Tiscali phone flat und asterisk

Chris78

Neuer User
Mitglied seit
2 Dez 2004
Beiträge
89
Punkte für Reaktionen
0
Punkte
0
Hallo.
ich bekomme leider den Asterisk nicht an den Tiscali sip server angemeldet, ich bekommme immer folgende fehlermeldungen.

Sip read:
SIP/2.0 479 Please use Tiscali-Access-Products for using this Service
Via: SIP/2.0/UDP 192.168.1.63:5060;branch=z9hG4bK13e1c738;received=84.174.xx.x
From: <sip:[email protected]>;tag=as6c91b928
To: <sip:[email protected]>;tag=17d15667d4a34409cba6e324d32e4f46.518d
Call-ID: [email protected]
CSeq: 106 REGISTER
Server: Sip EXpress router (0.8.14 (i386/freebsd))
Content-Length: 0
Warning: 392 62.26.xx.x:5060 "Noisy feedback tells: pid=20740 req_src_ip=84.174.xx.xx req_src_port=5060 in_uri=sip:tiscali.de out_uri=sip:tiscali.de via_cnt==1"


9 headers, 0 lines
-- Got SIP response 479 "Please use Tiscali-Access-Products for using this Service" back from 62.26.xx.x
Destroying call '[email protected]'
May 18 09:08:27 NOTICE[20464]: chan_sip.c:4018 sip_reg_timeout: Registration for '[email protected]' timed out, trying again
11 headers, 0 lines
Reliably Transmitting:
REGISTER sip:tiscali.de SIP/2.0
Via: SIP/2.0/UDP 192.168.1.63:5060;branch=z9hG4bK1ffea72a
From: <sip:[email protected]>;tag=as494d031c
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 107 REGISTER
User-Agent: Asterisk PBX
Expires: 120
Contact: <sip:[email protected]>
Event: registration
Content-Length: 0

ich habe in der sip.conf nat= yes und externip auch auf ein dyndns gelegt. aber leider klappt es nicht... Hat jemand eine Idee..?
 
Folgendes

Got SIP response 479 "Please use Tiscali-Access-Products for using this Service" back from 62.26.xx.x

würde mich stutzig machen. ;)
 
ot SIP response 479 "Please use Tiscali-Access-Products for using this Service" back from 62.26.64.3

warum?
 
Weil es sich ganz danach anhört, als wäre der SER auf Tiscali-Seite so konfiguriert, dass er anderen als speziell von Tiscali angepasste Geräten / Softphones die Anmeldung verweigert, z.B. dadurch, dass ein bestimmter String erwartet wird.

Was sagt denn der Support dazu?
 
ggf. fromuser / fromdomain in der sip.conf setzen und den realm statt asterisk mal auf was anderes setzen.

Wobei die Frage ist, wie das überprüft wird -- kann ja im Prinzip nur am "User-Agent: Asterisk PBX" liegen. Müsste man wissen was Tiscali an dieser Stelle hören will ;)

Aber schon seltsam das ...
 
habe das Problem gelöst und selbst verursacht, ich hatte zwar Tiscali als neuen Provider eingetragen aber kein rcnetwork restart gemacht. So war der Rechner noch mit t-online online...

jetzt hab ich noch Probleme mit der Anmeldung. über Xlite kann ich mich anmelden aber noch nicht über asterisk.

sip debug peer tiscali
SIP Debugging Enabled for IP: 62.26.64.3:5060
*CLI> May 18 14:07:53 NOTICE[21185]: chan_sip.c:4018 sip_reg_timeout: Registration for '[email protected]' timed out, trying again
-- parse_srv: SRV mapped to host sip.tiscali.de, port 5060
11 headers, 0 lines
Reliably Transmitting:
REGISTER sip:tiscali.de SIP/2.0
Via: SIP/2.0/UDP 192.168.1.63:5060;branch=z9hG4bK72b46cb0
From: <sip:[email protected]>;tag=as305d7e43
To: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 112 REGISTER
User-Agent: Asterisk PBX
Expires: 120
Contact: <sip:[email protected]>
Event: registration
Content-Length: 0

(no NAT) to 62.26.64.3:5060


Sip read:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.63:5060;branch=z9hG4bK72b46cb0;received=83.129.xx.xx
From: <sip:[email protected]>;tag=as305d7e43
To: <sip:[email protected]>;tag=17d15667d4a34409cba6e324d32e4f46.05d7
Call-ID: [email protected]
CSeq: 112 REGISTER
WWW-Authenticate: Digest realm="tiscali.de", nonce="428b301ebdc848d7cc77b986732584994e53a623"
Server: Sip EXpress router (0.8.14 (i386/freebsd))
Content-Length: 0
Warning: 392 62.26.64.3:5060 "Noisy feedback tells: pid=20743 req_src_ip=83.129.xx.xx req_src_port=5060 in_uri=sip:tiscali.de out_uri=sip:tiscali.de via_cnt==1"


