Sipgate raustelefonieren geht, angerufen werden kann ich nic

BlackSektor

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Hallo, ich kann nun mit meinem Asterisk raustelefonieren (über Sipgate)
angerufen werden kann ich jedoch nicht.

Vieleicht kann mir jemand helfen

Hier mein dialplan

Code:
[ Context 'sipgate' created by 'pbx_config' ]
  '800XXXX' =>      1. Dial(SIP/10|30|tr)                         [pbx_config]
                    2. Hangup()                                   [pbx_config]


[ Context 'ausgsipgate' created by 'pbx_config' ]
  '_0.' =>          1. Dial(SIP/${EXTEN:1}@sipgate|30|tr)         [pbx_config]
                    2. Playback(invalid)                          [pbx_config]
                    3. Hangup()                                   [pbx_config]


[ Context '11' created by 'pbx_config' ]
  '11' =>           1. Dial(SIP/11)                               [pbx_config]
                    2. Hangup()                                   [pbx_config]


[ Context '10' created by 'pbx_config' ]
  '10' =>           1. Dial(SIP/10)                               [pbx_config]
                    2. ()                                         [pbx_config]


[ Context 'default' created by 'pbx_config' ]
  Include =>        '10'                                          [pbx_config]
  Include =>        '11'                                          [pbx_config]
  Include =>        'ausgsipgate'                                 [pbx_config]

[ Context 'parkedcalls' created by 'res_features' ]
  '700' =>          1. Park()                                     [res_features]
und hier die konfigs

sip.conf

Code:
[general]
port=5060
bindaddr=192.168.6.1
context=default
srvlookup=yes
nat=yes
insecure=very
register=> 8006724:[email protected]/8006724


[sipgate]
type=friend
username=8006724
secret=XXXXXX
host=sipgate.de
fromuser=8006724
nat=yes
context=sipgate
canreinvite=no


[10]
type=friend
username=10
secret=10
host=dynamic
callerid="10"=<10>


[11]
type=friend
username=11
secret=11
host=dynamic
callerid="11"=<11>
extensions.conf

Code:
[general]
static=yes
writeprotect=no


[default]
include=> 10
include=> 11
include=> ausgsipgate



[10]
exten=>10,1,Dial(SIP/10)
exten=>10,2 Hangup


[11]
exten=>11,1,Dial(SIP/11)
exten=>11,2,Hangup

[ausgsipgate]
exten=> _0.,1,Dial(SIP/${EXTEN:1}@sipgate,30,tr)
exten=> _0.,2,Playback(invalid)
exten=> _0.,3,Hangup

[sipgate]
exten=> 800XXXX,1,Dial(SIP/10,30,tr)
exten=> 800XXXX,2,Hangup
hier meine peers

Code:
*CLI> sip show peers
Name/username    Host            Dyn Nat ACL Mask             Port     Status
11/11            192.168.6.5      D          255.255.255.255  5060     Unmonitored
10/10            (Unspecified)    D          255.255.255.255  0        Unmonitored
sipgate/8006724  217.10.79.9          N      255.255.255.255  5060     Unmonitored
Und jetzt die debug ausgaben beim anruf
Code:
*CLI> sip debug
SIP Debugging Enabled
*CLI>

Sip read:

0 headers, 0 lines


Sip read:
INVITE sip:[email protected]:5060 SIP/2.0
Record-Route: <sip:[email protected];ftag=as11b265de;lr=on>
Max-Forwards: 9
Record-Route: <sip:[email protected];ftag=as11b265de;lr=on>
Via: SIP/2.0/UDP 217.10.79.9;branch=z9hG4bK9668.85b5bd24.1
Via: SIP/2.0/UDP 217.10.79.8;branch=z9hG4bK9668.a2a5e1b1.0
Via: SIP/2.0/UDP 217.10.79.30:5060;branch=z9hG4bK247ba0f8
From: "08427985707" <sip:[email protected]>;tag=as11b265de
To: <sip:[email protected]>
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Fri, 03 Dec 2004 10:25:24 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 370
Sipgate-Authentication: accepted

v=0
o=root 3372 3372 IN IP4 217.10.79.30
s=session
c=IN IP4 217.10.79.9
t=0 0
m=audio 47494 RTP/AVP 8 0 3 10 97 18 2 5
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:10 L16/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:18 G729/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:5 DVI4/8000
a=silenceSupp:off - - - -
a=direction:active
a=nortpproxy:yes

18 headers, 17 lines
Using latest request as basis request
Sending to 217.10.79.9 : 5060 (NAT)
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 3
Found RTP audio format 10
Found RTP audio format 97
Found RTP audio format 18
Found RTP audio format 2
Found RTP audio format 5
Peer audio RTP is at port 217.10.79.9:47494
Found description format PCMA
Found description format PCMU
Found description format GSM
Found description format L16
Found description format iLBC
Found description format G729
Found description format G726-32
Found description format DVI4
Capabilities: us - 0x8000e(GSM|ULAW|ALAW|H263), peer - audio=0x57e(GSM|ULAW|ALAW|G726|ADPCM|SLINR|G729A|ILBC)/video=0x0(EMPTY), combined - 0xe(GSM|ULAW|ALAW)
Non-codec capabilities: us - 0x1(G723), peer - 0x0(EMPTY), combined - 0x0(EMPTY)
Found peer 'sipgate'
Reliably Transmitting (NAT):
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 217.10.79.9;branch=z9hG4bK9668.85b5bd24.1;received=217.10.79.9;rport=5060
Via: SIP/2.0/UDP 217.10.79.8;branch=z9hG4bK9668.a2a5e1b1.0
Via: SIP/2.0/UDP 217.10.79.30:5060;branch=z9hG4bK247ba0f8
From: "08427985707" <sip:[email protected]>;tag=as11b265de
To: <sip:[email protected]>;tag=as4dd550ef
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:[email protected]>
Proxy-Authenticate: Digest realm="asterisk", nonce="5b07b6b6"
Content-Length: 0


 to 217.10.79.9:5060
Scheduling destruction of call '[email protected]' in 15000 ms


Sip read:
INVITE sip:[email protected]:5060 SIP/2.0
Record-Route: <sip:[email protected];ftag=as11b265de;lr=on>
Max-Forwards: 9
Record-Route: <sip:[email protected];ftag=as11b265de;lr=on>
Via: SIP/2.0/UDP 217.10.79.9;branch=z9hG4bK9668.85b5bd24.1
Via: SIP/2.0/UDP 217.10.79.8;branch=z9hG4bK9668.a2a5e1b1.0
Via: SIP/2.0/UDP 217.10.79.30:5060;branch=z9hG4bK247ba0f8
From: "08427985707" <sip:[email protected]>;tag=as11b265de
To: <sip:[email protected]>
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Fri, 03 Dec 2004 10:25:24 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 370
Sipgate-Authentication: accepted

v=0
o=root 3372 3372 IN IP4 217.10.79.30
s=session
c=IN IP4 217.10.79.9
t=0 0
m=audio 47494 RTP/AVP 8 0 3 10 97 18 2 5
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:10 L16/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:18 G729/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:5 DVI4/8000
a=silenceSupp:off - - - -
a=direction:active
a=nortpproxy:yes

