So, that's not quite a debug log, but just the console log with Verbose+ output.
A debug log will show a lot more information, including what the media cache modules are trying to do when they go to get the file.
You can find information on getting debug information on the Asterisk here:
You may also want to verify that the res_http_media_cache module is loaded. That module is what actually does the work of pulling the remote file down for local playback.
On Fri, Jul 20, 2018 at 3:10 PM Naftoli Gugenheim <naftoligug@xxxxxxxxx> wrote:
_______________________________________________In one terminal tab:
$ sudo nc -kl 80In another (note: asterisk is running in docker with
--net=host):$ docker-compose exec asterisk cat /etc/hosts 127.0.0.1 localhost 127.0.0.1 example.com 127.0.1.1 naftoli-ThinkPad-W540 # The following lines are desirable for IPv6 capable hosts ::1 ip6-localhost ip6-loopback fe00::0 ip6-localnet ff00::0 ip6-mcastprefix ff02::1 ip6-allnodes ff02::2 ip6-allrouters $ docker-compose exec asterisk curl http://example.com/dummyfile.wav ^C⏎The HTTP request headers show up in
nc.However,
$ docker-compose exec asterisk asterisk -rvvvvvddddddT Seeding global EID '5c:51:4f:a5:bf:59' from 'wlp3s0' using 'siocgifhwaddr' Parsing /etc/asterisk/asterisk.conf Asterisk 15.5.0, Copyright (C) 1999 - 2016, Digium, Inc. and others. Created by Mark Spencer <<a href="" href="mailto:markster@xxxxxxxxxx" target="_blank">markster@xxxxxxxxxx" target="_blank">markster@xxxxxxxxxx</a>> Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'core show license' for details. ========================================================================= Connected to Asterisk 15.5.0 currently running on naftoli-ThinkPad-W540 (pid = 8) Core debug is still 6. [Jul 20 20:00:16] == Setting global variable 'SIPDOMAIN' to 'localhost' [Jul 20 20:00:16] -- Executing [1400@inbound:1] Set("PJSIP/local-0000004e", "JITTERBUFFER(adaptive)=default") in new stack [Jul 20 20:00:16] -- Executing [1400@inbound:2] AGI("PJSIP/local-0000004e", "agi://127.0.0.1/route") in new stack [Jul 20 20:00:16] > 0x7f9e8000cb00 -- Strict RTP learning after remote address set to: 173.124.23.24:7078 [Jul 20 20:00:16] > 0x7f9e8000cb00 -- Strict RTP qualifying stream type: audio [Jul 20 20:00:16] > 0x7f9e8000cb00 -- Strict RTP switching source address to 127.0.0.1:7078 [Jul 20 20:00:16] -- AGI Script Executing Application: (MixMonitor) Options: (/sounds/monitor-2018-07-20T20:00:16.992040Z.wav) [Jul 20 20:00:16] == Begin MixMonitor Recording PJSIP/local-0000004e [Jul 20 20:00:16] WARNING[6384][C-00000050]: file.c:772 ast_openstream_full: File http://example.com/dummyfile.wav does not exist in any format [Jul 20 20:00:17] -- <PJSIP/local-0000004e> Playing '/sounds/prompts/welcome-to.slin' (escape_digits=) (sample_offset 0) (language 'en') [Jul 20 20:00:17] WARNING[6384][C-00000050]: chan_iax2.c:1228 jb_warning_output: Resyncing the jb. last_delay 0, this delay -359631367, threshold 1000, new offset 359631367 [Jul 20 20:00:18] -- <PJSIP/local-0000004e> Playing '/sounds/prompts/some-org.slin' (escape_digits=) (sample_offset 0) (language 'en') [Jul 20 20:00:19] -- <PJSIP/local-0000004e> Playing '/sounds/prompts/press-2-now-to-use-a-phone-number-other-than-the-one-you-are-calling-from-.slin' (escape_digits=0123456789#*) (sample_offset 0) (language 'en') [Jul 20 20:00:20] WARNING[6370]: res_pjsip_registrar.c:957 find_registrar_aor: AOR '<REDACTED>' not found for endpoint 'local' [Jul 20 20:00:21] > 0x7f9e8000cb00 -- Strict RTP learning complete - Locking on source address 127.0.0.1:7078 [Jul 20 20:00:21] -- <PJSIP/local-0000004e>AGI Script agi://127.0.0.1/route completed, returning -1 [Jul 20 20:00:21] == MixMonitor close filestream (mixed) [Jul 20 20:00:21] == End MixMonitor Recording PJSIP/local-0000004eNothing shows up in
nc.P.S. I have no idea why it thinks the other prompts are .slin when in reality they are .wav
Thanks.
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Matthew Jordan
Digium, Inc. | CTO
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com & http://asterisk.org
Digium, Inc. | CTO
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com & http://asterisk.org
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