Asterisk 13 hinter Fritzbox

HrGesangsverein

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Hallo,

bestimmt bin ich nicht der erste, der das machen will, aber ich finde keine passende Lösung. Ich möchte Asterisk 13.7 in einer VM (IP 10.10.7.15) hinter einer fritzbox (IP 10.10.7.1, als Gateway) betreiben. Dazu habe ich ein Softphone auf einem Win-Rechner eingerichtet, Nebenstelle 102, mit der ich rein und raus telefonieren will. Kurz: rein geht, raus nicht. Die Ausgabe von Asterisk -vvvvv zeigt:
Code:
== Using SIP RTP CoS mark 5
    -- Executing [[email protected]:1] Dial("SIP/102-00000000", "SIP/[email protected],45,r") in new stack
  == Using SIP RTP CoS mark 5
    -- Called SIP/[email protected]
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Auto fallthrough, channel 'SIP/102-00000000' status is 'CHANUNAVAIL'
Im Softphone erhalte ich:
Call failure
SIP 503 - Service unavailable

Meine sip.conf:
Code:
[general]
disallow=all
allow=ulaw
allow=alaw
allow=gsm
qualify=yes
canreinvite=no
context=incoming
register => 620:[email protected]/XXXX519
register => 621:[email protected]/XXXX7096
register => 622:[email protected]/XXXX7097
register => 623:[email protected]/XXXX7098

[620]
user=620
type=peer
canreinvite=no
secret=sip0tel
host=10.10.7.1
nat=no
context=fritzbox-in
fromdomain=fritz.box
dtmfmode=rfc2833

[101]
type=friend
context=outgoing
host=dynamic
secret=sip0tel

[102]
type=friend
context=outgoing
host=dynamic
secret=sip0tel
und meine extensions.conf:
Code:
[fritzbox-in]
exten => XXXX519,1,Dial(SIP/101&SIP/102,60)
exten => XXXX519,n,Hangup()

[outgoing]
exten => _X.,1,Dial(SIP/${EXTEN}@620,45,r)
 
R

rentier-s

Guest
Ändere user in defaultuser, secret in remotesecret und ergänze authuser=620, dann sollte das klappen.

canreinvite heißt übrigens schon lange directmedia.
 

HrGesangsverein

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Hallo,

ich habe die Änderungen gemacht. Wofür sind die denn gut? Geändert hat sich aber nichts...

sip.conf:
Code:
[general]
disallow=all
allow=ulaw
allow=alaw
allow=gsm
qualify=yes
canreinvite=no
context=incoming
register => 620:[email protected]/7852519
register => 621:[email protected]/68007096
register => 622:[email protected]/68007097
register => 623:[email protected]/68007098

[620]
defaultuser=620
authuser=620
type=peer
directmedia=no
remotesecret=sip0tel
host=10.10.7.1
nat=no
context=fritzbox-in
fromdomain=fritz.box
dtmfmode=rfc2833

[101]
type=friend
context=outgoing
host=dynamic
secret=sip0tel

[102]
type=friend
context=outgoing
host=dynamic
secret=sip0tel
 
R

rentier-s

Guest
Naja, user gibt es nicht, das heißt halt defaultuser. authuser wäre zusätzlich bzw. explizit für die Authentifizierung. remotesecret bewirkt im Gegensatz zu secret, dass Du kein insecure setzen musst wenn defaultuser gesetzt ist.

Vielleicht braucht die Fritzbox noch fromuser=620, weiß ich grad nicht.

Zumindest fällt mir gerade sonst nichts mehr auf, mach doch bitte SIP Debug an (sip set debug peer 620) und probier es noch mal. Da kommt seeeehr viel Text, interessant sind die INVITE Pakete und die Antwort darauf. Mit sip set debuf off kannst Du das Debug wieder ausschalten.
 

HrGesangsverein

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sip set debug on

Hallo,

in der Tat kommen sehr viele Meldungen, die für mich schwer bis gar nicht zu interpretieren sind.
Ich erkenne 101 als die Nebenstelle, von der ich anrufe, 01795555555 ist die Nummer, die ich anrufen will.
10.10.7.15 ist die Adresse von Asterisk, 10.10.7.1 ist die Adresse der fritz box.
Interessanterweise sind die Nummern wild mit den IP Adressen vermischt. Das gibt für mich keinen Sinn...