18 headers, 17 lines
Ignoring this request
Found peer 'sipgate'


Sip read:
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 217.10.79.9;branch=z9hG4bK9668.85b5bd24.1
From: "08427985707" <sip:[email protected]>;tag=as11b265de
Call-ID: [email protected]
To: <sip:[email protected]>;tag=as4dd550ef
CSeq: 102 ACK
User-Agent: sipgate ser
Content-Length: 0


8 headers, 0 lines


Sip read:
INFO sip:[email protected]:5060 SIP/2.0
Record-Route: <sip:[email protected];ftag=as11b265de;lr=on>
Max-Forwards: 9
Record-Route: <sip:[email protected];ftag=as11b265de;lr=on>
Via: SIP/2.0/UDP 217.10.79.9;branch=z9hG4bKa668.cb0ddb76.1
Via: SIP/2.0/UDP 217.10.79.8;branch=z9hG4bKa668.ea40ee86.0
Via: SIP/2.0/UDP 217.10.79.30:5060;branch=z9hG4bK4b18c41f
From: "08427985707" <sip:[email protected]>;tag=as11b265de
To: <sip:[email protected]>
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 103 INFO
User-Agent: Asterisk PBX
Content-Type: application/dtmf-relay
Content-Length: 24
Sipgate-Authentication: accepted

Signal=5
Duration=250

16 headers, 2 lines
Receiving DTMF!
Transmitting (NAT):
SIP/2.0 481 Call leg/transaction does not exist
Via: SIP/2.0/UDP 217.10.79.9;branch=z9hG4bKa668.cb0ddb76.1;received=217.10.79.9;rport=5060
Via: SIP/2.0/UDP 217.10.79.8;branch=z9hG4bKa668.ea40ee86.0
Via: SIP/2.0/UDP 217.10.79.30:5060;branch=z9hG4bK4b18c41f
From: "08427985707" <sip:[email protected]>;tag=as11b265de
To: <sip:[email protected]>;tag=as4dd550ef
Call-ID: [email protected]
CSeq: 103 INFO
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:[email protected]>
Content-Length: 0


 to 217.10.79.9:5060
Destroying call '[email protected]'


Sip read:
INFO sip:[email protected]:5060 SIP/2.0
Record-Route: <sip:[email protected];ftag=as11b265de;lr=on>
Max-Forwards: 9
Record-Route: <sip:[email protected];ftag=as11b265de;lr=on>
Via: SIP/2.0/UDP 217.10.79.9;branch=z9hG4bK7668.4995d0b3.1
Via: SIP/2.0/UDP 217.10.79.8;branch=z9hG4bK7668.b33c6d73.0
Via: SIP/2.0/UDP 217.10.79.30:5060;branch=z9hG4bK5dfce868
From: "08427985707" <sip:[email protected]>;tag=as11b265de
To: <sip:[email protected]>
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 104 INFO
User-Agent: Asterisk PBX
Content-Type: application/dtmf-relay
Content-Length: 24
Sipgate-Authentication: accepted

Signal=8
Duration=250

16 headers, 2 lines
Receiving DTMF!
Transmitting (NAT):
SIP/2.0 481 Call leg/transaction does not exist
Via: SIP/2.0/UDP 217.10.79.9;branch=z9hG4bK7668.4995d0b3.1;received=217.10.79.9;rport=5060
Via: SIP/2.0/UDP 217.10.79.8;branch=z9hG4bK7668.b33c6d73.0
Via: SIP/2.0/UDP 217.10.79.30:5060;branch=z9hG4bK5dfce868
From: "08427985707" <sip:[email protected]>;tag=as11b265de
To: <sip:[email protected]>;tag=as1a4cc10a
Call-ID: [email protected]
CSeq: 104 INFO
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact:
Content-Length: 0


 to 217.10.79.9:5060
Destroying call '[email protected]'


Sip read:
INFO sip:[email protected]:5060 SIP/2.0
Record-Route: <sip:[email protected];ftag=as11b265de;lr=on>
Max-Forwards: 9
Record-Route: <sip:[email protected];ftag=as11b265de;lr=on>
Via: SIP/2.0/UDP 217.10.79.9;branch=z9hG4bK8668.c8b81785.1
Via: SIP/2.0/UDP 217.10.79.8;branch=z9hG4bK8668.e2ec3b54.0
Via: SIP/2.0/UDP 217.10.79.30:5060;branch=z9hG4bK37e7a872
From: "08427985707" <sip:[email protected]>;tag=as11b265de
To: <sip:[email protected]>
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 105 INFO
User-Agent: Asterisk PBX
Content-Type: application/dtmf-relay
Content-Length: 24
Sipgate-Authentication: accepted

Signal=0
Duration=250

16 headers, 2 lines
Receiving DTMF!
Transmitting (NAT):
SIP/2.0 481 Call leg/transaction does not exist
Via: SIP/2.0/UDP 217.10.79.9;branch=z9hG4bK8668.c8b81785.1;received=217.10.79.9;rport=5060
Via: SIP/2.0/UDP 217.10.79.8;branch=z9hG4bK8668.e2ec3b54.0
Via: SIP/2.0/UDP 217.10.79.30:5060;branch=z9hG4bK37e7a872
From: "08427985707" <sip:[email protected]>;tag=as11b265de
To: <sip:[email protected]>;tag=as73cea4b0
Call-ID: [email protected]
CSeq: 105 INFO
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact:
Content-Length: 0


 to 217.10.79.9:5060
Destroying call '[email protected]'


Sip read:
INFO sip:[email protected]:5060 SIP/2.0
Record-Route: <sip:[email protected];ftag=as11b265de;lr=on>
Max-Forwards: 9
Record-Route: <sip:[email protected];ftag=as11b265de;lr=on>
Via: SIP/2.0/UDP 217.10.79.9;branch=z9hG4bK3768.d837b7f7.1
Via: SIP/2.0/UDP 217.10.79.8;branch=z9hG4bK3768.afce1801.0
Via: SIP/2.0/UDP 217.10.79.30:5060;branch=z9hG4bK6dc8e001
From: "08427985707" <sip:[email protected]>;tag=as11b265de
To: <sip:[email protected]>
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 109 INFO
User-Agent: Asterisk PBX
Content-Type: application/dtmf-relay
Content-Length: 24
Sipgate-Authentication: accepted

Signal=2
Duration=250

16 headers, 2 lines
Receiving DTMF!
Transmitting (NAT):
SIP/2.0 481 Call leg/transaction does not exist
Via: SIP/2.0/UDP 217.10.79.9;branch=z9hG4bK3768.d837b7f7.1;received=217.10.79.9;rport=5060
Via: SIP/2.0/UDP 217.10.79.8;branch=z9hG4bK3768.afce1801.0
Via: SIP/2.0/UDP 217.10.79.30:5060;branch=z9hG4bK6dc8e001
From: "08427985707" <sip:[email protected]>;tag=as11b265de
To: <sip:[email protected]>;tag=as4d1f4c71
Call-ID: [email protected]
CSeq: 109 INFO
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact:
Content-Length: 0


 to 217.10.79.9:5060
Destroying call '[email protected]'


