TIPS Viva WAZO: A New Beginning

Discussion in 'Developers' Corner' started by rjaiswal, Apr 23, 2016.

  1. wardmundy

    wardmundy Nerd Uno

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    @arztde Let me address your international issues more specifically. First, like most lazy Americans, I only speak English. Second, our apps are intended to be a guide to help OTHERS build apps. We can't possibly support a dozen languages in everything we do. If you want German, by all means build it and share it with the rest of us. Finally, as to weather, believe me we've tried to build international versions at least a half dozen times. And every time, the provider magically changes things to break our code. The National Weather Service does it once every 10 years, and we've made it extremely difficult for them to even detect our presence. We've given up on others but, by all means, be our guest. These are open source GPL projects, not limitless charitable contributions on my part. We expect international visitors to at least learn how their dial strings differ from 1NXXNXXXXXX. If that's insurmountable, then a commercial product produced in your home country is probably a better fit for you. As you said, 90% of our users are from the U.S. so that's where we must focus our attention given our limited resources.
     
    #81 wardmundy, May 13, 2016
    Last edited: May 13, 2016
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  2. arztde

    arztde Active Member

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    It will be good if such international prequisits are in a extra article and every time it match just pointed with a link. In this way it can grow.
    I realy suggest you to take my offer and set up a German line and try it yoursself. I will forward you the necesary data for localphone and freevoipdeal and you will understand where are the hickhughs realy if you leave the US System.
    There are Background reasons i stay so long
    I have an idea and did set up for this a Sourceforge Project. But as long i do not have such a stable demo running it makes no sense to go on there. I am not a programmer as you know. But the app Idea will push and for this it needs realy a running demo and absolute international free access. The cost control is for the moment realy providers that stop access if server is realy hacked. If there is no more credit than its just 10 $ what is lost. This is the cheapest inshurance i ever can have and does not need firewall for the moment.
     
  3. arztde

    arztde Active Member

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    @wardmundy i realy have no problems that the things are not running whatever i try. At home i have Snom R9m and did set it up and works for my needs. In Germany we have the Fritzbox and its easy to set it up. My plan is somehow to get write some nice app. A code that give others for some time free access automaticely. With Gemeinschaft 3 exist also a system special for German needs preconfigured. The question still exists to run a basic international incredible pbx and from there to go on. I did check the instruction with a 44 number to does not work if you leave disa. thats i did offer you to make a real live test. What i mean incomming calls no block! Acess to this number from everywhere. without to free every country seperatelely and the same to outgoing calls. External calls without restriction.
     
  4. wardmundy

    wardmundy Nerd Uno

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    DISA for XiVO Pioneers

    We're adding DISA functionality through a dialplan script that you can append to the end of xivo_extrafeatures.conf if you want to experiment. You'll find the config file under Configuration Files in the XiVO GUI's IPBX section.

    Here's the code:
    Code:
    ;# // BEGIN DISA
    exten => 3472,1,Answer
    exten => 3472,n,Wait(1)
    exten => 3472,n,Set(TIMEOUT(digit)=7)
    exten => 3472,n,Set(TIMEOUT(response)=10)
    exten => 3472,n,GotoIf($["${CALLERID(number)}"="701"]?disago1)  ; Good guy
    exten => 3472,n,Goto(bad,1)
    exten => 3472,n(disago1),Background(enter-password)
    exten => 3472,n,Read(MYCODE,beep,9)
    exten => 3472,n,GotoIf($["${MYCODE}" = "12341234"]?disago2:bad,1)
    exten => 3472,n(disago2),Set(TIMEOUT(absolute)=3600) ; 3600 seconds =1 hour
    exten => 3472,n,Read(NUM2CALL,pls-entr-num-uwish2-call,10)
    exten => 3472,n,Background(calling)
    exten => 3472,n,SayDigits("${NUM2CALL}")
    exten => 3472,n,GotoIf($["${NUM2CALL}" = "0"]?bad,1)
    exten => 3472,n,Dial(Local/${NUM2CALL}@default)
    exten => 3472,n,Hangup
    exten => bad,1,Hangup
    ;# // END DISA
    Before using this, make the following changes:

