звонок от циски на астериск по сипу идёт в default context

dial-peer voice 91234567 voip
 destination-pattern 91234567
 session protocol sipv2
 session target ipv4:192.168.333.333
 codec g711alaw

[2921_8m]
type=peer
host=192.168.222.222
dtmfmode=rfc2833
context=allow_mezhg
disallow=all
allow=alaw
allow=ulaw
allow=g729
qualify=yes
jbforce=yes
insecure=invite

 == Using SIP RTP TOS bits 96
  == Using SIP RTP CoS mark 5
  == Using UDPTL TOS bits 96
  == Using UDPTL CoS mark 5
Sending to 192.168.333.333 : 5060 (no NAT)
Found RTP audio format 8
Found RTP audio format 19
Found audio description format PCMA for ID 8
Found audio description format CN for ID 19
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x8 (alaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x2 (CN), combined - 0x0 (nothing)
Peer audio RTP is at port 192.168.333.333:16400
Looking for 91234567 in sip (domain 192.168.222.222)

<--- Reliably Transmitting (no NAT) to 192.168.333.333:5060 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.333.333:5060;x-route-tag="tgrp:2292068";branch=z9hG4bK8E55;received=192.168.333.333
From: <sip:192.168.333.333>;tag=33B13570-1889
To: <sip:91234567@192.168.222.222>;tag=as7e419fb9
Call-ID: 6ADAC80A-BD6A11E2-B467C3BC-D8B58A70@192.168.333.333
CSeq: 101 INVITE
Server: Asterisk PBX 1.6.2.17.2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 0


<------------>
[May 16 18:21:44] NOTICE[11425]: chan_sip.c:20314 handle_request_invite: Call from '' to extension '91234567' rejected because extension not found in context 'sip'.