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[Problem] Telekom 15 Min Abbruch bei Ausgehend

December 8, 2016, 12:41 pm
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Hallo Leute,

ich stehe vor einem Problem, was mich aktuell wirklich verzweifeln lässt & ich hoffe Ihr könnt mir helfen.
Folgende Situation:

Asterisk 13.13.0 Server auf einer Debian-VM.
Sowohl I-Net Provider, als auch SIP-Provider ist die Deutsche Telekom.
Der Asterisk hängt hinter einem Router (NAT).

Ich habe exakt alle 15 Minuten Gesprächsabbrüche, wenn ich einen Kollegen, der einen Unitymedia Anschluss hat anrufe.
Wenn ich mein Handy anrufe (T-Mobile) habe ich keine Abbrüche.
Der Kollege hat eine ganz normale Standardkonfiguration (Fritzbox und direkt Telefone daran)

Rufe ich meinen Kollegen über mein Handy an, hält die Verbindung auch länger als 15 Min, es liegt also definitiv an mir (an meiner Asterisk Konfiguration)

Wenn mich allerdings mein Kollege von seinem Unitymedia anschluss anruft, haben wir keine Probleme, es ist wirklich nur, wenn ich ihn anrufe.

Meine SIP.conf :
Code:
[general]
session-timers=refuse
localnet=10.0.20.0/255.255.255.0
static=yes
writeprotect=no


udpbindaddr=10.0.20.12            ; IP address to bind UDP listen socket to (0.0.0.0 binds to all)
tcpenable=yes                    ; Enable server for incoming TCP connections (default is no)
tcpbindaddr=10.0.20.12            ; IP address for TCP server to bind to (0.0.0.0 binds to all interfaces)
transport=udp                  ; Set the default transports.  The order determines the primary default transport.
                                ; If tcpenable=no and the transport set is tcp, we will fallback to UDP.


srvlookup=yes                  ; Enable DNS SRV lookups on outbound calls
                                ; Note: Asterisk only uses the first host
                                ; in SRV records
                                ; Disabling DNS SRV lookups disables the
                                ; ability to place SIP calls based on domain
                                ; names to some other SIP users on the Internet
                                ; Specifying a port in a SIP peer definition or
                                ; when dialing outbound calls will supress SRV
                                ; lookups for that peer or call.


register => MEINE_TELEKOM_RUFNUMMER@tel.t-online.de/RUFNUMMER~480




[Telekom]
disallow=all
username=MEINE_TELEKOM_RUFNUMMER
type=peer
trustrpid=yes
sendrpid=yes
qualify=no
nat=force_rport,comedia
keepalive=30
insecure=port,invite
host=tel.t-online.de
fromdomain=tel.t-online.de
dtmfmode=rfc2833
directmedia=no
defaultexpiry=600
context=default
session-timers=refuse
allow=g722,alaw,ulaw



Ich habe ein Sip set debug on gemacht und erhalte nach 15 Minuten diese Meldungen:

Erklärung: 10.0.20.12 ist die IP von meinem Asterisk Server
Meine Rufnummer habe ich aus Datenschutzgründen auf: "MEINE_RUFNR_TELEKOM" umbenannt.
Die Rufnummer von meinem Kumpel der als Provider Unitymeida hat habe ich in "RUFNUMMER_UNITYMEDIA" umbenannt.

Ich hoffe Ihr könnt mir helfen, ich bin wirklich ratlos... !

Code:


ASTERISK*CLI>


<--- SIP read from UDP:217.0.23.100:5060 --->
INVITE sip:MEINE_RUFNR_TELEKOM@10.0.20.12:5060 SIP/2.0
Max-Forwards: 64
Via: SIP/2.0/UDP 217.0.23.100:5060;branch=z9hG4bKg3Zqkv7ih1exn2ewc9b6l4bz5udkg6sd4
To: "geforce28-phone" <sip:MEINE_RUFNR_TELEKOM@tel.t-online.de>;tag=as50d9b1fc
From: <sip:RUFNUMMER_UNITYMEDIA@tel.t-online.de>;tag=h7g4Esbg_p65548t1481227986m739317c1516376121s1_1140532393-736849955
Call-ID: 27e46154539f6322682232985a7e1ca3@tel.t-online.de
CSeq: 104 INVITE
Contact: <sip:sgc_c@217.0.23.100;transport=udp>;+g.3gpp.icsi-ref="urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel"
Min-Se: 900
Session-Expires: 1800;refresher=uac
Supported: timer
Supported: 100rel
Content-Type: application/sdp
Content-Length: 150
Allow: REGISTER, REFER, NOTIFY, SUBSCRIBE, INFO, PRACK, PUBLISH, UPDATE, INVITE, ACK, OPTIONS, CANCEL, BYE


v=0
o=- 798282883 1164185624 IN IP4 217.0.23.100
s=sip call
c=IN IP4 217.0.4.199
t=0 0
m=audio 3312 RTP/AVP 8
a=rtpmap:8 PCMA/8000
a=sendrecv
<------------->
--- (15 headers 8 lines) ---
Sending to 217.0.23.100:5060 (NAT)


