In beide Richtungen nichts zu hören

Sidebar Sidebar
Upgrade 3CX to v18 and get it hosted free!
D

D-Trix

Neuer User
Mitglied seit
11 Jul 2005
Beiträge
127
Punkte für Reaktionen
0
Punkte
0
Ich habe mich heute bei Sipnetworks.de (ehemals Interfoni) angemeldet.

Registerd ist der Account bereits im asterisk:
> sip show registry
> Host Username Refresh State
> siplogin.de:5060 49180xxxxxxx 1185 Registered

Bei Anrufversuchen klingelt es zwar, aber nach abheben ist in beiden Richtungen nichts zu hören. Auf den ersten Blick scheint das ja ein Firewall oder NAT- problem zu sein, jedoch funktonieren Sipgate und Sipsnip bereits einwandfrei. Im Sip debug Modus fällt jedoch die Retransmitting Meldung ins Auge:

Retransmitting #7 (no NAT) to 80.237.199.17:5060:
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.0.160:5060;branch=z9hG4bK2405cb88;rport
From: "ABC" <sip:[email protected]>;tag=as1a40237a
To: <sip:[email protected]>
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Tue, 11 Oct 2005 19:04:44 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Content-Type: application/sdp
Content-Length: 265


aktueller (12.Okt) Auszug der sip.conf:
Code: 
[general]
port = 5060
bindaddr = 192.168.0.160
;externip = 212.xxx.xxx.xxx
localnet = 192.168.0.0/255.255.255.0 :local network and mask
srvlookup = yes
context = default
language=de
;Uebertragunsart
disallow=all
;allow=ilbc
allow=alaw
allow=ulaw
allow=g726
allow=gsm
allow=slinear
canreinvite=no ;yes -only works if all devices are on the same side of the NAT
insecure=very
nat=yes
dtmfmode=inband
;tos = reliability
tos=0x18
maxexpirey = 3600
defaultexpirey = 1200
register => SIP-Account-ID:[email protected]/SIP-Account-ID 

[SIP-Account-ID]
type=friend
username=SIP-Account-ID
fromuser=SIP-Account-ID
secret=SIP-Passwort
host=siplogin.de
fromdomain=siplogin.de
insecure=very
nat=no

Auszug der sip.conf (alt):
Code: 
[general]
port = 5060
bindaddr = 0.0.0.0
;externip = 212.xxx.xxx.xxx
localnet = 192.168.0.0/255.255.255.0 :local network and mask
srvlookup = yes
context = default
language=de
;Uebertragunsart
disallow=all
;allow=ilbc
allow=alaw
allow=ulaw
allow=g726
allow=gsm
allow=slinear
canreinvite=no ;yes -only works if all devices are on the same side of the NAT
insecure=very
nat=yes
dtmfmode=inband
;tos = reliability
tos=0x18
maxexpirey = 3600
defaultexpirey = 1200
register => 49180xxxxx:[email protected]/sipnetworks



[sipnetworksout]
type=friend
username=49180xxxxxxxxx
secret=xxxxxxxxx
host=siplogin.de
fromuser=49180xxxxxxxxxx
;fromdomain=siplogin.de
context=default
canreinvite=no
qualify=no
;disallow=all
disallow=ilbc
;allow=alaw
;allow=gsm
insecure=very
nat=no
tos=reliability


Was bedeutet das Retransmitting überhaupt und wie kann ich das vermeiden?
 
wieso setzt Du im [general] nat=yes wenn Du es im [sipnetworksout] dann wieder abschaltest ? Bist Du denn nun hinter NAT oder nicht ?
 
das stammt noch aus meinen unzähligen probierversuchen. bei sipsnip und sipgate klappt es mit nat=no. ich bin hinter einem isa server 2004 und weiss offen gestanden nicht ob da NAT praktiziert wird oder nicht *schäm* eigentlich müsste es aber doch, da ich mehr pcs im netzwerk habe, als externe IPs?
 
