GMX: Verbindung zu 1&1 ohne Ton

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Crea

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Hi,
ich habe bei meinem Asterisk Probleme, wenn ich über meine GMX Flat mittels SIP einen 1&1 Teilnehmer anrufe.
Wir hören uns gegenseitig nicht.
Per Paketmitschnitt (angehängt) habe ich herausgefunden woran es liegt, mein Asterisk will eine den RTP-Stream über den GMX SIP-Server abwickeln, die Gegenstelle (in diesem Fall eine Fritzbox) will aber einen direkten Stream zum Asterisk.

Ausschnitt aus der sip.conf:
Code: 
[general]
Qualify=no
context=default		
bindport=5060			
bindaddr=0.0.0.0		
srvlookup=yes			
externhost=sXXXXXXXXp.ath.cx
externrefresh=100
localnet=192.168.0.0/255.255.255.0
nat=never

[gmxXXXX1]
username=49XXXXXXXX1
host=sip.gmx.net
type=peer
fromuser=49XXXXXXXX1
secret=YYYYYYYYYY
disallow=all
allow=ulaw
nat=yes
qualify=no
canreinvite=no
insecure=very
fromdomain=sip.gmx.net
canreinvite=no
pickupgroup=1
callgroup=1
Ohne nat=yes läuft kein ausgehender Ton.

ich habe IPTables als Firewall, mit dem Script wird sie aktiviert (ausschnitt):

Code: 
$IPTABLES -N Cid464F30ED28453.0
$IPTABLES -A OUTPUT -p tcp -m tcp  --sport 5004:5080  --dport 5004:5080  -m state --state NEW  -j Cid464F30ED28453.0 
$IPTABLES -A OUTPUT -p tcp -m tcp  --sport 10000:10100  --dport 10000:10100  -m state --state NEW  -j Cid464F30ED28453.0 
$IPTABLES -A OUTPUT -p udp -m udp  --sport 10000:10100  --dport 10000:10100  -m state --state NEW  -j Cid464F30ED28453.0 
$IPTABLES -A OUTPUT -p udp -m udp  --sport 5004:5080  --dport 5004:5080  -m state --state NEW  -j Cid464F30ED28453.0 
$IPTABLES -A INPUT -p tcp -m tcp  --sport 5004:5080  --dport 5004:5080  -m state --state NEW  -j ACCEPT 
$IPTABLES -A INPUT -p tcp -m tcp  --sport 10000:10100  --dport 10000:10100  -m state --state NEW  -j ACCEPT 
$IPTABLES -A INPUT -p udp -m udp  --sport 10000:10100  --dport 10000:10100  -m state --state NEW  -j ACCEPT 
$IPTABLES -A INPUT -p udp -m udp  --sport 5004:5080  --dport 5004:5080  -m state --state NEW  -j ACCEPT
$IPTABLES -A Cid464F30ED28453.0  -d 192.168.0.1  -j ACCEPT

Zu den PDFs, ich bin 49XXXXXXX1 bzw. 0XXXXXXX1 und die Gegenstelle 0YYYYYYY2 bzw. 49YYYYYYY2.
 

Anhänge

Zuletzt bearbeitet:
kannst Du nicht einfach mal auf dem Asterisk ein simples SIP Debug eines Verbindungsaufbaus machen und hier posten? Hab eigentlich keine Lust mich durch 100 Seiten nixsagendes TCP Dump zu wurschteln - genau dafür hat ja Asterisk die Debug-Möglichkeit.

Außerdem würde ich wetten, daß Du ein ganz anderes Problem hast - aber um das zu beurteilen, brauch ich das SIP Debug vom Asterisk.
 
OK hier die Ausgabe, ich bin der XXXXXX die gegenstelle YYYYYYY.

