kenmirrors/asterisk
1
0
Fork 0
mirror of https://github.com/asterisk/asterisk.git synced 2025-10-26 22:30:28 +00:00
Code Issues Packages Projects Releases Wiki Activity

Add 416 response to OPTIONS packet.

Browse Source 
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/branches/1.8@312889 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This commit is contained in:
Richard Mudgett
2011-04-05 16:19:35 +00:00
parent 0acdb60dbd
commit ddc3fac28b
1 changed files with 23 additions and 8 deletions
Download Patch File Download Diff File Expand all files Collapse all files

31
channels/chan_sip.c
View File

@@ -20700,6 +20700,8 @@ 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) {
@@ -20731,24 +20733,37 @@ 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,