[S] Change in osmo-msc[master]: msc_ho_send_handover_request: fix check_after_deref

osmith has submitted this change. ( https://gerrit.osmocom.org/c/osmo-msc/+/33398 ) Change subject: msc_ho_send_handover_request: fix check_after_deref ...................................................................... msc_ho_send_handover_request: fix check_after_deref Fixes: CID#211073 Change-Id: I9389cc9c4af987c11652c6fe9a0ff4289a7781c9 --- M src/libmsc/msc_ho.c 1 file changed, 12 insertions(+), 2 deletions(-) Approvals: Jenkins Builder: Verified pespin: Looks good to me, but someone else must approve fixeria: Looks good to me, approved diff --git a/src/libmsc/msc_ho.c b/src/libmsc/msc_ho.c index 2a5891a..ce3f100 100644 --- a/src/libmsc/msc_ho.c +++ b/src/libmsc/msc_ho.c @@ -401,8 +401,6 @@ /* Don't send AoIP Transport Layer Address for inter-MSC Handover */ .rtp_ran_local = (msc_a->ho.new_cell.type == MSC_NEIGHBOR_TYPE_LOCAL_RAN_PEER) ? call_leg_local_ip(msc_a->cc.call_leg, RTP_TO_RAN) : NULL, - .call_id_present = true, - .call_id = cc_trans->callref, }, }; @@ -422,6 +420,8 @@ return; } ran_enc_msg.handover_request.geran.channel_type = &channel_type; + ran_enc_msg.handover_request.call_id_present = true; + ran_enc_msg.handover_request.call_id = cc_trans->callref; sdp_audio_codecs_to_speech_codec_list(&scl, &cc_trans->cc.local.audio_codecs); if (!scl.len) { -- To view, visit https://gerrit.osmocom.org/c/osmo-msc/+/33398 To unsubscribe, or for help writing mail filters, visit https://gerrit.osmocom.org/settings Gerrit-Project: osmo-msc Gerrit-Branch: master Gerrit-Change-Id: I9389cc9c4af987c11652c6fe9a0ff4289a7781c9 Gerrit-Change-Number: 33398 Gerrit-PatchSet: 2 Gerrit-Owner: osmith <osmith(a)sysmocom.de> Gerrit-Reviewer: Jenkins Builder Gerrit-Reviewer: fixeria <vyanitskiy(a)sysmocom.de> Gerrit-Reviewer: osmith <osmith(a)sysmocom.de> Gerrit-Reviewer: pespin <pespin(a)sysmocom.de> Gerrit-MessageType: merged