Summary: | ASTERISK-20511: Directrtpsetup does not wrk in SVN-branch-1.8-r374177 | ||||||||||
Reporter: | Krzysztof Chmielewski (kristoff) | Labels: | |||||||||
Date Opened: | 2012-10-04 10:20:23 | Date Closed: | 2012-10-04 12:44:41 | ||||||||
Priority: | Major | Regression? | |||||||||
Status: | Closed/Complete | Components: | Channels/chan_sip/General | ||||||||
Versions: | 1.8.16.0 | Frequency of Occurrence | |||||||||
Related Issues: |
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Environment: | Attachments: | ||||||||||
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. |