----------------------------------------------------------- This is an automatically generated e-mail. To reply, visit: https://reviewboard.asterisk.org/r/4442/#review14522 -----------------------------------------------------------
I don't see the *_no_direction.xml files. ./asterisk/trunk/tests/channels/SIP/sip_hold/sipp/phone_B_IP_media_restrict.xml <https://reviewboard.asterisk.org/r/4442/#comment25076> Heh, that was sloppy ;) - wdoekes On Feb. 23, 2015, 9:38 p.m., Ashley Sanders wrote: > > ----------------------------------------------------------- > This is an automatically generated e-mail. To reply, visit: > https://reviewboard.asterisk.org/r/4442/ > ----------------------------------------------------------- > > (Updated Feb. 23, 2015, 9:38 p.m.) > > > Review request for Asterisk Developers. > > > Bugs: ASTERISK-24824 > https://issues.asterisk.org/jira/browse/ASTERISK-24824 > > > Repository: testsuite > > > Description > ------- > > This test is to ensure that Asterisk correctly applies the direction of the > media stream when a=<sendonly|recvonly|inactive|sendrecv> is missing from the > offer's SDP. The expected behavior is for Asterisk to apply "sendrecv" as the > direction of the media stream when no direction attribute is present in an > offer's SDP. According to RFC 4566 (Section 6. SDP Attributes): "If none of > the attributes "sendonly", "recvonly", "inactive", and "sendrecv" is present, > "sendrecv" SHOULD be assumed as the default for sessions that are not of the > conference type "broadcast" or "H332" [...]" > > The test scenario: > > 1. From Phone A, send an offer to Phone B to establish a call > 2. From Phone B, send an offer to Phone A to put the call on hold. > 3. Observe that the MOH start event occurs. > 4. From Phone B, send an offer to Phone A to 'un-hold' the call (ensure that > the direction attribute from the offer's SDP is omitted) > 5. Observe that the MOH stop event occurs. > > Presently, this test fails for certain versions of Asterisk. From what I can > tell, it is present from (at least) 1.8.21 up to the 11 branch. > > ***Note*** This is the test. It is only the test. The update to the Asterisk > source is coming soon to a review board near you (well, this review board). > > > Diffs > ----- > > > ./asterisk/trunk/tests/channels/SIP/sip_hold/sipp/phone_B_media_restrict.xml > 6458 > ./asterisk/trunk/tests/channels/SIP/sip_hold/sipp/phone_B_IP_restrict.xml > 6458 > > ./asterisk/trunk/tests/channels/SIP/sip_hold/sipp/phone_B_IP_media_restrict.xml > 6458 > ./asterisk/trunk/tests/channels/SIP/sip_hold/run-test 6458 > > Diff: https://reviewboard.asterisk.org/r/4442/diff/ > > > Testing > ------- > > > Thanks, > > Ashley Sanders > >
-- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-dev
