Summary:ASTERISK-20511: Directrtpsetup does not wrk in SVN-branch-1.8-r374177
Reporter:Krzysztof Chmielewski (kristoff)Labels:
Date Opened:2012-10-04 10:20:23Date Closed:2012-10-04 12:44:41
Versions: Frequency of
must be completed before resolvingASTERISK-20410 Asterisk Blockers
must be completed before resolvingASTERISK-20411 Asterisk 10.9.0 Blockers
causesASTERISK-20520 Write a test for the Asterisk Test Suite that covers directrtpsetup
is related toASTERISK-20409 sip_tech_info channels cannot be bridged, not even with themselves
Description:It seems that direct media setup does not work in SVN-branch-1.8-r374177. Asterisk sends Re-INVITE with new media address to user A and B, after user B answers the call

On 1.8.13 initial Invite (with option direct media setup in sip.conf) already has user A/B IP address in SDP.

Comments:By: Joshua C. Colp (jcolp) 2012-10-04 10:30:01.790-0500

You are actually referring to "directrtpsetup" here, not "directmedia". They are different.

By: Matt Jordan (mjordan) 2012-10-04 10:33:30.059-0500

We require a complete debug log to help triage the issue. This document will provide instructions on how to collect debugging logs from an Asterisk machine for the purpose of helping bug marshals troubleshoot an issue: https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information

Also provide your sip.conf, extensions.conf.  Make sure 'sip set debug on' is enabled in your DEBUG log.

By: Joshua C. Colp (jcolp) 2012-10-04 12:44:41.361-0500

Fixed in 1.8 as of revision 374456 and 10 as of revision 374457.

By: Krzysztof Chmielewski (kristoff) 2012-10-05 03:27:21.421-0500

Thank You for response.
Joshua You are right, I meant the directrtpsetup, I'm sorry for my mistake.

I've tested this scenario on 1.8 Revision 374523 and it works fine, thank You.