    1. Adjust the WhiteList to reflect "safe" CallerID numbers in line 5. Just change 701 to desired CallerID number. Clone the line to add more numbers to the WhiteList. We use 2-step security for DISA. You not only have to have a matching CallerID number when you dial in (yes, CallerID numbers can be spoofed!) but you also need to enter the 8-digit password.
    2. Set a very secure 8-digit password in line 9. It's your phone bill.
    3. Set the absolute timeout for DISA calls in line 10. 3600=1 hour
    4. Adjust maximum digits for outbound calls in line 11. NXXNXXXXXX = 10
    5. Once you save your changes, you can pick an IVR option such as 0 to call the DISA extension, 3472. Edit ivr-1.conf and change 0 option to:
    Code:
    exten => 0,1(ivrsel-0),Dial(Local/[email protected])
    Finally, add the traditional Asterisk sound files to your server:
    Code:
    cd /usr/share/asterisk/sounds/en
    wget http://downloads.asterisk.org/pub/telephony/sounds/asterisk-extra-sounds-en-ulaw-current.tar.gz
    wget http://downloads.asterisk.org/pub/telephony/sounds/asterisk-core-sounds-en-ulaw-current.tar.gz
    tar zxvf asterisk-extra-sounds-en-ulaw-current.tar.gz
    tar zxvf asterisk-core-sounds-en-ulaw-current.tar.gz
    chown asterisk:asterisk *.ulaw
    
    You obviously need an Outbound Trunk to make DISA calls, and the dial string must match the number of digits you set in step #4 above.
     
    #84 wardmundy, May 13, 2016
    Last edited: May 13, 2016
  5. henry

    henry Member

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    #85 henry, May 14, 2016
    Last edited by a moderator: May 14, 2016
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  6. arztde

    arztde Active Member

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    Outgoing calls to germany work like this. In this case you have to dial the number 383-the german number you like to call without the zero

    in my case 38323278381720


    major problem is that if you receive a call the number 023278381720 you see in the phone and to call back you need to type uncomfortable handy the calling number. additional is not clear how to spoof yours dialing number

    Outgouing calls -> Extern prefix = 49 Extension = 383XXXX
     
  7. ou812

    ou812 Guru

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    I have noticed that my voip.ms trunks does not play back ringing to the person calling in, I have not changed settings in the voip.ms portal I just unregistered trunk from Freepbx and registered on XIVO, so when calling in you hear silence until the call is answered, I can not find a setting in the sip trunks to enable this, anyone have a clue how to set this.

    Gary.
     
  8. ou812

    ou812 Guru

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    I have narrowed it down to the ring group I have it routed to, if set to a single user it plays back ring tone, if you dial ring group internally it does does not play ring back so I will look there.

    Gary
     
  9. ou812

    ou812 Guru

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    OK it turns out that when calling a group it immediately plays back the music on hold selection until the call is answered, so make sure MOH is set to blank and not default.

    Gary
     
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  10. arztde

    arztde Active Member

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    Got it to run :) Also with German number. Try 00498938038825.
     
  11. wardmundy

    wardmundy Nerd Uno

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    #91 wardmundy, May 15, 2016
    Last edited: May 16, 2016
  12. wardmundy

    wardmundy Nerd Uno

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    Deciphering SIP Trunk Settings

    Admittedly, setting up new SIP providers still is a little hit & miss when it comes to the appropriate XiVO Settings. The XiVO Devs are working to simplify this process. In the meantime, Pascal Cadotte (one of the primo XiVO Devs) has provided a huge hint on how to check your XiVO settings against the actual SIP settings generated for the trunk in the Asterisk/XiVO realtime environment.