Sip read:
INFO sip:[email protected]:5060 SIP/2.0
Record-Route: <sip:[email protected];ftag=as11b265de;lr=on>
Max-Forwards: 9
Record-Route: <sip:[email protected];ftag=as11b265de;lr=on>
Via: SIP/2.0/UDP 217.10.79.9;branch=z9hG4bK4768.a88a5623.1
Via: SIP/2.0/UDP 217.10.79.8;branch=z9hG4bK4768.bea05fd5.0
Via: SIP/2.0/UDP 217.10.79.30:5060;branch=z9hG4bK0ad1241f
From: "08427985707" <sip:[email protected]>;tag=as11b265de
To: <sip:[email protected]>
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 108 INFO
User-Agent: Asterisk PBX
Content-Type: application/dtmf-relay
Content-Length: 24
Sipgate-Authentication: accepted

Signal=7
Duration=250

16 headers, 2 lines
Receiving DTMF!
Transmitting (NAT):
SIP/2.0 481 Call leg/transaction does not exist
Via: SIP/2.0/UDP 217.10.79.9;branch=z9hG4bK4768.a88a5623.1;received=217.10.79.9;rport=5060
Via: SIP/2.0/UDP 217.10.79.8;branch=z9hG4bK4768.bea05fd5.0
Via: SIP/2.0/UDP 217.10.79.30:5060;branch=z9hG4bK0ad1241f
From: "08427985707" <sip:[email protected]>;tag=as11b265de
To: <sip:[email protected]>;tag=as35a4a495
Call-ID: [email protected]
CSeq: 108 INFO
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact:
Content-Length: 0


 to 217.10.79.9:5060
Destroying call '[email protected]'


Sip read:
INFO sip:[email protected]:5060 SIP/2.0
Record-Route: <sip:[email protected];ftag=as11b265de;lr=on>
Max-Forwards: 9
Record-Route: <sip:[email protected];ftag=as11b265de;lr=on>
Via: SIP/2.0/UDP 217.10.79.9;branch=z9hG4bKb478.6e124a07.1
Via: SIP/2.0/UDP 217.10.79.8;branch=z9hG4bKb478.8da83242.0
Via: SIP/2.0/UDP 217.10.79.30:5060;branch=z9hG4bK12f26619
From: "08427985707" <sip:[email protected]>;tag=as11b265de
To: <sip:[email protected]>
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 110 INFO
User-Agent: Asterisk PBX
Content-Type: application/dtmf-relay
Content-Length: 24
Sipgate-Authentication: accepted

Signal=4
Duration=250

16 headers, 2 lines
Receiving DTMF!
Transmitting (NAT):
SIP/2.0 481 Call leg/transaction does not exist
Via: SIP/2.0/UDP 217.10.79.9;branch=z9hG4bKb478.6e124a07.1;received=217.10.79.9;rport=5060
Via: SIP/2.0/UDP 217.10.79.8;branch=z9hG4bKb478.8da83242.0
Via: SIP/2.0/UDP 217.10.79.30:5060;branch=z9hG4bK12f26619
From: "08427985707" <sip:[email protected]>;tag=as11b265de
To: <sip:[email protected]>;tag=as5add0b60
Call-ID: [email protected]
CSeq: 110 INFO
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact:
Content-Length: 0


 to 217.10.79.9:5060
Destroying call '442ed31d3e481cde[email protected]'


Sip read:
INFO sip:[email protected]:5060 SIP/2.0
Record-Route: <sip:[email protected];ftag=as11b265de;lr=on>
Max-Forwards: 9
Record-Route: <sip:[email protected];ftag=as11b265de;lr=on>
Via: SIP/2.0/UDP 217.10.79.9;branch=z9hG4bK6668.cf3ea444.1
Via: SIP/2.0/UDP 217.10.79.8;branch=z9hG4bK6668.01c83766.0
Via: SIP/2.0/UDP 217.10.79.30:5060;branch=z9hG4bK325edae9
From: "08427985707" <sip:[email protected]>;tag=as11b265de
To: <sip:[email protected]>
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 107 INFO
User-Agent: Asterisk PBX
Content-Type: application/dtmf-relay
Content-Length: 24
Sipgate-Authentication: accepted

Signal=6
Duration=250

16 headers, 2 lines
Receiving DTMF!
Transmitting (NAT):
SIP/2.0 481 Call leg/transaction does not exist
Via: SIP/2.0/UDP 217.10.79.9;branch=z9hG4bK6668.cf3ea444.1;received=217.10.79.9;rport=5060
Via: SIP/2.0/UDP 217.10.79.8;branch=z9hG4bK6668.01c83766.0
Via: SIP/2.0/UDP 217.10.79.30:5060;branch=z9hG4bK325edae9
From: "08427985707" <sip:[email protected]>;tag=as11b265de
To: <sip:[email protected]>;tag=as41d93c90
Call-ID: [email protected]
CSeq: 107 INFO
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact:
Content-Length: 0


 to 217.10.79.9:5060
Destroying call '[email protected]'


Sip read:
INFO sip:[email protected]:5060 SIP/2.0
Record-Route: <sip:[email protected];ftag=as11b265de;lr=on>
Max-Forwards: 9
Record-Route: <sip:[email protected];ftag=as11b265de;lr=on>
Via: SIP/2.0/UDP 217.10.79.9;branch=z9hG4bK5668.05111d34.1
Via: SIP/2.0/UDP 217.10.79.8;branch=z9hG4bK5668.3f11a961.0
Via: SIP/2.0/UDP 217.10.79.30:5060;branch=z9hG4bK6280f632
From: "08427985707" <sip:[email protected]>;tag=as11b265de
To: <sip:[email protected]>
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 106 INFO
User-Agent: Asterisk PBX
Content-Type: application/dtmf-relay
Content-Length: 24
Sipgate-Authentication: accepted

Signal=0
Duration=250

16 headers, 2 lines
Receiving DTMF!
Transmitting (NAT):
SIP/2.0 481 Call leg/transaction does not exist
Via: SIP/2.0/UDP 217.10.79.9;branch=z9hG4bK5668.05111d34.1;received=217.10.79.9;rport=5060
Via: SIP/2.0/UDP 217.10.79.8;branch=z9hG4bK5668.3f11a961.0
Via: SIP/2.0/UDP 217.10.79.30:5060;branch=z9hG4bK6280f632
From: "08427985707" <sip:[email protected]>;tag=as11b265de
To: <sip:[email protected]>;tag=as06db8db7
Call-ID: [email protected]
CSeq: 106 INFO
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact:
Content-Length: 0


 to 217.10.79.9:5060
Destroying call '[email protected]'


