Atxfer while on hold

Asterisk Asterisk Support
1

I tried it but it didn’t behave I thought.

Specifically,
When the number 999 talks to anyone and 999 does hold and 999 does atxfer to 888,
999 can not hear any sound but 888 can hear what 999 talks.
At this time, 999 does unhold, then 999 and 888 talks correctly.

Is my thought wrong that 999 and 888 can talk correctly without unhold ?
This behavior is correct ?

2

Which channel technology and< for SIP, which channel technology driver?

How is hold signalled to Asterisk?

How is atxfer signalled to Asterisk?

3

This is the log which 12345 calls to 222 and 222 do hold and 222 do atxfer to 441.
Is this log enogh ?

Asterisk: 13.20.0

    -- Called SIP/222
    -- SIP/222-00000016 is ringing
       > 0x73e06b30 -- Strict RTP learning after remote address set to: 192.168.1.143:58595
       > 0x73e06b30 -- Strict RTP learning after remote address set to: 192.168.1.143:58595
    -- SIP/222-00000016 answered SIP/kkk-00000015
    -- Channel SIP/222-00000016 joined 'simple_bridge' basic-bridge <aa0a83af-c527-4f89-b22b-ae8e82ea2209>
    -- Channel SIP/kkk-00000015 joined 'simple_bridge' basic-bridge <aa0a83af-c527-4f89-b22b-ae8e82ea2209>
       > 0x73e06b30 -- Strict RTP learning after ICE completion
       > 0x73e06b30 -- Strict RTP switching to RTP target address 192.168.1.143:58595 as source
       > 0x74609ab0 -- Strict RTP switching to RTP target address XXX.XXX.XXX.XXX:31476 as source
    -- Started music on hold, class 'default', on channel 'SIP/kkk-00000015'
       > 0x74609ab0 -- Strict RTP learning complete - Locking on source address XXX.XXX.XXX.XXX:31476
    -- Channel SIP/222-00000016: Started DTMF attended transfer.
    -- <SIP/222-00000016> Playing 'pbx-transfer.gsm' (language 'ja')
       > 0x73e06b30 -- Strict RTP learning complete - Locking on source address 192.168.1.143:58595
    -- Executing [441@default:1] NoOp("Local/441@default-00000005;2", "CALLERID: 12345") in new stack
    -- Executing [441@default:2] NoOp("Local/441@default-00000005;2", "EXTEN: 441") in new stack
    -- Executing [441@default:3] Dial("Local/441@default-00000005;2", "SIP/441,30,rTtKk") in new stack
    -- Channel Local/441@default-00000005;1 joined 'simple_bridge' basic-bridge <977ee88c-1f21-47b5-b1f9-4a66b6d14848>
    -- Channel SIP/222-00000016 left 'simple_bridge' basic-bridge <aa0a83af-c527-4f89-b22b-ae8e82ea2209>
    -- Channel SIP/222-00000016 joined 'simple_bridge' basic-bridge <977ee88c-1f21-47b5-b1f9-4a66b6d14848>
  == DTLS ECDH initialized (secp256r1), faster PFS enabled
  == Using SIP RTP CoS mark 5
    -- Called SIP/441
    -- SIP/441-00000017 is ringing
       > 0x74213220 -- Strict RTP learning after remote address set to: 192.168.1.145:53056
    -- SIP/441-00000017 answered Local/441@default-00000005;2
    -- Channel SIP/441-00000017 joined 'simple_bridge' basic-bridge <7c4a5fbe-a332-4154-bac5-9953f633f449>
    -- Channel Local/441@default-00000005;2 joined 'simple_bridge' basic-bridge <7c4a5fbe-a332-4154-bac5-9953f633f449>
       > 0x74213220 -- Strict RTP learning after ICE completion
       > 0x74213220 -- Strict RTP switching to RTP target address 192.168.1.145:53056 as source
       > 0x74213220 -- Strict RTP learning complete - Locking on source address 192.168.1.145:53056