1und1 Probleme

Support
1

Hallo zusammen,

Versuche gerade verzweifelt meine FritzBox arbeitslos zu machen, scheitere aber wie folgt:

Konfiguration:
1 Amt:

Bezeichnung: 1und1.de
SIP Daten:
Type: Friend
User: 499913617xxx
Host: sip.1und1.de
Optionen:
fromuser=4999136174xx
fromdomain=sip.1und1.de

Ausgehende sowie Eingehende Regeln alle mit * und erlauben über das Amt.

Problematik:

Ausgehende Anrufe funktionieren, Eingehende nicht
“Die von Ihnen gewählte Rufnummer ist derzeit nicht erreichbar”

Sip show registry

Host Username Refresh State
sip.1und1.de:5060 4999136174xx 14385 Registered

Sip Show peers
Name/username Host Dyn Nat ACL Port Status
123/123 192.168.1.210 D N 5060 OK (22 ms)
4999136174xx/4999136174xx 212.227.67.134 N 5060 OK (86 ms)

Sonst ist leider überhaupt nichts im Log bei einem Anruf.
Firewalls etc. sind offen wie ein Scheunentor.
ExternIP sowie Localnet sind gesetzt (habe eine statische IP)

Hoffe Ihr könnt mir noch weiterhelfen.

Danke

[EDIT]
Hab’ gerade noch 'nen alten Sipgate Account ausgegraben, der funktioniert dank eurer Schnellanlage perfekt, also würd ich Firewalls etc. mal ausschließen…
[EDIT2]
Nach ein paar Versuchen bekomme ich jetzt folgenden Log:


Connected to Asterisk 1.2.30.4-BRIstuffed-0.3.0-PRE-1y-w currently running on mobydick (pid = 3925)
Verbosity is at least 3
    -- Remote UNIX connection
mobydick*CLI> sip show peer
Usage: sip show peer <name> [load]
       Lists all details on one SIP peer and the current status.
       Option "load" forces lookup of peer in realtime storage.
mobydick*CLI> sip show peers
Name/username              Host            Dyn Nat ACL Port     Status
123/123                    192.168.1.210    D   N      5060     OK (21 ms)
7936343/7936343            217.10.79.9          N      5060     OK (42 ms)
4999136174xx/4999136174xx  212.227.67.201       N      5060     OK (71 ms)
3 sip peers [3 online , 0 offline]
mobydick*CLI> sip debug peer 4999136174xx
SIP Debugging Enabled for IP: 212.227.67.201:5060
mobydick*CLI>
mobydick*CLI>
mobydick*CLI>
mobydick*CLI>
<-- SIP read from 212.227.67.201:5060:
INVITE sip:s@217.64.68.82 SIP/2.0
Record-Route: <sip:212.227.67.201;lr=on;ftag=914904559>
Record-Route: <sip:212.227.67.165;lr=on;ftag=914904559;did=26f.54d83e45>
Record-Route: <sip:212.227.67.138;lr=on;ftag=914904559>
Via: SIP/2.0/UDP 212.227.67.201;branch=z9hG4bKa9c4.3bb7312d97f058f2bf436327b55f0e2d.0
Via: SIP/2.0/UDP 212.227.67.165;branch=z9hG4bKa9c4.3bb7312d97f058f2bf436327b55f0e2d.0
Via: SIP/2.0/UDP 212.227.67.138;branch=z9hG4bKa9c4.5a2573c6.0
Via: SIP/2.0/UDP 217.188.44.83;branch=z9hG4bKa9c4.5a2573c6.0
Via: SIP/2.0/UDP 62.53.206.6:5060;branch=z9hG4bK4vh6ou306g3gslgqp471.1
From: +4917024279xx <sip:+4917024279xx@1und1-3.sip.mgc.voip.telefonica.de:5060;user=phone>;tag=914904559
To: +4999136174xx <sip:+4999136174xx@62.53.206.22:5060;user=phone>
Call-ID: 588022e3-36d8eaa8-4cb824f8-8f2f@1und1-3.sip.mgc.voip.telefonica.de
CSeq: 1 INVITE
Max-Forwards: 23
Supported: timer
Session-Expires: 1800
Min-SE: 1800
Contact: +4917024279xx <sip:+4917024279xx@62.53.206.6:5060;transport=udp>
Allow: INVITE,ACK,PRACK,SUBSCRIBE,BYE,CANCEL,NOTIFY,INFO,REFER,UPDATE
P-Asserted-Identity: +4917024279xx <sip:+4917024279xx@1und1-3.sip.mgc.voip.telefonica.de;user=phone>
Content-Type: application/sdp
Content-Length: 678