Sip read:
INFO sip:[email protected]:5060 SIP/2.0
Record-Route: <sip:[email protected];ftag=as11b265de;lr=on>
Max-Forwards: 9
Record-Route: <sip:[email protected];ftag=as11b265de;lr=on>
Via: SIP/2.0/UDP 217.10.79.9;branch=z9hG4bK4768.a88a5623.1
Via: SIP/2.0/UDP 217.10.79.8;branch=z9hG4bK4768.bea05fd5.0
Via: SIP/2.0/UDP 217.10.79.30:5060;branch=z9hG4bK0ad1241f
From: "08427985707" <sip:[email protected]>;tag=as11b265de
To: <sip:[email protected]>
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 108 INFO
User-Agent: Asterisk PBX
Content-Type: application/dtmf-relay
Content-Length: 24
Sipgate-Authentication: accepted

Signal=7
Duration=250

16 headers, 2 lines
Receiving DTMF!
Transmitting (NAT):
SIP/2.0 481 Call leg/transaction does not exist
Via: SIP/2.0/UDP 217.10.79.9;branch=z9hG4bK4768.a88a5623.1;received=217.10.79.9;rport=5060
Via: SIP/2.0/UDP 217.10.79.8;branch=z9hG4bK4768.bea05fd5.0
Via: SIP/2.0/UDP 217.10.79.30:5060;branch=z9hG4bK0ad1241f
From: "08427985707" <sip:[email protected]>;tag=as11b265de
To: <sip:[email protected]>;tag=as0097d44d
Call-ID: [email protected]
CSeq: 108 INFO
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact:
Content-Length: 0


 to 217.10.79.9:5060
Destroying call '[email protected]'


Sip read:
INFO sip:[email protected]:5060 SIP/2.0
Record-Route: <sip:[email protected];ftag=as11b265de;lr=on>
Max-Forwards: 9
Record-Route: <sip:[email protected];ftag=as11b265de;lr=on>
Via: SIP/2.0/UDP 217.10.79.9;branch=z9hG4bK6668.cf3ea444.1
Via: SIP/2.0/UDP 217.10.79.8;branch=z9hG4bK6668.01c83766.0
Via: SIP/2.0/UDP 217.10.79.30:5060;branch=z9hG4bK325edae9
From: "08427985707" <sip:[email protected]>;tag=as11b265de
To: <sip:[email protected]>
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 107 INFO
User-Agent: Asterisk PBX
Content-Type: application/dtmf-relay
Content-Length: 24
Sipgate-Authentication: accepted

Signal=6
Duration=250

16 headers, 2 lines
Receiving DTMF!
Transmitting (NAT):
SIP/2.0 481 Call leg/transaction does not exist
Via: SIP/2.0/UDP 217.10.79.9;branch=z9hG4bK6668.cf3ea444.1;received=217.10.79.9;rport=5060
Via: SIP/2.0/UDP 217.10.79.8;branch=z9hG4bK6668.01c83766.0
Via: SIP/2.0/UDP 217.10.79.30:5060;branch=z9hG4bK325edae9
From: "08427985707" <sip:[email protected]>;tag=as11b265de
To: <sip:[email protected]>;tag=as2b04d9bc
Call-ID: [email protected]
CSeq: 107 INFO
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact:
Content-Length: 0


 to 217.10.79.9:5060
Destroying call '[email protected]'


Sip read:
INFO sip:[email protected]:5060 SIP/2.0
Record-Route: <sip:[email protected];ftag=as11b265de;lr=on>
Max-Forwards: 9
Record-Route: <sip:[email protected];ftag=as11b265de;lr=on>
Via: SIP/2.0/UDP 217.10.79.9;branch=z9hG4bK5668.05111d34.1
Via: SIP/2.0/UDP 217.10.79.8;branch=z9hG4bK5668.3f11a961.0
Via: SIP/2.0/UDP 217.10.79.30:5060;branch=z9hG4bK6280f632
From: "08427985707" <sip:[email protected]>;tag=as11b265de
To: <sip:[email protected]>
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 106 INFO
User-Agent: Asterisk PBX
Content-Type: application/dtmf-relay
Content-Length: 24
Sipgate-Authentication: accepted

Signal=0
Duration=250

16 headers, 2 lines
Receiving DTMF!
Transmitting (NAT):
SIP/2.0 481 Call leg/transaction does not exist
Via: SIP/2.0/UDP 217.10.79.9;branch=z9hG4bK5668.05111d34.1;received=217.10.79.9;rport=5060
Via: SIP/2.0/UDP 217.10.79.8;branch=z9hG4bK5668.3f11a961.0
Via: SIP/2.0/UDP 217.10.79.30:5060;branch=z9hG4bK6280f632
From: "08427985707" <sip:[email protected]>;tag=as11b265de
To: <sip:[email protected]>;tag=as1b7a1d94
Call-ID: [email protected]
CSeq: 106 INFO
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact:
Content-Length: 0


 to 217.10.79.9:5060
Destroying call '[email protected]'


Sip read:

0 headers, 0 lines
    -- parse_srv: SRV mapped to host proxy.de.sipgate.net, port 5060
11 headers, 0 lines
Reliably Transmitting:
REGISTER sip:sipgate.de SIP/2.0
Via: SIP/2.0/UDP 192.168.6.1:5060;branch=z9hG4bK4dfd2398
From: <sip:[email protected]>;tag=as161f4ff1
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 106 REGISTER
User-Agent: Asterisk PBX
Expires: 120
Contact: <sip:[email protected]>
Event: registration
Content-Length: 0

 (no NAT) to 217.10.79.9:5060


Sip read:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.6.1:5060;branch=z9hG4bK4dfd2398;rport=5060;received=217.187.98.23
From: <sip:[email protected]>;tag=as161f4ff1
To: <sip:[email protected]>;tag=b11cb9bb270104b49a99a995b8c68544.45c2
Call-ID: [email protected]
CSeq: 106 REGISTER
WWW-Authenticate: Digest realm="sipgate.de", nonce="41b049624974c131304899dba5cc4de5f8a0c728"
Server: sipgate ser
Content-Length: 0
Warning: 392 217.10.79.9:5060 "Noisy feedback tells:  pid=8442 req_src_ip=217.187.98.23 req_src_port=5060 in_uri=sip:sipgate.de out_uri=sip:sipgate.de via_cnt==1"


10 headers, 0 lines
12 headers, 0 lines
Reliably Transmitting:
REGISTER sip:sipgate.de SIP/2.0
Via: SIP/2.0/UDP 192.168.6.1:5060;branch=z9hG4bK47de180b
From: <sip:[email protected]>;tag=as161f4ff1
To: <sip:[email protected]>;tag=b11cb9bb270104b49a99a995b8c68544.45c2
Call-ID: [email protected]
CSeq: 107 REGISTER
User-Agent: Asterisk PBX
Authorization: Digest username="8006724", realm="sipgate.de", algorithm=MD5, uri="sip:sipgate.de", nonce="41b049624974c131304899dba5cc4de5f8a0c728", response="634549faa5906b9c105ac86e6f2fbf88", opaque=""
Expires: 120
Contact: <sip:[email protected]>
Event: registration
Content-Length: 0

 (no NAT) to 217.10.79.9:5060


Sip read:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.6.1:5060;branch=z9hG4bK47de180b;rport=5060;received=217.187.98.23
From: <sip:[email protected]>;tag=as161f4ff1
To: <sip:[email protected]>;tag=b11cb9bb270104b49a99a995b8c68544.45c2
Call-ID: [email protected]
CSeq: 107 REGISTER
Contact: <sip:[email protected]:5060>;q=0.00;expires=120
Server: sipgate ser
Content-Length: 0
Warning: 392 217.10.79.9:5060 "Noisy feedback tells:  pid=8445 req_src_ip=217.187.98.23 req_src_port=5060 in_uri=sip:sipgate.de out_uri=sip:sipgate.de via_cnt==1"


10 headers, 0 lines
Destroying call '[email protected]'


Sip read:

0 headers, 0 lines
Die interne Nebenstelle kann ich aus irgendwelchen gründen auch nicht anwählen?

