Smbdy pls help: outgoing call failing?

Hi,

I want to make a ZAP outgoing call when SIP call lands on Asterisk

I have setup my dialplan as follows:

exten=>xyz,1,Dial(ZAP/1/xyz)

but I am getting following error:
though I am able to see SETUP message
but I am getting Release in reponse with error code as “Unallocated (unassigned) number”

can somebody help me, where am I going wrong

<— SIP read from 61.246.0.46:4060 —>
INVITE sip:01244176316@61.246.0.46 SIP/2.0
Via: SIP/2.0/UDP 61.246.0.46:4060;branch=z9hG4bK2705804096-24964
Route: sip:172.31.121.25:5060;lr
Max-Forwards: 70
Request-Disposition: no-fork
P-Asserted-Identity:sip:6623600@sipserver.com,sip:6623600@sipserver.com
Allow: INVITE,ACK,CANCEL,BYE,OPTIONS
Supported: timer
Accept: application/sdp
P-App-Service-Info:OrigServiceProcessed
Accept-Language:en
Accept-Encoding:identity
From: user sip:+911246623605@sipserver.com;tag=ICF_2685233681-10150
To: sip:01244176316@61.246.0.46
Call-ID: 38@61.246.0.46
CSeq: 1 INVITE
Contact: sip:61.246.0.46:4060
Content-Type: application/sdp
Content-Length: 131

v=0
o=user 12345 787 IN IP4 61.246.0.48

c=IN IP4 61.246.0.48
t=0 0
m=audio 12300 RTP/AVP 8
a=rtpmap:8 PCMA/8000/1

<------------->
— (19 headers 7 lines) —
Sending to 61.246.0.46 : 4060 (no NAT)
Using INVITE request as basis request - 38@61.246.0.46
Found no matching peer or user for '61.246.0.46:4060’
Found RTP audio format 8
Peer audio RTP is at port 61.246.0.48:12300
Found description format PCMA for ID 8
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port 61.246.0.48:12300
Looking for 01244176316 in default (domain 61.246.0.46)
list_route: hop: sip:61.246.0.46:4060

<— Transmitting (no NAT) to 61.246.0.46:4060 —>
SIP/2.0 100 Trying
rom: user sip:+911246623605@sipserver.com;tag=ICF_2685233681-10150
To: sip:01244176316@61.246.0.46
Call-ID: 38@61.246.0.46
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: sip:01244176316@172.31.121.25
Content-Length: 0

[b]
<------------>
– Executing [01244176316@default:1] Dial(“SIP/sipserver.com-08ec5c08”, “Zap/14/+911244095888”) in new stack
– Making new call for cr 32776
– Requested transfer capability: 0x00 - SPEECH

Protocol Discriminator: Q.931 (8) len=49
Call Ref: len= 2 (reference 8/0x8) (Originator)
Message type: SETUP (5)
[04 03 80 90 a3]
Bearer Capability (len= 5) [ Ext: 1 Q.931 Std: 0 Info transfer capability: Speech (0)
Ext: 1 Trans mode/rate: 64kbps, circuit-mode (16)
Ext: 1 User information layer 1: A-Law (35)
[18 03 a9 83 8e]
Channel ID (len= 5) [ Ext: 1 IntID: Implicit PRI Spare: 0 Exclusive Dchan: 0
ChanSel: Reserved
Ext: 1 Coding: 0 Number Specified Channel Type: 3
Ext: 1 Channel: 14 ]
[6c 0f 21 80 2b 39 31 31 32 34 36 36 32 33 36 30 35]
Calling Number (len=17) [ Ext: 0 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1)
Presentation: Presentation permitted, user number not screened (0) ‘+911246623605’ ]
[70 0e a1 2b 39 31 31 32 34 34 30 39 35 38 38 38]
Called Number (len=16) [ Ext: 1 TON: National Number (2) NPI: ISDN/Telephony Numbering Plan (E.164/E.163) (1) ‘+911244095888’ ]
[a1]
Sending Complete (len= 1)
q931.c:2879 q931_setup: call 32776 on channel 14 enters state 1 (Call Initiated) – Called 14/+911244095888
Audio is at 172.31.121.25 port 15842
Adding codec 0x8 (alaw) to SDP
[/b]
<— Transmitting (no NAT) to 61.246.0.46:4060 —>
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 61.246.0.46:4060;branch=z9hG4bK2705804096-24964;received=61.246.0.46
From: user sip:+911246623605@sipserver.com;tag=ICF_2685233681-10150
To: sip:01244176316@61.246.0.46;tag=as043e3cd5
Call-ID: 38@61.246.0.46
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: sip:01244176316@172.31.121.25
Content-Type: application/sdp
Content-Length: 186

v=0
o=root 30709 30709 IN IP4 172.31.121.25
s=session
m=audio 15842 RTP/AVP 8
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------>
< Protocol Discriminator: Q.931 (8) len=9
< Call Ref: len= 2 (reference 8/0x8) (Terminator)
< Message type: RELEASE COMPLETE (90)
< [08 02 84 81]
< Cause (len= 4) [ Ext: 1 Coding: CCITT (ITU) standard (0) Spare: 0 Location: Public network serving the remote user (4)
< Ext: 1 Cause: Unallocated (unassigned) number (1), class = Normal Event (0) ]
– Processing IE 8 (cs0, Cause)
q931.c:3501 q931_receive: call 32776 on channel 14 enters state 0 (Null)
– Channel 0/14, span 1 got hangup, cause 1
NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Null
NEW_HANGUP DEBUG: Destroying the call, ourstate Null, peerstate Null
– Hungup ‘Zap/14-1’
[Feb 4 19:43:25] NOTICE[30837]: cdr.c:434 ast_cdr_free: CDR on channel ‘Zap/14-1’ not posted
== Everyone is busy/congested at this time (1:0/0/1)
== Auto fallthrough, channel ‘SIP/sipserver.com-08ec5c08’ status is ‘CHANUNAVAIL’

<— Transmitting (no NAT) to 61.246.0.46:4060 —>
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 61.246.0.46:4060;branch=z9hG4bK2705804096-24964;received=61.246.0.46
From: user sip:+911246623605@sipserver.com;tag=ICF_2685233681-10150
To: sip:01244176316@61.246.0.46;tag=as043e3cd5
Call-ID: 38@61.246.0.46
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: sip:01244176316@172.31.121.25
Content-Length: 0
X-Asterisk-HangupCause: Unallocated (unassigned) number
X-Asterisk-HangupCauseCode: 1

just to add ,
I am using PRI line Q.931 signalling and a digium ISDN card.