v=0
o=- 22873570 0 IN IP4 62.52.144.28
s=Cisco SDP 0
c=IN IP4 62.52.144.28
t=0 0
m=audio 32966 RTP/AVP 8 0 18 101 102 103 104 105 4 106 3 107 108 109 125 99 100
a=rtpmap:101 G729a/8000
a=rtpmap:102 G726-16/8000
a=rtpmap:103 G726-24/8000
a=rtpmap:104 G726-32/8000
a=rtpmap:105 G7231-H/8000
a=rtpmap:106 G7231-L/8000
a=rtpmap:107 G729b/8000
a=rtpmap:108 G7231a-H/8000
a=rtpmap:109 G7231a-L/8000
a=rtpmap:125 GnX64/8000
a=rtpmap:99 telephone-event/8000
a=fmtp:99 0-15
a=rtpmap:100 X-NSE/8000
a=fmtp:100 192-194,200-202
a=X-sqn:0
a=X-cap: 1 audio RTP/AVP 100
a=X-cpar: a=rtpmap:100 X-NSE/8000
a=X-cpar: a=fmtp:100 192-194,200-202
a=X-cap: 2 image udptl t38

--- (22 headers 25 lines) ---
Ignoring this INVITE request
Transmitting (NAT) to 212.227.67.201:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 212.227.67.201;branch=z9hG4bKa9c4.3bb7312d97f058f2bf436327b55f0e2d.0;received=212.227.67.201
Via: SIP/2.0/UDP 212.227.67.165;branch=z9hG4bKa9c4.3bb7312d97f058f2bf436327b55f0e2d.0
Via: SIP/2.0/UDP 212.227.67.138;branch=z9hG4bKa9c4.5a2573c6.0
Via: SIP/2.0/UDP 217.188.44.83;branch=z9hG4bKa9c4.5a2573c6.0
Via: SIP/2.0/UDP 62.53.206.6:5060;branch=z9hG4bK4vh6ou306g3gslgqp471.1
From: +4917024279xx <sip:+4917024279xx@1und1-3.sip.mgc.voip.telefonica.de:5060;user=phone>;tag=914904559
To: +4999136174xx <sip:+4999136174xx@62.53.206.22:5060;user=phone>
Call-ID: 588022e3-36d8eaa8-4cb824f8-8f2f@1und1-3.sip.mgc.voip.telefonica.de
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:4999136174xx@217.64.68.82>
Content-Length: 0


---
    -- Executing Macro("SIP/5060-081d75a0", "mdc_emergency|4999136174xx") in new stack
    -- Executing NoOp("SIP/5060-081d75a0", "macro-mdc_emergency:: Exten: 4999136174xx") in new stack
    -- Executing Goto("SIP/5060-081d75a0", "4999136174xx|1") in new stack
    -- Goto (macro-mdc_emergency,4999136174xx,1)
    -- Executing NoOp("SIP/5060-081d75a0", "4999136174xx ist keine Notrufnummer") in new stack
    -- Executing Hangup("SIP/5060-081d75a0", "0") in new stack
  == Spawn extension (no-auth-in, 4999136174xx, 2) exited non-zero on 'SIP/5060-081d75a0'
Scheduling destruction of call '588022e3-36d8eaa8-4cb824f8-8f2f@1und1-3.sip.mgc.voip.telefonica.de' in 32000 ms
Reliably Transmitting (NAT) to 212.227.67.201:5060:
SIP/2.0 480 Temporarily unavailable
Via: SIP/2.0/UDP 212.227.67.134;branch=z9hG4bKa9c4.3bb7312d97f058f2bf436327b55f0e2d.1;received=212.227.67.201
Via: SIP/2.0/UDP 212.227.67.165;branch=z9hG4bKa9c4.3bb7312d97f058f2bf436327b55f0e2d.1
Via: SIP/2.0/UDP 212.227.67.138;branch=z9hG4bKa9c4.5a2573c6.0
Via: SIP/2.0/UDP 217.188.44.83;branch=z9hG4bKa9c4.5a2573c6.0
Via: SIP/2.0/UDP 62.53.206.6:5060;branch=z9hG4bK4vh6ou306g3gslgqp471.1
From: +4917024279xx <sip:+4917024279xx@1und1-3.sip.mgc.voip.telefonica.de:5060;user=phone>;tag=914904559
To: +4999136174xx <sip:+4999136174xx@62.53.206.22:5060;user=phone>;tag=as785fc21c
Call-ID: 588022e3-36d8eaa8-4cb824f8-8f2f@1und1-3.sip.mgc.voip.telefonica.de
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0


