Festnetz über Freenet: * ignoriert RTP von Freenet

saddamski

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Hallo zusammen!
Ich habe leider ein Problem mit * im Zusammenhang mit dem Freenet ISDN Gateway.

Der Aufbau ist folgender:
DECT Telefon -> isdn -> ITA -> ethernet -> Asterisk -> ethernet -> Router ->Internet -> Freenet -> isdn -> Mein Handy.

Dabei funktioniert die Audioverbindung von Handy zu DECT nicht. Die andere Richtung funktioniert tadellos. Das interessante dabei ist, dass die RTP Daten von Freenet aber den Asterisk erreichen. Nur scheint dieser die zu verwerfen bzw. zu ignorieren.
Ansonsten funktioniert alles. Auch Gespräche über das Internet funktionieren. Mit Sipgate habe ich z.B. keine Probleme. Was mir dabei aufgefallen ist, dass bei Sipgate die IP von SIP Proxy und ISDN Gateway gleich sind. Bei Freenet sind diese verschieden. Ob das ein Grund ist?
Ich benutze die Version 1.2.7.1 auf einer Debian Kiste.
Habe auch mal im Bugtracker gesucht, allerdings nix passendes gefunden. Wobei ich mir da auch sicher bin dass ich nach evtl. nach den falschen Wörtern gesucht habe.


Hier mal die ganzen Einstellungen und Logs

tcpdump
Code:
13:41:59.660438 IP 62.104.212.131.19024 > 192.168.0.3.12044: UDP, length 172
13:41:59.668426 IP 192.168.0.6.5010 > 192.168.0.3.12024: UDP, length 172
13:41:59.668621 IP 192.168.0.3.12044 > 62.104.212.131.19024: UDP, length 172
13:41:59.680816 IP 62.104.212.131.19024 > 192.168.0.3.12044: UDP, length 172
13:41:59.688520 IP 192.168.0.6.5010 > 192.168.0.3.12024: UDP, length 172
13:41:59.688691 IP 192.168.0.3.12044 > 62.104.212.131.19024: UDP, length 172
13:41:59.695860 IP 62.104.212.131.19024 > 192.168.0.3.12044: UDP, length 172
13:41:59.708596 IP 192.168.0.6.5010 > 192.168.0.3.12024: UDP, length 172
13:41:59.708789 IP 192.168.0.3.12044 > 62.104.212.131.19024: UDP, length 172
13:41:59.715952 IP 62.104.212.131.19024 > 192.168.0.3.12044: UDP, length 172
13:41:59.728497 IP 192.168.0.6.5010 > 192.168.0.3.12024: UDP, length 172
13:41:59.728682 IP 192.168.0.3.12044 > 62.104.212.131.19024: UDP, length 172
13:41:59.736311 IP 62.104.212.131.19024 > 192.168.0.3.12044: UDP, length 172
13:41:59.748371 IP 192.168.0.6.5010 > 192.168.0.3.12024: UDP, length 172
13:41:59.748566 IP 192.168.0.3.12044 > 62.104.212.131.19024: UDP, length 172
13:41:59.756083 IP 62.104.212.131.19024 > 192.168.0.3.12044: UDP, length 172
13:41:59.768400 IP 192.168.0.6.5010 > 192.168.0.3.12024: UDP, length 172
13:41:59.768597 IP 192.168.0.3.12044 > 62.104.212.131.19024: UDP, length 172
13:41:59.776580 IP 62.104.212.131.19024 > 192.168.0.3.12044: UDP, length 172
13:41:59.788487 IP 192.168.0.6.5010 > 192.168.0.3.12024: UDP, length 172
13:41:59.788657 IP 192.168.0.3.12044 > 62.104.212.131.19024: UDP, length 172
13:41:59.796765 IP 62.104.212.131.19024 > 192.168.0.3.12044: UDP, length 172
rtp debug ip 62.104.212.131 in der Asterisl CLI zeigt:
Code:
Sent RTP packet to 62.104.212.131:19024 (type 8, seq 46095, ts 372000, len 160)
Sent RTP packet to 62.104.212.131:19024 (type 8, seq 46096, ts 372160, len 160)
Sent RTP packet to 62.104.212.131:19024 (type 8, seq 46097, ts 372320, len 160)
Sent RTP packet to 62.104.212.131:19024 (type 8, seq 46098, ts 372480, len 160)
Sent RTP packet to 62.104.212.131:19024 (type 8, seq 46099, ts 372640, len 160)
Sent RTP packet to 62.104.212.131:19024 (type 8, seq 46100, ts 372800, len 160)
Sent RTP packet to 62.104.212.131:19024 (type 8, seq 46101, ts 372960, len 160)
Sent RTP packet to 62.104.212.131:19024 (type 8, seq 46102, ts 373120, len 160)
Sent RTP packet to 62.104.212.131:19024 (type 8, seq 46103, ts 373280, len 160)
Sent RTP packet to 62.104.212.131:19024 (type 8, seq 46104, ts 373440, len 160)
Sent RTP packet to 62.104.212.131:19024 (type 8, seq 46105, ts 373600, len 160)
Sent RTP packet to 62.104.212.131:19024 (type 8, seq 46106, ts 373760, len 160)
Sent RTP packet to 62.104.212.131:19024 (type 8, seq 46107, ts 373920, len 160)
Sent RTP packet to 62.104.212.131:19024 (type 8, seq 46108, ts 374080, len 160)
sip show channels während das Gespräch aufgebaut ist:
Code:
Peer             User/ANR    Call ID      Seq (Tx/Rx)  Form  Hold     Last Message
194.97.54.97     0173xxxxxx  5d84bf0d37b  00103/00000  alaw  No       Tx: ACK
192.168.0.6      132         0017364b04-  00101/05889  alaw  No       Rx: ACK
Die Ausgabe in der CLI:
Code:
    -- Executing Dial("SIP/132-b3af", "SIP/[email protected]_THO|45|") in new stack
    -- Called [email protected]_THO
    -- SIP/Freenet_THO-cc1e is ringing
    -- SIP/Freenet_THO-cc1e is making progress passing it to SIP/132-b3af
    -- SIP/Freenet_THO-cc1e answered SIP/132-b3af
    -- Attempting native bridge of SIP/132-b3af and SIP/Freenet_THO-cc1e
  == Spawn extension (Tel132, 0173xxxxxxx, 1) exited non-zero on 'SIP/132-b3af'
sip.conf
Code:
[general]
context=default           
bindport=5060             
bindaddr=0.0.0.0          
srvlookup=yes             
defaultexpirey=1800
language=de
disallow=all
allow=alaw
allow=ulaw
externhost=xxx.dyndns.org
localnet=192.168.0.0/255.255.255.0
nat=yes
;---------------------------------------------------------
;Sipgate xxxx
register => 65XXXXX:xxxsipgate.de/Sipgate_TVA

