No audio when calling from pap2, but can receive

Hi, im trying to make as home phone network with some old landline phones.

I have manged to get asterisk running at created numbers sip numbers for the 2 lines in pjsip, and the linksys pap2 can connect to them. If i call from my android phone via mizudroid, it works and you can talk in both directions. But if i call from the phones connected to the pap2 then they can only receive audio from the mobile but no audio is send from the landline phone.

Im totally new to asterisk, so im not sure if its a misconfiguration on either asterisk or pap2

Everything is running on a local network.

Here is my pap2 configuration

Here is my pjsip config

[6002]
type = endpoint
context = from-internal-custom
disallow = all
allow = ulaw
aors = 6002
auth = auth6002
direct_media=no
 
[6002]
type = aor
max_contacts = 999
 
[auth6002]
type=auth
auth_type=userpass
password=1234
username=6002

And here is my extension.conf regarding the extensions

[from-internal-custom]
exten => _6XXX,1,Dial(PJSIP/${EXTEN})

Hope somebody can help

You are missing rtp.conf and a verbose, full log, with pjsip set logger on enabled.

I dont have any rtp.conf file

unsure if this is log enough

[Aug  4 08:28:23]     -- Added contact 'sip:6002@192.168.1.77:5060' to AOR '6002' with expiration of 3600 seconds
[Aug  4 08:28:23]   == Endpoint 6002 is now Reachable
[Aug  4 08:28:23]     -- Added contact 'sip:6004@192.168.1.77:5060' to AOR '6004' with expiration of 3600 seconds
[Aug  4 08:28:23]   == Endpoint 6004 is now Reachable
[Aug  4 08:31:12]     -- Executing [6004@from-internal-custom:1] Dial("PJSIP/6004-00000000", "PJSIP/6004") in new stack
[Aug  4 08:31:12]     -- Called PJSIP/6004
[Aug  4 08:31:12]   == Everyone is busy/congested at this time (1:1/0/0)
[Aug  4 08:31:12]     -- Auto fallthrough, channel 'PJSIP/6004-00000000' status is 'BUSY'
[Aug  4 08:31:42]     -- Executing [6004@from-internal-custom:1] Dial("PJSIP/6002-00000002", "PJSIP/6004") in new stack
[Aug  4 08:31:42]     -- Called PJSIP/6004
[Aug  4 08:31:43]     -- PJSIP/6004-00000003 is ringing
[Aug  4 08:31:47]     -- PJSIP/6004-00000003 answered PJSIP/6002-00000002
[Aug  4 08:31:47]     -- Channel PJSIP/6004-00000003 joined 'simple_bridge' basic-bridge <6f08bd62-9e9e-4abc-9ed4-5a7e872e8f0d>
[Aug  4 08:31:47]     -- Channel PJSIP/6002-00000002 joined 'simple_bridge' basic-bridge <6f08bd62-9e9e-4abc-9ed4-5a7e872e8f0d>
[Aug  4 08:32:17]     -- Added contact 'sip:6003@192.168.1.91:13876;pn-provider=fcm;pn-param=com.mizuvoip.mizudroid.app;pn-prid=fuMoS2jORSyoZ9JoTL__Cr:APA91bGJZZe9iQ72QO890YXKivEsIdcJAn5XT5K2C0jlESj2UNcsrXMFSjqWNx5uMHt2QElLOHRV9vAyfDuhnHBSjP1UmqPFtERHcN9wwJDu9tAzMOh3U_90cmkLyiIrh0iiInScVXIO' to AOR '6003' with expiration of 720 seconds
[Aug  4 08:32:17]   == Endpoint 6003 is now Reachable
[Aug  4 08:32:19]     -- Channel PJSIP/6002-00000002 left 'native_rtp' basic-bridge <6f08bd62-9e9e-4abc-9ed4-5a7e872e8f0d>
[Aug  4 08:32:19]     -- Channel PJSIP/6004-00000003 left 'native_rtp' basic-bridge <6f08bd62-9e9e-4abc-9ed4-5a7e872e8f0d>
[Aug  4 08:32:19]   == Spawn extension (from-internal-custom, 6004, 1) exited non-zero on 'PJSIP/6002-00000002'
[Aug  4 08:32:33]     -- Executing [6003@from-internal-custom:1] Dial("PJSIP/6004-00000004", "PJSIP/6003") in new stack
[Aug  4 08:32:33]     -- Called PJSIP/6003
[Aug  4 08:32:33]     -- PJSIP/6003-00000005 is ringing
[Aug  4 08:32:38]     -- PJSIP/6003-00000005 answered PJSIP/6004-00000004
[Aug  4 08:32:38]     -- Channel PJSIP/6003-00000005 joined 'simple_bridge' basic-bridge <82145d89-1c6f-4a76-a2d2-d8a6ba7ba5a6>
[Aug  4 08:32:38]     -- Channel PJSIP/6004-00000004 joined 'simple_bridge' basic-bridge <82145d89-1c6f-4a76-a2d2-d8a6ba7ba5a6>
[Aug  4 08:32:52]     -- Channel PJSIP/6003-00000005 left 'native_rtp' basic-bridge <82145d89-1c6f-4a76-a2d2-d8a6ba7ba5a6>
[Aug  4 08:32:52]     -- Channel PJSIP/6004-00000004 left 'native_rtp' basic-bridge <82145d89-1c6f-4a76-a2d2-d8a6ba7ba5a6>
[Aug  4 08:32:52]   == Spawn extension (from-internal-custom, 6003, 1) exited non-zero on 'PJSIP/6004-00000004'
[Aug  4 08:36:53]     -- Removed contact 'sip:6003@192.168.1.91:13876;pn-provider=fcm;pn-param=com.mizuvoip.mizudroid.app;pn-prid=fuMoS2jORSyo

PJSIP/6004 is my landline phone and PJSIP/6003 is the softphone

There is no pjsip set logger on content.

how do I do that?

Run pjsip set logger on

Here is a full log from startup of the docker container and enabling logging for pjsip and first making a call from the softphone to the pap2 where sound is working, and afterwards making a call the otherway where sound dont work.