---
12 headers, 0 lines
Reliably Transmitting (NAT) to 212.227.67.201:5060:
OPTIONS sip:sip.1und1.de SIP/2.0
Via: SIP/2.0/UDP 217.64.68.82:5060;branch=z9hG4bK3c3efe9e;rport
From: "asterisk" <sip:asterisk@217.64.68.82>;tag=as0d4e0d23
To: <sip:sip.1und1.de>
Contact: <sip:asterisk@217.64.68.82>
Call-ID: 79a1711840d45353471555c150192536@217.64.68.82
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sun, 10 Oct 2010 17:39:38 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0


---
mobydick*CLI>
<-- SIP read from 212.227.67.201:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 217.64.68.82:5060;branch=z9hG4bK3c3efe9e;rport=5060
From: "asterisk" <sip:asterisk@217.64.68.82>;tag=as0d4e0d23
To: <sip:sip.1und1.de>;tag=f8f2ab2c1295e90ed7dbb499b30f44b2.d25d
Call-ID: 79a1711840d45353471555c150192536@217.64.68.82
CSeq: 102 OPTIONS
Server: UI Kamailio
Content-Length: 0


--- (8 headers 0 lines) ---
Destroying call '79a1711840d45353471555c150192536@217.64.68.82'
mobydick*CLI>
<-- SIP read from 212.227.67.201:5060:
CANCEL sip:s@217.64.68.82 SIP/2.0
Record-Route: <sip:212.227.67.201;lr=on;ftag=914904559>
Via: SIP/2.0/UDP 212.227.67.201;branch=z9hG4bKa9c4.3bb7312d97f058f2bf436327b55f0e2d.0
Via: SIP/2.0/UDP 212.227.67.165;branch=z9hG4bKa9c4.3bb7312d97f058f2bf436327b55f0e2d.0
From: +4917024279xx <sip:+4917024279xx@1und1-3.sip.mgc.voip.telefonica.de:5060;user=phone>;tag=914904559
Call-ID: 588022e3-36d8eaa8-4cb824f8-8f2f@1und1-3.sip.mgc.voip.telefonica.de
To: +4999136174xx <sip:+4999136174xx@62.53.206.22:5060;user=phone>
CSeq: 1 CANCEL
Max-Forwards: 69
User-Agent: Kamailio (1.5.4-tls (x86_64/linux))
Content-Length: 0


--- (11 headers 0 lines) ---
Sending to 212.227.67.201 : 5060 (NAT)
Reliably Transmitting (NAT) to 212.227.67.201:5060:
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 212.227.67.134;branch=z9hG4bKa9c4.3bb7312d97f058f2bf436327b55f0e2d.1;received=212.227.67.201
Via: SIP/2.0/UDP 212.227.67.165;branch=z9hG4bKa9c4.3bb7312d97f058f2bf436327b55f0e2d.1
Via: SIP/2.0/UDP 212.227.67.138;branch=z9hG4bKa9c4.5a2573c6.0
Via: SIP/2.0/UDP 217.188.44.83;branch=z9hG4bKa9c4.5a2573c6.0
Via: SIP/2.0/UDP 62.53.206.6:5060;branch=z9hG4bK4vh6ou306g3gslgqp471.1
From: +4917024279xx <sip:+4917024279xx@1und1-3.sip.mgc.voip.telefonica.de:5060;user=phone>;tag=914904559
To: +4999136174xx <sip:+4999136174xx@62.53.206.22:5060;user=phone>;tag=as785fc21c
Call-ID: 588022e3-36d8eaa8-4cb824f8-8f2f@1und1-3.sip.mgc.voip.telefonica.de
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0


