chan_sip: SIP route header is missing on UPDATE
if Asterisk need to send an UPDATE before answer on a channel that uses Record-Route: it will not include a Route header ASTERISK-29955 Change-Id: Id1920ecbfea7739a038b14dc94487ecfe7b57eef
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@ -12441,9 +12441,8 @@ static int reqprep(struct sip_request *req, struct sip_pvt *p, int sipmethod, ui
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* Send UPDATE to the same destination as CANCEL, if call is not in final state.
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*/
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if (!sip_route_empty(&p->route) &&
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!(sipmethod == SIP_CANCEL ||
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(sipmethod == SIP_ACK && (p->invitestate == INV_COMPLETED || p->invitestate == INV_CANCELLED)) ||
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(sipmethod == SIP_UPDATE && (p->invitestate == INV_PROCEEDING || p->invitestate == INV_EARLY_MEDIA)))) {
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!(sipmethod == SIP_CANCEL ||
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(sipmethod == SIP_ACK && (p->invitestate == INV_COMPLETED || p->invitestate == INV_CANCELLED)))) {
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if (p->socket.type != AST_TRANSPORT_UDP && p->socket.tcptls_session) {
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/* For TCP/TLS sockets that are connected we won't need
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* to do any hostname/IP lookups */
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@ -12451,6 +12450,11 @@ static int reqprep(struct sip_request *req, struct sip_pvt *p, int sipmethod, ui
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/* For NATed traffic, we ignore the contact/route and
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* simply send to the received-from address. No need
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* for lookups. */
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} else if (sipmethod == SIP_UPDATE && (p->invitestate == INV_PROCEEDING || p->invitestate == INV_EARLY_MEDIA)) {
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/* Calling set_destination for an UPDATE in early dialog
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* will result in mangling of the target for a subsequent
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* CANCEL according to ASTERISK-24628 so do not do it.
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*/
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} else {
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set_destination(p, sip_route_first_uri(&p->route));
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}
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