No Intern Call

Hi… i play now for 2 days with my asterisk/freepbx and dont get it right to work…

I created 2 Extensions:
202 = a Software Phone - Nero SIPPS
203 = a Targa DIP Phone 450 (the same as Siemens Gigaset C450IP)

When I call from 202 -> 203 I get “Service Unavailable” on the screen
When I call from 203 -> 202 I get “VoIP Status Code: 503”

A call to a external Number (Outbound) works fine from both Phone.
A call from a external Trunk (Inbound) rings only on the 202 but the Route a as follow:

any DID / any CID => Set Destination => Ring Groups Alle <600>
And the Ring Group Alle <600> has Ring strategy: ringall and 202 + 201 in the extension list.

I need some help… i got this fu*k Phone (203) not to ringing…

This is the debug output when i call 202 > 203:

[code:1]
Asterisk 1.2.16-BRIstuffed-0.3.0-PRE-1x, Copyright © 1999 - 2006 Digium, Inc. and others.
Created by Mark Spencer [email protected]
Asterisk comes with ABSOLUTELY NO WARRANTY; type ‘show warranty’ for details.
This is free software, with components licensed under the GNU General Public
License version 2 and other licenses; you are welcome to redistribute it under
certain conditions. Type ‘show license’ for details.

Connected to Asterisk 1.2.16-BRIstuffed-0.3.0-PRE-1x currently running on proxy (pid = 12003)
proxyCLI> set debug 20
proxy
CLI> set verbose 20
Core debug was 0 and is now 20
Verbosity was 0 and is now 20
proxy*CLI> agi debug
AGI Debugging Enabled
– Executing Macro(“SIP/202-081d3460”, “exten-vm|novm|203”) in new stack
– Executing Macro(“SIP/202-081d3460”, “user-callerid”) in new stack
– Executing NoOp(“SIP/202-081d3460”, “user-callerid: device 202”) in new stack
– Executing GotoIf(“SIP/202-081d3460”, “0?report”) in new stack
– Executing GotoIf(“SIP/202-081d3460”, “0?start”) in new stack
– Executing Set(“SIP/202-081d3460”, “REALCALLERIDNUM=202”) in new stack
– Executing NoOp(“SIP/202-081d3460”, “REALCALLERIDNUM is 202”) in new stack
– Executing Set(“SIP/202-081d3460”, “AMPUSER=202”) in new stack
– Executing Set(“SIP/202-081d3460”, “AMPUSERCIDNAME=SIPPS”) in new stack
– Executing GotoIf(“SIP/202-081d3460”, “0?report”) in new stack
– Executing Set(“SIP/202-081d3460”, “CALLERID(all)=“SIPPS” <202>”) in new stack
– Executing Set(“SIP/202-081d3460”, “REALCALLERIDNUM=202”) in new stack
– Executing NoOp(“SIP/202-081d3460”, “TTL: ARG1: novm”) in new stack
– Executing GotoIf(“SIP/202-081d3460”, “0?continue”) in new stack
– Executing Set(“SIP/202-081d3460”, “_TTL=64”) in new stack
– Executing GotoIf(“SIP/202-081d3460”, “1?continue”) in new stack
– Goto (macro-user-callerid,s,21)
– Executing NoOp(“SIP/202-081d3460”, “Using CallerID “SIPPS” <202>”) in new stack
– Executing Set(“SIP/202-081d3460”, “FROMCONTEXT=exten-vm”) in new stack
– Executing Set(“SIP/202-081d3460”, “VMBOX=novm”) in new stack
– Executing Set(“SIP/202-081d3460”, “EXTTOCALL=203”) in new stack
– Executing Set(“SIP/202-081d3460”, “CFUEXT=”) in new stack