---
Transmitting (NAT) to 212.227.67.201:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 212.227.67.201;branch=z9hG4bKa9c4.3bb7312d97f058f2bf436327b55f0e2d.0;received=212.227.67.201
Via: SIP/2.0/UDP 212.227.67.165;branch=z9hG4bKa9c4.3bb7312d97f058f2bf436327b55f0e2d.0
Record-Route: <sip:212.227.67.201;lr=on;ftag=914904559>
From: +4917024279xx <sip:+4917024279xx@1und1-3.sip.mgc.voip.telefonica.de:5060;user=phone>;tag=914904559
To: +4999136174xx <sip:+4999136174xx@62.53.206.22:5060;user=phone>;tag=as785fc21c
Call-ID: 588022e3-36d8eaa8-4cb824f8-8f2f@1und1-3.sip.mgc.voip.telefonica.de
CSeq: 1 CANCEL
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:4999136174xx@217.64.68.82>
Content-Length: 0


---
mobydick*CLI>
2

Hi!

Kann es sein, dass du für die s-Extension keine eingehende Regel erstellt hast? :confused:
Du brauchst bei der Quelle ein * und beim Ziel s, als Durchwahl dann das Ziel für den Anruf eintragen.
Schon sollte es eigentlich funktionieren…

Ne kurze Rückmeldung wäre super :wink:

Beste Grüße :cool:

Dom

3

schon geschehen, aber keine wirkung…

4

Hmm, mittendrin ging’s jetzt 2 mal hintereinander, sehr komisch

Der Log eines erfolgreichen Anrufes:


mobydick*CLI> sip debug peer 4999136174xx
SIP Debugging Enabled for IP: 212.227.67.204:5060
mobydick*CLI>
<-- SIP read from 212.227.67.204:5060:
INVITE sip:s@217.64.68.82 SIP/2.0
Record-Route: <sip:212.227.67.204;lr=on;ftag=425229367>
Record-Route: <sip:212.227.67.223;lr=on;ftag=425229367;did=15a.1da2dd8>
Record-Route: <sip:212.227.67.199;lr=on;ftag=425229367>
Via: SIP/2.0/UDP 212.227.67.204;branch=z9hG4bKfb0a.0f6d85fcb0aca3b7e7f26aa009a5a5bf.0
Via: SIP/2.0/UDP 212.227.67.223;branch=z9hG4bKfb0a.0f6d85fcb0aca3b7e7f26aa009a5a5bf.0
Via: SIP/2.0/UDP 212.227.67.199;branch=z9hG4bKfb0a.4db75db.0
Via: SIP/2.0/UDP 195.71.47.156;branch=z9hG4bKfb0a.4db75db.0
Via: SIP/2.0/UDP 62.53.206.6:5060;branch=z9hG4bK7qtls7309oq11kceq101.1
From: +491702427925 <sip:+491702427925@1und1-3.sip.mgc.voip.telefonica.de:5060;user=phone>;tag=425229367
To: +4999136174xx <sip:+4999136174xx@62.53.206.22:5060;user=phone>
Call-ID: 373339a6-55b3f254-6170d663-67d1@1und1-3.sip.mgc.voip.telefonica.de
CSeq: 1 INVITE
Max-Forwards: 23
Supported: timer
Session-Expires: 1800
Min-SE: 1800
Contact: +491702427925 <sip:+491702427925@62.53.206.6:5060;transport=udp>
Allow: INVITE,ACK,PRACK,SUBSCRIBE,BYE,CANCEL,NOTIFY,INFO,REFER,UPDATE
P-Asserted-Identity: +491702427925 <sip:+491702427925@1und1-3.sip.mgc.voip.telefonica.de;user=phone>
Content-Type: application/sdp
Content-Length: 680