Hier noch mal ein SIP-Log von einem "auf-beiden-seiten-lautlos-Gespräch":

Code: 
    -- Executing Dial("CAPI/contr1/99-1", "SIP/0238xxxxxx@sipnetworksout|60") in new stack


We're at 192.168.0.160 port 14516
Answering/Requesting with root capability 0x8 (alaw)
Answering with preferred capability 0x4 (ulaw)
Answering with preferred capability 0x10 (g726)
Answering with preferred capability 0x2 (gsm)
Answering with preferred capability 0x40 (slin)
12 headers, 12 lines

Reliably Transmitting (no NAT) to 80.237.199.17:5060:
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.0.160:5060;branch=z9hG4bK7d72a53b;rport
From: "AB" <sip:[email protected]>;tag=as5405c760
To: <sip:[email protected]>
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Wed, 12 Oct 2005 12:53:59 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Content-Type: application/sdp
Content-Length: 265

v=0
o=root 15867 15867 IN IP4 192.168.0.160
s=session
c=IN IP4 192.168.0.160
t=0 0
m=audio 14516 RTP/AVP 8 0 111 3 10
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:3 GSM/8000
a=rtpmap:10 L16/8000
a=silenceSupp:off - - - -

---
    -- Called 0238xxxxxxx@sipnetworksout
Retransmitting #1 (no NAT) to 80.237.199.17:5060:
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.0.160:5060;branch=z9hG4bK7d72a53b;rport
From: "AB" <sip:[email protected]>;tag=as5405c760
To: <sip:[email protected]>
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Wed, 12 Oct 2005 12:53:59 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Content-Type: application/sdp
Content-Length: 265

v=0
o=root 15867 15867 IN IP4 192.168.0.160
s=session
c=IN IP4 192.168.0.160
t=0 0
m=audio 14516 RTP/AVP 8 0 111 3 10
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:3 GSM/8000
a=rtpmap:10 L16/8000
a=silenceSupp:off - - - -

---
Retransmitting #2 (no NAT) to 80.237.199.17:5060:
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.0.160:5060;branch=z9hG4bK7d72a53b;rport
From: "AB" <sip:[email protected]>;tag=as5405c760
To: <sip:[email protected]>
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Wed, 12 Oct 2005 12:53:59 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Content-Type: application/sdp
Content-Length: 265

v=0
o=root 15867 15867 IN IP4 192.168.0.160
s=session
c=IN IP4 192.168.0.160
t=0 0
m=audio 14516 RTP/AVP 8 0 111 3 10
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:3 GSM/8000
a=rtpmap:10 L16/8000
a=silenceSupp:off - - - -

---
Retransmitting #3 (no NAT) to 80.237.199.17:5060:
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.0.160:5060;branch=z9hG4bK7d72a53b;rport
From: "AB" <sip:[email protected]>;tag=as5405c760
To: <sip:[email protected]>
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Wed, 12 Oct 2005 12:53:59 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Content-Type: application/sdp
Content-Length: 265

v=0
o=root 15867 15867 IN IP4 192.168.0.160
s=session
c=IN IP4 192.168.0.160
t=0 0
m=audio 14516 RTP/AVP 8 0 111 3 10
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:3 GSM/8000
a=rtpmap:10 L16/8000
a=silenceSupp:off - - - -

---
Retransmitting #4 (no NAT) to 80.237.199.17:5060:
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.0.160:5060;branch=z9hG4bK7d72a53b;rport
From: "AB" <sip:[email protected]>;tag=as5405c760
To: <sip:[email protected]>
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Wed, 12 Oct 2005 12:53:59 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Content-Type: application/sdp
Content-Length: 265

v=0
o=root 15867 15867 IN IP4 192.168.0.160
s=session
c=IN IP4 192.168.0.160
t=0 0
m=audio 14516 RTP/AVP 8 0 111 3 10
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:3 GSM/8000
a=rtpmap:10 L16/8000
a=silenceSupp:off - - - -

---
Retransmitting #5 (no NAT) to 80.237.199.17:5060:
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.0.160:5060;branch=z9hG4bK7d72a53b;rport
From: "AB" <sip:[email protected]>;tag=as5405c760
To: <sip:[email protected]>
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Wed, 12 Oct 2005 12:53:59 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Content-Type: application/sdp
Content-Length: 265

v=0
o=root 15867 15867 IN IP4 192.168.0.160
s=session
c=IN IP4 192.168.0.160
t=0 0
m=audio 14516 RTP/AVP 8 0 111 3 10
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:3 GSM/8000
a=rtpmap:10 L16/8000
a=silenceSupp:off - - - -

---
Retransmitting #6 (no NAT) to 80.237.199.17:5060:
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.0.160:5060;branch=z9hG4bK7d72a53b;rport
From: "AB" <sip:[email protected]>;tag=as5405c760
To: <sip:[email protected]>
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Wed, 12 Oct 2005 12:53:59 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Content-Type: application/sdp
Content-Length: 265

v=0
o=root 15867 15867 IN IP4 192.168.0.160
s=session
c=IN IP4 192.168.0.160
t=0 0
m=audio 14516 RTP/AVP 8 0 111 3 10
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:3 GSM/8000
a=rtpmap:10 L16/8000
a=silenceSupp:off - - - -

