No Audio when attended the call

Hi,
We are trying with asterisk 20.5. When we make a call from Soft phone “A1”, we could able to hear the ring. But when the other party ( with soft phone “A2”) attended the call, no audio is hearing from both sides. It was working last day. but now it is not working (without any changes). What could be the reason for this (we have enabled the required ports in this case)
please help.

Hi,
Please share the pjsip.conf and extension.conf configuration

Regards,
Partha

Thanks for the help.
Seems, this is intermittent. sometimes it work, sometimes not.

pjsip.conf

[transport-udp]
type=transport
protocol=udp
bind=0.0.0.0:5060
external_media_address=PUBLIC_IP_OF_SERVER
external_signaling_address=PUBLIC_IP_OF_SERVER
symmetric_transport=yes
;====================================================
; Lagos 
;====================================================
[auth200]
type=auth
username=200
password=200
auth_type=userpass

[200]
type=aor
max_contacts=2
qualify_frequency=60

[200]
type=endpoint
context=internal
auth=auth200
aors=200
rewrite_contact=yes
rtp_symmetric=yes
direct_media=no
disallow=all
allow=ulaw
allow=g722
transport=transport-udp


;Line 2 

[auth201]
type=auth
username=201
password=201
auth_type=userpass

[201]
type=aor
max_contacts=2
qualify_frequency=60

[201]
type=endpoint
context=internal
auth=auth201
aors=201
rewrite_contact=yes
rtp_symmetric=yes
direct_media=no
disallow=all
allow=ulaw
allow=g722
transport=transport-udp

extension.conf

[general]
static=yes
writeprotect=no
autofallthrough=yes


[internal]
exten => _NXX,1,Verbose(call was placed..)
exten => _NXX,n,Answer()
exten => _NXX,n,Dial(PJSIP/${EXTEN},60))
exten => _NXX,n,Hangup()

Hi,
You are using media server. Are you behind a firewall? Could you share the debug log .
Regards,
Partha

Sure, It is an intermittent issue. Next time when we get this issue, will collect the logs and share. Thanks.