[asterisk-bugs] [JIRA] (ASTERISK-24657) directmedia=no does not work in Asterisk 13.1.0

[Matt Jordan (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-24657?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=224285#comment-224285 ] 

Matt Jordan commented on ASTERISK-24657:
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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

Make sure your log has 'sip set debug on' enabled.

> directmedia=no does not work in Asterisk 13.1.0
> -----------------------------------------------
>
>                 Key: ASTERISK-24657
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-24657
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>    Affects Versions: 13.1.0
>         Environment: The Asterisk server runs under Fedora 20, with static ipv4 addresses and working ipv6.
> The yealinkphone is a sip client of the asterisk server.  The yealink phone has a non-routable ipv4 address and a routable ipv6 address, but preferentially uses the ipv6 address.
> The service provider, callwithus, does not have an ipv6 address.
>            Reporter: Thomas B. Clark
>            Assignee: Thomas B. Clark
>
> I just upgraded from Asterisk 11.14.2 to asterisk 13.1.0, and that caused directmedia=no to stop working.  It worked in 11.14, and using the same sip.conf it does not work in 13.1.0.  Here is the relevant part of sip.conf:
> [yealinkphone]
> type=friend
> host=dynamic
> context=outgoing
> secret=xxxxxx
> defaultuser=xxxxxxx
> insecure=port,invite
> directmedia=no
> qualify=yes
> And here is what happens:
>     -- SIP/callwithus-00000003 is making progress passing it to SIP/yealinkphone-00000002
>        > 0x7fa784016470 -- Probation passed - setting RTP source address to 69.85.185.222:21708
>     -- SIP/callwithus-00000003 answered SIP/yealinkphone-00000002
> -- Channel SIP/yealinkphone-00000002 joined 'simple_bridge' basic-bridge <5fb81915-9eaa-4b69-9be8-26f8a1db1fff>
>     -- Channel SIP/callwithus-00000003 joined 'simple_bridge' basic-bridge <5fb81915-9eaa-4b69-9be8-26f8a1db1fff>
>        > Bridge 5fb81915-9eaa-4b69-9be8-26f8a1db1fff: switching from simple_bridge technology to native_rtp
>        > 0x7fa70400afd0 -- Probation passed - setting RTP source address to [2601:8:9181:4800:215:65ff:fe27:ac8e]:11786
> Even if I had directmedia=yes, Asterisk shouldn't issue a reinvite because the yealinkphone is using ipv6 and the service provider, callwithus, is using ipv4. However, as I stated, I have directmedia=no, so nothing else should matter, but the statement is being ignored.
> I have worked around the issue by setting ,,t at the end of all of the Dial statements, and that successfully prevents switching to native_rtp.  Nothing else I tried does.



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