[Asterisk-Users] SIP call doesn't execute the 's'-extension

[Tobias Wolf]

Hi,

i have just started to configure access to the * over SIP-Phones. 
Therefore I have defined this SIP-Phone in sip.conf:

[tobias]
type=friend
username=tobias
secret=tobias
auth=md5
host=dynamic
reinvite=no
dtmfmode=inband
callerid="Tobias" <1087006>
allow=all
context=javaAgi
dtmfmode=rfc2833


As you can see i am directing calls from this user to the context 
[javaAgi] which is defined here in extension.conf:

[javaAgi]
exten => s,1,Answer()
exten => s,2,Playback(code1000)
exten => s,3,Hangup()
exten => 1,1,Answer()
exten => 1,2,Playback(code1000)
exten => 1,3,Hangup()

If i dial 1 on my SIP Phone everything works as suspected, the call is 
answered and the gsm-file is played. My understanding of the 
's'-extension is, that it is executed then a call comes in an there is 
no extension wich matches the called number. But if i dial a random 
number i get an "404 Not found" error.

Here is an snippet of what * tells me on sip debug, but i can't get a 
clue out of it:


12 headers, 13 lines
Using latest request as basis request
Sending to 10.3.4.98 : 5060 (non-NAT)
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 98
Found RTP audio format 97
Found RTP audio format 101
Peer audio RTP is at port 10.3.4.98:8000
Found description format pcmu
Found description format pcma
Found description format gsm
Found description format iLBC
Found description format speex
Found description format telephone-event
Capabilities: us - 0x8000e(GSM|ULAW|ALAW|H263), peer - 
audio=0x60e(GSM|ULAW|ALAW|SPEEX|ILBC)/video=0x0(EMPTY), combined - 
0xe(GSM|ULAW|ALAW)
Non-codec capabilities: us - 0x1(G723), peer - 0x1(G723), combined - 
0x1(G723)
Found user 'tobias'
Looking for 2 in javaAgi
Reliably Transmitting (no NAT):
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 
10.3.4.98:5060;branch=z9hG4bK102F7CD4855F4C4E927C3398E3C57BF4
From: Tobias <sip:tobias at 10.3.1.6>;tag=2760968676
To: <sip:2 at 10.3.1.6>;tag=as396962de
Call-ID: 79A5523F-AFCA-4DBE-9AA2-F51377E8B5AE at 10.3.4.98
CSeq: 58303 INVITE
User-Agent: evision PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:2 at 10.3.1.6>
Content-Length: 0

Perhaps anyone can point me to the right direction ??

Tobias