TIPS Viva WAZO: A New Beginning

wardmundy

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UPDATE: You can try it for yourself now on the demo, option 1 and say "American Airlines" or "Delta Airlines."
demoline.gif
 
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wardmundy

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Just build a call range for each area you wish to support. The U.S./Canada one is what you have. For Germany, add something like 491000000000-499999999999. Then do something similar for iNUM (I think they all look like 8835100XXXXXXXX, e.g. 883510000000000-883510099999999.
 

wardmundy

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RingPlus SIP Trunk Setup Instructions:

1. In General Settings -> SIP Protocol:
CiHDs-KXAAEasuh.jpg


2. In Trunk Management -> SIP Protocol, click + to add a new SIP trunk and make it look like this using your RingPlus WiFi FluidCall credentials:

Ch9kaaAW0AA7luu.jpg

Ch9k2JLWUAQDFqI.jpg

Chs0f7PWwAA_Wy3.jpg


CjYwM1EWYAA4hHX.jpg


3. In Call Management -> Call Permissions, create an Everybody Group with your Users, Incoming Call Trunk, and Outgoing Call Context (outcall).

4. In (1) Contexts, edit (2) from-external context, (3) click on Incoming Calls tab, and fill in the remaining blanks (4, 5, and 6) as shown below and click (7) Save.

CjYo588WUAA9CFL.jpg


5. In Call Management -> Incoming Calls, create a new entry for your new SIP trunk with Call Permissions for Everybody and specify where to route incoming calls:

CiHHiUSXIAAovHC.jpg


6. In Call Management -> Outgoing Calls, create a new entry for your new SIP trunk and specify your Extension Dial String and CallerID matching your RingPlus number for Outbound Calls. If you leave CallerID blank, outbound calls will fail. Set Call Permissions for Everybody.

CiHGJOgWMAAdBDC.jpg

CiHG5kiWwAAwlsW.jpg


7. Check Asterisk CLI to be sure RingPlus Trunk is registered: asterisk -rx "sip show registry"
 
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MGD4me

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IPtables - security

If anyone ever doubts that exposing a PBX to the big bad internet without having strong protection is totally unsafe, try this for fun...

I wanted to give XiVO a whirl, but I have a lot of reading to do first. Especially regarding the web interface and vastly different approach to menu structures. But, I digress.

I followed Ward's tutorial to install XiVO on a Cloud@Cost server. I did NOT install IPtables (TM3) deliberately, as I was curious to see how long it would take to be discovered. I setup two extensions just to prove that the PBX was functional, but no trunks, and nowhere else to go. I left the PBX exposed for two days, then scrolled thorough the logs.

I was amazed at just how soon the PBX was discovered, with hundreds of attempts being made to authenticate common extensions, I guess to place free calls. Of course, I could be wrong ;o)

I did this to satisfy my curiosity, as an experiment. I have since deleted the server, and will eventually re-install an XiVO sandbox to trial further, but will follow the tutorial to lock it down. But, if anyone else has the time, or has an interest for the need to be paranoid, I found what I did to be a worthwhile exercise. It was 'interesting' to see the extent of the attacks, and the countries trying to get in by 'a reverse IP' search.

It was kind of like when your mother tells you not to do something, and you still want to try it, just to find out 'why'.
 

wardmundy

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Perhaps @Sylvain Boily would agree to host the raspi3 repo if we could get him a copy. There shouldn't be a problem finding a suitable alternative for the second one.
 

Sylvain Boily

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Hello, the raspivo distribution is maintained by a community member, not by us. So I'm not in charge. @TiJof is the maintainer of raspivo.
 
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wardmundy

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I did a fresh install and used Firefox this time. Firefox also complains about an invalid security certificate, so I had it make an exception so I could proceed.

Everything went smoothly this time all the way through the installation/validation process, but I'm still unable to log in at the authentication screen that follows. I used the password I set on the second line and tried numerous things on the first line, including leaving the word 'Login' there. There's a little browser activity each time, but I'm left on the authentication screen.

You're using root for the username??
 

wardmundy

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No, should I?

I assumed it was admin or the name I entered for Entity during the installation/validation.

I think it's always root and whatever password you set up in the web-based phase of the install.
 

RPi-Fan

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I think it's always root and whatever password you set up in the web-based phase of the install.
'root' was about the only thing I didn't try for a username. That got me in and everything appears to be up and running.

Now to get an extension and provider registered and see if I can make a call.

Thanks for the help!
 

wardmundy

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Working on FreeVoipDeal as we speak. Stay tuned. I only have zwei hands.

[update] Here are the FreeVoipDeal Settings for XiVO. Should be similar for any of the Betamax companies. Getting dangerous :arabia:

WARNING: Be sure to read the recent Nerd Vittles tutorial for the In's & Out's & Gotchas of Using Betamax trunks with Asterisk!

Trunk Setup

1. In General tab, use your acctname as Name and Authentication username and use your real Password. Then fill in the other blanks as shown using sip.freevoipdeal.com as provider address.

CiWmWZRXIAQfIcL.jpg

2. Skip the Register tab.

3. In the Signaling tab, DTMF=RFC2833 and Monitoring=Yes. Customize Codec to G.711 u-law.

4. In the Advanced tab, Insecure=All and Port=5060.

5. Save your setup.


Outgoing Call Setup

1. In General tab, make it look like this moving your FVD trunk entry to left (selected) column.

CiWopYPXEAEue95.jpg


2. In Exten tab, for U.S. calls only, it would look like this to dial 10-digit numbers using 383 (FVD) prefix for the outbound calls. Adjust as needed for International calling. Note that this setup strips off 383 before placing a call AND adds a 1-prefix to outgoing calls to comport with FVD dialing rules. Be sure to also plug in the spoofed Callerid you wish to use on the outbound calls.

CiWpkvxW0AENGqH.jpg
 
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wardmundy

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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.
 
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wardmundy

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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/3472@default)

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.
 
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ou812

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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.
 

ou812

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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
 

ou812

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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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