Asterisk 14 - Connection breaks off at call acceptance

Hello,

I am just setting up a Asterisk 14 on a RaspberryPi3 with Debian Jessie. As a provider serves me the Telekom with their VoIP. In addition, I have for testing on my desktop Zoiper (6001) installed and registered at the Asterisk. My Asterisk can register with the Telekom. I can call numbers outside and my Zoiper client is also ringing. The connection is terminated only every time the other side or I lift off.
On my OpenWRT router is loaded nf-nat-sip and nf_conntrack_sip, as well as for RTP the Portrange 30000-310000 of 217.0.23.4 (tel.t-online.de) to 192.168.0.200 (Asterisk) opened. The same is configured in the rtp.conf.
Here is my pjsip.conf:
[global]
type=global
user_agent=Asterisk
endpoint_identifier_order=ip,username
default_from_user=rufnummer_mit_vorwahl

[transport-udp]
type=transport
protocol=udp
bind=0.0.0.0
local_net=192.168.0.0/24

[transport-tcp]
type=transport
protocol=tcp
bind=0.0.0.0
local_net=192.168.0.0/24

[telekom_rufnummer_mit_vorwahl]
type=registration
transport=transport-udp
outbound_auth=telekom_rufnummer_mit_vorwahl_auth
server_uri=sip:tel.t-online.de
client_uri=sip:+49rufnummer_mit_vorwahl@tel.t-online.de
contact_user=rufnummer_mit_vorwahl
retry_interval=60
forbidden_retry_interval=300
expiration=480
auth_rejection_permanent=false

[telekom_rufnummer_mit_vorwahl_auth]
type=auth
auth_type=userpass
password=secret
username=rufnummer_mit_vorwahl
realm=tel.t-online.de

[telekom_out]
type=endpoint
transport=transport-udp
context=unspecified
disallow=all
allow=g722
allow=alaw
outbound_auth=telekom_rufnummer_mit_vorwahl_auth
aors=telekom_out
callerid=rufnummer_mit_vorwahl
from_user=rufnummer_mit_vorwahl
from_domain=tel.t-online.de

[telekom_out]
type=aor
contact=sip:rufnummer_mit_vorwahl@tel.t-online.de

[telekom_in]
type=endpoint
transport=transport-udp
context=telekom_in
disallow=all
allow=g722
allow=alaw
outbound_auth=telekom_rufnummer_mit_vorwahl_auth

[telekom_in]
type=identify
endpoint=telekom_in
match=217.0.0.0/13

[6001]
type=endpoint
transport=transport-udp
context=internalsip
disallow=all
allow=g722
allow=alaw
auth=auth6001
aors=6001
mailboxes=wie in voicemail.conf definiert

[auth6001]
type=auth
auth_type=userpass
password=password
username=6001
realm=example.com

[6001]
type=aor
max_contacts=1
remove_existing=true

[6001]
type=identify
endpoint=6001
match=192.168.0.2

[acl]
type=acl
deny=0.0.0.0/0.0.0.0
; Telekom
permit=217.0.0.0/13
; eigenes LAN
permit=192.168.0.0/16

and the extension.conf:

[general]
static=yes
writeprotect=yes
autofallthrough=yes
extenpatternmatchnew=no
clearglobalvars=no
userscontext=unspecified

[unspecified]
; wer hier landet ist entweder schlecht konfiguriert oder hat keine "Rechte"

exten => _X.,1,Answer()
exten => _X.,2,Verbose(D E F A U L T ==> ${CALLERID(num)} kam um ${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)} in UNSPECIFIED an als er versuchte die Nummer ${EXTEN} anzurufen.)
exten => _X.,3,Hangup()

[internalsip]
; in den Kontext gelangt man, wenn man einen Call
; von den internen Telefonen startet

; direkt einzelne User anwaehlen
exten => contact_name1,1,Dial(PJSIP/contact_name1)
exten => contact_name2,1,Dial(PJSIP/contact_name2)

;Mailboxabfrage von intern ohne PIN
; exten => mailboxname,1,VoiceMailMain(mailboxname@mailboxcontext,s)

;national, mit +49 gewaehlt
exten => _+49X.,1,Dial(PJSIP/telekom_out/sip:0${EXTEN:3}@tel.t-online.de,60)
exten => _+49X.,n,Hangup()

;international lassen wir nicht zu
exten => _+X.,1,Hangup() 
exten => _00X.,1,Hangup() 

;national, mit 0 vorneweg
exten => _0Z.,1,Dial(PJSIP/telekom_out/sip:${EXTEN}@tel.t-online.de,60)
exten => _0Z.,n,Hangup() 

; Ortsnetz
exten => _Z.,1,Dial(PJSIP/telekom_out/sip:Ortsnetzkennzahl-mit-0${EXTEN}@tel.t-online.de,60)
exten => _Z.,n,Hangup() 

; Notrufe gehen immer
exten => 110,1,Dial(PJSIP/telekom_out/sip:110@tel.t-online.de,60)
exten => 110,n,Hangup() 
exten => 112,1,Dial(PJSIP/telekom_out/sip:112@tel.t-online.de,60)
exten => 112,n,Hangup()

[telekom_in] 
; Anrufe von extern via Telekom
; 30 Sekunden klingen
exten => rufnummer_mit_vorwahl,1,Dial(PJSIP/6001,30) 
; danach auf die Mailbox umleiten
; exten => rufnummer_mit_vorwahl,n,VoiceMail(mailboxname@mailboxcontext)
exten => rufnummer_mit_vorwahl,n,Hangup()

Additionally a short passage of the Asterisk Cli during a call:
[Dec 1 17:36:52] WARNING[1931]: res_pjsip_pubsub.c:639 subscription_get_handler_from_rdata: No registered subscribe handler for event presence.winfo
– Executing [rufnummer_mit_vorwahl@telekom_in:1] Dial(“PJSIP/telekom_in-00000008”, “PJSIP/6001,30”) in new stack
– Called PJSIP/6001
– PJSIP/6001-00000009 is ringing
== Spawn extension (telekom_in, rufnummer_mit_vorwahl, 1) exited non-zero on ‘PJSIP/telekom_in-00000008’
[Dec 1 17:42:16] WARNING[1941]: res_pjsip_pubsub.c:3085 pubsub_on_rx_publish_request: No registered publish handler for event presence
[Dec 1 17:42:16] WARNING[1942]: res_pjsip_pubsub.c:639 subscription_get_handler_from_rdata: No registered subscribe handler for event presence.winfo
– Executing [rufnummer_mit_vorwahl@telekom_in:1] Dial(“PJSIP/telekom_in-0000000a”, “PJSIP/6001,30”) in new stack
– Called PJSIP/6001
– PJSIP/6001-0000000b is ringing
== Spawn extension (telekom_in, rufnummer_mit_vorwahl, 1) exited non-zero on ‘PJSIP/telekom_in-0000000a’

Can someone help me, why the connections always abort immediately?

Best regards,

Diani

You would need to provide the output of “pjsip set logger on” to see what the SIP signaling is doing.