10 headers, 0 lines
12 headers, 0 lines
Reliably Transmitting:
REGISTER sip:tiscali.de SIP/2.0
Via: SIP/2.0/UDP 192.168.1.63:5060;branch=z9hG4bK4e922831
From: <sip:[email protected]>;tag=as305d7e43
To: <sip:[email protected]>;tag=17d15667d4a34409cba6e324d32e4f46.05d7
Call-ID: [email protected]
CSeq: 113 REGISTER
User-Agent: Asterisk PBX
Authorization: Digest username="user", realm="tiscali.de", algorithm=MD5, uri="sip:tiscali.de", nonce="428b301ebdc848d7cc77b986732584994e53a623", response="5929c9b2b9e5f4603110bceb314a2d62", opaque=""
Expires: 120
Contact: <sip:[email protected]>
Event: registration
Content-Length: 0

einstellungen in der sip.conf:

register => user:[email protected]/tiscali


[tiscali]
type=friend
realm=tiscali.de
host=sip.tiscali.de
username=christoph.hehl
fromdomain=tiscali.de
fromuser=christoph.hehl
context=incoming
secret=password
canreinvite=no
;qualify=yes
;disallow=all
allow=gsm
insecure=very
nat=yes
dtmfmode=info
tos=0x18
 
Schon mal versucht, ein paar Codecs mehr dazuzupacken?
 
hallo.
habe leider immernoch schwierigkeiten:
- Executing Dial("Zap/2-1", "SIP/@{EXTEN:1}@tisphone||tr") in new stack
May 20 09:38:02 WARNING[17373]: chan_sip.c:1389 create_addr: No such host: {EXTEN
May 20 09:38:02 NOTICE[17373]: app_dial.c:759 dial_exec: Unable to create channel of type 'SIP'

meine Sip.conf
[general]

bindaddr = 0.0.0.0
localnet = 192.168.1.0/24
externhost = dyndns
port = 5060
context = default
maxexpirey = 3600
defaultexpirey = 120
srvlookup = yes
tos = 0x18
disallow = all
allow = gsm
allow = alaw
allow = ulaw
allow = g729


[tisphone]
host = tiscali.de
realm = tiscali.de
fromuser = user
fromdomain = tiscali.de
user = user
username = user
secret = password
qualify = no
nat = yes
allow = ulaw
allow = gsm
allow = g729
type = friend


meine Extensions:
exten => _2.,1,Dial(SIP/@{EXTEN:1}@tisphone,,tr)
exten => _2.,2,Hangup
 
exten => _2.,1,Dial(SIP/@{EXTEN:1}@tisphone,,tr)
das ist falsch, da ist ein @ zuviel:
exten => _2.,1,Dial(SIP/{EXTEN:1}@tisphone,,tr)
 
Leider lässt das nächste Problem nicht auf sich warten.: ( mit Kphone kein problem)
== CDR updated on Zap/2-1
-- Executing Dial("Zap/2-1", "SIP/{EXTEN:1}@tisphone||tr") in new stack
-- Called {EXTEN:1}@tisphone
May 20 11:17:51 WARNING[18281]: chan_sip.c:685 retrans_pkt: Maximum retries exceeded on call [email protected] for seqno 102 (Critical Request)
== No one is available to answer at this time
-- Executing Hangup("Zap/2-1", "") in new stack



Code:
sip debug peer tisphone
SIP Debugging Enabled for IP: 62.26.64.21:5060
  == CDR updated on Zap/2-1
    -- Executing Dial("Zap/2-1", "SIP/{EXTEN:1}@tisphone||tr") in new stack
We're at 192.168.1.12 port 14206
Answering/Requesting with root capability 8
Answering with preferred capability 0x2 (gsm)
Answering with preferred capability 0x4 (ulaw)
Answering with preferred capability 0x100 (g729)
Answering with non-codec capability 0x1 (telephone-event)
12 headers, 13 lines
Reliably Transmitting:
INVITE sip:{EXTEN:1}@tiscali.de SIP/2.0
Via: SIP/2.0/UDP 192.168.1.12:5060;branch=z9hG4bK26e56fbf;rport
From: "CID withheld" <sip:[email protected]>;tag=as4f0b5f76
To: <sip:{EXTEN:1}@tiscali.de>
Contact: <sip:[email protected]>
Call-ID: [email][email protected][/email]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Fri, 20 May 2005 09:13:33 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 289

