Asterisk stürzt ab nachdem aufgelgt wurde

[SeAcabo]

der titel beschreibt das problem ja bereits
ich benutzte asterisk ausschließlich mit sip
welche konfigurationsdatei könnte falsch sein
meine extensions.conf:

[general]
static=yes
writeprotect=no

[globals]


[default]
exten => 1,1,Dial(SIP/testusera)
exten => 2,1,Dial(SIP/testuserb)

meine sip.conf:

[general]
context=testproxy
recordhistory=yes
port = 5060 ; Port to bind to
bindaddr = 0.0.0.0 ; Address to bind to
context = default ; Default for incoming calls
srvlookup=yes ; Enable SRV lookups on outbound calls
dtmfmode=inband
allow=alaw ; Allow all codecs
language=de
realm=godot.de
useragent=godot
trustrpid = yes
nat=no
register => 677921:[email protected]/EXTEN

[testusera]
type=friend
username=testusera
secret=1234
host=dynamic
nat=no
canreinvite=no
allow=alaw
callerid=slu
regexten=1234 ; When they register, create extension 1234

[testuserb]
type=friend
username=testuserb
secret=1234
host=dynamic
nat=no
canreinvite=no
allow=alaw
callerid=ben
regexten=1223 ; When they register, create extension 1223

[testpeer]
type=peer
username=testpeer
secret=1234
host=localhost

könnte es eventuell an einer andern stelle das problem entstehen?

 

[SeAcabo]

also die ausgabe sagt :

2005-07-15 13:32:24 DEBUG[4142]: Check for res for testuserb
2005-07-15 13:32:24 DEBUG[4142]: Call from user 'testuserb' is 1 out of 0
2005-07-15 13:32:24 DEBUG[4142]: build_route: Contact hop: "ben" <sip:[email protected];transport=udp>
2005-07-15 13:32:24 VERBOSE[4142]: -- Executing [1;36;40mDial[0;37;40m("[1;35;40mSIP/testuserb-bbba[0;37;40m", "[1;35;40mSIP/testusera[0;37;40m") in new stack
2005-07-15 13:32:24 DEBUG[4142]: Setting NAT on RTP to 0
2005-07-15 13:32:24 DEBUG[4142]: Outgoing Call for testusera
2005-07-15 13:32:24 DEBUG[4142]: Call from user 'testusera' is 1 out of 0
2005-07-15 13:32:24 VERBOSE[4142]: -- Called testusera
2005-07-15 13:32:24 DEBUG[4142]: (Provisional) Stopping retransmission (but retaining packet) on '[email protected]' Request 102: Found
2005-07-15 13:32:24 VERBOSE[4142]: -- SIP/testusera-7aa0 is ringing
2005-07-15 13:32:27 DEBUG[4142]: update_user_counter(testusera) - decrement outUse counter
2005-07-15 13:32:27 DEBUG[4142]: Exiting with DIALSTATUS=CANCEL.
2005-07-15 13:32:27 VERBOSE[4142]: == Spawn extension (default, 1, 1) exited non-zero on 'SIP/testuserb-bbba'

vielleicht hilft das weiter