Hello, I’m facing a problem when I try to execute the dial plan described below.
[test_dial_plan] exten => _X.,1,Verbose(Testing Started) exten => _X.,n,Playback(http://192.168.127.27/asterisk/gsm/tt-monkeys.gsm) exten => _X.,n,Hangup()
The dial plan is returning warnings when it tries to execute the Playback method and the call does not hang up. Here are the logs returned by the Asterisk.
== Setting global variable 'SIPDOMAIN' to 'XXX.XXX.XXX.XXX' -- Executing [123@test_dial_plan:1] Verbose("PJSIP/test_lele-00000020", "Testing Started") in new stack Testing Started -- Executing [123@test_dial_plan:2] Playback("PJSIP/test_lele-00000020", "http://192.168.127.27/asterisk/gsm/tt-monkeys.gsm") in new stack > 0x7f6338038760 -- Strict RTP learning after remote address set to: XXX.XXX.XXX.XXX:8000 > 0x7f6338038760 -- Strict RTP qualifying stream type: audio > 0x7f6338038760 -- Strict RTP switching source address to 192.168.150.1:8000 [Mar 8 14:18:20] WARNING[C-00000021]: file.c:779 ast_openstream_full: File http://192.168.127.27/asterisk/gsm/tt-monkeys.gsm does not exist in any format [Mar 8 14:18:20] WARNING[C-00000021]: file.c:1252 ast_streamfile: Unable to open http://192.168.127.27/asterisk/gsm/tt-monkeys.gsm (format (ulaw)): No such file or directory [Mar 8 14:18:20] WARNING[C-00000021]: app_playback.c:492 playback_exec: Playback failed on PJSIP/test_lele-00000020 for http://192.168.127.27/asterisk/gsm/tt-monkeys.gsm -- Executing [123@test_dial_plan:3] Hangup("PJSIP/test_lele-00000020", "") in new stack == Spawn extension (test_dial_plan, 123, 3) exited non-zero on 'PJSIP/test_lele-00000020'
I already tried the approaches below.
• Set the cache settings on the media server
• Use different file extensions like (ulaw, wav, gsm)
• Omit the file extensions.