[Gelöst] keine ausgehende Telefonie möglich (Kabel Deutschland) / All Circuits are busy now

[D4735]

Hallo,

ich hoffe es kann mir jemand weiterhelfen.
Ich komme einfach mit den ausgehenden Telefonaten nicht klar, eingehend funktioniert.

Einige Daten:

Daten aus der Fritzbox:

Daten aus der Fritzbox:
                username = "+49510000000";
                authname = "222222222_KAV_1";
                passwd = "xxxxxxxxxxx";
                registrar = "reg181.kabelphone.de";

sip show peers

Asterisk*CLI> sip show peers
Name/username             Host                                    Dyn Forcerport Comedia    ACL Port     Status      Description        
51000000000/222222222_KAV 88.134.209.1                                Yes        Yes            5060     OK (25 ms)                     
700/700                   192.168.73.51                            D  No         No          A  5060     OK (19 ms)

Anruf bei der Kabel-Deutschland-Support-Hotline ;-)

root@Asterisk:~# asterisk -r
Asterisk 13.13.1, Copyright (C) 1999 - 2014, Digium, Inc. and others.
Created by Mark Spencer <[email protected]>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type 'core show license' for details.
=========================================================================
Connected to Asterisk 13.13.1 currently running on Asterisk (pid = 2020)

<--- SIP read from UDP:192.168.73.51:5060 --->
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.73.51:5060;branch=z9hG4bK1947317668
From: "Arbeit" <sip:[email protected]:5060>;tag=3504696826
To: <sip:[email protected]:5060>
Call-ID: [email protected]
CSeq: 1 INVITE
Contact: <sip:[email protected]:5060>
Content-Type: application/sdp
Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE
Max-Forwards: 70
User-Agent: Yealink SIP-T46G 28.81.0.91
Allow-Events: talk,hold,conference,refer,check-sync
Supported: replaces
Content-Length: 308

v=0
o=- 21045 21045 IN IP4 192.168.73.51
s=SDP data
c=IN IP4 192.168.73.51
t=0 0
m=audio 11836 RTP/AVP 9 0 8 18 101
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=ptime:20
a=sendrecv
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<------------->
--- (14 headers 15 lines) ---
Sending to 192.168.73.51:5060 (NAT)
Sending to 192.168.73.51:5060 (NAT)
Using INVITE request as basis request - [email protected]
Found peer '700' for '700' from 192.168.73.51:5060

<--- Reliably Transmitting (no NAT) to 192.168.73.51:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.73.51:5060;branch=z9hG4bK1947317668;received=192.168.73.51
From: "Arbeit" <sip:[email protected]:5060>;tag=3504696826
To: <sip:[email protected]:5060>;tag=as69eb7529
Call-ID: [email protected]
CSeq: 1 INVITE
Server: Windows-Phone
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1ea701c8"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '[email protected]' in 6400 ms (Method: INVITE)

<--- SIP read from UDP:192.168.73.51:5060 --->
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.73.51:5060;branch=z9hG4bK1947317668
From: "Arbeit" <sip:[email protected]:5060>;tag=3504696826
To: <sip:[email protected]:5060>;tag=as69eb7529
Call-ID: [email protected]
CSeq: 1 ACK
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---

<--- SIP read from UDP:192.168.73.51:5060 --->
INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.73.51:5060;branch=z9hG4bK1947873010
From: "Arbeit" <sip:[email protected]:5060>;tag=3504696826
To: <sip:[email protected]:5060>
Call-ID: [email protected]
CSeq: 2 INVITE
Contact: <sip:[email protected]:5060>
Authorization: Digest username="700", realm="asterisk", nonce="1ea701c8", uri="sip:[email protected]:5060", response="bab06bd1febe422615f6baa98073d3f9", algorithm=MD5
Content-Type: application/sdp
Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE
Max-Forwards: 70
User-Agent: Yealink SIP-T46G 28.81.0.91
Allow-Events: talk,hold,conference,refer,check-sync
Supported: replaces
Content-Length: 308

v=0
o=- 21045 21045 IN IP4 192.168.73.51
s=SDP data
c=IN IP4 192.168.73.51
t=0 0
m=audio 11836 RTP/AVP 9 0 8 18 101
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=ptime:20
a=sendrecv
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<------------->
--- (15 headers 15 lines) ---
Sending to 192.168.73.51:5060 (no NAT)
Using INVITE request as basis request - [email protected]
Found peer '700' for '700' from 192.168.73.51:5060
Found RTP audio format 9
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 101
Found audio description format G722 for ID 9
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 101
Capabilities: us - (g722|g729|ulaw|alaw|gsm), peer - audio=(ulaw|alaw|g722|g729)/video=(nothing)/text=(nothing), combined - (g722|g729|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.73.51:11836
Looking for 08005266625 in from-internal (domain 192.168.73.17)
sip_route_dump: route/path hop: <sip:[email protected]:5060>