v=0
o=root 18236 18236 IN IP4 192.168.1.12
s=session
c=IN IP4 192.168.1.12
t=0 0
m=audio 14206 RTP/AVP 8 3 0 18 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
 (NAT) to 62.26.64.21:5060
    -- Called {EXTEN:1}@tisphone
Retransmitting #1 (NAT):
INVITE sip:{EXTEN:1}@tiscali.de SIP/2.0
Via: SIP/2.0/UDP 192.168.1.12:5060;branch=z9hG4bK26e56fbf;rport
From: "CID withheld" <sip:[email protected]>;tag=as4f0b5f76
To: <sip:{EXTEN:1}@tiscali.de>
Contact: <sip:[email protected]>
Call-ID: [email][email protected][/email]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Fri, 20 May 2005 09:13:33 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 289

v=0
o=root 18236 18236 IN IP4 192.168.1.12
s=session
c=IN IP4 192.168.1.12
t=0 0
m=audio 14206 RTP/AVP 8 3 0 18 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

 to 62.26.64.21:5060
Retransmitting #2 (NAT):
INVITE sip:{EXTEN:1}@tiscali.de SIP/2.0
Via: SIP/2.0/UDP 192.168.1.12:5060;branch=z9hG4bK26e56fbf;rport
From: "CID withheld" <sip:[email protected]>;tag=as4f0b5f76
To: <sip:{EXTEN:1}@tiscali.de>
Contact: <sip:[email protected]>
Call-ID: [email][email protected][/email]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Fri, 20 May 2005 09:13:33 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 289

v=0
o=root 18236 18236 IN IP4 192.168.1.12
s=session
c=IN IP4 192.168.1.12
t=0 0
m=audio 14206 RTP/AVP 8 3 0 18 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

 to 62.26.64.21:5060
Retransmitting #3 (NAT):
INVITE sip:{EXTEN:1}@tiscali.de SIP/2.0
Via: SIP/2.0/UDP 192.168.1.12:5060;branch=z9hG4bK26e56fbf;rport
From: "CID withheld" <sip:[email protected]>;tag=as4f0b5f76
To: <sip:{EXTEN:1}@tiscali.de>
Contact: <sip:[email protected]>
Call-ID: [email][email protected][/email]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Fri, 20 May 2005 09:13:33 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 289

v=0
o=root 18236 18236 IN IP4 192.168.1.12
s=session
c=IN IP4 192.168.1.12
t=0 0
m=audio 14206 RTP/AVP 8 3 0 18 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

 to 62.26.64.21:5060
Retransmitting #4 (NAT):
INVITE sip:{EXTEN:1}@tiscali.de SIP/2.0
Via: SIP/2.0/UDP 192.168.1.12:5060;branch=z9hG4bK26e56fbf;rport
From: "CID withheld" <sip:[email protected]>;tag=as4f0b5f76
To: <sip:{EXTEN:1}@tiscali.de>
Contact: <sip:[email protected]>
Call-ID: [email][email protected][/email]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Fri, 20 May 2005 09:13:33 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 289

v=0
o=root 18236 18236 IN IP4 192.168.1.12
s=session
c=IN IP4 192.168.1.12
t=0 0
m=audio 14206 RTP/AVP 8 3 0 18 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

 to 62.26.64.21:5060
Retransmitting #5 (NAT):
INVITE sip:{EXTEN:1}@tiscali.de SIP/2.0
Via: SIP/2.0/UDP 192.168.1.12:5060;branch=z9hG4bK26e56fbf;rport
From: "CID withheld" <sip:[email protected]>;tag=as4f0b5f76
To: <sip:{EXTEN:1}@tiscali.de>
Contact: <sip:[email protected]>
Call-ID: [email][email protected][/email]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Fri, 20 May 2005 09:13:33 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 289

v=0
o=root 18236 18236 IN IP4 192.168.1.12
s=session
c=IN IP4 192.168.1.12
t=0 0
m=audio 14206 RTP/AVP 8 3 0 18 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

 to 62.26.64.21:5060
May 20 11:13:39 WARNING[18236]: chan_sip.c:685 retrans_pkt: Maximum retries exceeded on call [email][email protected][/email] for seqno 102 (Critical Request)
 
Holen Sie sich 3CX - völlig kostenlos!
Verbinden Sie Ihr Team und Ihre Kunden Telefonie Livechat Videokonferenzen

Gehostet oder selbst-verwaltet. Für bis zu 10 Nutzer dauerhaft kostenlos. Keine Kreditkartendetails erforderlich. Ohne Risiko testen.

3CX
Für diese E-Mail-Adresse besteht bereits ein 3CX-Konto. Sie werden zum Kundenportal weitergeleitet, wo Sie sich anmelden oder Ihr Passwort zurücksetzen können, falls Sie dieses vergessen haben.