<--- Transmitting (NAT) to 217.0.23.100:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 217.0.23.100:5060;branch=z9hG4bKg3Zqkv7ih1exn2ewc9b6l4bz5udkg6sd4;received=217.0.23.100;rport=5060
From: <sip:RUFNUMMER_UNITYMEDIA@tel.t-online.de>;tag=h7g4Esbg_p65548t1481227986m739317c1516376121s1_1140532393-736849955
To: "geforce28-phone" <sip:MEINE_RUFNR_TELEKOM@tel.t-online.de>;tag=as50d9b1fc
Call-ID: 27e46154539f6322682232985a7e1ca3@tel.t-online.de
CSeq: 104 INVITE
Server: Asterisk PBX 13.13.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces
Contact: <sip:MEINE_RUFNR_TELEKOM@10.0.20.12:5060>
Content-Length: 0




<------------>
Audio is at 10156
Adding codec alaw to SDP
Adding codec g722 to SDP
Adding codec ulaw to SDP


<--- Reliably Transmitting (NAT) to 217.0.23.100:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 217.0.23.100:5060;branch=z9hG4bKg3Zqkv7ih1exn2ewc9b6l4bz5udkg6sd4;received=217.0.23.100;rport=5060
From: <sip:RUFNUMMER_UNITYMEDIA@tel.t-online.de>;tag=h7g4Esbg_p65548t1481227986m739317c1516376121s1_1140532393-736849955
To: "geforce28-phone" <sip:MEINE_RUFNR_TELEKOM@tel.t-online.de>;tag=as50d9b1fc
Call-ID: 27e46154539f6322682232985a7e1ca3@tel.t-online.de
CSeq: 104 INVITE
Server: Asterisk PBX 13.13.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces
Contact: <sip:MEINE_RUFNR_TELEKOM@10.0.20.12:5060>
Content-Type: application/sdp
Content-Length: 226


v=0
o=root 187727919 187727920 IN IP4 10.0.20.12
s=Asterisk PBX 13.13.0
c=IN IP4 10.0.20.12
t=0 0
m=audio 10156 RTP/AVP 8 9 0
a=rtpmap:8 PCMA/8000
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=maxptime:150
a=sendrecv


<------------>


<--- SIP read from UDP:217.0.23.100:5060 --->
ACK sip:MEINE_RUFNR_TELEKOM@10.0.20.12:5060 SIP/2.0
Max-Forwards: 64
Via: SIP/2.0/UDP 217.0.23.100:5060;branch=z9hG4bKg3Zqkv7i5liilv6uh34trf1vdc739bg6x
To: "geforce28-phone" <sip:MEINE_RUFNR_TELEKOM@tel.t-online.de>;tag=as50d9b1fc
From: <sip:RUFNUMMER_UNITYMEDIA@tel.t-online.de>;tag=h7g4Esbg_p65548t1481227986m739317c1516376121s1_1140532393-736849955
Call-ID: 27e46154539f6322682232985a7e1ca3@tel.t-online.de
CSeq: 104 ACK
Contact: <sip:sgc_c@217.0.23.100;transport=udp>;+g.3gpp.icsi-ref="urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel"
Content-Length: 0


<------------->
--- (9 headers 0 lines) ---


<--- SIP read from UDP:217.0.23.100:5060 --->
OPTIONS sip:MEINE_RUFNR_TELEKOM@10.0.20.12:5060 SIP/2.0
Max-Forwards: 70
Via: SIP/2.0/UDP 217.0.23.100:5060;branch=z9hG4bKg3Zqkv7i52einpm8k1163solodmbpcgng
To: <sip:MEINE_RUFNR_TELEKOM@tel.t-online.de>;tag=as50d9b1fc
From: <sip:RUFNUMMER_UNITYMEDIA@tel.t-online.de>;tag=h7g4Esbg_p65548t1481227986m739317c1516376121s1_1140532393-736849955
Call-ID: 27e46154539f6322682232985a7e1ca3@tel.t-online.de
CSeq: 105 OPTIONS
Content-Length: 0


<------------->
--- (8 headers 0 lines) ---


<--- Transmitting (NAT) to 217.0.23.100:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 217.0.23.100:5060;branch=z9hG4bKg3Zqkv7i52einpm8k1163solodmbpcgng;received=217.0.23.100;rport=5060
From: <sip:RUFNUMMER_UNITYMEDIA@tel.t-online.de>;tag=h7g4Esbg_p65548t1481227986m739317c1516376121s1_1140532393-736849955
To: <sip:MEINE_RUFNR_TELEKOM@tel.t-online.de>;tag=as50d9b1fc
Call-ID: 27e46154539f6322682232985a7e1ca3@tel.t-online.de
CSeq: 105 OPTIONS
Server: Asterisk PBX 13.13.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces
Contact: <sip:MEINE_RUFNR_TELEKOM@10.0.20.12:5060>
Accept: application/sdp
Content-Length: 0




<------------>
ASTERISK*CLI>
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