Vieleicht kann mir ja jemand helfen
 

Netview

IPPF-Promi
Mitglied seit
1 Apr 2004
Beiträge
3,366
Punkte für Reaktionen
0
Punkte
36
Du willst eine pbx über sip callen?

Dies geht nur über die capi z.B.

exten => 8006724,1,Dial(Capi/@37:40,60)

37 ist die msn der ISDN-Karte und 40 die Rufgruppe (gerufene Nummer an der Pbx)
 

Hupe

Aktives Mitglied
Mitglied seit
8 Apr 2004
Beiträge
2,586
Punkte für Reaktionen
0
Punkte
0
10/10 (Unspecified) D 255.255.255.255 0 Unmonitored
Also, eingehende Sipgate-Gespräche sollen wohl auf den Peer 10 weitergestelt werden, oder? Irgendwie sieht es nicht so aus, als ob der Client richtig registriert ist. Was ist das für ein Client. Setz doch mal "qualify=yes" bei den Peers dazu. Außerdem solltest Du mal codecs in der sip.conf definieren.
 

BlackSektor

Neuer User
Mitglied seit
3 Nov 2004
Beiträge
34
Punkte für Reaktionen
0
Punkte
0
Ein SNOM 190 und ein X-Lite Client (der ist nur zum Testen)
Richtig, die Gespräche sollen an den Client 10 und 11 ran.
Ich hab das ganze jetzt mal überarbeitet:

Code:
SIP Debugging Enabled for IP: 217.10.79.9:5060
*CLI>

Sip read:
INVITE sip:[email protected]:5060 SIP/2.0
Record-Route: <sip:[email protected];ftag=as36dc9bfc;lr=on>
Max-Forwards: 9
Record-Route: <sip:49211580[email protected];ftag=as36dc9bfc;lr=on>
Via: SIP/2.0/UDP 217.10.79.9;branch=z9hG4bK2e1.26f0cb67.1
Via: SIP/2.0/UDP 217.10.79.8;branch=z9hG4bK2e1.a9083e57.0
Via: SIP/2.0/UDP 217.10.79.30:5060;branch=z9hG4bK75d864b1
From: "08427985707" <sip:[email protected]>;tag=as36dc9bfc
To: <sip:[email protected]>
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Fri, 03 Dec 2004 16:33:44 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 372
Sipgate-Authentication: accepted

v=0
o=root 24194 24194 IN IP4 217.10.79.30
s=session
c=IN IP4 217.10.79.9
t=0 0
m=audio 45596 RTP/AVP 8 0 3 10 97 18 2 5
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:10 L16/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:18 G729/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:5 DVI4/8000
a=silenceSupp:off - - - -
a=direction:active
a=nortpproxy:yes

18 headers, 17 lines
Using latest request as basis request
Sending to 217.10.79.9 : 5060 (non-NAT)
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 3
Found RTP audio format 10
Found RTP audio format 97
Found RTP audio format 18
Found RTP audio format 2
Found RTP audio format 5
Peer audio RTP is at port 217.10.79.9:45596
Found description format PCMA
Found description format PCMU
Found description format GSM
Found description format L16
Found description format iLBC
Found description format G729
Found description format G726-32
Found description format DVI4
Capabilities: us - 0x80008(ALAW|H263), peer - audio=0x57e(GSM|ULAW|ALAW|G726|ADPCM|SLINR|G729A|ILBC)/video=0x0(EMPTY), combined - 0x8(ALAW)
Non-codec capabilities: us - 0x1(G723), peer - 0x0(EMPTY), combined - 0x0(EMPTY)
Found peer 'sipgate'
Reliably Transmitting (NAT):
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 217.10.79.9;branch=z9hG4bK2e1.26f0cb67.1;received=217.10.79.9;rport=5060
Via: SIP/2.0/UDP 217.10.79.8;branch=z9hG4bK2e1.a9083e57.0
Via: SIP/2.0/UDP 217.10.79.30:5060;branch=z9hG4bK75d864b1
From: "08427985707" <sip:[email protected]>;tag=as36dc9bfc
To: <sip:[email protected]>;tag=as4c4cb239
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:[email protected]>
Proxy-Authenticate: Digest realm="asterisk", nonce="0a1b4fe7"
Content-Length: 0


 to 217.10.79.9:5060
Scheduling destruction of call '[email protected]' in 15000 ms


Sip read:
INVITE sip:[email protected]:5060 SIP/2.0
Record-Route: <sip:[email protected];ftag=as36dc9bfc;lr=on>
Max-Forwards: 9
Record-Route: <sip:[email protected];ftag=as36dc9bfc;lr=on>
Via: SIP/2.0/UDP 217.10.79.9;branch=z9hG4bK2e1.26f0cb67.1
Via: SIP/2.0/UDP 217.10.79.8;branch=z9hG4bK2e1.a9083e57.0
Via: SIP/2.0/UDP 217.10.79.30:5060;branch=z9hG4bK75d864b1
From: "08427985707" <sip:[email protected]>;tag=as36dc9bfc
To: <sip:[email protected]>
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Fri, 03 Dec 2004 16:33:44 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 372
Sipgate-Authentication: accepted

v=0
o=root 24194 24194 IN IP4 217.10.79.30
s=session
c=IN IP4 217.10.79.9
t=0 0
m=audio 45596 RTP/AVP 8 0 3 10 97 18 2 5
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:10 L16/8000
a=rtpmap:97 iLBC/8000
a=rtpmap:18 G729/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:5 DVI4/8000
a=silenceSupp:off - - - -
a=direction:active
a=nortpproxy:yes

18 headers, 17 lines
Ignoring this request
Found peer 'sipgate'


Sip read:
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 217.10.79.9;branch=z9hG4bK2e1.26f0cb67.1
From: "08427985707" <sip:[email protected]>;tag=as36dc9bfc
Call-ID: [email protected]
To: <sip:[email protected]>;tag=as4c4cb239
CSeq: 102 ACK
User-Agent: sipgate ser
Content-Length: 0


8 headers, 0 lines
11 headers, 0 lines
Reliably Transmitting:
OPTIONS sip:217.10.79.9 SIP/2.0
Via: SIP/2.0/UDP 192.168.6.1:5060;branch=z9hG4bK7ec81138
From: "asterisk" <sip:[email protected]>;tag=as2d09910a
To: <sip:217.10.79.9>
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Date: Fri, 03 Dec 2004 18:13:29 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Length: 0