---
Reliably Transmitting (no NAT) to 80.237.199.17:5060:
CANCEL sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.0.160:5060;branch=z9hG4bK7d72a53b;rport
From: "AB" <sip:[email protected]>;tag=as5405c760
To: <sip:[email protected]>
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 CANCEL
User-Agent: Asterisk PBX
Content-Length: 0


---
Scheduling destruction of call '[email protected]' in 15000 ms
    -- CAPI Hangingup
    -- removed pipe for PLCI = 0x2501
Retransmitting #1 (no NAT) to 80.237.199.17:5060:
CANCEL sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.0.160:5060;branch=z9hG4bK7d72a53b;rport
From: "AB" <sip:[email protected]>;tag=as5405c760
To: <sip:[email protected]>
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 CANCEL
User-Agent: Asterisk PBX
Content-Length: 0


---

<-- SIP read from 80.237.199.17:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.160:5060;received=212.132.194.xx;received=212.132.194.xx;branch=z9hG4bK7d72a53b;rport=58350
From: "AB" <sip:[email protected]>;tag=as5405c760
To: <sip:[email protected]>;tag=as461cef1f
Call-ID: [email protected]
CSeq: 102 CANCEL
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Contact: <sip:[email protected]>
Content-Length: 0
 
Hier das ganze nochmal mit nat=yes :

Code: 
-- Executing Dial("CAPI/contr1/99-0", "SIP/0238xxxxxxx@sipnetworksout|60") in new stack
We're at 192.168.0.160 port 15834
Answering/Requesting with root capability 0x8 (alaw)
Answering with preferred capability 0x4 (ulaw)
Answering with preferred capability 0x10 (g726)
Answering with preferred capability 0x2 (gsm)
Answering with preferred capability 0x40 (slin)
12 headers, 12 lines
Reliably Transmitting (NAT) to 80.237.199.17:5060:
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.0.160:5060;branch=z9hG4bK4dd9229c;rport
From: "AB" <sip:[email protected]>;tag=as5ab37a23
To: <sip:[email protected]>
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Wed, 12 Oct 2005 13:04:57 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Content-Type: application/sdp
Content-Length: 265

v=0
o=root 15916 15916 IN IP4 192.168.0.160
s=session
c=IN IP4 192.168.0.160
t=0 0
m=audio 15834 RTP/AVP 8 0 111 3 10
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:3 GSM/8000
a=rtpmap:10 L16/8000
a=silenceSupp:off - - - -

---
    -- Called 0238xxxxxxx@sipnetworksout
Retransmitting #1 (NAT) to 80.237.199.17:5060:
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.0.160:5060;branch=z9hG4bK4dd9229c;rport
From: "AB" <sip:[email protected]>;tag=as5ab37a23
To: <sip:[email protected]>
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Wed, 12 Oct 2005 13:04:57 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Content-Type: application/sdp
Content-Length: 265

v=0
o=root 15916 15916 IN IP4 192.168.0.160
s=session
c=IN IP4 192.168.0.160
t=0 0
m=audio 15834 RTP/AVP 8 0 111 3 10
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:3 GSM/8000
a=rtpmap:10 L16/8000
a=silenceSupp:off - - - -

---
Retransmitting #2 (NAT) to 80.237.199.17:5060:
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.0.160:5060;branch=z9hG4bK4dd9229c;rport
From: "AB" <sip:[email protected]>;tag=as5ab37a23
To: <sip:[email protected]>
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Wed, 12 Oct 2005 13:04:57 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Content-Type: application/sdp
Content-Length: 265

v=0
o=root 15916 15916 IN IP4 192.168.0.160
s=session
c=IN IP4 192.168.0.160
t=0 0
m=audio 15834 RTP/AVP 8 0 111 3 10
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:3 GSM/8000
a=rtpmap:10 L16/8000
a=silenceSupp:off - - - -

---
Retransmitting #3 (NAT) to 80.237.199.17:5060:
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.0.160:5060;branch=z9hG4bK4dd9229c;rport
From: "AB" <sip:[email protected]>;tag=as5ab37a23
To: <sip:[email protected]>
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Wed, 12 Oct 2005 13:04:57 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Content-Type: application/sdp
Content-Length: 265

v=0
o=root 15916 15916 IN IP4 192.168.0.160
s=session
c=IN IP4 192.168.0.160
t=0 0
m=audio 15834 RTP/AVP 8 0 111 3 10
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:3 GSM/8000
a=rtpmap:10 L16/8000
a=silenceSupp:off - - - -

