Trunk Settings

Gronkstar

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#1
I am very knew to Asterisk PBX Server software. I have tried many different free PBX Server solution suites build as an iso with Cent OS but run into the same problem when trying to setup the Australian Carrier "BBPGlobal" as i am only given:

SIP SERVER Address: sip1.bbpglobal.com
PORT: 5190
ID: 613001313
Password: Disclosed

Yet with Asterisks you need a lot more well it seems it anyways. I have no idea on how to configure it. I have read the Elastrix without tears PDF's as i have with all the other Suites but they are all the same documents and if i use the supplied Peer & User Information( however changing the specified fields such as, sip address, port, user name and password ) it dosent work. I am a complete dummy when it comes to this the only thing i can see to work out and get right is the Registration string.

What i am looking to try and do is... send all incomming calls to an IVR and be able to make outgoing calls from any extension by dilaing "0" to get a free external line.

Can anyone help me get this sorted.

Useful Sources:

www.bbpglobal.com


Cheers

Matthew
 

rejil.rajan

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#2
Hi Mathew

Are you looking for someone to help you configure Asterisk with the Service provider or you want someone to help you configure the IVR and outgoing calls. The latter is simple when using elastix.
 

Gronkstar

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#3
i want help configuring Asterisk with the Service provider mainly so i can make in and out calls please email me mmatters@ausnetservers.net.au
 

rejil.rajan

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#4
Could you provide me with the screenshot configuring the Trunk and the result of sip show registry from Asterisk. Please could you also try configuring a softphone with this account and see if it is working.
 

Gronkstar

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#5
i have configured a softphone and a normal voip phone with these settings and they work. i have no idea how to configure the trunk settings its way to complex for me im only provided with a uysername sip port and sup address as well as the internal number some how i have to put that into the feilds and make it work but going by the pdf that i was reading " Elastrix without tears " it dosent work
 

rejil.rajan

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#6
Hi Matthew

It is very simple to configure in elastix. Try this configuration


User Settings:

Trunk Name: (Your Username)
username=XXXXXXXXX ( Your Username )
type=friend
secret=XXXXXXXXX ( Your Password )
qualify=yes
nat=no
insecure=very
host=sip2.bbpglobal.com
fromdomain=sip2.bbpglobal.com
dtmfmode=rfc2833
canreinvite=no

&

Peer Settings:
canreinvite=no
context=from-internal
fromuser=XXXXXXXXX ( Your username )
insecure=very
qualify=no
secret=XXXXXXXXX ( Your password )
type=user
username=XXXXXXXXX ( Covered For Security Purposes )


Register String

register=<Your username>:<Your Password>@sip2.bbpglobal.com/<Your Username>

After you do this, please login to the console of asterisk or from the web page use Asterisk CLI and provide the result of sip show registry
 

Gronkstar

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#7
when you talk about:
username=XXXXXXXXX ( Covered For Security Purposes )

are you referring to my carriers username or an internal extension? if its an internal extension i want every call to go to an IVR not to an extension

when it comes to the peer settings, im not sure if its the carriers user name and pass or the local servers extension details
 

Gronkstar

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#8
i am getting very confused, carnt i send you the details in a private message and the access to the server to so can setup the first part, im running out of time to get this out. sorry to be impatient
 

Patrick_elx

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#9
A trunk has nothing to do with extension.

You configure your trunk to go to and from your voip provider to your server.

What you do with inbound calls from these trunks is managed in inbound routes.
The way your extensions dial out to these trunk is managed in outbound routes.

Elastix without tears is really well written an explains these concepts in an easy way to understand.

Then what you put in the trunk settings is only only about your trunk. Your provider is completely ignorant regarding what your internal setup is and should stay that way.
 

Gronkstar

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#10
this is way to over my head..... this is a complex voip system i have delt with easy voip systems on windows. i have entered all the info into the trunk and click apply and applied the changes. now where do i go to see if its registered. Once thing i haven't done is the dial plans???
 

Patrick_elx

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#11

rejil.rajan

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#12
You can go into the console using an SSH client like putty.

Or go to PBX -> Tools -> Asterisk CLI and give the command sip show registry
 

Gronkstar

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#13
Host Username Refresh State Reg.Time
sip2.bbpglobal.com:5060 XXXXXXXXXX 105 Registered Thu, 09 Jul 2009 09:59:41

It seems to be registered now, how i do the incomming and outgoing trunks? so i can make and recive calls?
 

Patrick_elx

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#14
If you are register, then I believe that your trunk is setup. But as you are not providing us with more information it's difficult to answer you.


You need to create an inbound route to go to the extension or the ivr you want.

You need to create an outbound route to be able to call out.

Again if you had read Elastix without Tears everything is explained in chapters 8 and 9
 

Gronkstar

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#15
I have scanned the book it make no sence to me no point getting someone to read something they carnt understand a step by step guide would have been much better instead of the approach being taken. I have spent the last 2 hours trying to setup the inbound and outbound routes and probably made them worse from using what i can understand in the PDF u provided. If you cant help dont reply, or if you want to palm me off to a book i have tried reading and dont understand then dont reply.
 

rejil.rajan

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#16
Ok now what you have to do is configure the outbound route, say 9|XX. and select the trunk you configured.

Then based on the your plan prefix 9 before the dial plan and call the number you wish to call. The dial plan for BBP is

http://www.bbpglobal.com/global/dial.php
 

dicko

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#17
Gronkstar:

Then with due respect, Perhaps you should accept your limitations and go back to your simpler VOIP implementations under windowboxes.


Please don't get me wrong, this is not a criticism, Elastix/Asterisk can never be plug and play, it is way too sophisticated, It costs nothing to get it, it takes some effort to implement it however. "Elastix Without Tears" is IMHO a quintessential step by step script for guys like you, if you reject it after "scanning" it, then you just threw the baby out with the bathwater.

I fear you might risk possible disapproval here if your sense of "entitlement" enables you to castigate Patrick for his patent help and patient postings here, which he did for you and you alone. Maybe you get lucky and someone will do your homework for you but then again maybe not.

Perhaps a "time-out" would help as you give EWT and your brain another chance? (or $2500 to micro$oft)

dicko
 

Gronkstar

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#18
i have used the windows software and would continue to but my provider carnt seem to solve an issue with the windows pbx server software when you place someone on hold the on hold music dosent work nor does the transfer function work it tries to disconnect the call. I have read through chapters 8 and 9 of the tears booklet and it makes none of no sence to me. maybe i am overlooking what its trying to say im just really trying to get this back up and running ASAP. BBP Global really should have some configuration instructions to help in this situation. Is it possible to have a step by step guide with screen shots on how to configure the inbound and outbound routes, i have configured them from the booklet as i said but it dosent seem to be working ( more then likely i have configured it wrong )
 

dicko

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#19
I sense your pain,

Personally (and I suggest, with some logic) I would start at chapter 1.

What you get here is free, (and yes, you are welcome!)

If you remain paniced/confused then our gracious host Palosanto offer paid support.

http://store.palosanto.com/index.php/su ... d-8x5.html
 

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