v=0
o=- 2170486788 0 IN IP4 62.52.144.28
s=Cisco SDP 0
c=IN IP4 62.52.144.28
t=0 0
m=audio 31756 RTP/AVP 8 0 18 101 102 103 104 105 4 106 3 107 108 109 125 99 100
a=rtpmap:101 G729a/8000
a=rtpmap:102 G726-16/8000
a=rtpmap:103 G726-24/8000
a=rtpmap:104 G726-32/8000
a=rtpmap:105 G7231-H/8000
a=rtpmap:106 G7231-L/8000
a=rtpmap:107 G729b/8000
a=rtpmap:108 G7231a-H/8000
a=rtpmap:109 G7231a-L/8000
a=rtpmap:125 GnX64/8000
a=rtpmap:99 telephone-event/8000
a=fmtp:99 0-15
a=rtpmap:100 X-NSE/8000
a=fmtp:100 192-194,200-202
a=X-sqn:0
a=X-cap: 1 audio RTP/AVP 100
a=X-cpar: a=rtpmap:100 X-NSE/8000
a=X-cpar: a=fmtp:100 192-194,200-202
a=X-cap: 2 image udptl t38

--- (22 headers 25 lines) ---
Using INVITE request as basis request - 373339a6-55b3f254-6170d663-67d1@1und1-3.sip.mgc.voip.telefonica.de
Sending to 212.227.67.204 : 5060 (non-NAT)
Found peer '4999136174xx'
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 18
Found RTP audio format 101
Found RTP audio format 102
Found RTP audio format 103
Found RTP audio format 104
Found RTP audio format 105
Found RTP audio format 4
Found RTP audio format 106
Found RTP audio format 3
Found RTP audio format 107
Found RTP audio format 108
Found RTP audio format 109
Found RTP audio format 125
Found RTP audio format 99
Found RTP audio format 100
Peer audio RTP is at port 62.52.144.28:31756
Found description format G729a
Found description format G726-16
Found description format G726-24
Found description format G726-32
Found description format G7231-H
Found description format G7231-L
Found description format G729b
Found description format G7231a-H
Found description format G7231a-L
Found description format GnX64
Found description format telephone-event
Found description format X-NSE
Capabilities: us - 0xe (gsm|ulaw|alaw), peer - audio=0x10011f (g723|gsm|ulaw|alaw|g726|g729|h263p)/video=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Looking for s in mdc_incoming-1 (domain 217.64.68.82)
list_route: hop: <sip:212.227.67.204;lr=on;ftag=425229367>
list_route: hop: <sip:212.227.67.223;lr=on;ftag=425229367;did=15a.1da2dd8>
list_route: hop: <sip:212.227.67.199;lr=on;ftag=425229367>
Transmitting (NAT) to 212.227.67.204:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 212.227.67.204;branch=z9hG4bKfb0a.0f6d85fcb0aca3b7e7f26aa009a5a5bf.0;received=212.227.67.204
Via: SIP/2.0/UDP 212.227.67.223;branch=z9hG4bKfb0a.0f6d85fcb0aca3b7e7f26aa009a5a5bf.0
Via: SIP/2.0/UDP 212.227.67.199;branch=z9hG4bKfb0a.4db75db.0
Via: SIP/2.0/UDP 195.71.47.156;branch=z9hG4bKfb0a.4db75db.0
Via: SIP/2.0/UDP 62.53.206.6:5060;branch=z9hG4bK7qtls7309oq11kceq101.1
From: +491702427925 <sip:+491702427925@1und1-3.sip.mgc.voip.telefonica.de:5060;user=phone>;tag=425229367
To: +4999136174xx <sip:+4999136174xx@62.53.206.22:5060;user=phone>
Call-ID: 373339a6-55b3f254-6170d663-67d1@1und1-3.sip.mgc.voip.telefonica.de
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:s@217.64.68.82>
Content-Length: 0


