Sip to ISDN problem

I am trying to replace our Inbound DID Trunks with a Sip Trunk.

The inbound DID uses wink start and sends 7 digits via DTMF.

The CLI log indicates that I have the DID to ISDN working - and it does

The Sip Trunk and a Sip Phone used for testing only get rejected for cause 96

-- Starting simple switch on 'DAHDI/1-1' -- Executing [6@pstn-in:1] Answer("DAHDI/1-1", "") in new stack -- Executing [6@pstn-in:2] NoOp("DAHDI/1-1", "6") in new stack -- Executing [6@pstn-in:3] Read("DAHDI/1-1", "pstn-dn,silence/5,6,n,0,5") in new stack -- Accepting a maximum of 6 digits. -- <DAHDI/1-1> Playing 'silence/5.gsm' (language 'en') -- User entered '101024' -- Executing [6@pstn-in:4] NoOp("DAHDI/1-1", "101024") in new stack -- Executing [6@pstn-in:5] Set("DAHDI/1-1", "dn=6101024") in new stack -- Executing [6@pstn-in:6] NoOp("DAHDI/1-1", "6101024") in new stack -- Executing [6@pstn-in:7] Set("DAHDI/1-1", "inboundchan=DAHDI/g1") in new stack -- Executing [6@pstn-in:8] Set("DAHDI/1-1", "CALLERID(num)=3347372300") in new stack -- Executing [6@pstn-in:9] Dial("DAHDI/1-1", "DAHDI/g1/ 6101024,5") in new stack -- Requested transfer capability: 0x00 - SPEECH -- Called DAHDI/g1/ 6101024 -- DAHDI/i2/ 6101024-d is proceeding passing it to DAHDI/1-1 -- DAHDI/i2/ 6101024-d is ringing -- DAHDI/i2/ 6101024-d answered DAHDI/1-1 -- Native bridging DAHDI/1-1 and DAHDI/i2/ 6101024-d -- Hungup 'DAHDI/i2/ 6101024-d' == Spawn extension (pstn-in, 6, 9) exited non-zero on 'DAHDI/1-1' -- Hanging up on 'DAHDI/1-1' -- Hungup 'DAHDI/1-1' == Using SIP RTP CoS mark 5 -- Executing [6101024@siptest1:1] Answer("SIP/123-00000002", "") in new stack > 0xb7304e98 -- Probation passed - setting RTP source address to 192.168.1.141:11788 -- Executing [6101024@siptest1:2] Dial("SIP/123-00000002", "Dahdi/g1/6101024,5") in new stack -- Requested transfer capability: 0x00 - SPEECH -- Called Dahdi/g1/6101024 -- Span 2: Channel 0/1 got hangup, cause 96 -- Hungup 'DAHDI/i2/6101024-e' == Everyone is busy/congested at this time (1:0/0/1) -- Executing [6101024@siptest1:3] Hangup("SIP/123-00000002", "") in new stack == Spawn extension (siptest1, 6101024, 3) exited non-zero on 'SIP/123-00000002' == Using SIP RTP CoS mark 5 -- Executing [3346101024@inbound:1] Dial("SIP/vitel-inbound-00000003", "DAHDI/g1/3346101024,5") in new stack -- Requested transfer capability: 0x00 - SPEECH -- Called DAHDI/g1/3346101024 -- Span 2: Channel 0/1 got hangup, cause 96 -- Hungup 'DAHDI/i2/3346101024-f' == Everyone is busy/congested at this time (1:0/0/1) -- Auto fallthrough, channel 'SIP/vitel-inbound-00000003' status is 'CHANUNAVAIL' wolverine*CLI>

system,conf:

span=1,0,0,esf,b8zs
span=2,0,0,esf,b8zs

e&m=1-24
bchan=25-47
dchan=48

chan_dahdi.conf

[code][trunkgroups]

[channels]
context=pstn-in
signalling=em_w
channel =>1-24

context=isdn-callin
group=1
switchtype=4ess
signalling=pri_cpe
channel =>25-47
[/code]

sip.conf

[code][general]
register => usr:psw@inbound29.vitelity.net:5060

disallow=all
allow=all

[123]
context=siptest1
type=friend
host=dynamic
secret=testphone123

[vitel-inbound]
type=peer
dtmfmode=auto
host=inbound29.vitelity.net
context=inbound
username=usr
secret=psw
allow=all
insecure=port,invite
canreinvite=no

[vitel-outbound]
type=friend
dtmfmode=auto
host=outbound.vitelity.net
username=usr
fromuser=use
secret=psw
trustrpid=yes
sendrpid=yes
allow=all
canreinvite=no

[/code]

and extensions.conf

[general]

[pstn-in]
exten => _N,1,Answer()
	same => n,NoOp(${EXTEN})
	same => n,Read(pstn-dn,silence/5,6,n,0,5)
	same => n,NoOp(${pstn-dn})
	same => n,Set(dn=${EXTEN}${pstn-dn})
	same => n,NoOp(${dn})
	same => n,Set(inboundchan=DAHDI/g1)
	same => n,Set(CALLERID(num)=3347372300)
;	same => n,Set(CALLERID(name)="APCI")
	same => n,Dial(${inboundchan}/ ${dn},5)
	same => n,Hangup()


[inbound]
exten => _334610102x,1,Dial(DAHDI/g1/${EXTEN},5)

[siptest1]
exten => _Nxxxxxx,1,Answer()
	same => n,Dial(Dahdi/g1/${EXTEN},5)
	same => n,Hangup()

I’ve tried changing all sorts of settings for the Sip Phone and the inbound Sip Trunk but so far nothing seems to make this work.

Any usefull help would be appreciated
Larry

FOUND THE ISSUE!!!

In a nutshell the call was rejected because from the Sip Trunk and the SipPhone the CallerID(name) field 'Display" was being sent but there was no information for it. When the call originated on the inbound T1 the “Display” field was not included in the ISDN setup message.

FIX ---- add Set(CALLERID(name-valid)=no) before the Dial command.