Es scheint, dass auch meine externe IP auftaucht.

Code:
<--- SIP read from UDP:10.10.7.22:64444 --->
INVITE sip:[email protected];transport=UDP SIP/2.0
Via: SIP/2.0/UDP 93.220.19.170:64444;branch=z9hG4bK-d8754z-d1d628ec70933b7c-1---d8754z-
Max-Forwards: 70
Contact: <sip:[email protected]:64444;transport=UDP>
To: <sip:[email protected];transport=UDP>
From: <sip:[email protected];transport=UDP>;tag=f23b2d52
Call-ID: ODg3M2IyNTFkZGM4OTI0YTMzYzI2ZDZlODU3NzM2ZDM.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/sdp
Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri
User-Agent: Z 3.6.25251 r25476
Allow-Events: presence, kpml
Content-Length: 241

v=0
o=Z 0 0 IN IP4 93.220.19.170
s=Z
c=IN IP4 93.220.19.170
t=0 0
m=audio 8000 RTP/AVP 3 110 8 0 98 101
a=rtpmap:110 speex/8000
a=rtpmap:98 iLBC/8000
a=fmtp:98 mode=20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->
--- (14 headers 12 lines) ---
Sending to 10.10.7.22:64444 (NAT)
Sending to 10.10.7.22:64444 (NAT)
Using INVITE request as basis request - ODg3M2IyNTFkZGM4OTI0YTMzYzI2ZDZlODU3NzM2ZDM.
Found peer '101' for '101' from 10.10.7.22:64444

<--- Reliably Transmitting (NAT) to 10.10.7.22:64444 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 93.220.19.170:64444;branch=z9hG4bK-d8754z-d1d628ec70933b7c-1---d8754z-;received=10.10.7.22;rport=64444
From: <sip:[email protected];transport=UDP>;tag=f23b2d52
To: <sip:[email protected];transport=UDP>;tag=as4a068b61
Call-ID: ODg3M2IyNTFkZGM4OTI0YTMzYzI2ZDZlODU3NzM2ZDM.
CSeq: 1 INVITE
Server: Asterisk PBX 13.7.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2c12e0a5"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'ODg3M2IyNTFkZGM4OTI0YTMzYzI2ZDZlODU3NzM2ZDM.' in 6400 ms (Method: INVITE)

<--- SIP read from UDP:10.10.7.22:64444 --->
ACK sip:[email protected];transport=UDP SIP/2.0
Via: SIP/2.0/UDP 93.220.19.170:64444;branch=z9hG4bK-d8754z-d1d628ec70933b7c-1---d8754z-
Max-Forwards: 70
To: <sip:[email protected];transport=UDP>;tag=as4a068b61
From: <sip:[email protected];transport=UDP>;tag=f23b2d52
Call-ID: ODg3M2IyNTFkZGM4OTI0YTMzYzI2ZDZlODU3NzM2ZDM.
CSeq: 1 ACK
Content-Length: 0

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

<--- SIP read from UDP:10.10.7.22:64444 --->
INVITE sip:[email protected];transport=UDP SIP/2.0
Via: SIP/2.0/UDP 93.220.19.170:64444;branch=z9hG4bK-d8754z-ae1228db2871bad4-1---d8754z-
Max-Forwards: 70
Contact: <sip:[email protected]:64444;transport=UDP>
To: <sip:[email protected];transport=UDP>
From: <sip:[email protected];transport=UDP>;tag=f23b2d52
Call-ID: ODg3M2IyNTFkZGM4OTI0YTMzYzI2ZDZlODU3NzM2ZDM.
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/sdp
Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri
User-Agent: Z 3.6.25251 r25476
Authorization: Digest username="101",realm="asterisk",nonce="2c12e0a5",uri="sip:[email protected];transport=UDP",response="c9e2acf433e15a63bd281e9124ba68cb",algorithm=MD5
Allow-Events: presence, kpml
Content-Length: 241

