[Asterisk-Users] call from asterisk to SIP cisco 5300

[Ivan Vershigora]

i dial on my phone to to 80912222222
and convert it on asterisk to #00#70912222222
But Cisco says 404

============cisco peer=============
!
dial-peer voice 22 pots
 huntstop
 preference 5
 destination-pattern #00#......\*
 translate-outgoing calling 1
 direct-inward-dial
 port 0:D
 prefix 810
!
================================

============peer in sip.conf==========
[krdvox]
context=from-sip
type=peer
host=123.123.123.123
canreinvite=yes
dtmfmode=inband       
================================

============extensions.conf==========
exten => _.,1,SetCallerID("8612730000" <8612731107>[|a])
exten => _.,2,Dial(SIP/#00#7${EXTEN:1}@krdvox,60)
exten => _.,3,Congestion
================================

============Asterisk says===========
-- Executing Dial("SIP/201-2966", "SIP/#00#70912222222 at krdvox|60") in 
new stack
    -- Called #00#70912222222 at krdvox
    -- Got SIP response 404 "Not Found" back from XXX.XXX.XXX.XXX
    -- SIP/krdvox-3910 is circuit-busy
  == Everyone is busy/congested at this time
===============================

======CISCO debug ccsip ===========
Nov  3 16:10:03.516: Received:
INVITE sip:#00#70912222222 at 123.123.123.123 SIP/2.0
Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK6697eb34
From: "8612730000" <sip:8612730000 at 1.1.1.1>;tag=as74db268c
To: <sip:#00#70957555655 at 123.123.123.123>
Contact: <sip:8612730000 at 1.1.1.1>
Call-ID: 3ac14bc91f81edb732cc3681388b811d at 123.123.123.123
CSeq: 102 INVITE
User-Agent: CSCO/6
Date: Thu, 03 Nov 2005 13:10:06 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 235

.....

Nov  3 16:10:03.524: MatchNextPeer: Peer 999 matched
Nov  3 16:10:03.524: Using Voice Class Codec, tag=1

.....

Disconnect Cause (SIP)   : 404

===============================
Nov  3 16:10:03.524: MatchNextPeer: Peer 999 matched

Peer 999- wrong one !!!!!!!
why he cant find dial-peer voice 22


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