 (no NAT) to 217.10.79.9:5060
Retransmitting #1 (no NAT):
OPTIONS sip:217.10.79.9 SIP/2.0
Via: SIP/2.0/UDP 192.168.6.1:5060;branch=z9hG4bK7ec81138
From: "asterisk" <sip:[email protected]>;tag=as2d09910a
To: <sip:217.10.79.9>
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Date: Fri, 03 Dec 2004 18:13:29 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Length: 0


 to 217.10.79.9:5060


Sip read:
SIP/2.0 482 Loop Detected
Via: SIP/2.0/UDP 192.168.6.1:5060;branch=z9hG4bK7ec81138;rport=5060;received=217.187.98.23
From: "asterisk" <sip:[email protected]>;tag=as2d09910a
To: <sip:217.10.79.9>;tag=b11cb9bb270104b49a99a995b8c68544.1824
Call-ID: [email protected]
CSeq: 102 OPTIONS
Server: sipgate ser
Content-Length: 0
Warning: 392 217.10.79.9:5060 "Noisy feedback tells:  pid=8442 req_src_ip=217.187.98.23 req_src_port=5060 in_uri=sip:217.10.79.9 out_uri=sip:217.10.79.9 via_cnt==1"


9 headers, 0 lines
Destroying call '[email protected]'


Sip read:
INFO sip:[email protected]:5060 SIP/2.0
Record-Route: <sip:[email protected];ftag=as36dc9bfc;lr=on>
Max-Forwards: 9
Record-Route: <sip:[email protected];ftag=as36dc9bfc;lr=on>
Via: SIP/2.0/UDP 217.10.79.9;branch=z9hG4bK3e1.0ab748b7.1
Via: SIP/2.0/UDP 217.10.79.8;branch=z9hG4bK3e1.30ea5a67.0
Via: SIP/2.0/UDP 217.10.79.30:5060;branch=z9hG4bK36fccae9
From: "08427985707" <sip:[email protected]>;tag=as36dc9bfc
To: <sip:[email protected]>
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 103 INFO
User-Agent: Asterisk PBX
Content-Type: application/dtmf-relay
Content-Length: 24
Sipgate-Authentication: accepted

Signal=5
Duration=250

16 headers, 2 lines
Receiving DTMF!
Transmitting (NAT):
SIP/2.0 481 Call leg/transaction does not exist
Via: SIP/2.0/UDP 217.10.79.9;branch=z9hG4bK3e1.0ab748b7.1;received=217.10.79.9;rport=5060
Via: SIP/2.0/UDP 217.10.79.8;branch=z9hG4bK3e1.30ea5a67.0
Via: SIP/2.0/UDP 217.10.79.30:5060;branch=z9hG4bK36fccae9
From: "08427985707" <sip:[email protected]>;tag=as36dc9bfc
To: <sip:[email protected]>;tag=as4c4cb239
Call-ID: [email protected]
CSeq: 103 INFO
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:[email protected]>
Content-Length: 0


 to 217.10.79.9:5060
Destroying call '[email protected]'


Sip read:
INFO sip:[email protected]:5060 SIP/2.0
Record-Route: <sip:[email protected];ftag=as36dc9bfc;lr=on>
Max-Forwards: 9
Record-Route: <sip:[email protected];ftag=as36dc9bfc;lr=on>
Via: SIP/2.0/UDP 217.10.79.9;branch=z9hG4bK0e1.c63cac2.1
Via: SIP/2.0/UDP 217.10.79.8;branch=z9hG4bK0e1.678adc92.0
Via: SIP/2.0/UDP 217.10.79.30:5060;branch=z9hG4bK7d97ce0e
From: "08427985707" <sip:[email protected]>;tag=as36dc9bfc
To: <sip:[email protected]>
Contact: <sip:08427985707[email protected]>
Call-ID: [email protected]
CSeq: 104 INFO
User-Agent: Asterisk PBX
Content-Type: application/dtmf-relay
Content-Length: 24
Sipgate-Authentication: accepted

Signal=8
Duration=250

16 headers, 2 lines
Receiving DTMF!
Transmitting (no NAT):
SIP/2.0 481 Call leg/transaction does not exist
Via: SIP/2.0/UDP 217.10.79.9;branch=z9hG4bK0e1.c63cac2.1
Via: SIP/2.0/UDP 217.10.79.8;branch=z9hG4bK0e1.678adc92.0
Via: SIP/2.0/UDP 217.10.79.30:5060;branch=z9hG4bK7d97ce0e
From: "08427985707" <sip:[email protected]>;tag=as36dc9bfc
To: <sip:[email protected]>;tag=as093323a0
Call-ID: [email protected]
CSeq: 104 INFO
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact:
Content-Length: 0


 to 217.10.79.9:5060
Destroying call '[email protected]'


Sip read:
INFO sip:[email protected]:5060 SIP/2.0
Record-Route: <sip:[email protected];ftag=as36dc9bfc;lr=on>
Max-Forwards: 9
Record-Route: <sip:[email protected];ftag=as36dc9bfc;lr=on>
Via: SIP/2.0/UDP 217.10.79.9;branch=z9hG4bK1e1.848513e5.1
Via: SIP/2.0/UDP 217.10.79.8;branch=z9hG4bK1e1.36e770e6.0
Via: SIP/2.0/UDP 217.10.79.30:5060;branch=z9hG4bK1fe41bfd
From: "08427985707" <sip:[email protected]>;tag=as36dc9bfc
To: <sip:[email protected]>
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 105 INFO
User-Agent: Asterisk PBX
Content-Type: application/dtmf-relay
Content-Length: 24
Sipgate-Authentication: accepted

Signal=0
Duration=250

16 headers, 2 lines
Receiving DTMF!
Transmitting (no NAT):
SIP/2.0 481 Call leg/transaction does not exist
Via: SIP/2.0/UDP 217.10.79.9;branch=z9hG4bK1e1.848513e5.1
Via: SIP/2.0/UDP 217.10.79.8;branch=z9hG4bK1e1.36e770e6.0
Via: SIP/2.0/UDP 217.10.79.30:5060;branch=z9hG4bK1fe41bfd
From: "08427985707" <sip:[email protected]>;tag=as36dc9bfc
To: <sip:[email protected]>;tag=as4af662a5
Call-ID: [email protected]
CSeq: 105 INFO
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact:
Content-Length: 0


 to 217.10.79.9:5060
Destroying call '[email protected]'


Sip read:
INFO sip:[email protected]:5060 SIP/2.0
Record-Route: <sip:[email protected];ftag=as36dc9bfc;lr=on>
Max-Forwards: 9
Record-Route: <sip:[email protected];ftag=as36dc9bfc;lr=on>
Via: SIP/2.0/UDP 217.10.79.9;branch=z9hG4bKed1.968746b6.1
Via: SIP/2.0/UDP 217.10.79.8;branch=z9hG4bKed1.a82239f7.0
Via: SIP/2.0/UDP 217.10.79.30:5060;branch=z9hG4bK0a9bb766
From: "08427985707" <sip:[email protected]>;tag=as36dc9bfc
To: <sip:[email protected]>
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 106 INFO
User-Agent: Asterisk PBX
Content-Type: application/dtmf-relay
Content-Length: 24
Sipgate-Authentication: accepted