Code: 
    -- Executing Dial("SIP/15-081cc690", "SIP/gmxXXXX90/09XXXXXXXX72|600|T") in new stack
We're at 84.170.181.252 port 10070
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
13 headers, 10 lines
Reliably Transmitting (NAT) to 212.227.15.197:5060:
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 84.170.181.252:5060;branch=z9hG4bK515b3429;rport
From: "Simon" <sip:[email protected]>;tag=as5cbe259c
To: <sip:[email protected]>
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 22 May 2007 20:00:09 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 220

v=0
o=root 27607 27607 IN IP4 84.170.181.252
s=session
c=IN IP4 84.170.181.252
t=0 0
m=audio 10070 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
    -- Called gmx204490/09XXXXXXXX72
server*CLI>
<-- SIP read from 212.227.15.197:5060:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 212.227.15.197:5060;branch=z9hG4bK515b3429;rport=5060
From: "Simon" <sip:[email protected]>;tag=as5cbe259c
To: <sip:[email protected]>;tag=329cfeaa6ded039da25ff8cbb8668bd2.18df
Call-ID: [email protected]
CSeq: 102 INVITE
Proxy-Authenticate: Digest realm="sip-gmx.net", nonce="46534d3d5eb8216a3e4074f5d426fba7affa76c5"
Server: UI OpenSer
Content-Length: 0


--- (9 headers 0 lines) ---
Transmitting (NAT) to 212.227.15.197:5060:
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 84.170.181.252:5060;branch=z9hG4bK515b3429;rport
From: "Simon" <sip:[email protected]>;tag=as5cbe259c
To: <sip:[email protected]>;tag=329cfeaa6ded039da25ff8cbb8668bd2.18df
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
We're at 84.170.181.252 port 10070
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 212.227.15.197:5060:
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 84.170.181.252:5060;branch=z9hG4bK36880251;rport
From: "Simon" <sip:[email protected]>;tag=as5cbe259c
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="49XXXXXXXX90", realm="sip-gmx.net", algorithm=MD5, uri="sip:[email protected]", nonce="46534d3d5ebdsfsdfsdfsdffba7affa76c5", response="f34d661cf30bsdfsfds49XXXXXXXX9003ef01afe", opaque=""
Date: Tue, 22 May 2007 20:00:09 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 220

v=0
o=root 27607 27608 IN IP4 84.170.181.252
s=session
c=IN IP4 84.170.181.252
t=0 0
m=audio 10070 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
server*CLI>
<-- SIP read from 212.227.15.197:5060:
SIP/2.0 100 trying -- your call is important to us
Via: SIP/2.0/UDP 212.227.15.197:5060;branch=z9hG4bK36880251;rport=5060
From: "Simon" <sip:[email protected]>;tag=as5cbe259c
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 103 INVITE
Server: UI OpenSer
Content-Length: 0


--- (8 headers 0 lines) ---
server*CLI>
<-- SIP read from 212.227.15.197:5060:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 212.227.15.197:5060;branch=z9hG4bK36880251;rport=5060
Record-Route: <sip:212.227.15.225;ftag=as5cbe259c;lr=on>
From: "Simon" <sip:[email protected]>;tag=as5cbe259c
To: <sip:[email protected]>;tag=8D883AD983340C5D
Call-ID: [email protected]
CSeq: 103 INVITE
Contact: <sip:[email protected];uniq=4E9C920666FC3352ADC98CA3A034D>
User-Agent: AVM FRITZ!Box Fon WLAN 7050 (UI) 14.04.31 (Feb  5 2007)
Content-Length: 0
X-Route-Info: IP


--- (11 headers 0 lines) ---
    -- SIP/gmx204490-081d6260 is ringing
Transmitting (no NAT) to 192.168.0.32:5060:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.0.32:5060;branch=z9hG4bKkt5ikvff33k9bk2oni4ts43;rport;received=192.168.0.32
From: <sip:[email protected]>;tag=dgfrp2bacphc75ed1qia
To: <sip:[email protected];user=phone>;tag=as35a52289
Call-ID: WguYyvd1oIdNSwWA50zGt6DaF6fKZh
CSeq: 623 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:[email protected]>
Content-Length: 0