v=0
o=Z 0 0 IN IP4 93.220.19.170
s=Z
c=IN IP4 93.220.19.170
t=0 0
m=audio 8000 RTP/AVP 3 110 8 0 98 101
a=rtpmap:110 speex/8000
a=rtpmap:98 iLBC/8000
a=fmtp:98 mode=20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->
--- (15 headers 12 lines) ---
Sending to 10.10.7.22:64444 (NAT)
Using INVITE request as basis request - ODg3M2IyNTFkZGM4OTI0YTMzYzI2ZDZlODU3NzM2ZDM.
Found peer '101' for '101' from 10.10.7.22:64444
Found RTP audio format 3
Found RTP audio format 110
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 98
Found RTP audio format 101
Found audio description format speex for ID 110
Found audio description format iLBC for ID 98
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw|alaw|gsm), peer - audio=(ulaw|gsm|alaw|ilbc|speex)/video=(nothing)/text=(nothing), combined - (ulaw|alaw|gsm)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 93.220.19.170:8000
Looking for 01795555555 in outgoing (domain 10.10.7.15)
sip_route_dump: route/path hop: <sip:[email protected]:64444;transport=UDP>

<--- Transmitting (NAT) to 10.10.7.22:64444 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 93.220.19.170:64444;branch=z9hG4bK-d8754z-ae1228db2871bad4-1---d8754z-;received=10.10.7.22;rport=64444
From: <sip:[email protected];transport=UDP>;tag=f23b2d52
To: <sip:[email protected];transport=UDP>
Call-ID: ODg3M2IyNTFkZGM4OTI0YTMzYzI2ZDZlODU3NzM2ZDM.
CSeq: 2 INVITE
Server: Asterisk PBX 13.7.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:[email protected]:5060>
Content-Length: 0


<------------>
Audio is at 18390
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding codec gsm to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 10.10.7.1:5060:
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 10.10.7.15:5060;branch=z9hG4bK2188632c
Max-Forwards: 70
From: <sip:[email protected]>;tag=as6108bcfa
To: <sip:[email protected]>
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: Asterisk PBX 13.7.2
Date: Fri, 01 Apr 2016 09:39:11 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 290

v=0
o=root 68486405 68486405 IN IP4 10.10.7.15
s=Asterisk PBX 13.7.2
c=IN IP4 10.10.7.15
t=0 0
m=audio 18390 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv

---

<--- Transmitting (NAT) to 10.10.7.22:64444 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 93.220.19.170:64444;branch=z9hG4bK-d8754z-ae1228db2871bad4-1---d8754z-;received=10.10.7.22;rport=64444
From: <sip:[email protected];transport=UDP>;tag=f23b2d52
To: <sip:[email protected];transport=UDP>;tag=as7b515fa9
Call-ID: ODg3M2IyNTFkZGM4OTI0YTMzYzI2ZDZlODU3NzM2ZDM.
CSeq: 2 INVITE
Server: Asterisk PBX 13.7.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:[email protected]:5060>
Content-Length: 0


<------------>

<--- SIP read from UDP:10.10.7.1:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 10.10.7.15:5060;branch=z9hG4bK2188632c
From: <sip:[email protected]>;tag=as6108bcfa
To: <sip:[email protected]>;tag=8B5B796A96948761
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: FRITZ!OS
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Transmitting (no NAT) to 10.10.7.1:5060:
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 10.10.7.15:5060;branch=z9hG4bK2188632c
Max-Forwards: 70
From: <sip:[email protected]>;tag=as6108bcfa
To: <sip:[email protected]>;tag=8B5B796A96948761
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]
CSeq: 102 ACK
User-Agent: Asterisk PBX 13.7.2
Content-Length: 0