---
    -- Executing Goto("SIP/4999136174xx-081bfb10", "mdc_trunk-1|s|1") in new stack
    -- Goto (mdc_trunk-1,s,1)
    -- Executing Set("SIP/4999136174xx-081bfb10", "__MDC_DIALCALLERNUM=+491702427925") in new stack
    -- Executing Set("SIP/4999136174xx-081bfb10", "LANGUAGE()=de") in new stack
    -- Executing GotoIf("SIP/4999136174xx-081bfb10", "0?forward") in new stack
    -- Executing Goto("SIP/4999136174xx-081bfb10", "mdc_external-1|s|1") in new stack
    -- Goto (mdc_external-1,s,1)
    -- Executing Set("SIP/4999136174xx-081bfb10", "CALLERID(name)=+491702427925") in new stack
    -- Executing Goto("SIP/4999136174xx-081bfb10", "mdc_external|123|1") in new stack
    -- Goto (mdc_external,123,1)
    -- Executing Set("SIP/4999136174xx-081bfb10", "MDC_CHANNEL_PROTOCOL=SIP") in new stack
    -- Executing Set("SIP/4999136174xx-081bfb10", "MDC_CHANNEL_ID=4999136174xx-081bfb10") in new stack
    -- Executing Set("SIP/4999136174xx-081bfb10", "MDC_CHANNEL_TMP=4999136174xx-081bfb10") in new stack
    -- Executing Set("SIP/4999136174xx-081bfb10", "MDC_CHANNEL_NAME=4999136174xx") in new stack
    -- Executing GosubIf("SIP/4999136174xx-081bfb10", "1?mdc_initcall-ext|123|1") in new stack
    -- Executing NoOp("SIP/4999136174xx-081bfb10", "initial call") in new stack
    -- Executing Set("SIP/4999136174xx-081bfb10", "__MDC_DIALDESCENT=ext") in new stack
    -- Executing Set("SIP/4999136174xx-081bfb10", "__MDC_DIALCALLEENUM=123") in new stack
    -- Executing GotoIf("SIP/4999136174xx-081bfb10", "1?nozap") in new stack
    -- Goto (mdc_initcall-ext,123,7)
    -- Executing Set("SIP/4999136174xx-081bfb10", "__MDC_DIALCALLERNAME=+491702427925") in new stack
    -- Executing Return("SIP/4999136174xx-081bfb10", "") in new stack
    -- Executing Gosub("SIP/4999136174xx-081bfb10", "mdc_defcall|123|1") in new stack
    -- Executing GotoIf("SIP/4999136174xx-081bfb10", "1?nozap") in new stack
    -- Goto (mdc_defcall,123,4)
    -- Executing Set("SIP/4999136174xx-081bfb10", "__MDC_DIALCHANNELNAME=4999136174xx") in new stack
    -- Executing Return("SIP/4999136174xx-081bfb10", "") in new stack
    -- Executing SIPAddHeader("SIP/4999136174xx-081bfb10", ""Alert-Info:<http://www.notused.de>;info=alert-external;x-line-id=0"") in new stack
    -- Executing Macro("SIP/4999136174xx-081bfb10", "pre-main") in new stack
Oct 10 22:55:12 WARNING[7049]: app_macro.c:208 macro_exec: No such context 'macro-pre-main' for macro 'pre-main'
    -- Executing GosubIf("SIP/4999136174xx-081bfb10", "1?mdc_initloop|s|1") in new stack
    -- Executing NoOp("SIP/4999136174xx-081bfb10", "initial loop") in new stack
    -- Executing Set("SIP/4999136174xx-081bfb10", "MDC_ALIAS_HOP=0") in new stack
    -- Executing Return("SIP/4999136174xx-081bfb10", "") in new stack
    -- Executing Goto("SIP/4999136174xx-081bfb10", "main|123|1") in new stack
    -- Goto (main,123,1)
    -- Executing Set("SIP/4999136174xx-081bfb10", "_MDC_VOICEMAIL=123") in new stack
    -- Executing Set("SIP/4999136174xx-081bfb10", "_MDC_WATCHED_VOICEMAIL=0") in new stack
    -- Executing Goto("SIP/4999136174xx-081bfb10", "mdc_ident-1|123|1") in new stack
    -- Goto (mdc_ident-1,123,1)
    -- Executing NoOp("SIP/4999136174xx-081bfb10", "alias-check:: Rufumleitung von 123 - 0") in new stack
    -- Executing GotoIf("SIP/4999136174xx-081bfb10", "0?123-umleitung|1:123-dial|1") in new stack
    -- Goto (mdc_ident-1,123-dial,1)
    -- Executing Gosub("SIP/4999136174xx-081bfb10", "mdc_prefix-123-ext|123|1") in new stack
    -- Executing Return("SIP/4999136174xx-081bfb10", "") in new stack
    -- Executing Gosub("SIP/4999136174xx-081bfb10", "mdc_main-123-ext|123|1") in new stack
    -- Executing Dial("SIP/4999136174xx-081bfb10", "SIP/123||o") in new stack
    -- Called 123
    -- SIP/123-081c29e0 is ringing
Transmitting (NAT) to 212.227.67.204:5060:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 212.227.67.204;branch=z9hG4bKfb0a.0f6d85fcb0aca3b7e7f26aa009a5a5bf.0;received=212.227.67.204
Via: SIP/2.0/UDP 212.227.67.223;branch=z9hG4bKfb0a.0f6d85fcb0aca3b7e7f26aa009a5a5bf.0
Via: SIP/2.0/UDP 212.227.67.199;branch=z9hG4bKfb0a.4db75db.0
Via: SIP/2.0/UDP 195.71.47.156;branch=z9hG4bKfb0a.4db75db.0
Via: SIP/2.0/UDP 62.53.206.6:5060;branch=z9hG4bK7qtls7309oq11kceq101.1
From: +491702427925 <sip:+491702427925@1und1-3.sip.mgc.voip.telefonica.de:5060;user=phone>;tag=425229367
To: +4999136174xx <sip:+4999136174xx@62.53.206.22:5060;user=phone>;tag=as78c8b69b
Call-ID: 373339a6-55b3f254-6170d663-67d1@1und1-3.sip.mgc.voip.telefonica.de
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:s@217.64.68.82>
Content-Length: 0


