Asterisk Weiterleitung an Cisco Callmanager

[suchmich1983]

Hallo Forum,

ich komme nicht mehr weiter, daher suche ich nun hier um Hilfe.

Ich möchte Asterisk als IVR System nutzen. Als PBX nutzen wir den Cisco Callmanager in der Version 4.1. Den Trunk habe ich eingerichtet und ich erreiche beim wählen auch schon das IVR Menü auf dem Asterisk.
Wenn ich allerdings eine Nummer drücke und der Anruf an den Callmanager zurück gegeben werden soll, berkomme ich immer folgende Meldung:

-- Executing [1@voicemenu-custom-1:1] Dial("SIP/192.168.5.62-081ec870", "SIP/[email protected]") in new stack
    -- Called [email protected]
    -- SIP/192.168.5.62-081a2aa8 is circuit-busy
  == Everyone is busy/congested at this time (1:0/1/0)
  == Auto fallthrough, channel 'SIP/192.168.5.62-081ec870' status is 'CONGESTION'

Hier auch noch mal den Part aus meiner extension.conf und die Einträge in der sip.conf:
extension.conf

comment=NL
alias_exten=891000
exten=s,1,Answer
include=default
exten=s,2,Background(thank-you-for-calling)
exten=s,3,Background(if-u-know-ext-dial)
exten=s,7,Background(press-1)
exten=s,5,Background(for)
exten=s,6,Background(distribution)
exten=s,10,Background(press-2)
exten=s,8,Background(for)
exten=s,9,Background(administration)
exten=s,13,Background(press-3)
exten=s,11,Background(for)
exten=s,12,Background(customer-service)
exten=s,4,Background(press-0)
exten=s,14,Background(to-reach-operator)
exten=1,1,Dial(SIP/[email protected])
exten=4,4,Hangup
exten=5,102,Congestion

sip.conf

[general]
context = default
bindport = 5060
bindaddr = 0.0.0.0
srvlookup = yes

[callman01]
type = friend
context = default
host = 192.168.x.y
disallow = all
allow = ulaw
allow = alaw
nat = no
canreinvite = yes
qualify = yes

[callman02]
type = friend
context = default
host = 192.168.x.x
disallow = all
allow = ulaw
allow = alaw
nat = no
canreinvite = yes
qualify = yes

Kann mir hier jemand helfen? Muss beim Callmanager noch was eingetragen werden? Oder fehlen noch Einstellungen am Asterisk?

Danke und Gruß
Christian

 

[suchmich1983]

Ich habe mal ein Log mitlaufen lassen, wenn ich versuche das Gespräch an den Callmanager zurück zu geben. Vielleicht sagt das jemandem was? Wäre für jede Hilfe dankbar!!!

tack
Audio is at 192.168.5.74 port 15964
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 192.168.5.61:5060:
INVITE sip:491534@DESOWI0020 SIP/2.0
Via: SIP/2.0/UDP 192.168.5.74:5060;branch=z9hG4bK26c8987c;rport
From: "Pohle, Christian" <sip:[email protected]>;tag=as70382076
To: <sip:491534@DESOWI0020>
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 07 Nov 2008 10:08:41 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 285

v=0
o=root 2047 2047 IN IP4 192.168.5.74
s=session
c=IN IP4 192.168.5.74
t=0 0
m=audio 15964 RTP/AVP 0 3 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
    -- Called 491534@DESOWI0020
asterisk1*CLI>
<--- SIP read from 192.168.5.61:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.5.74:5060;branch=z9hG4bK26c8987c;rport
From: "Pohle, Christian" <sip:[email protected]>;tag=as70382076
To: <sip:491534@DESOWI0020>;tag=50871012
Date: Fri, 07 Nov 2008 10:10:18 GMT
Call-ID: [email protected]
CSeq: 102 INVITE
Allow-Events: telephone-event
Content-Length: 0


<------------->
--- (9 headers 0 lines) ---
asterisk1*CLI>
<--- SIP read from 192.168.5.61:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.5.74:5060;branch=z9hG4bK26c8987c;rport
From: "Pohle, Christian" <sip:[email protected]>;tag=as70382076
To: <sip:491534@DESOWI0020>;tag=50871012
Date: Fri, 07 Nov 2008 10:10:18 GMT
Call-ID: [email protected]
CSeq: 102 INVITE
Allow-Events: telephone-event
Content-Length: 0


<------------->
--- (9 headers 0 lines) ---
Transmitting (no NAT) to 192.168.5.61:5060:
ACK sip:491534@DESOWI0020 SIP/2.0
Via: SIP/2.0/UDP 192.168.5.74:5060;branch=z9hG4bK26c8987c;rport
From: "Pohle, Christian" <sip:[email protected]>;tag=as70382076
To: <sip:491534@DESOWI0020>;tag=50871012
Contact: <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
    -- SIP/DESOWI0020-08223060 is circuit-busy
  == Everyone is busy/congested at this time (1:0/1/0)
  == Auto fallthrough, channel 'SIP/192.168.5.62-0821f0e8' status is 'CONGESTION'
Really destroying SIP dialog '[email protected]' Method: INVITE
asterisk1*CLI>
<--- SIP read from 192.168.5.62:5060 --->
BYE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP  192.168.5.62:5060;branch=z9hG4bK31486290
From: "Pohle, Christian" <sip:[email protected]>;tag=34433101
To: <sip:[email protected]>;tag=as2f4c11e3
Date: Fri, 07 Nov 2008 10:10:08 GMT
Call-ID: [email protected]
User-Agent: Cisco-CCM4.1
Max-Forwards: 70
CSeq: 102 BYE
Content-Length: 0

 

[suchmich1983]

Hat wirklich niemand eine Idee?