Started by an SCM change00443543541.0077737773572c9a787ad1fde5f70beed8e763d6b12ccc2cd4a4bc9a787ad1fde5f70beed8e763d6b12ccc2cd4a4brefs/remotes/origin/masterc9a787ad1fde5f70beed8e763d6b12ccc2cd4a4bc9a787ad1fde5f70beed8e763d6b12ccc2cd4a4brefs/remotes/origin/masterc9a787ad1fde5f70beed8e763d6b12ccc2cd4a4bc9a787ad1fde5f70beed8e763d6b12ccc2cd4a4brefs/remotes/origin/masterhttps://gerrit.osmocom.org/docker-playgroundfalse#572354255update-osmo-ci-on-slaves-dp #572572falsefalse5721567121SUCCESS1717489868665https://jenkins.osmocom.org/jenkins/view/gerrit/job/update-osmo-ci-on-slaves-dp/572/build2-deb11build-ansiblettcn3-asterisk-ims-ue-test/asterisk/pjsip.confcc5ec183c0f4414116a9628025a1395655a67fa61716984544000https://jenkins.osmocom.org/jenkins/user/pespinPau Espin Pedrolpespin@sysmocom.deasterisk: pjsip.conf: Use TCP when forwarding INVITE to IMS core Change-Id: I67bc01e4d53d47eb805e81736378f292b2834fd2 2024-05-29 14:09:04 +0200cc5ec183c0f4414116a9628025a1395655a67fa6asterisk: pjsip.conf: Use TCP when forwarding INVITE to IMS coreeditttcn3-asterisk-ims-ue-test/asterisk/pjsip.confttcn3-asterisk-ims-ue-test/asterisk/pjsip.confc9a787ad1fde5f70beed8e763d6b12ccc2cd4a4b1717069261000https://jenkins.osmocom.org/jenkins/user/pespinPau Espin Pedrolpespin@sysmocom.deasterisk: pjsip.conf: Disable remote bridging between local SIP and IMS While implementing a first ttcn3 test validating the MO call scenario (SIP-UA -> Asterisk -> IMS-CORE) [1] I was running into the scenario where, after the first SIP INVITE + 200 OK + ACK, Asterisk was sending a RE-INVITE to both parties to attempt to remotely bridge them (RTP traffic flowing directly between them without passing through Asterisk). This happened in part because I'm so far configure A-LAW on both sides so asterisk figures out it can do so. I still need to change IMS-core to EVS only. In any case, regardless of the codecs used, my understanding is that asterisk should never attempt remote bridging when using the "volte_ims" endpoint, since that network segment is separate from the local network where the local SIP UAs are located. The "direct_media=no" option just addresses the issue; Asterisk no longer tries to re-invite after the call is established Related: SYS#6782 Change-Id: I4edea96151b31f02bf292b43b757922389375429 2024-05-30 13:41:01 +0200c9a787ad1fde5f70beed8e763d6b12ccc2cd4a4basterisk: pjsip.conf: Disable remote bridging between local SIP and IMSeditttcn3-asterisk-ims-ue-test/asterisk/pjsip.confgithttps://jenkins.osmocom.org/jenkins/user/pespinPau Espin Pedrol