[asterisk-dev] SIP/RTP issue fixed?

Steve Davies davies147 at gmail.com
Wed May 4 04:48:13 CDT 2011


I have raised this against 1.6.2.18, but I suspect it will apply
equally to 1.4 and 1.8, though I have not checked:

  https://issues.asterisk.org/view.php?id=19225

It covers a case where a directmedia call is "Redirect"ed back into
the dialplan, and the media path is not correctly reset.

Regards,
Steve



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