sip.web.de with asterisk

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Hello together,

perhaps one of you can help me.
I must confess, I know myself not yet all too well with asterisk from however internal discussions get I already times problem-free.
my problem is now the following:
I yesterday announced myself with web.de to freephone and with the mad down load of web.de can one also telephone, however has I mean data registered in my asterisk and it does now not too much.

I attach simply times one configs and the appropriate cli spent:

#######sip.conf:

[general]
port=5060
externip=<, I will replace my present IP however through dyndns>
bindaddr= 0.0.0.0
context=default
srvlookup=yes
localnet= 192.168.2.0/255.255.255.0
allow=gsm
disallow=all
canreinvite=no
language=de
register=< user>: <passwd> @ sip.web.de/012012< more meine_nummer>
nat=no
dtmfmode=info
tos=0x18

[web.de]
more type=peer
username=< user>
secret=< passwd>
host=sip.web.de
fromuser=< user>
fromdomain=sip.web.de
context=default
canreinvite=no
qualify=yes
disallow=all
allow=gsm
insecure=very
nat=no
dtmfmode=info
tos=0x18


; user “Tobias”
[200]
type=friend
secret=test
insecure=very
; host= 192.168.1.200
host=dynamic
; defaultip= 192.168.1.200
disable=all
allow=alaw
allow=ulaw
dtmfmode=rfc2833
username=200
context=200
qualify=1000
callerid= " Tobias " <200>
canreinvite=no
reinvite=no
nat=yes

; user “Christian”
[201]
type=friend
secret=test
“sip.conf” 79L, 1057C


#######extensions.conf:

[default]
would Inc.-load => call
[calls]
exten => 01212< my number>, 1, dial (Zap/2/200,60, tT)
exten => 01212< my number>, 2, Hangup



[200]
exten => _201,1, NoOp (“call for “$ {EXTEN})
exten => _201,2, dial (SIP/$ {EXTEN}, 60, tr)
exten => _201,3, Congestion


exten => _x., 1, SetCallerID, 4888124
exten => _x., 2, dial (SIP/& {EXTEN} sipgate out, 60, tr)
exten => _x., 3, Congestion
exten => _x., 102, Busy

[201]
exten => _200,1, NoOp (“call for “$ {EXTEN})
exten => _200,2, dial (SIP/$ {EXTEN}, 60, tr)
exten => _200,3, Congestion


exten => _x., 1, SetCallerID, 4888124
exten => _x., 2, dial (SIP/& {EXTEN} sipgate out, 60, tr)
exten => _x., 3, Congestion
exten => _x., 102, Busy







CLI log:

<-- SIP READ from 217.72.200.89: 5060:

-- (0 headers 0 LINEs) Nat keepalive --
localhost*CLI>
<-- SIP READ from 217.72.200.89: 5060:
INVITE sip: 012012302500499@ 80.145.203.33: 1024 SIP/2.0
Record route: <sip: 217.72.200.89; ftag=as0418ac4e; lr=on>
Via: SIP/2.0/UDP 217.72.200.89; branch=z9hG4bK642c.a01cbcc4.0
Via: SIP/2.0/UDP 217.72.200.73: 5060; branch=z9hG4bK5dcde227; rport=50 60
From: “<mine festnetz=> Caller> “ <sip: <mine festnetz=> Caller> @web.de>; tag=as0418ac4e
Ton: <sip: <my web.de user> @sip.web.de>
Contact: <sip: <mine festnetz=> Caller> @ 217.72.200.73: 5060>
Call ID: 6b6fb49a156e9a3d2f4ad7dc4a24beb3@web.de
CSeq: 102 INVITE
User agent: Asterisk PBX
Max Forwards: 16
DATE: Fri, 02 June 2006 14:59: 01 GMT
Allow: INVITE, ACK, CANCEL, OPTION, BYE, REFER, SUBSCRIBE, NOTIFY
Content type: application/sdp
Content length: 322
P-rear: usrloc applied

v=0
o=root 16429 16429 IN IP4 217.72.200.73
s=session
c=IN IP4 217.72.200.73
t=0 0
m=audio 10972 RTP/AVP 8 0 111 3 97 101
a=rtpmap: 8 PCMA/8000
a=rtpmap: 0 PCMU/8000
a=rtpmap: 111 G726-32/8000
a=rtpmap: 3 GSM/8000
a=rtpmap: 97 iLBC/8000
a=rtpmap: 101 telephone-event/8000
a=fmtp: 101 0-16
a=silenceSuppFF -- --

