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

(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

diff --git a/channels/chan_sip.c b/channels/chan_sip.c
index 1b7d355ffd42d6340296e673dd9e6eb12d4da2f0..f571a20bd6ec3fee357075c4eb8655ae56a0b969 100644 (file)
--- a/channels/chan_sip.c
+++ b/channels/chan_sip.c
@@ -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));