Peers are using wrong dialplan context

Asterisk Asterisk Support
1

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
2

Hi,
Can you post your asterisk cli output.

Thanks.
Suresh.

3

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.

4

Why don’t you sip show user?