Signal=0
Duration=250

16 headers, 2 lines
Receiving DTMF!
Transmitting (no NAT):
SIP/2.0 481 Call leg/transaction does not exist
Via: SIP/2.0/UDP 217.10.79.9;branch=z9hG4bKed1.968746b6.1
Via: SIP/2.0/UDP 217.10.79.8;branch=z9hG4bKed1.a82239f7.0
Via: SIP/2.0/UDP 217.10.79.30:5060;branch=z9hG4bK0a9bb766
From: "08427985707" <sip:[email protected]>;tag=as36dc9bfc
To: <sip:[email protected]>;tag=as510dbbc9
Call-ID: [email protected]
CSeq: 106 INFO
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact:
Content-Length: 0


 to 217.10.79.9:5060
Destroying call '[email protected]'


Sip read:
INFO sip:[email protected]:5060 SIP/2.0
Record-Route: <sip:[email protected];ftag=as36dc9bfc;lr=on>
Max-Forwards: 9
Record-Route: <sip:[email protected];ftag=as36dc9bfc;lr=on>
Via: SIP/2.0/UDP 217.10.79.9;branch=z9hG4bKfd1.335f65.1
Via: SIP/2.0/UDP 217.10.79.8;branch=z9hG4bKfd1.871d37b.0
Via: SIP/2.0/UDP 217.10.79.30:5060;branch=z9hG4bK3e208b68
From: "08427985707" <sip:[email protected]>;tag=as36dc9bfc
To: <sip:[email protected]>
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 107 INFO
User-Agent: Asterisk PBX
Content-Type: application/dtmf-relay
Content-Length: 24
Sipgate-Authentication: accepted

Signal=6
Duration=250

16 headers, 2 lines
Receiving DTMF!
Transmitting (no NAT):
SIP/2.0 481 Call leg/transaction does not exist
Via: SIP/2.0/UDP 217.10.79.9;branch=z9hG4bKfd1.335f65.1
Via: SIP/2.0/UDP 217.10.79.8;branch=z9hG4bKfd1.871d37b.0
Via: SIP/2.0/UDP 217.10.79.30:5060;branch=z9hG4bK3e208b68
From: "08427985707" <sip:[email protected]>;tag=as36dc9bfc
To: <sip:[email protected]>;tag=as3a181741
Call-ID: [email protected]
CSeq: 107 INFO
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact:
Content-Length: 0


 to 217.10.79.9:5060
Destroying call '[email protected]'


Sip read:
INFO sip:[email protected]:5060 SIP/2.0
Record-Route: <sip:[email protected];ftag=as36dc9bfc;lr=on>
Max-Forwards: 9
Record-Route: <sip:[email protected];ftag=as36dc9bfc;lr=on>
Via: SIP/2.0/UDP 217.10.79.9;branch=z9hG4bKde1.726924c7.1
Via: SIP/2.0/UDP 217.10.79.8;branch=z9hG4bKde1.9f80be33.0
Via: SIP/2.0/UDP 217.10.79.30:5060;branch=z9hG4bK228287fc
From: "08427985707" <sip:[email protected]>;tag=as36dc9bfc
To: <sip:[email protected]>
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 108 INFO
User-Agent: Asterisk PBX
Content-Type: application/dtmf-relay
Content-Length: 24
Sipgate-Authentication: accepted

Signal=7
Duration=250

16 headers, 2 lines
Receiving DTMF!
Transmitting (no NAT):
SIP/2.0 481 Call leg/transaction does not exist
Via: SIP/2.0/UDP 217.10.79.9;branch=z9hG4bKde1.726924c7.1
Via: SIP/2.0/UDP 217.10.79.8;branch=z9hG4bKde1.9f80be33.0
Via: SIP/2.0/UDP 217.10.79.30:5060;branch=z9hG4bK228287fc
From: "08427985707" <sip:[email protected]>;tag=as36dc9bfc
To: <sip:[email protected]>;tag=as6280ba31
Call-ID: [email protected]
CSeq: 108 INFO
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact:
Content-Length: 0


 to 217.10.79.9:5060
Destroying call '[email protected]'


Sip read:
INFO sip:[email protected]:5060 SIP/2.0
Record-Route: <sip:[email protected];ftag=as36dc9bfc;lr=on>
Max-Forwards: 9
Record-Route: <sip:[email protected];ftag=as36dc9bfc;lr=on>
Via: SIP/2.0/UDP 217.10.79.9;branch=z9hG4bKce1.4d28ba6.1
Via: SIP/2.0/UDP 217.10.79.8;branch=z9hG4bKce1.0d000fa5.0
Via: SIP/2.0/UDP 217.10.79.30:5060;branch=z9hG4bK3201dc42
From: "08427985707" <sip:[email protected]>;tag=as36dc9bfc
To: <sip:[email protected]>
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 109 INFO
User-Agent: Asterisk PBX
Content-Type: application/dtmf-relay
Content-Length: 24
Sipgate-Authentication: accepted

Signal=2
Duration=250

16 headers, 2 lines
Receiving DTMF!
Transmitting (no NAT):
SIP/2.0 481 Call leg/transaction does not exist
Via: SIP/2.0/UDP 217.10.79.9;branch=z9hG4bKce1.4d28ba6.1
Via: SIP/2.0/UDP 217.10.79.8;branch=z9hG4bKce1.0d000fa5.0
Via: SIP/2.0/UDP 217.10.79.30:5060;branch=z9hG4bK3201dc42
From: "08427985707" <sip:[email protected]>;tag=as36dc9bfc
To: <sip:[email protected]>;tag=as1992b0db
Call-ID: [email protected]
CSeq: 109 INFO
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact:
Content-Length: 0


 to 217.10.79.9:5060
Destroying call '[email protected]'


Sip read:
INFO sip:[email protected]:5060 SIP/2.0
Record-Route: <sip:[email protected];ftag=as36dc9bfc;lr=on>
Max-Forwards: 9
Record-Route: <sip:[email protected];ftag=as36dc9bfc;lr=on>
Via: SIP/2.0/UDP 217.10.79.9;branch=z9hG4bK4c2.9772d1e3.1
Via: SIP/2.0/UDP 217.10.79.8;branch=z9hG4bK4c2.78b08e04.0
Via: SIP/2.0/UDP 217.10.79.30:5060;branch=z9hG4bK66e9d372
From: "08427985707" <sip:[email protected]>;tag=as36dc9bfc
To: <sip:[email protected]>
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 110 INFO
User-Agent: Asterisk PBX
Content-Type: application/dtmf-relay
Content-Length: 24
Sipgate-Authentication: accepted

Signal=4
Duration=250

16 headers, 2 lines
Receiving DTMF!
Transmitting (no NAT):
SIP/2.0 481 Call leg/transaction does not exist
Via: SIP/2.0/UDP 217.10.79.9;branch=z9hG4bK4c2.9772d1e3.1
Via: SIP/2.0/UDP 217.10.79.8;branch=z9hG4bK4c2.78b08e04.0
Via: SIP/2.0/UDP 217.10.79.30:5060;branch=z9hG4bK66e9d372
From: "08427985707" <sip:[email protected]>;tag=as36dc9bfc
To: <sip:[email protected]>;tag=as3ae2bf16
Call-ID: [email protected]
CSeq: 110 INFO
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact:
Content-Length: 0


 to 217.10.79.9:5060
Destroying call '[email protected]'
Mir ist insbesondere das Aufgefallen
SIP/2.0 481 Call leg / transaction does not exists

Ports hab ich folgende Freigegeben (an den Asterisk
5060, 10000 - 32000, und 21
Fehlen da noch welche?

