3CXPhone/Win11 from Asterisk/docker/WSL2 no sound

[MikeF1]

3CXPhone 6.0.26523.0
Windows 11
Local Asterisk Asterisk 14.3.0
Docker image dougbtv/asterisk14
WSL2 20.04
sip.conf: disallow=all allow=ulaw allow=gsm
ports 5060, 8080, 10000-100019

Hi, I can here DTMF and line-available tone, but not voice files
Also WAV file is still transcoded into slin though SIP only allowes ULAW and GSM
3CXPhone account audio codec order: PCMU, PCMA, GSM
Could someone please explain how to add/adjust necessary codecs or pinpoint the problem?
Thank you

Asterisk log:
asterisk_1 | -- Registered SIP 'docker-sip' at 172.19.0.1:44644
asterisk_1 | [Nov 1 13:50:17] NOTICE[259]: chan_sip.c:28327 handle_request_subscribe: Received SIP subscribe for peer without mailbox: docker-sip
asterisk_1 | == Using SIP RTP CoS mark 5
asterisk_1 | [Nov 1 13:50:25] WARNING[259][C-00000001]: chan_sip.c:10306 process_sdp: Declining non-primary audio stream: audio 10008 RTP/AVP 0 8 3 101
asterisk_1 | -- Executing [1@inbound:1] NoOp("SIP/docker-sip-00000000", "Testing Asterisk Setup") in new stack
asterisk_1 | -- Executing [1@inbound:2] NoOp("SIP/docker-sip-00000000", "Testing voice is working, using Asterisk standard file") in new stack
asterisk_1 | -- Executing [1@inbound:3] Playback("SIP/docker-sip-00000000", "hello-world") in new stack
asterisk_1 | -- <SIP/docker-sip-00000000> Playing 'hello-world.gsm' (language 'en')
asterisk_1 | -- Executing [1@inbound:4] NoOp("SIP/docker-sip-00000000", "Testing Paymentus voice files are present") in new stack
asterisk_1 | -- Executing [1@inbound:5] Playback("SIP/docker-sip-00000000", "/var/lib/asterisk/sounds/qa-sounds/sorry-we-did-not-get-valid-inp") in new stack
asterisk_1 | -- <SIP/docker-sip-00000000> Playing '/var/lib/asterisk/sounds/qa-sounds/sorry-we-did-not-get-valid-inp.slin' (language 'en')
asterisk_1 | -- Executing [1@inbound:6] NoOp("SIP/docker-sip-00000000", "Testing Playback of number works") in new stack
asterisk_1 | -- Executing [1@inbound:7] SayNumber("SIP/docker-sip-00000000", "123") in new stack
asterisk_1 | -- <SIP/docker-sip-00000000> Playing 'digits/1.gsm' (language 'en')
asterisk_1 | -- <SIP/docker-sip-00000000> Playing 'digits/hundred.gsm' (language 'en')
asterisk_1 | -- <SIP/docker-sip-00000000> Playing 'digits/20.gsm' (language 'en')
asterisk_1 | -- <SIP/docker-sip-00000000> Playing 'digits/3.gsm' (language 'en')
asterisk_1 | -- Executing [1@inbound:8] Hangup("SIP/docker-sip-00000000", "") in new stack
asterisk_1 | == Spawn extension (inbound, 1, 8) exited non-zero on 'SIP/docker-sip-00000000'
asterisk_1 | [Nov 1 13:50:57] WARNING[209]: chan_sip.c:4076 retrans_pkt: Retransmission timeout reached on transmission NzIxYjU4ZmYwZmE1OGJmNGYyMmYyMzQyYjZmNDQ0ZDQ. for seqno 1 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions

 

[MikeF1]

Resolved.
sip.conf: disallow=all allow=ulaw allow=gsm was irrelevant
Solution:

• In sip.conf [general] define:

nat=yes
externip=<result of `hostname -I | awk '{print $1}'` performed in WSL2 outside of container>
localnet=<result of `ip a | grep inet | grep eth0 | awk '{ print $$2 }' ` performed inside container (could sed substitute in the docker-compose service command for asterisk)>

• In 3CXPhone:

In profile configuration setting I'm in the office, local IP: use the same value as for externip. Do not use 127.0.0.1, because the phone connects but no sound passes