SIP /SCCP call forwarding

Hi Guys, =

I have a strange problem. Perhaps someone had a similar problem and =

could solve it. I've got an asterisk server, a H.323 gateway and =

a 7940 SCCP Phone. The callmanager version is 4.1.3(Sr1).

Asterisk mailbox : 111
7940             : 222
ISDN30-Number    : 444.

For example if someone from the pstn wants to call the =

7940, then the user calls 444-222.

The Problem is, when I make a callforward all on the 7940 to
the asterisk number 111, all calls from external doesn't work.
If I make the callforward all to 0444-111, then the callforward works.
Did anyone experienced such a problem. Here a list of calls, which works.

PSTN -> mailbox (444-111)      =3D works
PSTN -> 7940 (444-222)         =3D works
7940 -> mailbox (111)          =3D works
7940 -> mailbox (0444-111)     =3D works

PSTN -> 7940 (444-222) call forward all is set to 111 =3D doesn't work
PSTN -> 7940 (444-222) call forward all is set to 0444-111 =3D  work

Here a cut of my config:
voice call convert-discpi-to-prog
voice rtp send-recv
!
voice service voip
allow-connections h323 to h323
allow-connections h323 to sip
allow-connections sip to h323
sip
!

dial-peer voice 800 voip
destination-pattern 111
session protocol sipv2
session target sip-server
dtmf-relay rtp-nte
codec g711ulaw

sip-ua
set sip-status 401 pstn-cause 127
set sip-status 407 pstn-cause 127
set sip-status 410 pstn-cause 22
set sip-status 415 pstn-cause 127
set sip-status 480 pstn-cause 19
set sip-status 503 pstn-cause 127
set sip-status 580 pstn-cause 127
retry invite 3
retry register 3
timers register 150
registrar ipv4:1.2.3.51 expires 3600
sip-server ipv4:1.2.3.51
!

During the debugging, I could see by the call, which doesn't work,
that the H.323 Gateway send a cancel message and the call is terminated.

1. solero01 -> astvoice: INVITE
2. astvoice -> solero01: SIP/2.0 100 Trying
3. astvoice -> solero01: SIP/2.0 200 OK
4. solero01 -> astvoice: CANCEL sip:111@1.2.3.51:5060 SIP/2.0
5. astvoice -> solero01: SIP/2.0 487 Request Terminated
6. astvoice -> solero01: SIP/2.0 200 OK [...] CSeq: 101 CANCEL
7. solero01 -> astvoice: ACK sip:111@1.2.3.51:5060 SIP/2.0

I made the same debugging on the H.323 Gateway, but I could not see a reason
for the cancel. =

Regards,
Ahmad

-- =

Ahmad Cheikh-Moussa =

NetUSE AG
Dr.-Hell-Stra=DFe, 24107 Kiel, Germany
Telefon: +49 431 2390 400 --  Telefax: +49 431 2390 499
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