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Asterisk (with NodePhone) - Siemens C470IP configuration

Discussion in 'Other Operating Systems' started by birdie, Aug 23, 2011.

  1. birdie

    birdie Member

    Joined:
    Jun 18, 2002
    Messages:
    2,876
    Location:
    Bundaberg, Queensland
    Ok I've never used Asterisk before... but I've setup Asterisk following these settings: http://www.internode.on.net/pdf/products/np-business-trunks-user-manual.pdf

    I'm with NodePhone so I have configured.

    Now I'm trying to setup my Siemens C470IP VoIP phone to talk to the Asterisk box...

    This is my sip.conf
    Code:
    [general]
    ;For registration use your Service ID and password to register on behalf of the devices.
    register => 07xxxxxxxx:xxxxxxxx@sip.internode.on.net/s
    registertimeout=60
    context=default
    allowoverlap=no
    bindport=5060
    bindaddr=0.0.0.0
    srvlookup=yes
    
    [nodephone]
    context=nodephone
    type=friend
    username=07xxxxxxxx
    fromuser=07xxxxxxxx
    secret=xxxxxxxxxx
    fromdomain=sip.internode.on.net
    host=sipconnect.internode.on.net 
    dtmfmode=rfc2833
    disallow=all
    allow=alaw 
    allow=ulaw
    nat=no
    insecure=very
    
    [100]
    ;HomeLine 
    context=nodephone-outbound
    type=friend
    host=dynamic
    secret=xxxxxxxxxxx
    nat=no
    canreinvite=no
    disallow=all
    allow=ulaw
    allow=alaw
                
    
    This is my extensions.conf
    Code:
    [globals]
    ;----------------------------- 
    ; Users 
    ;----------------------------- 
    HomeLine=SIP/100
    
    [nodephone]
    ;send incoming calls from specified number to the appropriate extension 
    exten =>07xxxxxxxx,1,Dial(${HomeLine},20)
    exten =>07xxxxxxxx,n,Hangup
    
    Obviously I haven't got xxxx where my phone number and passwords are...

    Now on my C470IP what I have tried is:

    Code:
    Authentication Name: 100
    Authentication Password: xxxxxxxx
    Username: 100
    Display name: Asterisk
    
    Domain: <server ip>
    Proxy server address: (blank)
    Proxy server address: 5060
    Registrar server: <server ip>
    Registrar server port: 5060
    Registration refresh time: 180sec
    
    STUN disabled
    Any suggestions as where to start looking or what I have wrong?

    EDIT: Ok, I just rang my VoIP line from my mobile and the Asterisk box answered saying congratulations on setting up the asterisk install... so it seems as it is partially configured but its not completely working seeing that it went straight to that? or is that because my C470IP isn't logged in yet?
     
    Last edited: Aug 23, 2011
  2. OP
    OP
    birdie

    birdie Member

    Joined:
    Jun 18, 2002
    Messages:
    2,876
    Location:
    Bundaberg, Queensland
    Ok I've got my C470IP configured... not 100% what I had wrong... but its registered now...

    but I'm still getting the demo-congrats playing when i dial in to the home phone using my mobile...

    I also don't know how to configure the outgoing calls to use nodephone yet...

    So I have:

    Asterisk box <=> NodePhone - working (as I get the default asterisk demo message when i dial in from my mobile)
    C470IP VoIP Phone <=> Asterisk box - working as it is now registered and logged in...

    So i still need to get working:
    Outgoing calls via NodePhone - I think this might be because its not configured in extensions.conf? but I'm not sure how to do that?

    Incoming calls need to actually go to the handset rather than playing the demo-congrats file... which i'm not sure where its coming from as i started with a blank sip.conf and extensions.conf files... so it must be coming from another conf file...

    Any suggestions?
     
  3. Primüs

    Primüs Member

    Joined:
    Apr 1, 2003
    Messages:
    3,409
    Location:
    CFS
    Did you simply install asterisk on an existing linux server? I'd highly recommend using a specialised distribution such as Elastix, for the pure fact of easy configuration and maintainability. It comes with a nice web interface to set up trunks and whatnot.

    If you are against this idea for whatever reason, my guess is that you have not set up the appropriate extension/context for inbound calls. Specifically in your extensions.conf, where you have 07xxxx etc, you are saying send calls from that specific number, to your homeline extension. What you really want:

    Code:
    [globals]
    ;----------------------------- 
    ; Users 
    ;----------------------------- 
    HomeLine=SIP/100
    
    [nodephone]
    ;send incoming calls from specified number to the appropriate extension 
    exten =>_X.,1,Dial(${HomeLine},20)
    exten =>_X.,n,Hangup
    
    My pattern may be a little off but that should match most things.
     
  4. OP
    OP
    birdie

    birdie Member

    Joined:
    Jun 18, 2002
    Messages:
    2,876
    Location:
    Bundaberg, Queensland
    Yes the reason I'm running Ubuntu Server rather than a dedicated Asterisk distro, is I want it to do other tasks as well rather than just being a dedicated asterisk box...

    For incoming calls I had to change:
    register => 07xxxxxxxx:xxxxxxxx@sip.internode.on.net/s
    to
    register => 07xxxxxxxx:xxxxxxxx@sip.internode.on.net/07xxxxxxxx

    And use this for outgoing call setup:
    exten => _X.,1,Dial(SIP/nodephone/${EXTEN})

    So I've now got outgoing working...

    Incoming isn't quite yet though...