    1. Figure out what your existing settings for a trunk provider should be in FreePBX or Asterisk. HINT: Our spreadsheet cheat sheet is still available.
    2. Create new SIP trunk using one of the existing SIP trunk setups that we already have working.
    3. Check how your settings got translated using the XiVO Decoder Badge: xivo-confgen asterisk/sip.conf.
    4. Compare the results with #1, above. Make adjustments as necessary, stir & repeat.
     
  13. wardmundy

    wardmundy Nerd Uno

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    Vitelity Outbound Trunk Settings

    See Nerd Vittles for Special $3.99 DID Offer with 4 Channels!


    In the General tab, you may have to change the Static Outbound FQDN for Vitelity depending upon when your account was created. Context should work fine with Outcalls (to-extern) if your setup doesn't allow Default entry. NAT may need to be changed to Yes depending upon where your server sits. Try both. One will work, and one won't. ;)

    [​IMG]

    Leave Registration tab blank.

    Signaling tab should be cloned as shown below:
    [​IMG]

    Clone the Advanced tab as shown below except From field-User entry below should be your Vitelity subaccount username:

    [​IMG]

    Call Management -> Outgoing Calls setup is the usual drill using whatever CallerID number you can legally spoof.

    [​IMG]
    [​IMG]
    [​IMG]

    Note: See Below for Incoming Trunk Settings
     
    #93 wardmundy, May 15, 2016
    Last edited: May 16, 2016
  14. wardmundy

    wardmundy Nerd Uno

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    Vitelity Incoming Trunk Settings

    See Nerd Vittles for Special $3.99 DID Offer with 4 Channels!


    In Vitelity portal, point your DID to the SubAccount to be registered with XiVO. Then, in XiVO GUI, create a new SIP Trunk in Trunk Management -> SIP Protocol.

    In the General tab, change the Inbound FQDN to match the server assigned to your account. Enter your credentials and DID for your Vitelity subaccount:

    [​IMG]

    In the Register tab, make it look like the following using your credentials and inbound FQDN:

    [​IMG]

    In Signaling tab, set DTMF, Monitoring, and Codec:

    [​IMG]

    In Advanced tab, set Insecure=ALL, Port=5060, and From field-User to your Subaccount name:

    [​IMG]

    SAVE your Trunk settings.

    Create a new entry in Call Management -> Incoming Calls.

    In General tab, set your DID number assigned by Vitelity and choose a destination for the incoming calls:

    [​IMG]

    In Call Permissions tab, authorize Everybody:

    [​IMG]

    SAVE your Inbound Route and place a test call from an outside phone to your Vitelity DID.
     
    #94 wardmundy, May 16, 2016
    Last edited: May 16, 2016
  15. wardmundy

    wardmundy Nerd Uno

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    Activating DNS Manager with XiVO

    You'll need this to reliably use SIP providers that have deployed round-robin IP addressing, e.g. SIP2SIP, CallCentric, Simonics. Here's how to set it up.

    Code:
    echo [general] > /etc/asterisk/dnsmgr.conf
    echo enable=yes >> /etc/asterisk/dnsmgr.conf
    chown asterisk:www-data /etc/asterisk/dnsmgr.conf
    /etc/init.d/asterisk restart
    asterisk -rx "dnsmgr status"
    
     
  16. wardmundy

    wardmundy Nerd Uno

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  17. gotel

    gotel Member

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  18. gotel

    gotel Member

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    Ward, I'm getting stuck on the 3rd step of RingPlus setup. I keep getting an error.
     
  19. Bryan Hiller

    Bryan Hiller New Member

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    When I followed the tutorial in post #96 I could not see configuration files from xivo GUI after
    Installing Dial Plan Code for Sample Incredible PBX Applications.
    Chmod 0775 and Chown asterisk:www-data on /etc/asterisk/extensions_extra.d fixed this. Hope this is correct. thanks
     
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  20. Sylvain Boily

    Sylvain Boily Active Member

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    Hello Bryan, have you created file on the web interface and directly via ssh ? It is normal if you change the owner file, it disappear on web interface.