How to make asterisk record incoming video?

Question

Asterisk

We have ubuntu + asterisk 17 + freepbx the last
one, you need to record the incoming call to it, ideally via h.323, but you can
achieve the following via sip:

Asterisk 17.5.1, Copyright (C) 1999 - 2018, Digium, Inc. and others.
Created by Mark Spencer
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.
================================================= =======================
...............
[2020-07-13 15:34:00] ERROR[13117]: chan_ooh323.c:1972 ooh323_onReceivedSetup: Unacceptable ip 172.30.3.11
== Using SIP VIDEO TOS bits 136
== Using SIP VIDEO CoS mark 6
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
> 0x7f4ec0013700 -- Strict RTP learning after remote address set to: 172.30.3.11:49244
> 0x7f4ec0018900 -- Strict RTP learning after remote address set to: 172.30.3.11:49246
-- Executing [[email protected] from-sip-external:1] GotoIf("SIP/172.30.3.11-0000000e", "1?setlanguage:checkanon") in new stack
-- Goto (from-sip-external,s,2)
-- Executing [s @from-sip-external:2] Set("SIP/172.30.3.11-0000000e", "CHANNEL(language)=en") in new stack
-- Executing [[email protected]:3] GotoIf(" SIP/172.30.3.11-0000000e", "0?noanonymous") in new stack
-- Executing [[email protected]:4] Goto("SIP/172.30.3.11-0000000e", "from-trunk,, one") in new stack
-- Goto (from-trunk,s,1)
-- Executing [[email protected]:1] NoOp("SIP/172.30.3.11-0000000e", "No DID or CID Match") in new stack
-- Executing [[email protected]:2] Answer(" SIP/172.30.3.11-0000000e", "") in new stack
> 0x7f4ec0013700 -- Strict RTP switching to RTP target address 172.30.3.11:49244 as source
-- Executing [[email protected]:3] Log("SIP/ 172.30.3.11-0000000e", "WARNING,Friendly Scanner from 172.30.3.11") in new stack
[2020-07-13 15:34:06] WARNING[13165][C-0000000f]: Ext. s:3 @ from-trunk: Friendly Scanner from 172.30.3.11
-- Executing [[email protected]:4] Wait("SIP/172.30.3.11-0000000e", "2") in new stack
-- Executing [s @from-trunk:5] Playback("SIP/172.30.3.11-0000000e", "
> 0x7f4ec0013700 -- Strict RTP learning complete - Locking on source address 172.30.3.11:49244
> 0x7f4ec0018900 -- Strict RTP switching to RTP target address 172.30.3.11:49246 as source
> 0x7f4ec0018900 -- Strict RTP learning complete - Locking on source address 172.30 .3.11:49246
-- Executing [[email protected]:6] SayAlpha("SIP/172.30.3.11-0000000e", "") in new stack
-- Executing [[email protected]:7] Hangup("SIP /172.30.3.11-0000000e", "") in new stack
== Spawn extension (from-trunk, s, 7) exited non-zero on 'SIP/172.30.3.11-0000000e'
[2020-07-13 15:34 :13] WARNING[13165][C-0000000f]: pbx.c:2927 pbx_extension_helper: No application 'Macro' for extension (from-trunk, h, 1)
== Spawn extension (from-trunk, h, 1) exited non-zero on 'SIP/172.30.3.11-0000000e'

The maximum was able to achieve only this ... and then after a couple of seconds he hangs up.
how to make the asterisk pick up the phone and record all the video that comes to it?

Answer 1

Asterisk does not have the ability to write videos.
Recording media traffic in Asterisk is done by the Record application.
You still have a problem even in something else: calls simply do not go through, not to mention the records.
I recommend installing FreePBX Distro, everything will work out of the box there.
But recording will still be possible in Asterisk only audio