---
    -- SIP/123-081c29e0 is ringing
    -- SIP/123-081c29e0 is ringing
mobydick*CLI>
<-- SIP read from 212.227.67.204:5060:
CANCEL sip:s@217.64.68.82 SIP/2.0
Record-Route: <sip:212.227.67.204;lr=on;ftag=425229367>
Via: SIP/2.0/UDP 212.227.67.204;branch=z9hG4bKfb0a.0f6d85fcb0aca3b7e7f26aa009a5a5bf.0
Via: SIP/2.0/UDP 212.227.67.223;branch=z9hG4bKfb0a.0f6d85fcb0aca3b7e7f26aa009a5a5bf.0
From: +491702427925 <sip:+491702427925@1und1-3.sip.mgc.voip.telefonica.de:5060;user=phone>;tag=425229367
Call-ID: 373339a6-55b3f254-6170d663-67d1@1und1-3.sip.mgc.voip.telefonica.de
To: +4999136174xx <sip:+4999136174xx@62.53.206.22:5060;user=phone>
CSeq: 1 CANCEL
Max-Forwards: 69
User-Agent: Kamailio (1.5.4-tls (x86_64/linux))
Content-Length: 0


--- (11 headers 0 lines) ---
Sending to 212.227.67.204 : 5060 (NAT)
Reliably Transmitting (NAT) to 212.227.67.204:5060:
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 212.227.67.204;branch=z9hG4bKfb0a.0f6d85fcb0aca3b7e7f26aa009a5a5bf.0;received=212.227.67.204
Via: SIP/2.0/UDP 212.227.67.223;branch=z9hG4bKfb0a.0f6d85fcb0aca3b7e7f26aa009a5a5bf.0
Via: SIP/2.0/UDP 212.227.67.199;branch=z9hG4bKfb0a.4db75db.0
Via: SIP/2.0/UDP 195.71.47.156;branch=z9hG4bKfb0a.4db75db.0
Via: SIP/2.0/UDP 62.53.206.6:5060;branch=z9hG4bK7qtls7309oq11kceq101.1
From: +491702427925 <sip:+491702427925@1und1-3.sip.mgc.voip.telefonica.de:5060;user=phone>;tag=425229367
To: +4999136174xx <sip:+4999136174xx@62.53.206.22:5060;user=phone>;tag=as78c8b69b
Call-ID: 373339a6-55b3f254-6170d663-67d1@1und1-3.sip.mgc.voip.telefonica.de
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0