;Sipgate xxxx
register => 19xxxx:[email protected]/Sipgate_THO

;Freenet xxx
register => Elxxxxxx:[email protected]/Freenet_THO

[Sipgate_TVA]
type=peer
username=65xxxx
fromuser=65xxxx
secret=xxx
host=sipgate.de
fromdomain=sipgate.de
insecure=very
canreinvite=no
nat=yes

[Sipgate_THO]
type=peer
username=19xxx
fromuser=19xxx
secret=xxxx
host=sipgate.de
fromdomain=sipgate.de
insecure=very
canreinvite=no
nat=yes

[Freenet_THO]
type=friend
username=Elxxxx
fromuser=Elxxx
secret=xxx
host=freenet.de
fromdomain=freenet.de
insecure=very
canreinvite=no
nat=yes
port=5060

[freenet_in]
type=friend
fromdomain=freenet.de
host=freenet.de
context=ankommend
nat=yes
;qualify=yes
canreinvite=no

Hier folgen dann noch diverse Endgeräte
Ausschnitt aus der extensions.conf
Code:
[freenet_out_THO]
exten => _0XXX.,1,Dial,SIP/${EXTEN}@Freenet_THO|45|
sip debug peer Freenet_THO
Code:
We're at 80.130.237.89 port 12048
Adding codec 0x8 (alaw) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
13 headers, 11 lines
Reliably Transmitting (NAT) to 194.97.54.97:5060:
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 80.130.237.89:5060;branch=z9hG4bK5f40b4e9;rport
From: "131" <sip:[email protected]>;tag=as1566d63f
To: <sip:[email protected]>
Contact: <sip:[email protected]>
Call-ID: [email protected]
Seq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 19 Sep 2006 11:58:14 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 240