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

<--- Reliably Transmitting (NAT) to 10.10.7.22:64444 --->
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 93.220.19.170:64444;branch=z9hG4bK-d8754z-ae1228db2871bad4-1---d8754z-;received=10.10.7.22;rport=64444
From: <sip:[email protected];transport=UDP>;tag=f23b2d52
To: <sip:[email protected];transport=UDP>;tag=as7b515fa9
Call-ID: ODg3M2IyNTFkZGM4OTI0YTMzYzI2ZDZlODU3NzM2ZDM.
CSeq: 2 INVITE
Server: Asterisk PBX 13.7.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
X-Asterisk-HangupCause: Unallocated (unassigned) number
X-Asterisk-HangupCauseCode: 1
Content-Length: 0


<------------>

<--- SIP read from UDP:10.10.7.22:64444 --->
ACK sip:[email protected];transport=UDP SIP/2.0
Via: SIP/2.0/UDP 93.220.19.170:64444;branch=z9hG4bK-d8754z-ae1228db2871bad4-1---d8754z-
Max-Forwards: 70
To: <sip:[email protected];transport=UDP>;tag=as7b515fa9
From: <sip:[email protected];transport=UDP>;tag=f23b2d52
Call-ID: ODg3M2IyNTFkZGM4OTI0YTMzYzI2ZDZlODU3NzM2ZDM.
CSeq: 2 ACK
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog 'ODg3M2IyNTFkZGM4OTI0YTMzYzI2ZDZlODU3NzM2ZDM.' Method: ACK

<--- SIP read from UDP:10.10.7.22:64444 --->
PUBLISH sip:[email protected];transport=UDP SIP/2.0
Via: SIP/2.0/UDP 93.220.19.170:64444;branch=z9hG4bK-d8754z-80a5bfb7eb4187c2-1---d8754z-
Max-Forwards: 70
Contact: <sip:[email protected]:64444;transport=UDP>
To: <sip:[email protected];transport=UDP>
From: <sip:[email protected];transport=UDP>;tag=49020023
Call-ID: NzIwNjg1YjE3MzUxNWE4MTYxZWQ3YTAxYmRhNjI3MTE.
CSeq: 1 PUBLISH
Expires: 600
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/pidf+xml
Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri
User-Agent: Z 3.6.25251 r25476
Event: presence
Allow-Events: presence, kpml
Content-Length: 256

<?xml version="1.0" encoding="UTF-8"?>
<presence xmlns="urn:ietf:params:xml:ns:pidf" entity="sip:[email protected];transport=UDP"> <tuple id="101" > <status><basic>open</basic></status> <note>Online</note> </tuple>
</presence>
<------------->
--- (16 headers 3 lines) ---
Sending to 10.10.7.22:64444 (NAT)

<--- Transmitting (NAT) to 10.10.7.22:64444 --->
SIP/2.0 489 Bad Event
Via: SIP/2.0/UDP 93.220.19.170:64444;branch=z9hG4bK-d8754z-80a5bfb7eb4187c2-1---d8754z-;received=10.10.7.22;rport=64444
From: <sip:[email protected];transport=UDP>;tag=49020023
To: <sip:[email protected];transport=UDP>;tag=as17823c07
Call-ID: NzIwNjg1YjE3MzUxNWE4MTYxZWQ3YTAxYmRhNjI3MTE.
CSeq: 1 PUBLISH
Server: Asterisk PBX 13.7.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


<------------>
Really destroying SIP dialog 'NzIwNjg1YjE3MzUxNWE4MTYxZWQ3YTAxYmRhNjI3MTE.' Method: PUBLISH

<--- SIP read from UDP:10.10.7.22:64444 --->
SUBSCRIBE sip:[email protected];transport=UDP SIP/2.0
Via: SIP/2.0/UDP 93.220.19.170:64444;branch=z9hG4bK-d8754z-4bbd6edee9904508-1---d8754z-
Max-Forwards: 70
Contact: <sip:[email protected]:64444;transport=UDP>
To: <sip:[email protected];transport=UDP>
From: <sip:[email protected];transport=UDP>;tag=0261cc1e
Call-ID: YjRiY2M4OTVhMmIwNDdkYjY5NmIwYzhmYTQzYzAyZWU.
CSeq: 1 SUBSCRIBE
Expires: 600
Accept: application/watcherinfo+xml
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri
User-Agent: Z 3.6.25251 r25476
Event: presence.winfo
Allow-Events: presence, kpml
Content-Length: 0

