Hi, I have a series of setup for conference purpose:
configured a conference context with incall type with start and end numbers
setup a conference from Conferences menu
create an incall from Incalls menu with conference context and conference destination
From Lines menu, create a user with conference context
Create a Trunk with conference context and user type.
Now how can I test this configuration works?
I want to be able to allow anybody to mobile or land line phone to call in from a public number and allow them to have a conference together. by the way where to define the public phone number?
Thanks for help in advance.
configured a conference context with incall type with start and end numbers
You do not configure a Conference Context.
You add the Conference Range of numbers to the Internal Context
Then you create the Conference number (within the range defined above)
(which you can test by dialing the number from any internal device)
Then you setup a Trunk from an outside line with a DID
Then you direct an InCall route for the DID to the Conference
From Trunk menu, when I created a trunk, there is no field for DID, the current fields include: Name, Username, Password, Host, Type, Context for Trunk tab, for Register tab, only one checkbox: enable, for Options tab, only three option key value pairs: call-limit, amaflags, and subscribemwi, so have I missed anything here?
You first have to subscribe to a trunk provider and get a DID with a username (usually DID but not always) a Password (sometimes called secret) and a server address to which you will point your PBX.
The way you point your PBX to the server with your Trunks setup is as follows:
When you add a Trunk, you will see three tabs: Trunk, Register, Options
In the Trunk tab, you enter the information that lets your system call out
In the Register tab, you enter the information that lets your system accept incoming calls
In the Options tab, you enter any specific configuration options that your Trunk Provider says they need for connection to their system. These Options are in addition to the SIP (assuming you are using SIP connection) Options set in Advnaced -> SIP General Settings.
In the Trunk tab, set the Host Dropdown value to Static and you will then be given a field into which you put the trunk provider’s server address.
In the Trunk field, the trunk provider will tell you what the Type is but it is usually Peer for a Trunk.
In the Register tab, you will not see any entry fields until you Check the Enable checkbox, then the registration entries will become visible.
Transport is usually UDP.
In both the Trunk and Register tabs, make sure to use the trunk provider’s server address in the Host fields and the Trunk provider’s UserName (usually DID) and Password (sometimes called secret) in their respective fields. You may also need to enter your DID in the Callback Extension in the Register tab. Leave the SIP UserName blank in the Register tab.
I have a context of Internal type and with Conference range of start 2000 to end 2999
I created a Conference1 from Conferences menu, set extension 2001, and selected the conference label from the dropdown menu, and set PIN and admin PIN.
created a trunk with type User and selected the conference context of internal type and call-limit of 1000. there is no DID field in this step.
Created an Incall from Incalls menu, here I can only select an Incall type of context I created before (this context range: start 1000 - end 9999), also select a number of 00000000000000001502 (maybe I defined DID length 20) from the list of range from 00000000000000001001-00000000000000001999, select Conference as destination, and select the conference created earlier as Conference.
so I defined the start and end with 4 digits, should I change it to 10 digits like a mobile phone number, what about international phone number? does that mean the number I selected 00000000000000001502 is an outside public number? also what are the internal device?
Ok, thanks, I might have some beginner questions, first of all, how to subscribe to a trunk provider? do you mean some phone companies like Verizon? also what is the remote host and remote port? are they referring to the machine where the wazo server is installed? I thought by using this wazo software, if I configure right, anybody can call anybody else via this software on the internet, it seems like I need to subscribe something in order to make it work.
Sorry, I am brand new to this system and thanks for your patience.
Verizon is one provider of Trunks in the USA. Vitelity is one I use for US Trunks.
In Canada, I use Unlimitel/Primus.
The Trunk is what connects you to the Internet. Once connected, yes, you can call anyone anywhere.
So yes, the Remote Host is the Trunk Provider. You connect your PBX (server) to them via the Trunk setup.
Until you have a Trunk connected, the only people who can talk to each other are people who have a direct connection to your PBX. So if you have IP Phones or Softphones, you can connect them to your PBX and they can talk amongst themselves.
ok, much clearer, so I have verizon as my phone provider, the Remote Host will be the verizon’s ip address, is the trunk provider like verzion free of charge if I already have their service as my internet and phone provider? when you say IP phones, do you mean an internet enabled mobile phone?
I do not know what Verizon will charge, but I doubt very much it will be free.
From Verizon’s perspective (or any other provider of Trunks), they are providing you a legitimate, dialable phone number and the bandwidth that is required to support it.
There may be a one-time charge to set it up.
There likely will be a monthly charge, maybe two. Some companies charge a fixed fee per month plus usage and some (Vitelity) have a very small annual fee and then charge a usage rate (which in Vitelity’s case is something like $0.02 per minute of use, including long distance in the USA & Canada (and to other places) and a higher usage fee to some other places.
Note that if you do go with Vitelity, you will get one phone number but they like you to connect two seperate Trunks, one for incoming and one for outgoing calls (setup like above) and they allow 2 active calls per Trunk.
You could look at Primus/Unlimitel as well. They have DIDs for the USA.
There are a number of providers. Check them out and see which one looks best to you.
If you want to use your mobile phone, you would need to put a softphone on it. The Wazo team like Zoiper so take a look at that.
You can also use softphones on PCs and Tablets, assuming the software is available for those devices. Zoiper is.
Lots of other choices depending on the device on which you want to put the software.
As for your other question re internet phones, no, I meant a phone that is designed and manufactured to be used for working over a TCP/IP network (usually with Ethernet connection).
I use Aastra, which were bought by Mitel.
Others use a variety of types, but if you are going to get an Internet Phone (they are not cheap but they do make life easier and add functionality) make sure you get one that Wazo supports. Take a look at the brands and model numbers for which Wazo has plugins at Provisioning -> Plugins. They ahve Cisco, Digium, Grandstream Aastra, …
You could also get an ATA to convert your existing Analog phones to work over an Ethernet network. You lose some functionality that way but if you have a household full of Analog phones, it may be a way to dip your toe into this new world. It may be easier to just start with softphones.
Note that if you do use a softphone, which has no Mac Address and is not a “Device” you must use an all-numeric name. I am not sure why this is but it is what I had to do to get a softphone to work.
I suppose if you got into the guts of the system and had a private network you could change everything to TCP/IP, but you could not use that to communicate with others since they would be setup for UDP
UDP is a send-and-forget protocol. If what you send does not get through as intended, too bad, move on.
TCP/IP is an error-check protocol so for every transmission, there is a check at the other end to ensure that what was received is what was sent. If not, re-send.
For data transmission, TCP/IP is required since missing even a few bits is a problem.
For audio transmission, missing a few bits is not an issue - the human ear usually could not tell anyways and even if it did, worst case, the receipient says “Can you say that again?”. To use an error-check protocol on a real-time transmission like audio would impose unneeded and cpu-intensive load so why bother?
That is why the standard protocol for Voice Over IP is UDP.