Peers are using wrong dialplan context

Hello, everyone.
Sorry, if I’m missing something obvious here. I’m trying to setup asterisk 1.4 (tried the version from trunk and asterisk 1.4.2). I’ve configured a user entry in users.conf. One of the lines is “context = numberplan-custom-1”. I suppose that should make that user use the dialplan context [numberplan-custom-1]. I have [numberplan-custom-1] configured in extensions.conf. However the user uses [default].
Could anyone help me resolve this. Thanks in advance.

users.conf

[951XXXXXX]
callwaiting = yes
cid_number = 951XXXXXX
context = numberplan-custom-1
email =
fullname = New User
group =
hasagent = no
hasdirectory = no
hasiax = no
hasmanager = no
hassip = yes
hasvoicemail = yes
host = dynamic
mailbox = 951XXXXXX
secret = 000000
threewaycalling = yes
vmsecret = 1234
zapchan =
registeriax = no
registersip = yes

extensions.conf

[default]
exten = _X.,1,NoOp(“This is default”)

[numberplan-custom-1]
exten = _X.,1,NoOp(“This is numberplan-custom-1”)

Output of >sip show peer 951XXXXXX
CLI> sip show peer 951XXXXXX

  • Name : 951XXXXXX
    Secret :
    MD5Secret :
    Context : numberplan-custom-1
    Subscr.Cont. :
    Language :
    AMA flags : Unknown
    Transfer mode: open
    CallingPres : Presentation Allowed, Not Screened
    Callgroup : 1
    Pickupgroup : 1
    Mailbox : 951XXXXXX
    VM Extension : asterisk
    LastMsgsSent : 0/0
    Call limit : 0
    Dynamic : Yes
    Callerid : “New User” <951XXXXXX>
    MaxCallBR : 384 kbps
    Expire : 26
    Insecure : no
    Nat : RFC3581
    ACL : No
    T38 pt UDPTL : No
    CanReinvite : Yes
    PromiscRedir : No
    User=Phone : No
    Video Support: No
    Trust RPID : No
    Send RPID : No
    Subscriptions: Yes
    Overlap dial : No
    DTMFmode : rfc2833
    LastMsg : 0
    ToHost :
    Addr->IP : X.X.X.X Port 5060
    Defaddr->IP : 0.0.0.0 Port 5060
    Def. Username: 951XXXXXX
    SIP Options : (none)
    Codecs : 0x8000e (gsm|ulaw|alaw|h263)
    Codec Order : (none)
    Auto-Framing: No
    Status : Unmonitored
    Useragent : Sipura/SPA3000-2.0.11(GWg)
    Reg. Contact : sip:951XXXXXX@X.X.X.X:5060

Hi,
Can you post your asterisk cli output.

Thanks.
Suresh.

CLI>
– Executing [11111@default:1] NoOp(“SIP/80.1.61.21-092c23b0”,
"“This is default”") in new stack
== Auto fallthrough, channel ‘SIP/80.1.61.21-092c23b0’ status is
’UNKNOWN’

One strange thing I’ve noticed is that in lines like “SIP/80.1.61.21-092c23b0” before I used to see a number of extension that was calling. Now its my IP address, not a number. Another thing I didn’t mention is that I used GUI for the initial configuration.

Why don’t you sip show user?