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allow for custom URI options to be set (issue #4927 with mods)

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@6499 65c4cc65-6c06-0410-ace0-fbb531ad65f3
1.2-netsec
Kevin P. Fleming 20 years ago
parent 14c769230b
commit 6b76015469
2 changed files with 41 additions and 31 deletions
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69
channels/chan_sip.c
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@ -442,6 +442,7 @@ struct sip_invite_param {
char *distinctive_ring;
char *osptoken;
int addsipheaders;
char *uri_options;
char *vxml_url;
char *auth;
char *authheader;
@ -1941,6 +1942,8 @@ static int sip_call(struct ast_channel *ast, char *dest, int timeout)
/* Check whether there is a VXML_URL variable */
if (!p->options->vxml_url && !strcasecmp(ast_var_name(current), "VXML_URL")) {
p->options->vxml_url = ast_var_value(current);
} else if (!p->options->uri_options && !strcasecmp(ast_var_name(current), "SIP_URI_OPTIONS")) {
p->options->uri_options = ast_var_value(current);
} else if (!p->options->distinctive_ring && !strcasecmp(ast_var_name(current), "ALERT_INFO")) {
/* Check whether there is a ALERT_INFO variable */
p->options->distinctive_ring = ast_var_value(current);
@ -4388,9 +4391,11 @@ static void build_contact(struct sip_pvt *p)
}
/*--- initreqprep: Initiate new SIP request to peer/user ---*/
static void initreqprep(struct sip_request *req, struct sip_pvt *p, int sipmethod, char *vxml_url)
static void initreqprep(struct sip_request *req, struct sip_pvt *p, int sipmethod)
{
char invite[256] = "";
char invite_buf[256] = "";
char *invite = invite_buf;
size_t invite_max = sizeof(invite_buf);
char from[256];
char to[256];
char tmp[BUFSIZ/2];
@ -4460,41 +4465,45 @@ static void initreqprep(struct sip_request *req, struct sip_pvt *p, int sipmetho
}
if ((ourport != 5060) && ast_strlen_zero(p->fromdomain)) /* Needs to be 5060 */
snprintf(from, sizeof(from), "\"%s\" <sip:%s@%s:%d>;tag=as%08x", tmp, l, ast_strlen_zero(p->fromdomain) ? ast_inet_ntoa(iabuf, sizeof(iabuf), p->ourip) : p->fromdomain, ourport, p->tag);
snprintf(from, sizeof(from), "\"%s\" <sip:%s@%s:%d>;tag=as%08x", n, l, ast_strlen_zero(p->fromdomain) ? ast_inet_ntoa(iabuf, sizeof(iabuf), p->ourip) : p->fromdomain, ourport, p->tag);
else
snprintf(from, sizeof(from), "\"%s\" <sip:%s@%s>;tag=as%08x", n, l, ast_strlen_zero(p->fromdomain) ? ast_inet_ntoa(iabuf, sizeof(iabuf), p->ourip) : p->fromdomain, p->tag);
/* If we're calling a registred SIP peer, use the fullcontact to dial to the peer */
/* If we're calling a registered SIP peer, use the fullcontact to dial to the peer */
if (!ast_strlen_zero(p->fullcontact)) {
/* If we have full contact, trust it */
ast_copy_string(invite, p->fullcontact, sizeof(invite));
/* Otherwise, use the username while waiting for registration */
} else if (!ast_strlen_zero(p->username)) {
n = p->username;
if (pedanticsipchecking) {
ast_uri_encode(n, tmp, sizeof(tmp), 0);
n = tmp;
}
if (ntohs(p->sa.sin_port) != 5060) { /* Needs to be 5060 */
snprintf(invite, sizeof(invite), "sip:%s@%s:%d%s", n, p->tohost, ntohs(p->sa.sin_port), urioptions);
} else {
snprintf(invite, sizeof(invite), "sip:%s@%s%s", n, p->tohost, urioptions);
}
} else if (ntohs(p->sa.sin_port) != 5060) { /* Needs to be 5060 */
snprintf(invite, sizeof(invite), "sip:%s:%d%s", p->tohost, ntohs(p->sa.sin_port), urioptions);
