[Problem] Sipgate auf FBF Asterisk registriert aber keine Gespräche möglich.

baeckerman83

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Hiho!
Irgendwie komme ich mir etwas doof vor, weil ich hier laufend nachfragen muss. :( Aber irgendwie will das alles nicht so wie ich mir das vorstelle.
Jetzt geht mein Sipgate Account nicht. Auf der Sipgate Seite steht er sei online und unter 1. Gerät: Asterisk PBX 1.6.2.20 IP: sip:[email protected]:64703
also da sieht es gut aus, denn das ist auch meine IP. Diese ist dynamisch. Wenn ich aber auf meiner Sipgate Nummer anrufe, kommt "Der Teilnehmer ist nicht erreichbar". Wenn ich versuche abgehend zu telefonieren höre ich nichts.
Ich benutze zum abgehend telefonieren das Softphone Xlite4. Wenn ich intern anrufe funktioniert es und ich höre was. Dazu habe ich mir die 1002 als Testrufnummer gebaut, diese spielt einfach eine Ansage vor. Das geht. Nur Sipgate will nicht.
Jetzt habe ich was von Portweiterleitungen gelesen. Damit habe ich jetzt ein Problem. Man soll den Port 5060 auf die Asterisk Weiterleiten, mein SpeedportW920V sagt mir aber, der sei Intern belegt und kann nicht weiter geleitet werden. :( Diesen Speedport benötige ich wegen meinem VDSL Anschluss. Hieran habe ich dann die 7270v3 mit dem Asterisk angeschlossen. (Hier Lan1 benutzt).
Die IPs sind:
W920V: 192.168.2.1
7270v3: 192.168.2.111
dyndns ist vorhanden.

Hier dann mal meine sip.conf:
Code:
[general]
bindport=5060
bindaddr=0.0.0.0
localnet=192.168.2.0/255.255.255.0
externhost=dyndns.homeip.net
externrefresh = 10
srvlookup=yes
language=de
context = sonstige
register => 2929291:[email protected]/2929291

[30]
callerid=30
host=dynamic
domain=192.168.2.111
user=30
secret=geheim
type=friend
mailbox=30
nat=yes
canreinvite=yes

[2929291]
type=friend
username=2929291
fromuser=2929291
secret=PASSWORT
host=sipgate.de
fromdomain=sipgate.de
insecure=invite
canreinvite=no
nat=yes
disallow=all
allow=ulaw
context=sipgate_out

[sipgate_de_in]
type=friend
insecure=invite
nat=yes
fromdomain=sipgate.de
host=sipgate.de
disallow=all
allow=ulaw
context=ankommend
Und hier meine Extensions.conf
Code:
[general]
static=yes
writeprotect=no

; --------------------------------------------------------------------
; Es hat sich als gute Praxis erwiesen, die Inhalte der Datei
; extensions.conf modular aufzubauen. Diese Praxis wollen
; wir auch hier anwenden
;

[lokal]
; Erreichbarkeit der Nebenstellen 30-39
; untereinander herstellen

exten => 1001,1,Answer()
exten => 1001,2,Playback(/var/media/ftp/uStor01/asterisk/sound/de/hello-world)
exten => 1001,3,Hangup()

exten => 1002,1,Answer()
exten => 1002,2,Playback(hello-world)
exten => 1002,3,Hangup()

[sipgate_out]
exten => _0.,1,Dial(SIP/${EXTEN}@2929291/45/r)

[ankommend]
exten => 2929291,1,Dial(SIP/30)
; --------------------------------------------------------------------
;
; hier kommt der default-Context, in dem alle Geraete in der
; Grundkonfiguration erstmal laufen.
; Alle Geraete koennen sich gegenseitig anrufen

[sipgate_verbinden]
exten => 1,1,NoOP(Anwaehlen)
exten => 1,n,Dial(SIP/[email protected])

[sipgate]
exten => 2929291,1,Answer()
exten => 2929291,n,Playback(hello-world)


