Hilfe bei Einrichtung eines generischen SIP-Telefons

Support
1

Hallo zusammen,

ich hab nun dem Moby gesagt tue mal ein generic Phone einrichten. Daten eingeben und ab zum telefon (gigaset c610ip) auch hier die daten eingestellt.

Sprich: User / Passwort / IP

Nun sagt mir das Teil einfach nur Fehlgeschlagen. Hat jemand eine Idee?

2

Hi volvo1979,

telefonie auch angewendet ?

Grüße

Maik

3

Ja. Das habe ich. Im Telefon habe ich bei Registerserver / Proxyserver und Domain die server ip angegeben

4

Hi volvo1979,

dann schau mal auf der asterisk konsole. Dort sollte beim Anmeldeversuch eine Fehlermeldung kommen. Bitte dann den Output posten.

Grüße

Maik

5

<— SIP read from UDP:192.168.20.201:5060 —>
REGISTER sip:192.168.20.200 SIP/2.0
Via: SIP/2.0/UDP 192.168.20.201:5060;branch=z9hG4bK82361a166559df26957a71a2e780c50f;rport
From: “Daniel” <sip:daniel@192.168.20.200>;tag=1008141424
To: “Daniel” <sip:daniel@192.168.20.200>
Call-ID: 1523302202@192_168_20_201
CSeq: 193 REGISTER
Contact: <sip:daniel@192.168.20.201:5060>
Max-Forwards: 70
User-Agent: C610 IP/42.076.00.000.000
Expires: 180
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0

<------------->
— (12 headers 0 lines) —
Sending to 192.168.20.201:5060 (NAT)

<— Transmitting (NAT) to 192.168.20.201:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.20.201:5060;branch=z9hG4bK82361a166559df26957a71a2e780c50f;received=192.168.20.201;rport=5060
From: “Daniel” <sip:daniel@192.168.20.200>;tag=1008141424
To: “Daniel” <sip:daniel@192.168.20.200>;tag=as47e8fc5b
Call-ID: 1523302202@192_168_20_201
CSeq: 193 REGISTER
Server: Asterisk PBX 1.8.11.1-1digium1~squeeze
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce=“7fb31ddf”
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘1523302202@192_168_20_201’ in 32000 ms (Method: REGISTER)

<— SIP read from UDP:192.168.20.201:5060 —>
REGISTER sip:192.168.20.200 SIP/2.0
Via: SIP/2.0/UDP 192.168.20.201:5060;branch=z9hG4bKf85149838591af659e4891892bb191c;rport
From: “Daniel” <sip:daniel@192.168.20.200>;tag=1008141424
To: “Daniel” <sip:daniel@192.168.20.200>
Call-ID: 1523302202@192_168_20_201
CSeq: 194 REGISTER
Contact: <sip:daniel@192.168.20.201:5060>
Authorization: Digest username=“daniel”, realm=“asterisk”, algorithm=MD5, uri=“sip:192.168.20.200”, nonce=“7fb31ddf”, response=“9aacd8d10fdb2e0651f49bd7ca4a4863”
Max-Forwards: 70
User-Agent: C610 IP/42.076.00.000.000
Expires: 180
Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, INFO, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0

<------------->
— (13 headers 0 lines) —
Sending to 192.168.20.201:5060 (NAT)
– Registered SIP ‘daniel’ at 192.168.20.201:5060
Reliably Transmitting (NAT) to 192.168.20.201:5060:
OPTIONS sip:daniel@192.168.20.201:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.20.200:5060;branch=z9hG4bK2ea87ed2;rport
Max-Forwards: 70
From: “asterisk” <sip:asterisk@192.168.20.200>;tag=as3eb62cda
To: <sip:daniel@192.168.20.201:5060>
Contact: <sip:asterisk@192.168.20.200:5060>
Call-ID: 31673c88404b97f31fa233a25c930b17@192.168.20.200:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.11.1-1digium1~squeeze
Date: Fri, 18 Oct 2013 09:44:12 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<— Transmitting (NAT) to 192.168.20.201:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.20.201:5060;branch=z9hG4bKf85149838591af659e4891892bb191c;received=192.168.20.201;rport=5060
From: “Daniel” <sip:daniel@192.168.20.200>;tag=1008141424
To: “Daniel” <sip:daniel@192.168.20.200>;tag=as47e8fc5b
Call-ID: 1523302202@192_168_20_201
CSeq: 194 REGISTER
Server: Asterisk PBX 1.8.11.1-1digium1~squeeze
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Expires: 180
Contact: <sip:daniel@192.168.20.201:5060>;expires=180
Date: Fri, 18 Oct 2013 09:44:12 GMT
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘1523302202@192_168_20_201’ in 32000 ms (Method: REGISTER)

<— SIP read from UDP:192.168.20.201:5060 —>
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 192.168.20.200:5060;branch=z9hG4bK2ea87ed2;rport=5060
From: “asterisk” <sip:asterisk@192.168.20.200>;tag=as3eb62cda
To: <sip:daniel@192.168.20.201:5060>
Call-ID: 31673c88404b97f31fa233a25c930b17@192.168.20.200:5060
CSeq: 102 OPTIONS
User-Agent: C610 IP/42.076.00.000.000
Content-Length: 0

<------------->
— (8 headers 0 lines) —
[Oct 18 11:44:12] NOTICE[1379]: chan_sip.c:20792 handle_response_peerpoke: Peer ‘daniel’ is now Reachable. (110ms / 2000ms)
Really destroying SIP dialog ‘31673c88404b97f31fa233a25c930b17@192.168.20.200:5060’ Method: OPTIONS

6

Lach. Nn sagt es auf einmal doch, dass es angemeldet sei. Kann sich das teil mal entscheiden?

7

Hi volvo1979,

Du bist doch angemeldet "Peer ‘daniel’ is now Reachable. (110ms / 2000ms)"

Grüße Maik