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Fix a bug where the codecs of the called party leg were not properly sent back to the caller call leg when reinvited.
(closes issue #13569) Reported by: bkw918 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.4@195095 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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@@ -18559,11 +18559,8 @@ static int sip_set_rtp_peer(struct ast_channel *chan, struct ast_rtp *rtp, struc
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changed = 1;
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}
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if (codecs) {
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if ((p->redircodecs != codecs)) {
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if (p->redircodecs != codecs && (p->jointcapability & codecs) != p->jointcapability) {
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p->redircodecs = codecs;
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changed = 1;
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}
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if ((p->capability & codecs) != p->capability) {
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p->jointcapability &= codecs;
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p->capability &= codecs;
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changed = 1;
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@@ -3019,7 +3019,8 @@ static enum ast_bridge_result bridge_native_loop(struct ast_channel *c0, struct
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oldcodec1 = codec1;
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}
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if ((inaddrcmp(&t0, &ac0)) ||
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(vp0 && inaddrcmp(&vt0, &vac0))) {
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(vp0 && inaddrcmp(&vt0, &vac0)) ||
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(codec0 != oldcodec0)) {
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if (option_debug > 1) {
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ast_log(LOG_DEBUG, "Oooh, '%s' changed end address to %s:%d (format %d)\n",
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c0->name, ast_inet_ntoa(t0.sin_addr), ntohs(t0.sin_port), codec0);
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