    I'm getting this error:
    Call from '' (xxx.xxx.xxx.xxx:5060) to extension '07xxxxxxxx' rejected because extension not found in context 'default'.
     
  5. Primüs

    Primüs Member

    Joined:
    Apr 1, 2003
    Messages:
    3,409
    Location:
    CFS
    My comment still stands it looks like on your incoming call context which appears to be [globals] you only have a matching extension of your FNN (07xxxxxxx), you need the extension number to be a pattern to catch what you want, and what to dial it on to. So for instance the context which handles your incoming calls will require the dial plan in my previous post.

    As for wanting it to run other tasks, I'd install elastix in VirtualBox. You seriously wouldn't regret it. You can do a lot more tasks very quickly and very easy to maintain. I haven't worked with Asterisk dial plans for a while, but when i knew them back to front, even then they were annoying to make and follow by hand, having the GUI's made life so much easier.
     
  6. OP
    OP
    birdie

    birdie Member

    Joined:
    Jun 18, 2002
    Messages:
    2,876
    Location:
    Bundaberg, Queensland
    Ok I have it working but I'm not 100% sure that is correct...

    This is what I have now:

    sip.conf
    Code:
    [general]
    ;For registration use your Service ID and password to register on behalf of the devices.
    register => 07xxxxxxxx:xxxxxxxxxxxx@sip.internode.on.net/07xxxxxxxx
    registertimeout=60
    context=nodephone-inbound
    allowoverlap=no
    bindport=5060
    bindaddr=0.0.0.0
    tcpbindaddr=0.0.0.0
    tcpenable=yes
    srvlookup=yes
    
    [nodephone]
    context=nodephone-inbound
    type=friend
    username=07xxxxxxxx
    fromuser=07xxxxxxxx
    secret=xxxxxxxxxxx
    fromdomain=sip.internode.on.net
    host=sipconnect.internode.on.net 
    dtmfmode=rfc2833
    disallow=all
    allow=alaw 
    allow=ulaw 
    nat=no
    insecure=very
    
    [1001]
    ;HomeLine 
    callerid=NodePhone Line 1
    context=nodephone-outbound
    type=friend
    host=dynamic
    secret=1001
    mailbox=1001
    dtmfmode=rfc2833
    nat=no
    canreinvite=no
    disallow=all
    allow=ulaw
    allow=alaw
    extensions.conf
    Code:
    [globals]
    ;----------------------------- 
    ; Users 
    ;----------------------------- 
    HomeLine=SIP/1001
    
    [nodephone-inbound]
    ;send incoming calls from specified number to the appropriate extension 
    exten => _X.,1,Dial(${HomeLine},20)
    exten => _X.,n,Hangup
    
    [nodephone-outbound]
    exten => _X.,1,Dial(SIP/nodephone/${EXTEN})
    So I changed the context from global to nodephone-inbound in sip.conf... but I've got a funny feeling its not even using the nodephone portion of sip.conf for some reason? should the register part be all in nodephone rather than global?
     
  7. OP
    OP
    birdie

    birdie Member

    Joined:
    Jun 18, 2002
    Messages:
    2,876
    Location:
    Bundaberg, Queensland
    Ok well after still having issues with my Asterisk install on Ubuntu still not working 100%... (A call in would get audio both ways, but a call out would only get outgoing audio)

    So I decided to give in, and I wiped and reinstalled a clean install of Ubuntu Server 11.04, then installed VirtualBox (headless without the GUI) and installed phpvirtualbox to manage it via a web browser... awesome! :D

    Anyway so I did a install of Elastix (loosely following this guide: http://dumbme.mbit.com.au/elastix/elastix_without_tears.pdf) in a VM, I have a couple of extensions setup (one for the C470IP and another for my laptop for testing, will eventually get a Cisco desk phone (one of the ones with the big screen they use in offices... :D) for my office) and the extensions are all working internally...

    I've also setup the trunk for my NodePhone but when I go to System tab of the web interface it tells me: Trunks (1) : (0 Registered) (0 Not Registered) (1 Unknown) and as a result calls are not going via the VoIP either way...

    I'm not sure what's wrong or how to troubleshoot it with Elastix...

    My trunk settings are:
    Code:
    Trunk Description: NodePhone1
    Outbound Caller ID: 07xxxxxxxx
    Dial rules:
    61+4XXXXXXXX
    61+XXXXXXXX
    61+XXXXXXXXX
    
    Trunk Name: NodePhone1
    
    PEER Detail:
    type=peer
    username=07xxxxxxxx
    fromuser=07xxxxxxxx
    secret=xxxxxxxxxxxxxx
    fromdomain=sip.internode.on.net
    host=sipconnect.internode.on.net
    dtmfmode=rfc2833
    disallow=all
    allow=ulaw&alaw
    nat=no
    insecure=very
    
    User Context: 07xxxxxxxx
    
    context=from-trunk
    host=sip.internode.on.net
    secret=xxxxxxxxxxxxxx
    type=user
    username=07xxxxxxxx
    insecure=very
    qualify=no
    canreinvite=no
    
    Register String:
    07xxxxxxxx:xxxxxxxxxxxxx@sip.internode.on.net/07xxxxxxxx
    
    I'm unsure what I've got wrong... any suggestions?
     
  8. OP
    OP
    birdie

    birdie Member

    Joined:
    Jun 18, 2002
    Messages:
    2,876
    Location:
    Bundaberg, Queensland
    Primüs: Any ideas as what might be going on?
     

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