- Mitglied seit
- 3 Nov 2004
- Beiträge
- 34
- Punkte für Reaktionen
- 0
- Punkte
- 0
Hallo, ich kann nun mit meinem Asterisk raustelefonieren (über Sipgate)
angerufen werden kann ich jedoch nicht.
Vieleicht kann mir jemand helfen
Hier mein dialplan
und hier die konfigs
sip.conf
extensions.conf
hier meine peers
Und jetzt die debug ausgaben beim anruf
Die interne Nebenstelle kann ich aus irgendwelchen gründen auch nicht anwählen?
Vieleicht kann mir ja jemand helfen
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:49211580067[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 '[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]17.10.79.30>;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]9.30'
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