[sonstige]
include => lokal
include => sipgate_out
include => ankommend
Ankommendes Sip Protokoll
Code:
SIP Debugging enabled

<--- SIP read from UDP:217.10.79.9:5060 --->

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

<--- SIP read from UDP:217.10.79.9:5060 --->
INVITE sip:[email protected]93.196.11.11 SIP/2.0
Record-Route: <sip:217.10.79.9;lr;ftag=as699eeb62>
Record-Route: <sip:172.20.40.3;lr=on>
Record-Route: <sip:217.10.79.9;lr;ftag=as699eeb62>
Via: SIP/2.0/UDP 217.10.79.9:5060;branch=z9hG4bKf2b3.4809c5f4.0
Via: SIP/2.0/UDP 172.20.40.3;branch=z9hG4bKf2b3.4809c5f4.0
Via: SIP/2.0/UDP 217.10.79.9:5060;received=217.10.68.222;branch=z9hG4bK37cf87bb
Via: SIP/2.0/UDP 217.116.117.70:5060;received=217.116.117.70;branch=z9hG4bK37cf87bb;rport=5060
Max-Forwards: 67
From: "01716411111" <sip:[email protected]>;tag=as699eeb62
To: <sip:[email protected]>
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 464

v=0
o=root 67260353 67260353 IN IP4 217.116.117.70
s=sipgate VoIP GW
c=IN IP4 217.10.77.41
t=0 0
m=audio 45172 RTP/AVP 8 0 3 97 18 112 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:112 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
a=direction:active
a=nortpproxy:yes

<------------->
--- (18 headers 21 lines) ---
  == Using SIP RTP CoS mark 5
Sending to 217.10.79.9 : 5060 (no NAT)
Using INVITE request as basis request - [email protected]
Found peer '2929291' for '01716411111' from 217.10.79.9:5060
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 3
Found RTP audio format 97
Found RTP audio format 18
Found RTP audio format 112
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format PCMU for ID 0
Found audio description format GSM for ID 3
Found audio description format iLBC for ID 97
Found audio description format G729 for ID 18
Found audio description format G726-32 for ID 112
Found audio description format telephone-event for ID 101
Capabilities: us - 0x4 (ulaw), peer - audio=0xd0e (gsm|ulaw|alaw|g726|g729|ilbc)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 217.10.77.41:45172
Looking for 2929291 in sipgate_out (domain 93.196.11.11)

<--- Reliably Transmitting (NAT) to 217.10.79.9:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 217.10.79.9:5060;branch=z9hG4bKf2b3.4809c5f4.0;received=217.10.79.9
Via: SIP/2.0/UDP 172.20.40.3;branch=z9hG4bKf2b3.4809c5f4.0
Via: SIP/2.0/UDP 217.10.79.9:5060;received=217.10.68.222;branch=z9hG4bK37cf87bb
Via: SIP/2.0/UDP 217.116.117.70:5060;received=217.116.117.70;branch=z9hG4bK37cf87bb;rport=5060
From: "01716411111" <sip:[email protected]>;tag=as699eeb62
To: <sip:[email protected]>;tag=as411ac7a0
Call-ID: [email protected]
CSeq: 102 INVITE
Server: Asterisk PBX 1.6.2.20
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '[email protected]' in 32000 ms (Method: INVITE)

<--- SIP read from UDP:217.10.79.9:5060 --->
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 217.10.79.9:5060;branch=z9hG4bKf2b3.4809c5f4.0
Via: SIP/2.0/UDP 172.20.40.3;branch=z9hG4bKf2b3.4809c5f4.0
From: "01716411111" <sip:[email protected]>;tag=as699eeb62
Call-ID: [email protected]
To: <sip:[email protected]>;tag=as411ac7a0
CSeq: 102 ACK
Max-Forwards: 69
Content-Length: 0
X-hint: rr-enforced


