allow for custom URI options to be set (issue #4927 with mods)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@6499 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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@ -442,6 +442,7 @@ struct sip_invite_param {
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char *distinctive_ring;
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char *osptoken;
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int addsipheaders;
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char *uri_options;
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char *vxml_url;
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char *auth;
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char *authheader;
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@ -1941,6 +1942,8 @@ static int sip_call(struct ast_channel *ast, char *dest, int timeout)
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/* Check whether there is a VXML_URL variable */
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if (!p->options->vxml_url && !strcasecmp(ast_var_name(current), "VXML_URL")) {
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p->options->vxml_url = ast_var_value(current);
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} else if (!p->options->uri_options && !strcasecmp(ast_var_name(current), "SIP_URI_OPTIONS")) {
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p->options->uri_options = ast_var_value(current);
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} else if (!p->options->distinctive_ring && !strcasecmp(ast_var_name(current), "ALERT_INFO")) {
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/* Check whether there is a ALERT_INFO variable */
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p->options->distinctive_ring = ast_var_value(current);
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@ -4388,9 +4391,11 @@ static void build_contact(struct sip_pvt *p)
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}
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/*--- initreqprep: Initiate new SIP request to peer/user ---*/
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static void initreqprep(struct sip_request *req, struct sip_pvt *p, int sipmethod, char *vxml_url)
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static void initreqprep(struct sip_request *req, struct sip_pvt *p, int sipmethod)
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{
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char invite[256] = "";
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char invite_buf[256] = "";
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char *invite = invite_buf;
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size_t invite_max = sizeof(invite_buf);
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char from[256];
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char to[256];
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char tmp[BUFSIZ/2];
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@ -4460,41 +4465,45 @@ static void initreqprep(struct sip_request *req, struct sip_pvt *p, int sipmetho
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}
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if ((ourport != 5060) && ast_strlen_zero(p->fromdomain)) /* Needs to be 5060 */
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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);
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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);
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else
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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);
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/* If we're calling a registred SIP peer, use the fullcontact to dial to the peer */
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/* If we're calling a registered SIP peer, use the fullcontact to dial to the peer */
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if (!ast_strlen_zero(p->fullcontact)) {
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/* If we have full contact, trust it */
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ast_copy_string(invite, p->fullcontact, sizeof(invite));
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/* Otherwise, use the username while waiting for registration */
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} else if (!ast_strlen_zero(p->username)) {
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n = p->username;
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if (pedanticsipchecking) {
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ast_uri_encode(n, tmp, sizeof(tmp), 0);
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n = tmp;
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}
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if (ntohs(p->sa.sin_port) != 5060) { /* Needs to be 5060 */
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snprintf(invite, sizeof(invite), "sip:%s@%s:%d%s", n, p->tohost, ntohs(p->sa.sin_port), urioptions);
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} else {
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snprintf(invite, sizeof(invite), "sip:%s@%s%s", n, p->tohost, urioptions);
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}
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} else if (ntohs(p->sa.sin_port) != 5060) { /* Needs to be 5060 */
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snprintf(invite, sizeof(invite), "sip:%s:%d%s", p->tohost, ntohs(p->sa.sin_port), urioptions);
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ast_build_string(&invite, &invite_max, "%s", p->fullcontact);
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} else {
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snprintf(invite, sizeof(invite), "sip:%s%s", p->tohost, urioptions);
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/* Otherwise, use the username while waiting for registration */
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ast_build_string(&invite, &invite_max, "sip:");
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if (!ast_strlen_zero(p->username)) {
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n = p->username;
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if (pedanticsipchecking) {
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ast_uri_encode(n, tmp, sizeof(tmp), 0);
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n = tmp;
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}
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ast_build_string(&invite, &invite_max, "%s@", n);
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}
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ast_build_string(&invite, &invite_max, "%s", p->tohost);
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if (ntohs(p->sa.sin_port) != 5060) /* Needs to be 5060 */
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ast_build_string(&invite, &invite_max, ":%d", ntohs(p->sa.sin_port));
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ast_build_string(&invite, &invite_max, "%s", urioptions);
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}
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ast_copy_string(p->uri, invite, sizeof(p->uri));
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/* If there is a VXML URL append it to the SIP URL */
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if (vxml_url) {
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snprintf(to, sizeof(to), "<%s>;%s", invite, vxml_url);
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} else {
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snprintf(to, sizeof(to), "<%s>", invite);
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/* If custom URI options have been provided, append them */
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if (p->options && p->options->uri_options)
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ast_build_string(&invite, &invite_max, ";%s", p->options->uri_options);
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ast_copy_string(p->uri, invite_buf, sizeof(p->uri));
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/* If there is a VXML URL append it to the SIP URL */
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if (p->options && p->options->vxml_url) {
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snprintf(to, sizeof(to), "<%s>;%s", p->uri, p->options->vxml_url);
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} else {
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snprintf(to, sizeof(to), "<%s>", p->uri);
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}
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memset(req, 0, sizeof(struct sip_request));
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init_req(req, sipmethod, invite);
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init_req(req, sipmethod, p->uri);
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snprintf(tmp, sizeof(tmp), "%d %s", ++p->ocseq, sip_methods[sipmethod].text);
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add_header(req, "Via", p->via);
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@ -4520,7 +4529,7 @@ static int transmit_invite(struct sip_pvt *p, int sipmethod, int sdp, int init)
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/* Bump branch even on initial requests */
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p->branch ^= rand();
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build_via(p, p->via, sizeof(p->via));
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initreqprep(&req, p, sipmethod, p->options ? p->options->vxml_url : (char *) NULL);
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initreqprep(&req, p, sipmethod);
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} else
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reqprep(&req, p, sipmethod, 0, 1);
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@ -4783,7 +4792,7 @@ static int transmit_notify_with_mwi(struct sip_pvt *p, int newmsgs, int oldmsgs,
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char *t = tmp;
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size_t maxbytes = sizeof(tmp);
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initreqprep(&req, p, SIP_NOTIFY, NULL);
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initreqprep(&req, p, SIP_NOTIFY);
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add_header(&req, "Event", "message-summary");
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add_header(&req, "Content-Type", default_notifymime);
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@ -8272,7 +8281,7 @@ static int sip_notify(int fd, int argc, char *argv[])
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continue;
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}
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initreqprep(&req, p, SIP_NOTIFY, NULL);
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initreqprep(&req, p, SIP_NOTIFY);
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for (var = varlist; var; var = var->next)
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add_header(&req, var->name, var->value);
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@ -628,13 +628,14 @@ ${CALLINGSUBADDR} * Called PRI Subaddress
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${FAXEXTEN} * The extension called before being redirected to "fax"
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${PRIREDIRECTREASON} * Reason for redirect, if a call was directed
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The SIP channel sets the following variables:
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The SIP channel uses the following variables:
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---------------------------------------------------------
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${SIPCALLID} * SIP Call-ID: header verbatim (for logging or CDR matching)
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${SIPDOMAIN} * SIP destination domain of an inbound call (if appropriate)
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${SIPUSERAGENT} * SIP user agent
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${SIPURI} * SIP uri
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${SIP_CODEC} Set the SIP codec for a call
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${SIP_URI_OPTIONS} * additional options to add to the URI for an outgoing call
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The Agent channel uses the following variables:
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---------------------------------------------------------
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