<--- Transmitting (no NAT) to 192.168.73.51:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.73.51:5060;branch=z9hG4bK1947873010;received=192.168.73.51
From: "Arbeit" <sip:[email protected]:5060>;tag=3504696826
To: <sip:[email protected]:5060>
Call-ID: [email protected]
CSeq: 2 INVITE
Server: Windows-Phone
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:[email protected]:5060>
Content-Length: 0


<------------>
[2019-07-09 09:03:19] WARNING[2292]: func_cdr.c:383 cdr_write_callback: CDR requires a value (CDR(variable)=value)
[2019-07-09 09:03:19] WARNING[15065][C-00000018]: app_dial.c:2525 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)
Audio is at 28620
Adding codec g722 to SDP
Adding codec g729 to SDP
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 192.168.73.51:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.73.51:5060;branch=z9hG4bK1947873010;received=192.168.73.51
From: "Arbeit" <sip:[email protected]:5060>;tag=3504696826
To: <sip:[email protected]:5060>;tag=as0e310a1c
Call-ID: [email protected]
CSeq: 2 INVITE
Server: Windows-Phone
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:[email protected]:5060>
Content-Type: application/sdp
Content-Length: 349

v=0
o=root 1405485322 1405485322 IN IP4 192.168.73.17
s=Asterisk PBX 13.13.1
c=IN IP4 192.168.73.17
t=0 0
m=audio 28620 RTP/AVP 9 18 0 8 101
a=rtpmap:9 G722/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

<------------>
Really destroying SIP dialog '[email protected]:5060' Method: INVITE

<--- SIP read from UDP:192.168.73.51:5060 --->
ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.73.51:5060;branch=z9hG4bK2218398477
From: "Arbeit" <sip:[email protected]:5060>;tag=3504696826
To: <sip:[email protected]:5060>;tag=as0e310a1c
Call-ID: [email protected]
CSeq: 2 ACK
Contact: <sip:[email protected]:5060>
Max-Forwards: 70
User-Agent: Yealink SIP-T46G 28.81.0.91
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---

Trunk-Outgoing

username=222222222_KAV_1
type=friend
secret=xxxxxxxxxxx
qualify=yes
outboundproxy=prox01.kabelphone.de
nat=yes
insecure=port,invite
host=prox01.kabelphone.de
fromuser=+49510000000
fromdomain=reg181.kabelphone.de
disallow=all
canreinvite=no
caninvite=no
allow=alaw,ulaw

General-Sip-Settings sind u. a.
NAT=YES
und ExterneIP vergeben.
BindAddr=0.0.0.0
(Sorry, ich nutze FreePBX)



Gefühlt habe ich alles schon 3* umgestellt - hat jemand eine Idee?? (Danke,danke,danke,danke)

 

[sparkie]

muss das nicht 'kabelfon' statt 'kabelphone' heissen? Bei mir sieht der Komplett-Setup so aus: #9
soweit keine Probleme hier...

 

[D4735]

Mittlerweile hat das irgendwann (warum auch immer funktioniert), mit den Daten:

Outgoing:

username=222222222_KAV_1
type=peer
secret=xxxxxxxxxxx
qualify=yes
outboundproxy=prox01.kabelphone.de
nat=yes
insecure=port,invite
host=prox01.kabelphone.de
fromuser=+49510000000
fromdomain=reg181.kabelphone.de
disallow=all
canreinvite=no
caninvite=no
allow=alaw,ulaw

Incoming (wahrscheinlich überflüssiges drinnen):

username=222222222_KAV_1
type=peer
session-timers=refuse
session-expires=240
secret=xxxxxxxxxxx
qualify=yes
outboundproxy=prox01.kabelphone.de
nat=yes
insecure=invite
host=prox01.kabelphone.de
fromuser=+49510000000
fromdomain=reg181.kabelphone.de
dtmfmode=rfc2833&auto
disallow=all
directmedia=no
canreinvite=no
allow=alaw

Registrierung:

[email protected]:"xxxxxxxxxxx":[email protected]/0510000000

Diese Server sind die "alten" Kabel-Deutschland-Server, mittlerweile hatte ich dem technischen Wissen der Vodafone-Hotline vertraut, mit dem Ergebnis des Vertragswechsels, VOC (und kein VOIP) und dann ganz zum Schluss mit einem selbst gekauften Kabelmodem, sodass ich jetzt sogar offiziell SIP-Daten habe.
Dann auch mit den Server "sip.kabelfon.vodafone.de" anstelle von "kabelphone.de"