<------------->
--- (10 headers 0 lines) ---
Really destroying SIP dialog '[email protected]' Method: ACK

<--- SIP read from UDP:217.10.79.9:5060 --->

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

<--- SIP read from UDP:192.168.2.210:21844 --->



<------------->
Really destroying SIP dialog '[email protected]' Method: REGISTER
Und hier der Code für abgehend
Code:
<--- SIP read from UDP:217.10.79.9:5060 --->

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

<--- SIP read from UDP:192.168.2.210:21844 --->
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.2.210:21844;branch=z9hG4bK-d8754z-2304e50c4ae808c9-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:[email protected]:21844>
To: <sip:[email protected]>
From: "30"<sip:[email protected]>;tag=8aeb419b
Call-ID: YWNmZGQ4MDVkZDNjY2VlOTFjMTgxNTBjOGIwY2Q2MTQ.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
Supported: replaces
User-Agent: X-Lite 4 release 4.1 stamp 63214
Content-Length: 232

v=0
o=- 12972604281338287 1 IN IP4 192.168.2.210
s=CounterPath X-Lite 4.1
c=IN IP4 192.168.2.210
t=0 0
m=audio 58904 RTP/AVP 107 0 8 101
a=rtpmap:107 BV32/16000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv

<------------->
--- (13 headers 10 lines) ---
  == Using SIP RTP CoS mark 5
Sending to 192.168.2.210 : 21844 (no NAT)
Using INVITE request as basis request - YWNmZGQ4MDVkZDNjY2VlOTFjMTgxNTBjOGIwY2Q2MTQ.
Found peer '30' for '30' from 192.168.2.210:21844

<--- Reliably Transmitting (NAT) to 192.168.2.210:21844 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.2.210:21844;branch=z9hG4bK-d8754z-2304e50c4ae808c9-1---d8754z-;received=192.168.2.210;rport=21844
From: "30"<sip:[email protected]>;tag=8aeb419b
To: <sip:[email protected]>;tag=as6d01ba86
Call-ID: YWNmZGQ4MDVkZDNjY2VlOTFjMTgxNTBjOGIwY2Q2MTQ.
CSeq: 1 INVITE
Server: Asterisk PBX 1.6.2.20
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="7ab6e1c4"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'YWNmZGQ4MDVkZDNjY2VlOTFjMTgxNTBjOGIwY2Q2MTQ.' in 32000 ms (Method: INVITE)

<--- SIP read from UDP:192.168.2.210:21844 --->
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.2.210:21844;branch=z9hG4bK-d8754z-2304e50c4ae808c9-1---d8754z-;rport
Max-Forwards: 70
To: <sip:[email protected]>;tag=as6d01ba86
From: "30"<sip:[email protected]>;tag=8aeb419b
Call-ID: YWNmZGQ4MDVkZDNjY2VlOTFjMTgxNTBjOGIwY2Q2MTQ.
CSeq: 1 ACK
Content-Length: 0


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

<--- SIP read from UDP:192.168.2.210:21844 --->
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.2.210:21844;branch=z9hG4bK-d8754z-2304414062c1d7f2-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:[email protected]:21844>
To: <sip:[email protected]>
From: "30"<sip:[email protected]>;tag=8aeb419b
Call-ID: YWNmZGQ4MDVkZDNjY2VlOTFjMTgxNTBjOGIwY2Q2MTQ.
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
Supported: replaces
User-Agent: X-Lite 4 release 4.1 stamp 63214
Authorization: Digest username="30",realm="asterisk",nonce="7ab6e1c4",uri="sip:[email protected]",response="4fdd9143fce59bc13f02b4e9aec37f63",algorithm=MD5
Content-Length: 232

v=0
o=- 12972604281338287 1 IN IP4 192.168.2.210
s=CounterPath X-Lite 4.1
c=IN IP4 192.168.2.210
t=0 0
m=audio 58904 RTP/AVP 107 0 8 101
a=rtpmap:107 BV32/16000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv

<------------->
--- (14 headers 10 lines) ---
Sending to 192.168.2.210 : 21844 (NAT)
Using INVITE request as basis request - YWNmZGQ4MDVkZDNjY2VlOTFjMTgxNTBjOGIwY2Q2MTQ.
Found peer '30' for '30' from 192.168.2.210:21844
Found RTP audio format 107
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Found audio description format BV32 for ID 107
Found audio description format telephone-event for ID 101
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 192.168.2.210:58904
Looking for 051111111111 in sonstige (domain 192.168.2.111)
list_route: hop: <sip:[email protected]:21844>

<--- Transmitting (NAT) to 192.168.2.210:21844 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.2.210:21844;branch=z9hG4bK-d8754z-2304414062c1d7f2-1---d8754z-;received=192.168.2.210;rport=21844
From: "30"<sip:[email protected]>;tag=8aeb419b
To: <sip:[email protected]>
Call-ID: YWNmZGQ4MDVkZDNjY2VlOTFjMTgxNTBjOGIwY2Q2MTQ.
CSeq: 2 INVITE
Server: Asterisk PBX 1.6.2.20
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: <sip:[email protected]>
Content-Length: 0


<------------>
    -- Executing [[email protected]:1] Dial("SIP/30-0000000b", "SIP/[email protected]/45/r") in new stack
  == Using SIP RTP CoS mark 5
Audio is at 93.196.11.11 port 7140
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 62.157.140.133:5060:
INVITE sip:[email protected]/45/r SIP/2.0
Via: SIP/2.0/UDP 93.196.11.11:5060;branch=z9hG4bK48177cd1;rport
Max-Forwards: 70
From: "30" <sip:[email protected]>;tag=as4f75ce03
To: <sip:[email protected]/45/r>
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.2.20
Date: Wed, 01 Feb 2012 21:09:57 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 310

v=0
o=root 1492412816 1492412816 IN IP4 93.196.11.11
s=Asterisk PBX 1.6.2.20
c=IN IP4 93.196.11.11
t=0 0
m=audio 7140 RTP/AVP 0 3 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
    -- Called [email protected]/45/r
Retransmitting #1 (no NAT) to 62.157.140.133:5060:
INVITE sip:[email protected]/45/r SIP/2.0
Via: SIP/2.0/UDP 93.196.11.11:5060;branch=z9hG4bK48177cd1;rport
Max-Forwards: 70
From: "30" <sip:[email protected]>;tag=as4f75ce03
To: <sip:[email protected]/45/r>
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.2.20
Date: Wed, 01 Feb 2012 21:09:57 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 310

v=0
o=root 1492412816 1492412816 IN IP4 93.196.11.11
s=Asterisk PBX 1.6.2.20
c=IN IP4 93.196.11.11
t=0 0
m=audio 7140 RTP/AVP 0 3 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---

<--- SIP read from UDP:192.168.2.210:21844 --->
CANCEL sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.2.210:21844;branch=z9hG4bK-d8754z-2304414062c1d7f2-1---d8754z-;rport
Max-Forwards: 70
To: <sip:[email protected]>
From: "30"<sip:[email protected]>;tag=8aeb419b
Call-ID: YWNmZGQ4MDVkZDNjY2VlOTFjMTgxNTBjOGIwY2Q2MTQ.
CSeq: 2 CANCEL
User-Agent: X-Lite 4 release 4.1 stamp 63214
Authorization: Digest username="30",realm="asterisk",nonce="7ab6e1c4",uri="sip:[email protected]",response="3628207e55eb6fff781262c96f2ab3c4",algorithm=MD5
Content-Length: 0


<------------->
--- (10 headers 0 lines) ---
Sending to 192.168.2.210 : 21844 (NAT)

