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Merged revisions 312889 via svnmerge from 

https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r312889 | rmudgett | 2011-04-05 11:19:35 -0500 (Tue, 05 Apr 2011) | 5 lines
  
  Add 416 response to OPTIONS packet.
  
  RFC3261 Section 11.2 says the response code to an OPTIONS packet needs to
  be the same as if it were an INVITE.
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@312897 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Richard Mudgett 2011-04-05 16:21:28 +00:00
parent e005f07b7d
commit ad30fa7569
1 changed files with 23 additions and 8 deletions
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31
channels/chan_sip.c
View File

@ -21015,6 +21015,8 @@ static int handle_request_notify(struct sip_pvt *p, struct sip_request *req, str
*/
static int handle_request_options(struct sip_pvt *p, struct sip_request *req, struct ast_sockaddr *addr, const char *e)
{
const char *msg;
enum sip_get_dest_result gotdest;
int res;
if (p->lastinvite) {
@ -21046,24 +21048,37 @@ static int handle_request_options(struct sip_pvt *p, struct sip_request *req, st
}
/* must go through authentication before getting here */
res = (get_destination(p, req, NULL) == SIP_GET_DEST_EXTEN_FOUND ? 0 : -1);
gotdest = get_destination(p, req, NULL);
build_contact(p);
if (ast_strlen_zero(p->context))
ast_string_field_set(p, context, sip_cfg.default_context);
if (ast_shutting_down())
transmit_response_with_allow(p, "503 Unavailable", req, 0);
else if (res < 0)
transmit_response_with_allow(p, "404 Not Found", req, 0);
else
transmit_response_with_allow(p, "200 OK", req, 0);
if (ast_shutting_down()) {
msg = "503 Unavailable";
} else {
msg = "404 Not Found";
switch (gotdest) {
case SIP_GET_DEST_INVALID_URI:
msg = "416 Unsupported URI scheme";
break;
case SIP_GET_DEST_PICKUP_EXTEN_FOUND:
case SIP_GET_DEST_REFUSED:
case SIP_GET_DEST_EXTEN_NOT_FOUND:
//msg = "404 Not Found";
break;
case SIP_GET_DEST_EXTEN_FOUND:
msg = "200 OK";
break;
}
}
transmit_response_with_allow(p, msg, req, 0);
/* Destroy if this OPTIONS was the opening request, but not if
it's in the middle of a normal call flow. */
sip_scheddestroy(p, DEFAULT_TRANS_TIMEOUT);
return res;
return 0;
}
/*! \brief Handle the transfer part of INVITE with a replaces: header,