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asterisk
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Code Issues Releases Wiki Activity

Code clean up, inspired by rizzo's comments in issue 5978.

Browse Source 
- Don't check for ignore if ignore is always negative
- Add comments to explain what's going on


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@8728 65c4cc65-6c06-0410-ace0-fbb531ad65f3
1.4
Olle Johansson 20 years ago
parent 7707b953e1
commit db97a8d9ce
1 changed files with 39 additions and 38 deletions
Show Stats Download Patch File Download Diff File

77
channels/chan_sip.c
Unescape View File

@ -7036,9 +7036,12 @@ static int get_rpid_num(char *input,char *output, int maxlen)
}
/*! \brief Check if matching user or peer is defined */
/* Match user on From: user name and peer on IP/port */
/* This is used on first invite (not re-invites) and subscribe requests */
/*! \brief Check if matching user or peer is defined
Match user on From: user name and peer on IP/port
This is used on first invite (not re-invites) and subscribe requests
\return 0 on success, -1 on failure, and 1 on challenge sent
-2 on authentication error from chedck_auth()
*/
static int check_user_full(struct sip_pvt *p, struct sip_request *req, int sipmethod, char *uri, int reliable, struct sockaddr_in *sin, int ignore, char *mailbox, int mailboxlen)
{
struct sip_user *user = NULL;
@ -10411,70 +10414,64 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, int
} else if (debug)
ast_verbose("Ignoring this INVITE request\n");
if (!p->lastinvite && !ignore && !p->owner) {
/* Handle authentication if this is our first invite */
res = check_user(p, req, SIP_INVITE, e, 1, sin, ignore);
if (res) {
if (res < 0) {
ast_log(LOG_NOTICE, "Failed to authenticate user %s\n", get_header(req, "From"));
if (ignore)
transmit_response(p, "403 Forbidden", req);
else
transmit_response_reliable(p, "403 Forbidden", req, 1);
ast_set_flag(p, SIP_NEEDDESTROY);
ast_string_field_free(p, theirtag);
}
if (res > 0) /* We have challenged the user for auth */
return 0;
if (res < 0) { /* Something failed in authentication */
ast_log(LOG_NOTICE, "Failed to authenticate user %s\n", get_header(req, "From"));
transmit_response_reliable(p, "403 Forbidden", req, 1);
ast_set_flag(p, SIP_NEEDDESTROY);
ast_string_field_free(p, theirtag);
return 0;
}
/* Process the SDP portion */
/* We have a succesful authentication, process the SDP portion if there is one */
if (!ast_strlen_zero(get_header(req, "Content-Type"))) {
if (process_sdp(p, req)) {
transmit_response(p, "488 Not acceptable here", req);
/* Unacceptable codecs */
transmit_response_reliable(p, "488 Not acceptable here", req, 1);
ast_set_flag(p, SIP_NEEDDESTROY);
return -1;
}
} else {
p->jointcapability = p->capability;
ast_log(LOG_DEBUG, "Hm.... No sdp for the moment\n");
if (option_debug > 1)
ast_log(LOG_DEBUG, "No SDP in Invite, third party call control\n");
}
/* Queue NULL frame to prod ast_rtp_bridge if appropriate */
if (p->owner)
ast_queue_frame(p->owner, &af);
/* Initialize the context if it hasn't been already */
if (ast_strlen_zero(p->context))
ast_string_field_set(p, context, default_context);
/* Check number of concurrent calls -vs- incoming limit HERE */
ast_log(LOG_DEBUG, "Checking SIP call limits for device %s\n", p->username);
if (option_debug)
ast_log(LOG_DEBUG, "Checking SIP call limits for device %s\n", p->username);
res = update_call_counter(p, INC_CALL_LIMIT);
if (res) {
if (res < 0) {
ast_log(LOG_NOTICE, "Failed to place call for user %s, too many calls\n", p->username);
if (ignore)
transmit_response(p, "480 Temporarily Unavailable (Call limit)", req);
else
transmit_response_reliable(p, "480 Temporarily Unavailable (Call limit) ", req, 1);
transmit_response_reliable(p, "480 Temporarily Unavailable (Call limit) ", req, 1);
ast_set_flag(p, SIP_NEEDDESTROY);
}
return 0;
}
/* Get destination right away */
gotdest = get_destination(p, NULL);
get_rdnis(p, NULL);
extract_uri(p, req);
build_contact(p);
gotdest = get_destination(p, NULL); /* Get destination right away */
get_rdnis(p, NULL); /* Get redirect information */
extract_uri(p, req); /* Get the Contact URI */
build_contact(p); /* Build our contact header */
if (gotdest) {
if (gotdest < 0) {
if (ignore)
transmit_response(p, "404 Not Found", req);
else
transmit_response_reliable(p, "404 Not Found", req, 1);
transmit_response_reliable(p, "404 Not Found", req, 1);
update_call_counter(p, DEC_CALL_LIMIT);
} else {
if (ignore)
transmit_response(p, "484 Address Incomplete", req);
else
transmit_response_reliable(p, "484 Address Incomplete", req, 1);
transmit_response_reliable(p, "484 Address Incomplete", req, 1);
update_call_counter(p, DEC_CALL_LIMIT);
}
ast_set_flag(p, SIP_NEEDDESTROY);
@ -10482,7 +10479,7 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, int
/* If no extension was specified, use the s one */
if (ast_strlen_zero(p->exten))
ast_string_field_set(p, exten, "s");
/* Initialize tag */
/* Initialize our tag */
make_our_tag(p->tag, sizeof(p->tag));
/* First invitation */
c = sip_new(p, AST_STATE_DOWN, ast_strlen_zero(p->username) ? NULL : p->username);
@ -10496,8 +10493,12 @@ static int handle_request_invite(struct sip_pvt *p, struct sip_request *req, int
}
} else {
if (option_debug > 1 && sipdebug)
ast_log(LOG_DEBUG, "Got a SIP re-invite for call %s\n", p->callid);
if (option_debug > 1 && sipdebug) {
if (!ignore)
ast_log(LOG_DEBUG, "Got a SIP re-invite for call %s\n", p->callid);
else
ast_log(LOG_DEBUG, "Got a SIP re-transmit of INVITE for call %s\n", p->callid);
}
c = p->owner;
}
if (!ignore && p)

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