[asterisk-commits] trunk r8728 - /trunk/channels/chan_sip.c
asterisk-commits at lists.digium.com
asterisk-commits at lists.digium.com
Thu Jan 26 12:38:13 MST 2006
Author: oej
Date: Thu Jan 26 13:38:11 2006
New Revision: 8728
URL: http://svn.digium.com/view/asterisk?rev=8728&view=rev
Log:
Code clean up, inspired by rizzo's comments in issue 5978.
- Don't check for ignore if ignore is always negative
- Add comments to explain what's going on
Modified:
trunk/channels/chan_sip.c
Modified: trunk/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/trunk/channels/chan_sip.c?rev=8728&r1=8727&r2=8728&view=diff
==============================================================================
--- trunk/channels/chan_sip.c (original)
+++ trunk/channels/chan_sip.c Thu Jan 26 13:38:11 2006
@@ -7036,9 +7036,12 @@
}
-/*! \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 @@
} 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 @@
/* 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 @@
}
} 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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