Das ist meine Nebenstelle 11


Code:
*CLI> sip show peer 11


  * Name       : 11
  Secret       : <Set>
  MD5Secret    : <Not set>
  Context      : default
  Language     :
  FromUser     :
  FromDomain   :
  Callgroup    :  (0)
  Pickupgroup  :  (0)
  Mailbox      :
  LastMsgsSent : -1
  Dynamic      : Yes
  Expire       : 2
  Expiry       : 900
  Insecure     : No
  Nat          : No
  ACL          : No
  CanReinvite  : No
  PromiscRedir : No
  DTMFmode     : inband
  LastMsg      : 0
  ToHost       :
  Addr->IP     : 192.168.6.5 Port 5060
  Defaddr->IP  : 0.0.0.0 Port 5060
  Username     : 11
  Codecs       : ULAW ALAW
  Status       : OK (942 ms)
  Useragent    :
  Full Contact : sip:[email protected]:5060;line=pxtx8eem
Und das die Verbindung zu sipgate

Code:
*CLI> sip show peer sipgate


  * Name       : sipgate
  Secret       : <Set>
  MD5Secret    : <Not set>
  Context      : sipgate
  Language     :
  FromUser     : 8006724
  FromDomain   :
  Callgroup    :  (0)
  Pickupgroup  :  (0)
  Mailbox      :
  LastMsgsSent : -1
  Dynamic      : No
  Expire       : -1
  Expiry       : 900
  Insecure     : No
  Nat          : Always
  ACL          : No
  CanReinvite  : No
  PromiscRedir : No
  DTMFmode     : rfc2833
  LastMsg      : 0
  ToHost       : sipgate.de
  Addr->IP     : 217.10.79.9 Port 5060
  Defaddr->IP  : 0.0.0.0 Port 0
  Username     : 8006724
  Codecs       : ALAW H.263
  Status       : OK (99 ms)
  Useragent    :
  Full Contact :
Alle Nebenstellen

Code:
*CLI> sip show peers
Name/username    Host            Dyn Nat ACL Mask             Port     Status
11/11            192.168.6.5      D          255.255.255.255  5060     OK (43 ms)
10/10            (Unspecified)    D          255.255.255.255  0        Unmonitored
sipgate/8006724  217.10.79.9          N      255.255.255.255  5060     OK (97 ms)
Meine überarbeitete sip.conf
Code:
[general]
port=5060
bindaddr=192.168.6.1
context=default
srvlookup=yes
insecure=very
disallow=gsm
allow=alaw
disallow=ulaw
register=> 8006724:******@sipgate.de/8006724


[sipgate]
type=friend
username=8006724
secret=*******               
host=sipgate.de
fromuser=8006724
nat=yes
context=sipgate
canreinvite=no
qualify=yes

[10]
type=friend
username=10
secret=10
host=dynamic
callerid="10"=<10>
nat=no

[11]
type=friend
username=11
secret=11
host=dynamic
callerid="11"=<11>
dtmfmode=inband
disallow=all
allow=ulaw
allow=alaw
qualify=yes
canreinvite=no
nat=no
Die extensions.conf

Code:
[general]
static=yes
writeprotect=no


[default]
include=> 10
include=> 11
include=> ausgsipgate



[10]
exten=>10,1,Dial(SIP/10)
exten=>10,2 Hangup


[11]
exten=>11,1,Dial(SIP/11)
exten=>11,2,Hangup

[ausgsipgate]
exten=> _0.,1,Dial(SIP/${EXTEN:1}@sipgate,30,tr)
exten=> _0.,2,Playback(invalid)
exten=> _0.,3,Hangup

[sipgate]
exten=> h,1,Hangup
exten=> 800XXXX,1,Dial(SIP/11,30,tr)
Ich weis echt nicht mehr weiter, Ich hock wirklich schon seit heute vormittag am Problem rum und weis nicht was falsch ist.
 

BlackSektor

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Mein Router ist UPNP fähig. Deshalb hab ich jetzt mal das ganze Portforwarding rausgemacht.
Es funktioniert immer noch nicht.
 

Hupe

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Hmm, das tut mir echt leid, dass es nicht funzt. Ein paar kleinigkeiten habe ich noch:
ergänze mal in der sip conf:
bei [sipgate]
insecure=very
fromdomain=sipgate.de
disallow=all
allow=g726
allow=alaw
allow=ulaw

außerdem sollte bei [general]:
disallow=all und nicht disallow=gsm stehen.
Sonst ist alles außer GSM erlaubt.
 

BlackSektor

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Hallo, jetzt komm ich gar nicht mehr raus.

Könnte mir mal jemand, der sipgate erfolgrich mit Nebenstellen am laufen hat seine sip.conf und seinen dialplan zuschicken.

Außdem wäre es mal interessant welche Ports ich am Netgear WGR 614 v. 4 (neueste Firmware) freigeben muß. Bei UPNP hätte ich gedacht, daß ich gar keine freigeben muß.

Ich hab auch schon versucht, die DMZ auf dem Asterisk zu legen. Funktioniert auch nicht.



Bitte die confs per PN oder Board-E-Mail bzw. Info, dann gebe ich die richtige Adresse.

Danke schonmal.

Edit von Christoph: Habe die E-Mail-Adresse zum Schutz vor Spam entfernt.


OK, bitte konfigs am mein Board Postfach Danke!
 

BlackSektor

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Keiner da, der mir seine Konfigs per PN zukommen lassen kann (Natürlich ohne Passwort)
 

Netview

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damit klingeln dann beide Geräte bei eingehenden sipgate- Rufen:

in der extensions.conf:

exten => 8006724,1,Dial(SIP/10&SIP/11,60)
 

BlackSektor

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So, raus bzw. reintelefonieren geht. Blos ich höre den anderen gesprächspartner nicht und es wird jeweils aufgelegt. Meine configs stimmen soweit (denke ich). Mein * befindet sich hinter dem DSL Router.
Giebts den Router, die das korrekt durchleiten können, oder muß ich wirklich den * Server auch zum DSL Router machen?
 

Netview

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Folgende ports sind freizugeben:

10000-20000 (udp) gemäss rtp.conf
5060 (udp) gemäss sip.conf (port=5060 unter general:)
NAT=yes setzen (für alle clients und peers)!
 

3CX PBX - GRATIS
Linux / Win / Cloud

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