SN2400: SIP-User ist circuit busy

[VOIPdani]

Hallo,

Ich möchte mittels cron jede Stunde einen Call absetzen: /usr/sbin/asterisk -rx 'dial 1503@FIRMAINTERN'

Jedoch habe ich nun das Problem, dass mein Asterisk folgendes sagt:

Executing Dial("IAX2/172.21.31.51:4569-1", "SIP/1503@OUTGOING|20|tr") in new stack
    -- Called 1503@OUTGOING
    -- SIP/OUTGOING-6d3f is circuit-busy
  == Everyone is busy/congested at this time (1:0/1/0)

Versuch ich nun mit einem Softphone 1503 zu wählen, funktioniert dies tadellos.

Executing SetCallerID("SIP/LetD-3fa3", "7326****") in new stack
    -- Executing Dial("SIP/LetD-3fa3", "SIP/1503@OUTGOING|20|tr") in new stack
    -- Called 1503@OUTGOING
    -- SIP/OUTGOING-9030 answered SIP/LetD-3fa3
    -- Attempting native bridge of SIP/LetD-3fa3 and SIP/OUTGOING-9030

Ich hab mal ein paar Debug-Monitore mitlaufen lassen, wenn ich versuche einen Call über /usr/sbin/asterisk -rx 'dial 1503@FIRMAINTERN' abzusetzen:

From: "Unknown" <sip:[email protected]>;tag=as64c87981
To: <sip:[email protected]>
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 04 Sep 2006 13:42:22 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 238

v=0
o=root 9902 9902 IN IP4 172.21.0.160
s=session
c=IN IP4 172.21.0.160
t=0 0
m=audio 19708 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

15:42:21  SIP_TR> [GW] > Stack: to 172.21.0.160:5060
SIP/2.0 100 Trying
Call-ID: [email protected]
CSeq: 102 INVITE
From: "Unknown" <sip:[email protected]>;tag=as64c87981
To: <sip:[email protected]>;tag=7ac5da560fc6e29
Via: SIP/2.0/UDP 172.21.0.160:5060;branch=z9hG4bK52f01092;rport
Content-Length: 0
User-Agent: Patton SN2400 MxSF v3.2.7.44 00A0BA01E76E

15:42:21  SIP_TR> [GW] > Stack: to 172.21.0.160:5060
SIP/2.0 404 Not Found
Call-ID: [email protected]
CSeq: 102 INVITE
From: "Unknown" <sip:[email protected]>;tag=as64c87981
To: <sip:[email protected]>;tag=7ac5da560fc6e29
Via: SIP/2.0/UDP 172.21.0.160:5060;branch=z9hG4bK52f01092;rport
Content-Length: 0
User-Agent: Patton SN2400 MxSF v3.2.7.44 00A0BA01E76E

15:42:22  SIP_TR> [GW] > Stack: to 172.21.0.160:5060
SIP/2.0 404 Not Found
Call-ID: [email protected]
CSeq: 102 INVITE
From: "Unknown" <sip:[email protected]>;tag=as64c87981
To: <sip:[email protected]>;tag=7ac5da560fc6e29
Via: SIP/2.0/UDP 172.21.0.160:5060;branch=z9hG4bK52f01092;rport
Content-Length: 0
User-Agent: Patton SN2400 MxSF v3.2.7.44 00A0BA01E76E

Kennt jemand dieses Phänomän, was kann ich dagegen tun?

Danke im Voraus, lg Dani