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Re-instate sip_addheader() while waiting for a dialplan function. (Issue 6317)

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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@8481 f38db490-d61c-443f-a65b-d21fe96a405b
This commit is contained in:
oej 2006-01-23 16:21:51 +00:00
parent 4f95d6a546
commit 16f2c3d62d
1 changed files with 47 additions and 0 deletions
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47
channels/chan_sip.c
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@ -12788,6 +12788,18 @@ 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 sip_dtmfmode: change the DTMFmode for a SIP call (application) */
static int sip_dtmfmode(struct ast_channel *chan, void *data)
{
@ -12838,6 +12850,39 @@ static int sip_dtmfmode(struct ast_channel *chan, void *data)
return 0;
}
/*! \brief sip_addheader: Add a SIP header */
static int sip_addheader(struct ast_channel *chan, void *data)
{
int no = 0;
int ok = 0;
char varbuf[30];
char *inbuf = (char *) data;
if (ast_strlen_zero(inbuf)) {
ast_log(LOG_WARNING, "This application requires the argument: Header\n");
return 0;
}
ast_mutex_lock(&chan->lock);
/* Check for headers */
while (!ok && no <= 50) {
no++;
snprintf(varbuf, sizeof(varbuf), "_SIPADDHEADER%.2d", no);
if( (pbx_builtin_getvar_helper(chan, (const char *) varbuf) == (const char *) NULL) )
ok = 1;
}
if (ok) {
pbx_builtin_setvar_helper (chan, varbuf, inbuf);
if (sipdebug)
ast_log(LOG_DEBUG,"SIP Header added \"%s\" as %s\n", inbuf, varbuf);
} else {
ast_log(LOG_WARNING, "Too many SIP headers added, max 50\n");
}
ast_mutex_unlock(&chan->lock);
return 0;
}
/*! \brief sip_sipredirect: Transfer call before connect with a 302 redirect */
/* Called by the transfer() dialplan application through the sip_transfer() */
/* pbx interface function if the call is in ringing state */
@ -13050,6 +13095,7 @@ int load_module()
/* 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);
/* Register dialplan functions */
ast_custom_function_register(&sip_header_function);
@ -13085,6 +13131,7 @@ int unload_module()
ast_custom_function_unregister(&checksipdomain_function);
ast_unregister_application(app_dtmfmode);
ast_unregister_application(app_sipaddheader);
ast_cli_unregister_multiple(my_clis, sizeof(my_clis) / sizeof(my_clis[0]));