v=0
o=root 3018 3018 IN IP4 80.130.237.89
s=session
c=IN IP4 80.130.237.89
t=0 0
m=audio 12048 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
    -- Called [email protected]_THO
asterisk*CLI>
<-- SIP read from 194.97.54.97:5060:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 80.130.237.89:5060;branch=z9hG4bK5f40b4e9;rport=5060
From: "131" <sip:[email protected]>;tag=as1566d63f
To: <sip:[email protected]>;tag=36a431905c116a7e58f13c316b42f021.3910
Call-ID: [email protected]
CSeq: 102 INVITE
Proxy-Authenticate: Digest realm="freenet.de", nonce="450fdc81e3797e48848abf485651594b42f30064", qop="auth"
Content-Length: 0


--- (8 headers 0 lines)---
Transmitting (NAT) to 194.97.54.97:5060:
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 80.130.237.89:5060;branch=z9hG4bK5f40b4e9;rport
From: "131" <sip:[email protected]>;tag=as1566d63f
To: <sip:[email protected]>;tag=36a431905c116a7e58f13c316b42f021.3910
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
We're at 80.130.237.89 port 12048
Adding codec 0x8 (alaw) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 194.97.54.97:5060:
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 80.130.237.89:5060;branch=z9hG4bK54e9824c;rport
From: "131" <sip:[email protected]>;tag=as1566d63f
To: <sip:[email protected]>
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Proxy-Authorization: Digest username="Elxxxx", realm="freenet.de", algorithm=MD5, uri="sip:[email protected]", nonce="xxxxxx", response="xxxxxxx", opaque="", qop=auth, cnonce="1b7c5116", nc=00000001
Date: Tue, 19 Sep 2006 11:58:14 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 240

v=0
o=root 3018 3019 IN IP4 80.130.237.89
s=session
c=IN IP4 80.130.237.89
t=0 0
m=audio 12048 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
asterisk*CLI>
<-- SIP read from 194.97.54.97:5060:
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP 80.130.237.89:5060;branch=z9hG4bK54e9824c;rport=5060
From: "131" <sip:[email protected]>;tag=as1566d63f
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 103 INVITE
Content-Length: 0


--- (7 headers 0 lines)---
asterisk*CLI>
<-- SIP read from 194.97.54.97:5060:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 80.130.237.89:5060;branch=z9hG4bK54e9824c;rport=5060
From: "131" <sip:[email protected]>;tag=as1566d63f
To: <sip:[email protected]>;tag=008082384A4502248FAE00000F77
Call-ID: [email protected]
CSeq: 103 INVITE
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, INFO, REFER, NOTIFY, SUBSCRIBE, UPDATE
Content-Type: application/sdp
Content-Length:   183

v=0
o=- 3106003 0 IN IP4 62.104.212.131
s=session
c=IN IP4 62.104.212.131
t=0 0
m=audio 19192 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=sendrecv

--- (9 headers 9 lines)---
Found RTP audio format 8
Found RTP audio format 101
Peer audio RTP is at port 62.104.212.131:19192
Found description format PCMA
Found description format telephone-event
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
    -- SIP/Freenet_THO-af80 is ringing
    -- SIP/Freenet_THO-af80 is making progress passing it to SIP/132-b6e5
asterisk*CLI>
<-- SIP read from 194.97.54.97:5060:
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 80.130.237.89:5060;branch=z9hG4bK54e9824c;rport=5060
Record-Route: <sip:194.97.59.151;ftag=as1566d63f;lr=on>
Record-Route: <sip:194.97.54.97;ftag=as1566d63f;lr=on>
From: "131" <sip:[email protected]>;tag=as1566d63f
To: <sip:[email protected]>;tag=008082384A4502248FAE00000F77
Call-ID: [email protected]
CSeq: 103 INVITE
Contact: <sip:195.4.21.244:5060>
Allow-Events: refer
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, INFO, REFER, NOTIFY, SUBSCRIBE, UPDATE
Content-Type: application/sdp
Supported: 100rel, timer, replaces
Content-Length:   183

v=0
o=- 3106003 0 IN IP4 62.104.212.131
s=session
c=IN IP4 62.104.212.131
t=0 0
m=audio 19192 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=sendrecv