ast_build_string(&invite, &invite_max, "%s", p->fullcontact);
} else {
snprintf(invite, sizeof(invite), "sip:%s%s", p->tohost, urioptions);
/* Otherwise, use the username while waiting for registration */
ast_build_string(&invite, &invite_max, "sip:");
if (!ast_strlen_zero(p->username)) {
n = p->username;
if (pedanticsipchecking) {
ast_uri_encode(n, tmp, sizeof(tmp), 0);
n = tmp;
}
ast_build_string(&invite, &invite_max, "%s@", n);
}
ast_build_string(&invite, &invite_max, "%s", p->tohost);
if (ntohs(p->sa.sin_port) != 5060) /* Needs to be 5060 */
ast_build_string(&invite, &invite_max, ":%d", ntohs(p->sa.sin_port));
ast_build_string(&invite, &invite_max, "%s", urioptions);
}
ast_copy_string(p->uri, invite, sizeof(p->uri));
/* If there is a VXML URL append it to the SIP URL */
if (vxml_url) {
snprintf(to, sizeof(to), "<%s>;%s", invite, vxml_url);
} else {
snprintf(to, sizeof(to), "<%s>", invite);
/* If custom URI options have been provided, append them */
if (p->options && p->options->uri_options)
ast_build_string(&invite, &invite_max, ";%s", p->options->uri_options);
ast_copy_string(p->uri, invite_buf, sizeof(p->uri));
/* If there is a VXML URL append it to the SIP URL */
if (p->options && p->options->vxml_url) {
snprintf(to, sizeof(to), "<%s>;%s", p->uri, p->options->vxml_url);
} else {
snprintf(to, sizeof(to), "<%s>", p->uri);
}
memset(req, 0, sizeof(struct sip_request));
init_req(req, sipmethod, invite);
init_req(req, sipmethod, p->uri);
snprintf(tmp, sizeof(tmp), "%d %s", ++p->ocseq, sip_methods[sipmethod].text);
add_header(req, "Via", p->via);
@ -4520,7 +4529,7 @@ static int transmit_invite(struct sip_pvt *p, int sipmethod, int sdp, int init)
/* Bump branch even on initial requests */
p->branch ^= rand();
build_via(p, p->via, sizeof(p->via));
initreqprep(&req, p, sipmethod, p->options ? p->options->vxml_url : (char *) NULL);
initreqprep(&req, p, sipmethod);
} else
reqprep(&req, p, sipmethod, 0, 1);
@ -4783,7 +4792,7 @@ static int transmit_notify_with_mwi(struct sip_pvt *p, int newmsgs, int oldmsgs,
char *t = tmp;
size_t maxbytes = sizeof(tmp);
initreqprep(&req, p, SIP_NOTIFY, NULL);
initreqprep(&req, p, SIP_NOTIFY);
add_header(&req, "Event", "message-summary");
add_header(&req, "Content-Type", default_notifymime);
@ -8272,7 +8281,7 @@ static int sip_notify(int fd, int argc, char *argv[])
continue;
}
initreqprep(&req, p, SIP_NOTIFY, NULL);
initreqprep(&req, p, SIP_NOTIFY);
for (var = varlist; var; var = var->next)
add_header(&req, var->name, var->value);

3
doc/README.variables
Unescape View File

@ -628,13 +628,14 @@ ${CALLINGSUBADDR} * Called PRI Subaddress
${FAXEXTEN} * The extension called before being redirected to "fax"
${PRIREDIRECTREASON} * Reason for redirect, if a call was directed
The SIP channel sets the following variables:
The SIP channel uses the following variables:
---------------------------------------------------------
${SIPCALLID} * SIP Call-ID: header verbatim (for logging or CDR matching)
${SIPDOMAIN} * SIP destination domain of an inbound call (if appropriate)
${SIPUSERAGENT} * SIP user agent
${SIPURI} * SIP uri
${SIP_CODEC} Set the SIP codec for a call
${SIP_URI_OPTIONS} * additional options to add to the URI for an outgoing call
The Agent channel uses the following variables:
---------------------------------------------------------

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