TIPS Viva WAZO: A New Beginning

wardmundy

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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.
 

wardmundy

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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. ;)

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Leave Registration tab blank.

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

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

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Note: See Below for Incoming Trunk Settings
 
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wardmundy

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Vitelity Incoming Trunk Settings

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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:

CilL38WXIAEn51j.jpg


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

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

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

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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:

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In Call Permissions tab, authorize Everybody:

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SAVE your Inbound Route and place a test call from an outside phone to your Vitelity DID.
 
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wardmundy

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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"
 

gotel

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RingPlus SIP Trunk Setup Instructions:

1. In General Settings -> SIP Protocol:
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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:

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

4. 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


5. In Call Management -> Outgoing Calls, create a new entry for your new SIP trunk and specify your Extension Dial String for Outbound Calls. Set Call Permissions for Everybody:

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

gotel

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

Bryan Hiller

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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
 

Sylvain Boily

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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.
 

wardmundy

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There were some bugs and permission problems that crept into the ivr-xivo.tar.gz tarball. These now have been addressed. Just log into your XiVO PBX as root and reinstall:

Code:
cd /
rm -f ivr-xivo.tar.gz
wget http://incrediblepbx.com/ivr-xivo.tar.gz
tar zxvf ivr-xivo.tar.gz
/etc/init.d/asterisk restart

Don't forget to reenter your Speech Recognition API key in /var/lib/asterisk/agi-bin/speech-recog.agi.
 
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wardmundy

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Simultaneous Cellphone Ringing for Inbound Calls

Unless you're using a RingPlus trunk where incoming calls to that number ring both your cellphone AND your PBX, there's another way to achieve the same result with XiVO.

Keep in mind that outbound calls in XiVO are routed out using dialing prefixes. If you have set up a trunk with a provider that allows CallerID spoofing such as Vitelity, Anveo Direct, or VoIP.ms, then you can preserve the caller's original CallerID number on the forwarded call to your mobile phone provided the dial string for your cellphone number matches the format you set up for the trunk you wish to use. For example, if Exten for Vitelity is 8NXXNXXXXXX, then you would enter the number for your cellphone with an 8 prefix: 89991234567.

In the User setup for one of your extensions:

1. Enter your cellphone number in the Mobile Phone Number field. Be sure it includes any necessary dial prefix so that it's routed out through the correct trunk.

2. On the same screen, you'll find a Preprocess subroutine field. Enter the following there: pre-mobility

3. SAVE your changes.
 

wardmundy

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@Sylvain Boily: Very nice... I tried adding this to example.conf, and it worked like a champ!

Code:
[xivo-subrgbl-outcall]
exten = s,1,NoOp(### Outbound Call Subroutine ###)
same = n,NoOp(Calling FROM: ${XIVO_SRCNUM})
same = n,NoOp(Calling TRNK: ${INTERFACE})
same = n,NoOp(Calling DEST: ${TRUNKEXTEN})
; for testing, add 511 to Outgoing Call exten list to test
; add your email address below instead of [email protected]
; dial 511 to be sure you get the email
; then change Outgoing Call exten to 911 and change 511 in next line to 911
same = n,GotoIf($["${TRUNKEXTEN}"="511"]?sendemail:return) ; change to 911 after testing
same = n(sendemail),System(echo "A 911 call was just placed from extension ${XIVO_SRCNUM}" | mail -s 911Emergency [email protected]) ; add your email address here
same = n(return),Return()

A practical use would be to send email/SMS alerts to supervisors whenever someone dialed 911. Or you could interface calls to a Time Slips application for billing purposes IF we only had a hangup subroutine to hook onto so we could calculate the length of the call. :arabia:
 
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Sylvain Boily

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