[cisco-voip] SIP /SCCP call forwarding

[Vince Loschiavo]

I think you'll need a media termination point for doing forwards, transfers, etc with h323. 

Regards,

Vince Loschiavo
Senior Cisco Engineer
DATACORP
7862821164

Sent via BlackBerry from T-Mobile    

-----Original Message-----
From: Ahmad Cheikh-Moussa <acm at netuse.de>
Date: Sun, 28 Jan 2007 15:26:38 
To:cisco-voip at puck.nether.net
Subject: [cisco-voip] 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)      = works
PSTN -> 7940 (444-222)         = works
7940 -> mailbox (111)          = works
7940 -> mailbox (0444-111)     = works

PSTN -> 7940 (444-222) call forward all is set to 111 = doesn't work
PSTN -> 7940 (444-222) call forward all is set to 0444-111 =  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 at 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 at 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-Straße, 24107 Kiel, Germany
Telefon: +49 431 2390 400 --  Telefax: +49 431 2390 499
Service: Service at NetUSE.DE --  http://NetUSE.DE/
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