[asterisk-users] Voice broken during calls (again...)

[Luca Bertoncello]

Am 23.06.2020 um 17:04 schrieb Marek Greško:
> I interchanged LAN and LTE in the sentence.

OK...

> Do you have some kind of NAT in fron of asterisk? Or is your asterisk

No, Asterisk has a public IP. No NAT in front of Asterisk...

> having public IP? Could you share sip.conf (without passwords)? One
> LAN client, one LTE and general section.

Of course:

my outgoing configuration:
[pbxluca]
type=peer
defaultuser=<my username @telekom>
secret=<my very password>
dtmfmode=rfc2833
host=tel.t-online.de
context=luca_incoming
outboundproxy=tel.t-online.de
port=5060
fromuser=03511111111
fromdomain=tel.t-online.de
usereqphone=yes
canreinvite=yes
insecure=port,invite
nat=no
qualify=yes
qualifyfreq=600
disallow=all
allow=alaw
allow=ulaw

my phone configuration:
; Lucas Telefon
[00493511111111]
fullname = 00493511111111
secret = <very secret password>
hassip = yes
hasiax = no
hash323 = no
hasmanager = no
callwaiting = no
context = default
host = dynamic
dtmfmode=rfc2833
canreinvite=no
sendrpid=pai
type=friend
nat=force_rport,comedia
qualify=yes
qualifyfreq=60
avpf=no
force_avp=no
icesupport=no
encryption=no
callgroup=1
pickupgroup=1
deny=0.0.0.0/0.0.0.0
permit=192.168.200.0/255.255.255.0
dial=SIP/00493511111111

my mobile phone:
; Lucas Handy
[00491772222222]
fullname = 00491772222222
secret = <secret>
hassip = yes
hasiax = no
hash323 = no
hasmanager = no
callwaiting = no
context = default
host = dynamic
dtmfmode=rfc2833
canreinvite=no
sendrpid=pai
type=friend
nat=force_rport,comedia
qualify=yes
qualifyfreq=60
avpf=no
force_avp=no
icesupport=no
encryption=no
callgroup=1
pickupgroup=1
dial=SIP/00491772222222
allow = all

sip.conf:
[general]
context=default
allowoverlap=no
udpbindaddr=0.0.0.0:25572
tcpenable=yes
tcpbindaddr=0.0.0.0:25572
tlsenable=no
tlsbindaddr=0.0.0.0:25573
transport=udp
srvlookup=no
minexpiry=480
defaultexpiry=480
disallow=all
allow=alaw
allow=ulaw
allow=ilbc
allow=g729
allow=g723
allow=gsm
language=de
alwaysauthreject = yes
tlscertfile=/etc/asterisk/ssl/asterisk.pem
tlscafile=/etc/asterisk/ssl/ca.crt
tlscipher=ALL
tlsclientmethod=tlsv1
callcounter = yes
t38pt_udptl = yes
faxdetect = no
register =>
03511111111:xxxxxxxx:333333333333-0001 at t-online.de@pbxluca/00493511111111
register =>
03511111112:xxxxxxxx:333333333333-0001 at t-online.de@pbxfax/00493511111112
register =>
03511111113:xxxxxxxx:333333333333-0001 at t-online.de@pbxanika/00493511111113
register => 555555555:xxxxxxxx at messagenet/555555555
register => lucabertoncello:xxxxxxxxx at rebvoice/lucabertoncello
jbenable = no
jbmaxsize = 200
jbresyncthreshold = 1000
jbimpl = fixed

Thanks
Luca Bertoncello
(lucabert at lucabert.de)