--- (14 headers 9 lines)---
Found RTP audio format 8
Found RTP audio format 101
Peer audio RTP is at port 62.104.212.131:19192
Found description format PCMA
Found description format telephone-event
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
list_route: hop: <sip:194.97.54.97;ftag=as1566d63f;lr=on>
list_route: hop: <sip:194.97.59.151;ftag=as1566d63f;lr=on>
set_destination: Parsing <sip:194.97.54.97;ftag=as1566d63f;lr=on> for address/port to send to
set_destination: set destination to 194.97.54.97, port 5060
Transmitting (NAT) to 194.97.54.97:5060:
ACK sip:195.4.21.244:5060 SIP/2.0
Via: SIP/2.0/UDP 80.130.237.89:5060;branch=z9hG4bK02af255f;rport
Route: <sip:194.97.54.97;ftag=as1566d63f;lr=on>,<sip:194.97.59.151;ftag=as1566d63f;lr=on>
From: "131" <sip:[email protected]>;tag=as1566d63f
To: <sip:[email protected]>;tag=008082384A4502248FAE00000F77
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 103 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
    -- SIP/Freenet_THO-af80 answered SIP/132-b6e5
    -- Attempting native bridge of SIP/132-b6e5 and SIP/Freenet_THO-af80
asterisk*CLI>
<-- SIP read from 194.97.54.97:5060:
BYE sip:[email protected] SIP/2.0
Record-Route: <sip:194.97.54.97;ftag=008082384A4502248FAE00000F77;lr=on>
Record-Route: <sip:194.97.59.151;ftag=008082384A4502248FAE00000F77;lr=on>
Via: SIP/2.0/UDP 194.97.54.97;branch=0
Via: SIP/2.0/UDP 194.97.59.151;branch=z9hG4bK5e65.cc686e41.0
Via: SIP/2.0/UDP 195.4.21.244:5060;branch=z9hG4bK008082384A4502249639000012F8
From: <sip:[email protected]>;tag=008082384A4502248FAE00000F77
To: "131" <sip:[email protected]>;tag=as1566d63f
Call-ID: [email protected]
CSeq: 8543 BYE
Contact: <sip:[email protected]:5060>
Max-Forwards: 15
Content-Length:     0


--- (13 headers 0 lines)---
Sending to 194.97.54.97 : 5060 (NAT)
Transmitting (NAT) to 194.97.54.97:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 194.97.54.97;branch=0;received=194.97.54.97
Via: SIP/2.0/UDP 194.97.59.151;branch=z9hG4bK5e65.cc686e41.0
Via: SIP/2.0/UDP 195.4.21.244:5060;branch=z9hG4bK008082384A4502249639000012F8
ecord-Route: <sip:194.97.54.97;ftag=008082384A4502248FAE00000F77;lr=on>
Record-Route: <sip:194.97.59.151;ftag=008082384A4502248FAE00000F77;lr=on>
From: <sip:[email protected]>;tag=008082384A4502248FAE00000F77
To: "131" <sip:[email protected]>;tag=as1566d63f
Call-ID: [email protected]
CSeq: 8543 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:[email protected]>
Content-Length: 0
X-Asterisk-HangupCause: Normal Clearing


---
  == Spawn extension (Tel132, 0173xxxxxxx, 1) exited non-zero on 'SIP/132-b6e5'
Destroying call '[email protected]'
asterisk*CLI> sip no debug
Gibts jemanden der dazu eine Idee hat?
Habe beim Asterisk auch mal mit dem ethereal geloggt.
Die RTP's von Freenet kommen wirklich an und sind auch korrekt. Wenn ich die RTPs im ethereal extrahiere kann ich mir die auch anhören. Die Audiodaten kommen also ohne Probleme. Ansonsten funktioniert der Betrieb ja auch..nur Freenet macht ärger.
Ach ja..in der /etc/hosts ist der Eintrag "194.97.54.97 freenet.de" drin.

Tobias
 

saddamski

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Mitglied seit
28 Apr 2006
Beiträge
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Niemand der eine Idee hat?
Sind die Einstellungen denn wenigstens richtig? Es würde mir schon sehr helfen wenn ich wüssste dass es nicht an mir liegt :D
Die Einstellungen habe ich aus dem Kurs angepasst. Hier und da sind ein paar Sachen aus dem Forum abgeguckt. Also eigentlich sollte die Konfiguration laufen...zumindest tut sie es anscheinend bei anderen.

Gruß, Tobias
 

3CX PBX - GRATIS
Linux / Win / Cloud

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