Merge (for the time being) the alert code...
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@1107 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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@ -515,6 +515,25 @@ static int dial_exec(struct ast_channel *chan, void *data)
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}
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}
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}
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/* Check for ALERT_INFO in the SetVar list. This is for */
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/* SIP distinctive ring as per the RFC. For Cisco 7960s, */
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/* SetVar(ALERT_INFO=<x>) where x is an integer. However, */
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/* the RFC says it should be a URL. -- km- */
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if (strcasecmp(tech,"SIP")==0)
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{
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headp=&chan->varshead;
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AST_LIST_TRAVERSE(headp,current,entries) {
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/* Search for ALERT_INFO */
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if (strcasecmp(ast_var_name(current),"ALERT_INFO")==0)
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{
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newvar=ast_var_assign(ast_var_name(current),ast_var_value(current));
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newheadp=&tmp->chan->varshead;
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AST_LIST_INSERT_HEAD(newheadp,newvar,entries);
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break;
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}
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}
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}
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tmp->chan->appl = "AppDial";
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tmp->chan->data = "(Outgoing Line)";
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@ -365,7 +365,7 @@ static int transmit_response_with_sdp(struct sip_pvt *p, char *msg, struct sip_r
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static int transmit_response_with_auth(struct sip_pvt *p, char *msg, struct sip_request *req, char *rand, int reliable);
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static int transmit_request(struct sip_pvt *p, char *msg, int inc, int reliable);
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static int transmit_request_with_auth(struct sip_pvt *p, char *msg, int inc, int reliable);
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static int transmit_invite(struct sip_pvt *p, char *msg, int sendsdp, char *auth, char *vxml_url);
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static int transmit_invite(struct sip_pvt *p, char *msg, int sendsdp, char *auth, char *vxml_url,char *distinctive_ring);
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static int transmit_reinvite_with_sdp(struct sip_pvt *p, struct ast_rtp *rtp);
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static int transmit_info_with_digit(struct sip_pvt *p, char digit);
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static int transmit_message_with_text(struct sip_pvt *p, char *text);
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@ -774,6 +774,7 @@ static int sip_call(struct ast_channel *ast, char *dest, int timeout)
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int res;
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struct sip_pvt *p;
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char *vxml_url = NULL;
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char *distinctive_ring = NULL;
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struct varshead *headp;
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struct ast_var_t *current;
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@ -782,20 +783,27 @@ static int sip_call(struct ast_channel *ast, char *dest, int timeout)
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ast_log(LOG_WARNING, "sip_call called on %s, neither down nor reserved\n", ast->name);
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return -1;
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}
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/* Check whether there is vxml_url, distinctive ring variables */
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/* Check whether there is a VXML_URL variable */
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headp=&ast->varshead;
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AST_LIST_TRAVERSE(headp,current,entries) {
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/* Check whether there is a VXML_URL variable */
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if (strcasecmp(ast_var_name(current),"VXML_URL")==0)
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{
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vxml_url = ast_var_value(current);
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break;
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}
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/* Check whether there is a ALERT_INFO variable */
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if (strcasecmp(ast_var_name(current),"ALERT_INFO")==0)
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{
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distinctive_ring = ast_var_value(current);
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break;
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}
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}
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res = 0;
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p->outgoing = 1;
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transmit_invite(p, "INVITE", 1, NULL, vxml_url);
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transmit_invite(p, "INVITE", 1, NULL, vxml_url,distinctive_ring);
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if (p->maxtime) {
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/* Initialize auto-congest time */
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p->initid = ast_sched_add(sched, p->maxtime * 2, auto_congest, p);
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@ -2380,18 +2388,24 @@ static void initreqprep(struct sip_request *req, struct sip_pvt *p, char *cmd, c
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add_header(req, "User-Agent", "Asterisk PBX");
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}
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static int transmit_invite(struct sip_pvt *p, char *cmd, int sdp, char *auth, char *vxml_url)
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static int transmit_invite(struct sip_pvt *p, char *cmd, int sdp, char *auth, char *vxml_url, char *distinctive_ring)
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{
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struct sip_request req;
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initreqprep(&req, p, cmd, vxml_url);
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if (auth)
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add_header(&req, "Proxy-Authorization", auth);
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if (distinctive_ring)
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{
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add_header(&req, "Alert-info",distinctive_ring);
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}
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if (sdp) {
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add_sdp(&req, p, NULL);
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} else {
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add_header(&req, "Content-Length", "0");
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add_blank_header(&req);
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}
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if (!p->initreq.headers) {
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/* Use this as the basis */
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copy_request(&p->initreq, &req);
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@ -3761,7 +3775,7 @@ static int do_proxy_auth(struct sip_pvt *p, struct sip_request *req) {
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/* No way to authenticate */
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return -1;
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}
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return transmit_invite(p,"INVITE",1,digest, NULL);
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return transmit_invite(p,"INVITE",1,digest, NULL,NULL);
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}
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static int reply_digest(struct sip_pvt *p, struct sip_request *req, char *header, char *orig_header, char *digest, int digest_len) {
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@ -4830,9 +4844,9 @@ static int sip_poke_peer(struct sip_peer *peer)
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p->outgoing = 1;
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#ifdef VOCAL_DATA_HACK
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strncpy(p->username, "__VOCAL_DATA_SHOULD_READ_THE_SIP_SPEC__", sizeof(p->username));
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transmit_invite(p, "INVITE", 0, NULL, NULL);
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transmit_invite(p, "INVITE", 0, NULL, NULL,NULL);
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#else
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transmit_invite(p, "OPTIONS", 0, NULL, NULL);
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transmit_invite(p, "OPTIONS", 0, NULL, NULL,NULL);
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#endif
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gettimeofday(&peer->ps, NULL);
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peer->pokeexpire = ast_sched_add(sched, DEFAULT_MAXMS * 2, sip_poke_noanswer, peer);
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