---
Transmitting (NAT) to 212.227.67.204:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 212.227.67.204;branch=z9hG4bKfb0a.0f6d85fcb0aca3b7e7f26aa009a5a5bf.0;received=212.227.67.204
Via: SIP/2.0/UDP 212.227.67.223;branch=z9hG4bKfb0a.0f6d85fcb0aca3b7e7f26aa009a5a5bf.0
Record-Route: <sip:212.227.67.204;lr=on;ftag=425229367>
From: +491702427925 <sip:+491702427925@1und1-3.sip.mgc.voip.telefonica.de:5060;user=phone>;tag=425229367
To: +4999136174xx <sip:+4999136174xx@62.53.206.22:5060;user=phone>;tag=as78c8b69b
Call-ID: 373339a6-55b3f254-6170d663-67d1@1und1-3.sip.mgc.voip.telefonica.de
CSeq: 1 CANCEL
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:s@217.64.68.82>
Content-Length: 0


---
  == Spawn extension (mdc_main-123-ext, 123, 1) exited non-zero on 'SIP/4999136174xx-081bfb10'
    -- Executing Macro("SIP/4999136174xx-081bfb10", "hangup||CANCEL|") in new stack
    -- Executing NoOp("SIP/4999136174xx-081bfb10", ">>>macro-hangup:: EXTEN:  DIALSTATUS: CANCEL QUEUESTATUS: ") in new stack
mobydick*CLI>
<-- SIP read from 212.227.67.204:5060:
ACK sip:s@217.64.68.82 SIP/2.0
Record-Route: <sip:212.227.67.204;lr=on;ftag=425229367>
Via: SIP/2.0/UDP 212.227.67.204;branch=z9hG4bKfb0a.0f6d85fcb0aca3b7e7f26aa009a5a5bf.0
Via: SIP/2.0/UDP 212.227.67.223;branch=z9hG4bKfb0a.0f6d85fcb0aca3b7e7f26aa009a5a5bf.0
From: +491702427925 <sip:+491702427925@1und1-3.sip.mgc.voip.telefonica.de:5060;user=phone>;tag=425229367
Call-ID: 373339a6-55b3f254-6170d663-67d1@1und1-3.sip.mgc.voip.telefonica.de
To: +4999136174xx <sip:+4999136174xx@62.53.206.22:5060;user=phone>;tag=as78c8b69b
CSeq: 1 ACK
Max-Forwards: 69
User-Agent: Kamailio (1.5.4-tls (x86_64/linux))
Content-Length: 0


--- (11 headers 0 lines) ---
Destroying call '373339a6-55b3f254-6170d663-67d1@1und1-3.sip.mgc.voip.telefonica.de'
12 headers, 0 lines
Reliably Transmitting (NAT) to 212.227.67.204:5060:
OPTIONS sip:sip.1und1.de SIP/2.0
Via: SIP/2.0/UDP 217.64.68.82:5060;branch=z9hG4bK06f65a08;rport
From: "asterisk" <sip:asterisk@217.64.68.82>;tag=as4c0c6401
To: <sip:sip.1und1.de>
Contact: <sip:asterisk@217.64.68.82>
Call-ID: 7328d14036c6308a0e00e7a773d0513d@217.64.68.82
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sun, 10 Oct 2010 20:55:31 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0


---
mobydick*CLI>
<-- SIP read from 212.227.67.204:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 217.64.68.82:5060;branch=z9hG4bK06f65a08;rport=5060
From: "asterisk" <sip:asterisk@217.64.68.82>;tag=as4c0c6401
To: <sip:sip.1und1.de>;tag=f8f2ab2c1295e90ed7dbb499b30f44b2.b1c2
Call-ID: 7328d14036c6308a0e00e7a773d0513d@217.64.68.82
CSeq: 102 OPTIONS
Server: UI Kamailio
Content-Length: 0


--- (8 headers 0 lines) ---
Destroying call '7328d14036c6308a0e00e7a773d0513d@217.64.68.82'
mobydick*CLI>
5

Hallo,

wenn es mal geht und mal nicht dann machst Du evtl. eine registrierung mit deiner internen IP die dann nach einer Zeit ungültig wird. Versuche mal bitte das: http://community.pascom.net/showthread.php?31-Verbindungsproblem-mit-Sipgate

LG
Mathias

6

Tag,
Hatte ich eigentlich schon, nur leider “externhost” statt “externip”

naja, schauma mal ob’s was bringt