---
server*CLI>
<-- SIP read from 212.227.15.197:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 212.227.15.197:5060;branch=z9hG4bK36880251;rport=5060
Record-Route: <sip:212.227.15.225;ftag=as5cbe259c;lr=on>
From: "Simon" <sip:[email protected]>;tag=as5cbe259c
To: <sip:[email protected]>;tag=8D883AD983340C5D
Call-ID: [email protected]
CSeq: 103 INVITE
Contact: <sip:[email protected];uniq=4E9C920666FC3352ADC98CA3A034D>
User-Agent: AVM FRITZ!Box Fon WLAN 7050 (UI) 14.04.31 (Feb  5 2007)
Supported: 100rel,replaces,timer
Allow-Events: telephone-event,refer
Allow: INVITE,ACK,OPTIONS,CANCEL,BYE,UPDATE,PRACK,INFO,SUBSCRIBE,NOTIFY,REFER,MESSAGE
Content-Type: application/sdp
Accept: application/sdp, multipart/mixed
Accept-Encoding: identity
Content-Length:   223
X-Route-Info: IP

v=0
o=user 11766735 11766735 IN IP4 212.227.15.197
s=session
c=IN IP4 212.227.15.197
t=0 0
m=audio 7082 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11
a=sendrecv
a=rtcp:7083

--- (17 headers 11 lines) ---
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port 212.227.15.197:7082
Found description format PCMU
Found description format telephone-event
Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
list_route: hop: <sip:212.227.15.225;ftag=as5cbe259c;lr=on>
set_destination: Parsing <sip:212.227.15.225;ftag=as5cbe259c;lr=on> for address/port to send to
set_destination: set destination to 212.227.15.225, port 5060
Transmitting (NAT) to 212.227.15.197:5060:
ACK sip:[email protected];uniq=4E9C920666FC3352ADC98CA3A034D SIP/2.0
Via: SIP/2.0/UDP 84.170.181.252:5060;branch=z9hG4bK7ee499d4;rport
Route: <sip:212.227.15.225;ftag=as5cbe259c;lr=on>
From: "Simon" <sip:[email protected]>;tag=as5cbe259c
To: <sip:[email protected]>;tag=8D883AD983340C5D
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 103 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
    -- SIP/gmx204490-081d6260 answered SIP/15-081cc690
We're at 192.168.0.1 port 10082
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 192.168.0.32:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.32:5060;branch=z9hG4bKkt5ikvff33k9bk2oni4ts43;rport;received=192.168.0.32
From: <sip:[email protected]>;tag=dgfrp2bacphc75ed1qia
To: <sip:[email protected];user=phone>;tag=as35a52289
Call-ID: WguYyvd1oIdNSwWA50zGt6DaF6fKZh
CSeq: 623 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:[email protected]>
Content-Type: application/sdp
Content-Length: 235

v=0
o=root 27607 27607 IN IP4 192.168.0.1
s=session
c=IN IP4 192.168.0.1
t=0 0
m=audio 10082 RTP/AVP 0 8 98
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:98 telephone-event/8000
a=fmtp:98 0-16
a=silenceSupp:off - - - -

---
server*CLI>
<-- SIP read from 192.168.0.32:5060:
ACK sip:[email protected];transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.0.32:5060;branch=z9hG4bKnfpb368jmpee7k2oni4g52r;rport
To: <sip:[email protected];user=phone>;tag=as35a52289
From: <sip:[email protected]>;tag=dgfrp2bacphc75ed1qia
Call-ID: WguYyvd1oIdNSwWA50zGt6DaF6fKZh
CSeq: 623 ACK
Supported: sec-agree
Max-Forwards: 70
Proxy-Authorization: Digest realm="asterisk",nonce="27d4096a",algorithm=MD5,username="15",uri="sip:[email protected];user=phone",response="17cb456d9sfddsffdsfs8d4eabbdf169"
Content-Length: 0


--- (10 headers 0 lines) ---
server*CLI>
<-- SIP read from 192.168.0.32:5060:
BYE sip:[email protected];transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.0.32:5060;branch=z9hG4bK2j3bp2dmvlhc6qocggno86f;rport
To: <sip:[email protected];user=phone>;tag=as35a52289
From: <sip:[email protected]>;tag=dgfrp2bacphc75ed1qia
Call-ID: WguYyvd1oIdNSwWA50zGt6DaF6fKZh
CSeq: 624 BYE
Supported: sec-agree
Max-Forwards: 70
Proxy-Authorization: Digest realm="asterisk",nonce="27d4096a",algorithm=MD5,username="15",uri="sip:[email protected];transport=UDP",response="b1d44ffcdsfdgweff59266c514f"
Content-Length: 0