---
Retransmitting #4 (NAT) to 80.237.199.17:5060:
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.0.160:5060;branch=z9hG4bK4dd9229c;rport
From: "AB" <sip:[email protected]>;tag=as5ab37a23
To: <sip:[email protected]>
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Wed, 12 Oct 2005 13:04:57 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Content-Type: application/sdp
Content-Length: 265

v=0
o=root 15916 15916 IN IP4 192.168.0.160
s=session
c=IN IP4 192.168.0.160
t=0 0
m=audio 15834 RTP/AVP 8 0 111 3 10
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:3 GSM/8000
a=rtpmap:10 L16/8000
a=silenceSupp:off - - - -

---
Retransmitting #5 (NAT) to 80.237.199.17:5060:
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.0.160:5060;branch=z9hG4bK4dd9229c;rport
From: "AB" <sip:[email protected]>;tag=as5ab37a23
To: <sip:[email protected]>
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Wed, 12 Oct 2005 13:04:57 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Content-Type: application/sdp
Content-Length: 265

v=0
o=root 15916 15916 IN IP4 192.168.0.160
s=session
c=IN IP4 192.168.0.160
t=0 0
m=audio 15834 RTP/AVP 8 0 111 3 10
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:111 G726-32/8000
a=rtpmap:3 GSM/8000
a=rtpmap:10 L16/8000
a=silenceSupp:off - - - -

---
Reliably Transmitting (NAT) to 80.237.199.17:5060:
CANCEL sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.0.160:5060;branch=z9hG4bK4dd9229c;rport
From: "AB" <sip:[email protected]>;tag=as5ab37a23
To: <sip:[email protected]>
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 CANCEL
User-Agent: Asterisk PBX
Content-Length: 0


---
Scheduling destruction of call '[email protected]' in 15000 ms
    -- CAPI Hangingup
    -- removed pipe for PLCI = 0x2601

<-- SIP read from 80.237.199.17:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.160:5060;received=212.132.194.xx;received=212.132.194.xx;branch=z9hG4bK4dd9229c;rport=58266
From: "AB" <sip:[email protected]>;tag=as5ab37a23
To: <sip:[email protected]>;tag=as1a790d3e
Call-ID: [email protected]
CSeq: 102 CANCEL
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY
Contact: <sip:[email protected]>
Content-Length: 0


--- (10 headers 0 lines)---


Werden noch weitere Informationen gebraucht?


/edit: ich habs jetzt mal in der anderen Richtung versucht, ruft man über die sipnetworks nummer (1805xxxxxxxxxxx) Asterisk an, kann man den anrufer hören, umgekehrt jedoch nix
 
D-Trix schrieb:
Ich habe mich heute bei Sipnetworks.de (ehemals Interfoni) angemeldet.
Zum Vergrößern anklicken....

Nunja, eigentlich gibts den Laden schon länger als Interfoni ... wollte ich nur mal bemerken.

Ich kenn mich zwar nicht so mit Asterisk aus, aber die einseitige Sprachverbindung hört sich für mich irgendwie nach einem NAT-Problem an.
 
Um antworten zu können musst du eingeloggt sein.

Zurzeit aktive Besucher

Gesamt: 528 (Mitglieder: 26, Gäste: 502)

Neueste Beiträge

Statistik des Forums

Themen
244,862
Beiträge
2,219,699
Mitglieder
371,579
Neuestes Mitglied
frederik90
Holen Sie sich 3CX - völlig kostenlos!
Verbinden Sie Ihr Team und Ihre Kunden Telefonie Livechat Videokonferenzen

Gehostet oder selbst-verwaltet. Für bis zu 10 Nutzer dauerhaft kostenlos. Keine Kreditkartendetails erforderlich. Ohne Risiko testen.

3CX
Für diese E-Mail-Adresse besteht bereits ein 3CX-Konto. Sie werden zum Kundenportal weitergeleitet, wo Sie sich anmelden oder Ihr Passwort zurücksetzen können, falls Sie dieses vergessen haben.
Ihr E-Mail-Adresse verifizieren

Ihr E-Mail-Adresse verifizieren

Wir haben Ihnen eine E-Mail geschickt. Klicken Sie auf die Schaltfläche im E-Mail-Text, um Ihre E-Mail-Adresse zu verifizieren – (Können Sie diese nicht finden können, dann überprüfen Sie Ihren Spam-Ordner).

IPPF im Überblick

Neueste Beiträge

Direktlinks Telefonanlage

Website-Sponsoren

3CX sponsor
KONTAKTIEREN SIE UNS BEI INTERESSE
| Style by ThemeHouse
XenPorta 2 PRO © Jason Axelrod of 8WAYRUN
Oben Unten
Zurück