SOLVED Adding Vitelity Trunk to Incredible Pi

rcalv002

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Hi everyone,

First off thank you for the wonderful work you're doing here to bring lower cost telephony to all! I setup my pbxiaf installation on a raspberrypi B device. Used the link for the raspberrypi installation (Incredible PBX 3.11 for the Raspberry Pi featuring the very latest Asterisk 11 and FreePBX 2.11 versions for a near perfect telephony platform) put it on a sdcard it boots everythings up. I used the google motif to add a google account to the system and it worked after a bit of fidgeting. Now I ported my house number to vitelity and its complete. I don't know exactly what I'm suposed to do to add it to my system and have it all work out. This is the only info I get from Vitelity

Please add the following configuration to your /etc/asterisk/sip.conf

Add the following at the bottom of your sip.conf


Code:
[inbound]
type=friend
dtmfmode=auto
host=inbound29.vitelity.net
context=inbound
allow=all
insecure=port,invite
 
[outbound]
type=friend
dtmfmode=auto
host=outbound.vitelity.net
context=outbound
insecure=port,invite
allow=all


Please add the following configuration to your /etc/asterisk/extensions.conf

Add the following to the bottom of your extensions.conf


Code:
[outbound]
exten => _1NXXNXXXXXX,1,Dial(SIP/$\{EXTEN}@outbound)
exten => _011.,1,Dial(SIP/$\{EXTEN}@outbound)
 
; e911 must be enabled. see DIDs > NPANXXNXXX > Action > e911
exten => _911,1,Dial(SIP/911@outbound)
 
[inbound]
exten => mynumberishere,1,Answer


...I looked in these files wasnt sure whether i should manually add these settings or not, and there is no mention of usernames or passwords in that configuration offered by vitelity. So I used the GUI in PBXIAF to add an extension 702, trunk, inbound, outbound route. I'm not sure the settings I added there were correct. I'm also not sure the settings on the vitelity portal are correct either. I do have a subaccount created on vitelity and it says it is online, I tried it out with YATE and it connects and dials out to my verizon cellphone, however I get no audio either way. I have opened up a ton of those ports as it says in a pdf i found on pbxiaf website that said to open 5060-5082 and 10000:20000. In other places it says to not open any ports at all to the device, so I'm confused on what I'm supposed to do. I would welcome any help from someone with experience on vitelity or pbxiaf for raspberry or pbx in general.

Thank you all in advance, hope to hear from you soon!

Edit ----

I added this information straight into the files it asks for and saved/restarted but none of the info is pulled up in the gui.

At the top of your /etc/asterisk/sip.conf right after the general settings add:
register =>myusername:[email protected]:5060

Near the bottom add:

Code:
[inbound]
type=friend
dtmfmode=auto
host=inbound29.vitelity.net
context=inbound
username=myusername
secret=mypass
allow=all
insecure=port,invite
canreinvite=no
 
[outbound]
type=friend
dtmfmode=auto
host=outbound.vitelity.net
username=myusername
fromuser=myusername
secret=mypass
trustrpid=yes
sendrpid=yes
allow=all
canreinvite=no
 

hbonath

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Take a look at the vitelity portal under support and there are instructions for adding to freepbx through the GUI.
Their instructions show a very old version of freepbx but it should translate to the current version just fine.
The screen you want to be on in freepbx is under connectivity -> trunks.
Once that's in there you can move to configuring inbound and outbound routes.
 

rcalv002

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Hello!

Thanks for the response. I went ahead and wiped it clean and started over. I added the trunk, then an extension 702, inbound, outbound route. I can make calls from my softphone to my cellphone and I can hear 2 way audio just fine. However when I make calls to my home phone thats now been ported to vitelity all I get is a busy signal. I went to DID - Main, routed the number to my only subaccount. When I go to DID - Subaccount it says that account is online registered and it is logged in on my softphone. What is the problem with the busy signal? This is all I get from the CLI when I try to call in to it.

Code:
Connected to Asterisk 11.3.0-rc1 currently running on incrediblepbx (pid = 2909)
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
incrediblepbx*CLI>

Also, I am not using any portforwards now and its all working just fine, except for the busy signal. By the way, can someone point me to a guide on inbound and outbound routes to make sure I'm not goofing that up?

Thanks again!
 

hbonath

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Yep sounds like you're missing your inbound route. Go to connectivity - inbound routes and add a new one. Under DID enter your phone number in the format of: 2125551212 and under destination point to your extension if that's what you want to ring.
Make sure your vitelity inbound trunk has a registration string matching your subaccount auth info so that vitelity knows how to find your PBX.
 

rcalv002

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Hi,

Thanks for the help I really appreciate it. When I go to inbound route theres no registration string, that was only under trunk and i set it there. is there anything else i need to change?..