-- (16 headers 14 LINEs)--
Using INVITE request as basis request - 6b6fb49a156e9a3d2f4ad7dc4a24beb3@web.de
Sending ton of 217.72.200.89: 5060 (non NAT)
Found more peer “web.de”
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 111
Found RTP audio format 3
Found RTP audio format 97
Found RTP audio format 101
Peer audio RTP is RK haven 217.72.200.73: 10972
Found description format PCMA
Found description format PCMU
Found description format G726-32
Found description format GSM
Found description format iLBC
Found description format telephone vent
Capabilities: US - 0x2 (gsm), more peer - audio=0x41e (gsm|ulaw|alaw|g726|ilbc) /video=0x0 (emergency-hung), combined - 0x2 (gsm)
Non codec of capabilities: US - 0x1 (telephone vent), more peer - 0x1 (telephone vent), combined - 0x1 (telephone vent)
Looking for 012012302500499 in default (domain 80.145.203.33)
Reliably Transmitting (NO NAT) ton of 217.72.200.89: 5060:
SIP/2.0 404 emergency Found
Via: SIP/2.0/UDP 217.72.200.89; branch=z9hG4bK642c.a01cbcc4.0; receiv ed= 217.72.200.89
Via: SIP/2.0/UDP 217.72.200.73: 5060; branch=z9hG4bK5dcde227; rport=50 60
From: “<mine festnetz=> Caller> “ <sip: <mine festnetz=> Caller> @web.de>; tag=as0418ac4e
Ton: <sip: <my web.de user> @sip.web.de>; tag=as6369f83a
Call ID: 6b6fb49a156e9a3d2f4ad7dc4a24beb3@web.de
CSeq: 102 INVITE
User agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTION, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip: 012012302500499@ 80.145.203.33>
Content length: 0


---
localhost*CLI>
<-- SIP READ from 217.72.200.89: 5060:
ACK sip: 012012302500499@ 80.145.203.33: 1024 SIP/2.0
Via: SIP/2.0/UDP 217.72.200.89; branch=z9hG4bK642c.a01cbcc4.0
From: “<mine festnetz=> Caller> “ <sip: <mine festnetz=> Caller> @web.de>; tag=as0418ac4e
Call ID: 6b6fb49a156e9a3d2f4ad7dc4a24beb3@web.de
Ton: <sip: <my web.de user> @sip.web.de>; tag=as6369f83a
CSeq: 102 ACK
User agent: Sip express rout (0.9.4 (i386/linux))
Content length: 0


-- (8 headers 0 LINEs)--
Destroying call “6b6fb49a156e9a3d2f4ad7dc4a24beb3@web.de”
localhost*CLI>





I would be genuinly very very gratefully for a solution or at least an analysis I despair here still.

Thank you already times for the numerous answer

---

Your problem is nevertheless completely clearly described in the SIP log:

Code:

Looking for 012012302500499 in default (domain 80.145.203.33)

 SIP/2.0 404 emergency Found

The call arrives on your Asterisk, but one does not find extension the 012012302500499 in the context [default] extensions.conf

Probably you mistyped simply in extensions.conf.

---

Yes, looks like tip by:

Code:

[default]
 would Inc.-load => calls
 [calls]
 exten => 01212< my number>, 1, dial (Zap/2/200,60, tT)
 exten => 01212< my number>, 2, Hangup

should probably:

[default]
would Inc.-load => call
[calls]
exten => 012012< my number>, 1, dial (Zap/2/200,60, tT)
exten => 012012< my number>, 2, Hangup

read.

Greeting,
Tin

---

Hello I again,

it tunes genuinly a typing error I can from externally my softphones call now

now I must wait only to web.de me my Web cents de-energise thereby I also times to call can.

Many thank you for your fast answer.
Genuinly super of you thanks