<--- Reliably Transmitting (NAT) to 192.168.2.210:21844 --->
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 192.168.2.210:21844;branch=z9hG4bK-d8754z-2304414062c1d7f2-1---d8754z-;received=192.168.2.210;rport=21844
From: "30"<sip:[email protected]>;tag=8aeb419b
To: <sip:[email protected]>;tag=as3f0242c1
Call-ID: YWNmZGQ4MDVkZDNjY2VlOTFjMTgxNTBjOGIwY2Q2MTQ.
CSeq: 2 INVITE
Server: Asterisk PBX 1.6.2.20
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


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

<--- Transmitting (NAT) to 192.168.2.210:21844 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.210:21844;branch=z9hG4bK-d8754z-2304414062c1d7f2-1---d8754z-;received=192.168.2.210;rport=21844
From: "30"<sip:[email protected]>;tag=8aeb419b
To: <sip:[email protected]>;tag=as3f0242c1
Call-ID: YWNmZGQ4MDVkZDNjY2VlOTFjMTgxNTBjOGIwY2Q2MTQ.
CSeq: 2 CANCEL
Server: Asterisk PBX 1.6.2.20
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '[email protected]' in 32000 ms (Method: INVITE)
  == Spawn extension (sonstige, 051111111111, 1) exited non-zero on 'SIP/30-0000000b'

<--- SIP read from UDP:192.168.2.210:21844 --->
ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.2.210:21844;branch=z9hG4bK-d8754z-2304414062c1d7f2-1---d8754z-;rport
Max-Forwards: 70
To: <sip:[email protected]>;tag=as3f0242c1
From: "30"<sip:[email protected]>;tag=8aeb419b
Call-ID: YWNmZGQ4MDVkZDNjY2VlOTFjMTgxNTBjOGIwY2Q2MTQ.
CSeq: 2 ACK
Content-Length: 0


<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog 'YWNmZGQ4MDVkZDNjY2VlOTFjMTgxNTBjOGIwY2Q2MTQ.' Method: ACK
Retransmitting #2 (no NAT) to 62.157.140.133:5060:
INVITE sip:[email protected]/45/r SIP/2.0
Via: SIP/2.0/UDP 93.196.11.11:5060;branch=z9hG4bK48177cd1;rport
Max-Forwards: 70
From: "30" <sip:[email protected]>;tag=as4f75ce03
To: <sip:[email protected]/45/r>
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.6.2.20
Date: Wed, 01 Feb 2012 21:09:57 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 310

v=0
o=root 1492412816 1492412816 IN IP4 93.196.11.11
s=Asterisk PBX 1.6.2.20
c=IN IP4 93.196.11.11
t=0 0
m=audio 7140 RTP/AVP 0 3 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
Braucht ihr noch mehr Infos? Ich hoffe ihr könnt mir helfen. :)
 
Zuletzt bearbeitet:
R

rentier-s

Guest
Ich versuch mich noch mal dran, vielleicht wird aus dem Bier dann doch ein Colada ;-)

In der sip.conf muss im [2929291] das insecure und context raus.

Damit Du ein Portforwarding einrichten kannst, musst Du den bindport auf zB. 5061 stellen. In XLite dann dementsprechend 192.168.2.111:5061 als Server angeben. (Hatten wir schon ein paar Mal im Bereich [email protected])
Außerdem brauchst Du ein Portforwarding für die RTP Portrange, die findest Du in der rtp.conf (alles UDP).
 

baeckerman83

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So das hat auch geklappt. :) Ich musste die Ports ja auf der 7270v3 auch noch Intern weiterleiten. Und auch die 10.000 - 20.000 UDP Ports. Die standen zwar net in der rtp.conf, aber danach geht es jetzt.
*colada rüberschieb* :)
Jetzt geht es erst mal nicht weiter, habe das alles per VPN gemacht. Jetzt muss ich nen UMTS Stick anschließen und ISDN, dann versuche ich das capi und telefonieren per UMTS Gateway hinzubekommen. Solange geb ich Ruhe. :) Danke für deine Hilfe schon mal!