--- (10 headers 0 lines) ---
Sending to 192.168.0.32 : 5060 (NAT)
Transmitting (NAT) to 192.168.0.32:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.32:5060;branch=z9hG4bK2j3bp2dmvlhc6qocggno86f;rport;received=192.168.0.32
From: <sip:[email protected]>;tag=dgfrp2bacphc75ed1qia
To: <sip:[email protected];user=phone>;tag=as35a52289
Call-ID: WguYyvd1oIdNSwWA50zGt6DaF6fKZh
CSeq: 624 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


---
Scheduling destruction of call '[email protected]' in 32000 ms
set_destination: Parsing <sip:212.227.15.225;ftag=as5cbe259c;lr=on> for address/port to send to
set_destination: set destination to 212.227.15.225, port 5060
Reliably Transmitting (NAT) to 212.227.15.197:5060:
BYE sip:[email protected];uniq=4E9C920666FC3352ADC98CA3A034D SIP/2.0
Via: SIP/2.0/UDP 84.170.181.252:5060;branch=z9hG4bK366653c9;rport
Route: <sip:212.227.15.225;ftag=as5cbe259c;lr=on>
From: "Simon" <sip:[email protected]>;tag=as5cbe259c
To: <sip:[email protected]>;tag=8D883AD983340C5D
Call-ID: [email protected]
CSeq: 104 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Proxy-Authorization: Digest username="49XXXXXXXX90", realm="sip-gmx.net", algorithm=MD5, uri="sip:[email protected]", nonce="46534d3d5ebdsfggrrega7affa76c5", response="95587547sdfsfsdffdsdfds44640f6b96", opaque=""
Content-Length: 0


---
  == Spawn extension (Simon, 09XXXXXXXX72, 2) exited non-zero on 'SIP/15-081cc690'
    -- Executing Hangup("SIP/15-081cc690", "") in new stack
  == Spawn extension (Simon, h, 1) exited non-zero on 'SIP/15-081cc690'
server*CLI>
<-- SIP read from 212.227.15.197:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 212.227.15.197:5060;branch=z9hG4bK366653c9;rport=5060
From: "Simon" <sip:[email protected]>;tag=as5cbe259c
To: <sip:[email protected]>;tag=8D883AD983340C5D
Call-ID: [email protected]
CSeq: 104 BYE
X-RTP-Stat: PS=327;OS=78480;SP=0/0;SO=0;PR=0;OR=0;CR=0;SR=0;PL=0;BL=0;EN=PCMU;DE=;JI=0
User-Agent: AVM FRITZ!Box Fon WLAN 7050 (UI) 14.04.31 (Feb  5 2007)
Supported: 100rel,replaces,timer
Allow-Events: telephone-event,refer
Content-Length: 0


--- (11 headers 0 lines) ---
Destroying call '[email protected]'
Destroying call 'WguYyvd1oIdNSwWA50zGt6DaF6fKZh'
12 headers, 0 lines
Reliably Transmitting (no NAT) to 192.168.0.32:5060:
OPTIONS sip:[email protected]:5060;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK10234359
From: "asterisk" <sip:[email protected]>;tag=as52d60c69
To: <sip:[email protected]:5060;transport=UDP>
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 22 May 2007 20:00:29 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0


---
server*CLI>
<-- SIP read from 192.168.0.32:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK10234359
To: <sip:[email protected]>;tag=qdrbp2fv69hc6nq10opl
From: "asterisk" <sip:[email protected]>;tag=as52d60c69
Call-ID: [email protected]
CSeq: 102 OPTIONS
Content-Length: 0


--- (7 headers 0 lines) ---
Destroying call '[email protected]'
Destroying call '[email protected]'
Destroying call '[email protected]'
 
Ich hab jetzt noch ein bisschen mit der Konfiguration rumgespielt, aber leider bekomme ich es nicht zum laufen.
Kann vielleicht jemand auch mal seine iptables config posten, weil das setzten von nat=yes soll ja zu problemen führen, leider geht es nicht ohne!
 
Habs inzwischen selbst rausgefunden, bei iptables waren die Sourceports gesetzt, aber es sollten nur die destinationports gesetzt werden.
Jetzt läuft alles.
 
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