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Merged revisions 113118 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r113118 | qwell | 2008-04-07 13:00:09 -0500 (Mon, 07 Apr 2008) | 8 lines Allow playback with noanswer (and add earlyrtp option). (closes issue #9077) Reported by: pj Patches: earlyrtp.diff uploaded by wedhorn (license 30) Tested by: pj, qwell, DEA, wedhorn ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@113119 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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@@ -1223,6 +1223,7 @@ static struct skinny_device {
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int lastlineinstance;
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int lastcallreference;
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int capability;
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int earlyrtp;
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struct sockaddr_in addr;
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struct in_addr ourip;
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struct skinny_line *lines;
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@@ -2902,6 +2903,7 @@ static struct skinny_device *build_device(const char *cat, struct ast_variable *
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else
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memset(device_vmexten, 0, sizeof(device_vmexten));
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d->earlyrtp = 1;
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while(v) {
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if (!strcasecmp(v->name, "host")) {
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if (ast_get_ip(&d->addr, v->value)) {
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@@ -2928,6 +2930,8 @@ static struct skinny_device *build_device(const char *cat, struct ast_variable *
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ast_copy_string(d->version_id, v->value, sizeof(d->version_id));
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} else if (!strcasecmp(v->name, "canreinvite")) {
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canreinvite = ast_true(v->value);
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} else if (!strcasecmp(v->name, "earlyrtp")) {
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d->earlyrtp = ast_true(v->value);
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} else if (!strcasecmp(v->name, "nat")) {
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nat = ast_true(v->value);
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} else if (!strcasecmp(v->name, "callerid")) {
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@@ -3148,6 +3152,9 @@ static void *skinny_newcall(void *data)
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l->hidecallerid ? "" : l->cid_name,
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c->cid.cid_ani ? NULL : l->cid_num);
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ast_setstate(c, AST_STATE_RING);
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if (!sub->rtp) {
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start_rtp(sub);
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}
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res = ast_pbx_run(c);
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if (res) {
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ast_log(LOG_WARNING, "PBX exited non-zero\n");
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@@ -3602,45 +3609,61 @@ static int skinny_indicate(struct ast_channel *ast, int ind, const void *data, s
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case AST_CONTROL_RINGING:
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if (ast->_state != AST_STATE_UP) {
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if (!sub->progress) {
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transmit_tone(s, SKINNY_ALERT, l->instance, sub->callid);
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if (!d->earlyrtp) {
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transmit_tone(s, SKINNY_ALERT, l->instance, sub->callid);
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}
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transmit_callstate(s, l->instance, SKINNY_RINGOUT, sub->callid);
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transmit_dialednumber(s, exten, l->instance, sub->callid);
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transmit_displaypromptstatus(s, "Ring Out", 0, l->instance, sub->callid);
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transmit_callinfo(s, ast->cid.cid_name, ast->cid.cid_num, exten, exten, l->instance, sub->callid, 2); /* 2 = outgoing from phone */
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sub->ringing = 1;
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break;
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if (!d->earlyrtp) {
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break;
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}
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}
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}
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return -1;
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return -1; /* Tell asterisk to provide inband signalling */
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case AST_CONTROL_BUSY:
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if (ast->_state != AST_STATE_UP) {
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transmit_tone(s, SKINNY_BUSYTONE, l->instance, sub->callid);
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if (!d->earlyrtp) {
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transmit_tone(s, SKINNY_BUSYTONE, l->instance, sub->callid);
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}
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transmit_callstate(s, l->instance, SKINNY_BUSY, sub->callid);
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sub->alreadygone = 1;
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ast_softhangup_nolock(ast, AST_SOFTHANGUP_DEV);
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break;
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if (!d->earlyrtp) {
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break;
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}
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}
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return -1;
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return -1; /* Tell asterisk to provide inband signalling */
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case AST_CONTROL_CONGESTION:
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if (ast->_state != AST_STATE_UP) {
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transmit_tone(s, SKINNY_REORDER, l->instance, sub->callid);
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if (!d->earlyrtp) {
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transmit_tone(s, SKINNY_REORDER, l->instance, sub->callid);
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}
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transmit_callstate(s, l->instance, SKINNY_CONGESTION, sub->callid);
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sub->alreadygone = 1;
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ast_softhangup_nolock(ast, AST_SOFTHANGUP_DEV);
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break;
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if (!d->earlyrtp) {
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break;
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}
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}
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return -1;
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return -1; /* Tell asterisk to provide inband signalling */
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case AST_CONTROL_PROGRESS:
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if ((ast->_state != AST_STATE_UP) && !sub->progress && !sub->outgoing) {
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transmit_tone(s, SKINNY_ALERT, l->instance, sub->callid);
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if (!d->earlyrtp) {
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transmit_tone(s, SKINNY_ALERT, l->instance, sub->callid);
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}
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transmit_callstate(s, l->instance, SKINNY_PROGRESS, sub->callid);
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transmit_displaypromptstatus(s, "Call Progress", 0, l->instance, sub->callid);
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transmit_callinfo(s, ast->cid.cid_name, ast->cid.cid_num, exten, exten, l->instance, sub->callid, 2); /* 2 = outgoing from phone */
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sub->progress = 1;
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break;
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if (!d->earlyrtp) {
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break;
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}
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}
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return -1;
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case -1:
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return -1; /* Tell asterisk to provide inband signalling */
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case -1: /* STOP_TONE */
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transmit_tone(s, SKINNY_SILENCE, l->instance, sub->callid);
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break;
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case AST_CONTROL_HOLD:
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@@ -3656,7 +3679,7 @@ static int skinny_indicate(struct ast_channel *ast, int ind, const void *data, s
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break;
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default:
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ast_log(LOG_WARNING, "Don't know how to indicate condition %d\n", ind);
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return -1;
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return -1; /* Tell asterisk to provide inband signalling */
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}
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return 0;
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}
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@@ -63,6 +63,11 @@ keepalive=120
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;jblog = no ; Enables jitterbuffer frame logging. Defaults to "no".
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;-----------------------------------------------------------------------------------
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;----------------------------------- DEVICE OPTIONS --------------------------------
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;earlyrtp=1 ; whether audio signalling should be provided by asterisk
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; (earlyrtp=1) or device generated (earlyrtp=0).
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; defaults to earlyrtp=1
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;-----------------------------------------------------------------------------------
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; Typical config for 12SP+
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;[florian]
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