Certified versions of Asterisk are only for users with support contracts and I suspect, as Asterisk 13 is end of life, anyone on such a contract would have been moved to at least Asterisk 18. chan_sip is no longer supported for certified versions.
Please upgrade to at least the current (non-certified) release of Asterisk 18, and, if at all possible, to chan_pjsip.
As well as using variables which you have not defined, your dialplan contains a missing $. This isn’t an artefact of the forum, as the raw forum code is also missing the $.
It’s sort of behaving as though it works for direct media, but not otherwise, which is rather the opposite of what I’d expect. I think you need to provide “pjsip set logger on” output, if the problem persists after you make the above upgrades.
Here is correct syntax,
it’s voice listenable well without MixMonitor but problem while start MixMonitor.
it was working since long, but problem arise from last week. Also same dialplan working ok on other asterisk server.