The registration string does have the auth info for my one and only sub account.
 

hbonath

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Sorry for the confusion, my ADHD kicks in sometimes! I was referring to the inbound route under the vitelity "inbound" trunk, in the trunks section.
As long as that's good and freepbx system status shows a trunk registration, your inbound route pointing to your extension should be all you need.
 

rcalv002

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Ah,

This is what it says.

Code:
FreePBX Connections
IP Phones Online
1
IP Trunks Online
2
IP Trunk Registrations
1
 

hbonath

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That looks good. Is your inbound route working now?
 

hbonath

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Have you opened up the asterisk cli and watched the logs when you call in?
From Linux command line type asterisk -r then call your home number.
Do you see anything in the log? If possible, paste that part of the log in a post.

Also do you have insecure=port,invite set in the incoming trunk?
 

rcalv002

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I had pasted this up earlier in the thread, this is what pops up when i call, with asterisk -rvvvvvvvvv
Code:
Connected to Asterisk 11.3.0-rc1 currently running on incrediblepbx (pid = 2909)
  == Using SIP RTP TOS bits 184
  == Using SIP RTP CoS mark 5
incrediblepbx*CLI>

Yes, I have insecure=port,invite in the incoming section of the trunk.

I received this message from support.

Code:
Hello,
 
Your PBX or device is returning a 404 not found.
 
66.241.99.28 108.x.x.7 7203750306 ####### Jul 24 18:19:53.644 PM Jul 24 18:19:53.737 PM 404 Not Found
Jul 24 18:19:53.644 PM 66.241.99.28 > 108.x.x.7 INVITE sip:########@108.x.x.7:49885 SIP/2.0
Jul 24 18:19:53.737 PM 66.241.99.28 < 108.x.x.7 SIP/2.0 404 Not Found
Jul 24 18:19:53.737 PM 66.241.99.28 > 108.x.x.7 ACK sip:########@108.x.x.7:49885 SIP/2.0
 
Please ensure you have a proper inbound route set for this DID.

I must have setup my route incorrectly.
 

geopeterwc

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First off, delete anything that you've added to sip.conf and extensions.conf. The information best followed from Vitelity Support Page instructions for FreePBX - the GUI interface to Asterisk that's included in your PIAF installation.

Step 1: Create your trunk definition for the Vitelity DID
Step 2: Set up how calls are to be handled for incoming calls (ie., ring group)
Step 3: Define the INBOUND ROUTE for calls
Step4: Define the OUTBOUND ROUTE for calls (this step not covered below)

Using the FreePBX GUI, place the following (modify for your credentials) in the trunk configuration for your Vitelity trunk:

Trunk Name: [AnythingYouWant, ie., vitel1234]

Outbound Caller ID: [your DID number]

CID Options: [Allow Any CID]

Maximum Channels: [2]

OUTGOING SETTINGS
Trunk Name [give your trunk a name that you can recall, ie., vitel1234-out]

PEER DETAILS
type=friend
dtmfmode=auto
username=user_sub1
secret=Password
fromuser=user_sub1
trustrpid=yes
sendrpid=yes
context=from-trunk ; (this could be ext-did or from-pstn as well)
canreinvite=no
host=outbound.vitelity.net

INCOMING SETTINGS
USER Context: [whatever you wish, ie., vitel1234-in]

USER DETAILS:
type=friend
dtmfmode=auto
username=user_sub1
secret=Password
context=from-trunk ; (this could be ext-did or from-pstn as well)
insecure=very
canreinvite=no
host=inbound29.vitelity.net

Register String:
user_sub1:[email protected]:5060
-------------------------------------------------------

Create a RING GROUP (600)
Group Description: [All Phones]

Extension List:
place any/all extensions that you want to ring when a call comes in on this DID into the Extension List
and define which of these extensions will answer (voicemail) if no answer.

NOTE: If you have only one extension defined for your PIAF PBX, you can skip configuration of a ring group. Doing so ensures flexibility in the PIAF configuration though - ie., more extensions.
-------------------------------------------------------

Create an INBOUND ROUTE

Description: [incoming-any DID]

In the Set Destination field, select:
[Ring Groups] - [all phones <600>]

If only one extension, select:
[Extensions] - [extension number]

-------------------------------------------------------
Left to your imagination is the configuration of the Outbound Route where you will configure how the trunk will handle dialed calls. Vitelity requires that the dialed format is 1NXXXXXXXXX, and there are any number of ways that you can configure the Dial Patterns depending on how you are accustomed to dialing local and out-of-the-area calls.

ALSO, be certain that the Vitelity trunk(s) are configured to route to support [SIP] | [PBX Server or Softswitch] (You may configure any of the following DID numbers to route to a PBX, Switch or ATA device/softphone, configure call failover, call forwarding and much more on this page.)

Hope this helps ... works for me on 11 Vitelity DIDs, one of which is a Toll-Free DID.

/Pete./
 

rcalv002

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Yea!

The route was fine the only thing i had to change was context=from-trunk in the incoming section of the trunk and tada.
Thank you both very much for your help, I'm sure we'll chat again soon ;)
 

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