header" required="false" />
</syntax>
<description>
<para>SIPRemoveHeader() allows you to remove headers which were previously
added with SIPAddHeader(). If no parameter is supplied, all previously added
headers will be removed. If a parameter is supplied, only the matching headers
will be removed.</para>
<para>For example you have added these 2 headers:</para>
<para>SIPAddHeader(P-Asserted-Identity: sip:foo@bar);</para>
<para>SIPAddHeader(P-Preferred-Identity: sip:bar@foo);</para>
<para></para>
<para>// remove all headers</para>
<para>SIPRemoveHeader();</para>
<para>// remove all P- headers</para>
<para>SIPRemoveHeader(P-);</para>
<para>// remove only the PAI header (note the : at the end)</para>
<para>SIPRemoveHeader(P-Asserted-Identity:);</para>
<para></para>
<para>Always returns <literal>0</literal>.</para>
</description>
</application>
<function name="SIP_HEADER" language="en_US">
<synopsis>
Gets the specified SIP header.
@@ -23901,6 +23927,7 @@ static int sip_set_rtp_peer(struct ast_channel *chan, struct ast_rtp *rtp, struc
static char *app_dtmfmode = "SIPDtmfMode";
static char *app_sipaddheader = "SIPAddHeader";
static char *app_sipremoveheader = "SIPRemoveHeader";
/*! \brief Set the DTMFmode for an outbound SIP call (application) */
static int sip_dtmfmode(struct ast_channel *chan, void *data)
@@ -23999,6 +24026,38 @@ static int sip_addheader(struct ast_channel *chan, void *data)
return 0;
}
/*! \brief Remove SIP headers added previously with SipAddHeader application */
static int sip_removeheader(struct ast_channel *chan, void *data)
{
struct ast_var_t *newvariable;
struct varshead *headp;
int removeall = 0;
char *inbuf = (char *) data;
if (ast_strlen_zero(inbuf)) {
removeall = 1;
}
ast_channel_lock(chan);
headp=&chan->varshead;
AST_LIST_TRAVERSE_SAFE_BEGIN (headp, newvariable, entries) {
if (strncasecmp(ast_var_name(newvariable), "SIPADDHEADER", strlen("SIPADDHEADER")) == 0) {
if (removeall || (!strncasecmp(ast_var_value(newvariable),inbuf,strlen(inbuf)))) {
if (sipdebug)
ast_log(LOG_DEBUG,"removing SIP Header \"%s\" as %s\n",
ast_var_value(newvariable),
ast_var_name(newvariable));
AST_LIST_REMOVE_CURRENT(entries);
ast_var_delete(newvariable);
}
}
}
AST_LIST_TRAVERSE_SAFE_END;
ast_channel_unlock(chan);
return 0;
}
/*! \brief Transfer call before connect with a 302 redirect
\note Called by the transfer() dialplan application through the sip_transfer()
pbx interface function if the call is in ringing state
@@ -24282,6 +24341,7 @@ static int load_module(void)
/* Register dialplan applications */
ast_register_application_xml(app_dtmfmode, sip_dtmfmode);
ast_register_application_xml(app_sipaddheader, sip_addheader);
ast_register_application_xml(app_sipremoveheader, sip_removeheader);
/* Register dialplan functions */
ast_custom_function_register(&sip_header_function);
@@ -24343,6 +24403,7 @@ static int unload_module(void)
/* Unregister dial plan applications */
ast_unregister_application(app_dtmfmode);
ast_unregister_application(app_sipaddheader);
ast_unregister_application(app_sipremoveheader);
/* Unregister CLI commands */
ast_cli_unregister_multiple(cli_sip, ARRAY_LEN(cli_sip));