Not found in context with allowexternaldomains = no

Hi there. Please help me with the situation:
There are 2 SIP phones defined in asterisk 1.8.1.1 sip.conf and a cisco 5350 as a peer:
[general]
allowexternaldomains = no
domain=192.168.2.14

[363778]
host = 192.168.2.18
type=friend
context=sip-phones

[303999]
type = friend
host = 192.168.2.15
context = sip-phones

[cisco-voip]
type = peer
host = 192.168.2.2
context = sip-phones
insecure = port

I can make calls between SIP phones without a hitch, but calls from cisco to any of SIP phones are NOT available.
Cisco is acting as a simple gateway from PSTN to Voip -> Asterisk.
Here is the part of cisco config:
!
dial-peer voice 1001 voip
destination-pattern 363778
session protocol sipv2
session target sip-server
!
sip-ua
retry options 0
sip-server ipv4:192.168.2.14
!
Here is the asterisk console output for call from PSTN to SIP phone 303999:

<— SIP read from UDP:192.168.2.2:50867 —>
INVITE sip:303999@192.168.2.14:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.2:5060;x-route-tag="cid:main@192.168.2.2";branch=z9hG4bK1362586
Remote-Party-ID: sip:9187599048@192.168.2.2;party=calling;screen=yes;privacy=off
From: sip:7599048@192.168.2.2;tag=859D217C-1288
To: sip:303999@192.168.2.14
Date: Fri, 07 Jan 2011 08:50:50 GMT
Call-ID: FCF8BF4-197211E0-8A31B83B-913D6D01@192.168.2.2
Supported: 100rel,timer,resource-priority,replaces
Min-SE: 1800
Cisco-Guid: 265220044-426906080-2518614035-2151582848
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER, SUBSCRIBE, NOTIFY, INFO, REGISTER
CSeq: 101 INVITE
Max-Forwards: 70
Timestamp: 1294390250
Contact: sip:7599048@192.168.2.2:5060
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Disposition: session;handling=required
Content-Length: 304

v=0
o=CiscoSystemsSIP-GW-UserAgent 223 7041 IN IP4 192.168.2.2
s=SIP Call
c=IN IP4 192.168.2.2
t=0 0
m=audio 19036 RTP/AVP 18 8 0 4
c=IN IP4 192.168.2.2
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:4 G723/8000
a=fmtp:4 bitrate=6.3;annexa=yes
<------------->
— (21 headers 13 lines) —
== Using UDPTL CoS mark 5
Sending to 192.168.2.2:5060 (no NAT)
Using INVITE request as basis request - FCF8BF4-197211E0-8A31B83B-913D6D01@192.168.2.2
Found peer ‘cisco-voip’ for ‘7599048’ from 192.168.2.2:50867
== Using SIP RTP CoS mark 5
Found RTP audio format 18
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 4
Found audio description format G729 for ID 18
Found audio description format PCMA for ID 8
Found audio description format PCMU for ID 0
Found audio description format G723 for ID 4
Capabilities: us - 0x50c (ulaw|alaw|g729|ilbc), peer - audio=0x10d (g723|ulaw|alaw|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x10c (ulaw|alaw|g729)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port 192.168.2.2:19036

<— Reliably Transmitting (no NAT) to 192.168.2.2:5060 —>
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.2.2:5060;x-route-tag="cid:main@192.168.2.2";branch=z9hG4bK1362586;received=192.168.2.2
From: sip:9187599048@192.168.2.2;tag=859D217C-1288
To: sip:303999@192.168.2.14;tag=as5bda0cbe
Call-ID: FCF8BF4-197211E0-8A31B83B-913D6D01@192.168.2.2
CSeq: 101 INVITE
Server: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<------------>
[Jan 7 11:50:50] NOTICE[16286]: chan_sip.c:21321 handle_request_invite: Call from ‘cisco-voip’ to extension ‘303999’ rejected because extension not found in context ‘sip-phones’.
Scheduling destruction of SIP dialog ‘FCF8BF4-197211E0-8A31B83B-913D6D01@192.168.2.2’ in 32000 ms (Method: INVITE)

Here is the sip-phone context from extension.conf
[sip-phones]
exten => _3XXXXX, 1, Dial(SIP/${EXTEN}/${EXTEN})
exten => _3XXXXX, n, Hangup()

and some more useful info:
dialplan show 303999@sip-phones
[ Context ‘sip-phones’ created by ‘pbx_config’ ]
’_3XXXXX’ => 1. Dial(SIP/${EXTEN}/${EXTEN}) [pbx_config]
2. Hangup() [pbx_config]

When I change option allowexternaldomains to yes - the problem disappears at all, and calls from cisco are accepted ok.
So I have a question -why it DOES NOT work with allowexternaldomains = no ???
192.168.2.14 - is the asterisk itself and there some local domain defined in sip.conf:

sip show domains
Our local SIP domains: Context Set by
192.168.2.14 (default) [Configured]
X.X.X.X (default) [Automatic]

Please help with this theory question. May be a have to define some other domain as local to make it work?
I tried to add domain=192.168.2.2 but nothing changes…
Thanks a lot for any help

To whom it may concerm:
The Invite message contains: INVITE sip:303999@192.168.2.14:5060 SIP/2.0
The problem solved by adding domain=192.168.2.14:5060 in sip.conf
It appears that 192.168.2.14 and 192.168.2.14:5060 are different domains for asterisk.
So, it became works with allowexternaldomains = no