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chan_sip: SIP route header is missing on UPDATE

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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
This commit is contained in:
Mark Petersen
2022-03-07 15:11:41 +01:00
committed by Kevin Harwell
parent 1937f30202
commit 73fb22714e
1 changed files with 7 additions and 3 deletions
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10
channels/chan_sip.c
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@@ -12443,9 +12443,8 @@ static int reqprep(struct sip_request *req, struct sip_pvt *p, int sipmethod, ui
* Send UPDATE to the same destination as CANCEL, if call is not in final state.
*/
if (!sip_route_empty(&p->route) &&
!(sipmethod == SIP_CANCEL ||
(sipmethod == SIP_ACK && (p->invitestate == INV_COMPLETED || p->invitestate == INV_CANCELLED)) ||
(sipmethod == SIP_UPDATE && (p->invitestate == INV_PROCEEDING || p->invitestate == INV_EARLY_MEDIA)))) {
!(sipmethod == SIP_CANCEL ||
(sipmethod == SIP_ACK && (p->invitestate == INV_COMPLETED || p->invitestate == INV_CANCELLED)))) {
if (p->socket.type != AST_TRANSPORT_UDP && p->socket.tcptls_session) {
/* For TCP/TLS sockets that are connected we won't need
* to do any hostname/IP lookups */
@@ -12453,6 +12452,11 @@ static int reqprep(struct sip_request *req, struct sip_pvt *p, int sipmethod, ui
/* For NATed traffic, we ignore the contact/route and
* simply send to the received-from address. No need
* for lookups. */
} else if (sipmethod == SIP_UPDATE && (p->invitestate == INV_PROCEEDING || p->invitestate == INV_EARLY_MEDIA)) {
/* Calling set_destination for an UPDATE in early dialog
* will result in mangling of the target for a subsequent
* CANCEL according to ASTERISK-24628 so do not do it.
*/
} else {
set_destination(p, sip_route_first_uri(&p->route));
}