Как заставить asterisk записывать входящее видео?

Имеем ubuntu+ asterisk 17 + freepbx последний
нужно записать поступающий на него звонок в идеале по h.323 но можно и по sip
добился следующего:

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 [s@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 [s@from-sip-external:3] GotoIf("SIP/172.30.3.11-0000000e", "0?noanonymous") in new stack
-- Executing [s@from-sip-external:4] Goto("SIP/172.30.3.11-0000000e", "from-trunk,,1") in new stack
-- Goto (from-trunk,s,1)
-- Executing [s@from-trunk:1] NoOp("SIP/172.30.3.11-0000000e", "No DID or CID Match") in new stack
-- Executing [s@from-trunk: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 [s@from-trunk: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 [s@from-trunk:4] Wait("SIP/172.30.3.11-0000000e", "2") in new stack
-- Executing [s@from-trunk:5] Playback("SIP/172.30.3.11-0000000e", "ss-noservice") in new stack
-- Playing 'ss-noservice.gsm' (language 'en')
> 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 [s@from-trunk:6] SayAlpha("SIP/172.30.3.11-0000000e", "") in new stack
-- Executing [s@from-trunk: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'


максимум смог добиться только этого... и то через пару секунду кладет трубку.
как сделать так чтоб астериск поднимал трубку и записывал всё видео что на него поступает?
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