<------------->
--- (16 headers 0 lines) ---
Sending to 10.10.7.22:64444 (NAT)
Creating new subscription
Sending to 10.10.7.22:64444 (NAT)
sip_route_dump: route/path hop: <sip:[email protected]:64444;transport=UDP>
Found peer '101' for '101' from 10.10.7.22:64444

<--- Transmitting (NAT) to 10.10.7.22:64444 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 93.220.19.170:64444;branch=z9hG4bK-d8754z-4bbd6edee9904508-1---d8754z-;received=10.10.7.22;rport=64444
From: <sip:[email protected];transport=UDP>;tag=0261cc1e
To: <sip:[email protected];transport=UDP>;tag=as2a528d6a
Call-ID: YjRiY2M4OTVhMmIwNDdkYjY5NmIwYzhmYTQzYzAyZWU.
CSeq: 1 SUBSCRIBE
Server: Asterisk PBX 13.7.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="11327fba"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'YjRiY2M4OTVhMmIwNDdkYjY5NmIwYzhmYTQzYzAyZWU.' in 6400 ms (Method: SUBSCRIBE)

<--- SIP read from UDP:10.10.7.22:64444 --->
SUBSCRIBE sip:[email protected];transport=UDP SIP/2.0
Via: SIP/2.0/UDP 93.220.19.170:64444;branch=z9hG4bK-d8754z-0ca9c6767479335e-1---d8754z-
Max-Forwards: 70
Contact: <sip:[email protected]:64444;transport=UDP>
To: <sip:[email protected];transport=UDP>
From: <sip:[email protected];transport=UDP>;tag=0261cc1e
Call-ID: YjRiY2M4OTVhMmIwNDdkYjY5NmIwYzhmYTQzYzAyZWU.
CSeq: 2 SUBSCRIBE
Expires: 600
Accept: application/watcherinfo+xml
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri
User-Agent: Z 3.6.25251 r25476
Authorization: Digest username="101",realm="asterisk",nonce="11327fba",uri="sip:[email protected];transport=UDP",response="512df5db5c7d62ddc02739abe52ad86b",algorithm=MD5
Event: presence.winfo
Allow-Events: presence, kpml
Content-Length: 0

<------------->
--- (17 headers 0 lines) ---
Creating new subscription
Sending to 10.10.7.22:64444 (NAT)
Found peer '101' for '101' from 10.10.7.22:64444

<--- Transmitting (NAT) to 10.10.7.22:64444 --->
SIP/2.0 489 Bad Event
Via: SIP/2.0/UDP 93.220.19.170:64444;branch=z9hG4bK-d8754z-0ca9c6767479335e-1---d8754z-;received=10.10.7.22;rport=64444
From: <sip:[email protected];transport=UDP>;tag=0261cc1e
To: <sip:[email protected];transport=UDP>;tag=as2a528d6a
Call-ID: YjRiY2M4OTVhMmIwNDdkYjY5NmIwYzhmYTQzYzAyZWU.
CSeq: 2 SUBSCRIBE
Server: Asterisk PBX 13.7.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


<------------>
Really destroying SIP dialog 'YjRiY2M4OTVhMmIwNDdkYjY5NmIwYzhmYTQzYzAyZWU.' Method: SUBSCRIBE
asterisk*CLI> sip set debug off
SIP Debugging Disabled
 
R

rentier-s

Guest
Von der Fritzbox kommt ein 404 Not Found, das heißt schlichtweg, die Nummer die Du anrufen möchtest gibt es nicht bzw. die Fritzbox weiß nichts damit anzufangen.
 

HrGesangsverein

Neuer User
Mitglied seit
23 Okt 2015
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Wenn ich aber mein SIP phone dierekt an der fritz box registriere (620:[email protected]) kann ich aber hinaus anrufen. Ich nehme an, dass Asterisk irgendwas anders macht als das SIP phone.
 
R

rentier-s

Guest
In From und Contact steht noch Dein Softphone, das wäre jetzt das einzige was mir auffällt. Hast Du fromuser=620 eingebaut?