– Executing Set(“SIP/202-081d3460”, “CFBEXT=”) in new stack
– Executing Set(“SIP/202-081d3460”, “RT=”) in new stack
– Executing Macro(“SIP/202-081d3460”, “record-enable|203|IN”) in new stack
– Executing GotoIf(“SIP/202-081d3460”, “0?2:4”) in new stack
– Goto (macro-record-enable,s,4)
– Executing DeadAGI(“SIP/202-081d3460”, “recordingcheck|20070409-042339|asterisk-12003-1176085419.0”) in new stack
– Launched AGI Script /usr/share/asterisk/agi-bin/recordingcheck
AGI Tx >> agi_request: recordingcheck
AGI Tx >> agi_channel: SIP/202-081d3460
AGI Tx >> agi_language: de
AGI Tx >> agi_type: SIP
AGI Tx >> agi_uniqueid: asterisk-12003-1176085419.0
AGI Tx >> agi_callerid: 202
AGI Tx >> agi_calleridname: SIPPS
AGI Tx >> agi_callingpres: 0
AGI Tx >> agi_callingani2: 0
AGI Tx >> agi_callington: 0
AGI Tx >> agi_callingtns: 0
AGI Tx >> agi_dnid: 203
AGI Tx >> agi_rdnis: unknown
AGI Tx >> agi_context: macro-record-enable
AGI Tx >> agi_extension: s
AGI Tx >> agi_priority: 4
AGI Tx >> agi_enhanced: 0.0
AGI Tx >> agi_accountcode:
AGI Tx >>
– AGI Script recordingcheck completed, returning 0
– Executing NoOp(“SIP/202-081d3460”, “No recording needed”) in new stack
– Executing Macro(“SIP/202-081d3460”, “dial||t|203”) in new stack
– Executing DeadAGI(“SIP/202-081d3460”, “dialparties.agi”) in new stack
– Launched AGI Script /usr/share/asterisk/agi-bin/dialparties.agi
AGI Tx >> agi_request: dialparties.agi
AGI Tx >> agi_channel: SIP/202-081d3460
AGI Tx >> agi_language: de
AGI Tx >> agi_type: SIP
AGI Tx >> agi_uniqueid: asterisk-12003-1176085419.0
AGI Tx >> agi_callerid: 202
AGI Tx >> agi_calleridname: SIPPS
AGI Tx >> agi_callingpres: 0
AGI Tx >> agi_callingani2: 0
AGI Tx >> agi_callington: 0
AGI Tx >> agi_callingtns: 0
AGI Tx >> agi_dnid: 203
AGI Tx >> agi_rdnis: unknown
AGI Tx >> agi_context: macro-dial
AGI Tx >> agi_extension: s
AGI Tx >> agi_priority: 1
AGI Tx >> agi_enhanced: 0.0
AGI Tx >> agi_accountcode:
AGI Tx >>
– AGI Script dialparties.agi completed, returning 0
– Executing NoOp(“SIP/202-081d3460”, “Returned from dialparties with no extensions to call”) in new stack
– Executing NoOp(“SIP/202-081d3460”, "DIALSTATUS is ") in new stack
– Executing GosubIf(“SIP/202-081d3460”, “0?docfu|1”) in new stack
– Executing GosubIf(“SIP/202-081d3460”, “0?docfb|1”) in new stack
– Executing NoOp(“SIP/202-081d3460”, “Voicemail is novm”) in new stack
– Executing GotoIf(“SIP/202-081d3460”, “1?s-|1”) in new stack
– Goto (macro-exten-vm,s-,1)
– Executing PlayTones(“SIP/202-081d3460”, “congestion”) in new stack
– Executing Congestion(“SIP/202-081d3460”, “10”) in new stack
== Spawn extension (macro-exten-vm, s-, 2) exited non-zero on ‘SIP/202-081d3460’ in macro ‘exten-vm’
== Spawn extension (macro-exten-vm, s-, 2) exited non-zero on ‘SIP/202-081d3460’
– Saved useragent “TARGA IP FON 1020” for peer 203
– Got SIP response 405 “Method Not Allowed” back from 192.168.99.140
[/code:1]

Thanks for Help,
Thomas