Get help with installing, upgrading and running a PBX such as Asterisk.
User avatar
By newone
#4589
Hi everyone,

First let me say sorry for what's to come, usually I am a quick learner but as always I got lost in the sea of information, posts, tutorials and I forgot that you always have start from the basics. As a side project to my usual job, I have been tasked with implementing the VoIP solution at the company I work for. Now the truth is I don't know anything about what I am doing so I hope I can count on your support.

Now, for the first part I think my setup is as good as I can get it in a test environment. It's a virtual machine running FreePBX all updated, basic setup is done, I can call extensions, have a test sip trunk set up so I can receive and make calls, played a bit with the settings and configurations, all good until now. There are some things I can't find out by myself so please bear with me:
  • How do the modules correlate to each other? is there a certain sequence that must be followed? any hints where I can find some info? the bare minimum I got is trunk - IVR(group,etx.) - voicemail but the rest? the trunk I have has 3 phone numbers assigned to it, how do I set them up for different IVR's?
  • Do I follow the 3:1 rule (50 users 12-13 trunks)?
  • Concurrent calls? how many calls can I make or receive on one number? don't know anything about this.. any tips on reading material?
  • How do I setup call forwarding? I used Xphone and didn't see an option, but conference works.
  • What offers do the UK providers (good SLA,uptime,what else?) have and what should I look for (50 users)?
  • What should I look out for in my setup and what are the best practices to follow?
  • Mixture of hard and soft-phones any tips?
  • What about redundancy? Can i have 2 analog lines as backup to the sip trunk?
  • Should I think about buying the specialized hardware or is it ok if I run the setup in a VM (Type 1 Hypervizor) or bare metal. I guess intel I7, ssd and 16 gb ram will do for 20 recorded calls at the same time and all system requirements?
My setup should be like: 3 numbers each one IVR or queue-extension, voice recording and possibility to redirect the call to non recording extension (can the caller be sent back to the recorded call? how?). I know it's allot but at this point I need any advice I can so please can you help a poor lost sheep ? Even a drawing on a piece of paper will do. :dead:

Thank you and I hope that, in time, I'll bring my contribution to this forum to.
User avatar
By WelshPaul
#4591
Hello and welcome to the forum newone. I hope you don't mind, I have edited your original post so that it's easier to read and doesn't look like a wall of text. :)
newone wrote: Tue 16th May 2017, 23:12How do the modules correlate to each other? is there a certain sequence that must be followed? any hints where I can find some info? the bare minimum I got is trunk - IVR(group,etx.) - voicemail but the rest?
Modules just add additional features to FreePBX. You can add, remove and update modules by navigating to Admin > Module Admin within the FreePBX Administration web based GUI. Some modules may require another module be installed first, if it does, it will tell you and you won’t be able to install the module you want until you install the additional module it requires first. Only install the modules you want or need, no need to install them all.

To see what modules are available and install one, navigate to Admin > Module Admin within the FreePBX Administration web based GUI. Click the “Check Online” tab, here you will see a list of modules available for you to install. Simply click on the module that interests you and select “Download and Install” followed by clicking on the “Process” tab to install the module.

To check for module updates and install them, navigate to Admin > Module Admin within the FreePBX Administration web based GUI. Click the “Check Online” tab, put a tick in the box next to “Show only upgradeable” option. If there are any updates, you will see them listed here. Simply click on the “Upgrade All” tab to update your installed modules.
newone wrote: Tue 16th May 2017, 23:12the trunk I have has 3 phone numbers assigned to it, how do I set them up for different IVR's?
You would need to create some Inbound Routes.

First, navigate to Applications > IVR and create three IVR's. Once done, navigate to Connectivity > Inbound Routes and create three inbound routes, each one pointing to one of the three IVR's you set up.

Under the “DID Number” option located within the Inbound route configuration page, enter the DID number you want that Inbound Route to handle.

Of course, this assumes that your VoIP provider passes the DID through to FreePBX.
newone wrote: Tue 16th May 2017, 23:12Do I follow the 3:1 rule (50 users 12-13 trunks)?
No, if all your DID’s are supplied by the same VoIP provider, under the same account then you only need the one trunk.

If you have 50 users, you want 50 extensions.
newone wrote: Tue 16th May 2017, 23:12Concurrent calls? how many calls can I make or receive on one number? don't know anything about this.. any tips on reading material?
You need to ask your VoIP provider this question, some place a limit, others don’t. Voipfone will allow you to make and receive as many calls as your hardware and internet connection will allow. See: http://www.voipfone.co.uk/Simultaneous_Calls.php
newone wrote: Tue 16th May 2017, 23:12How do I setup call forwarding? I used Xphone and didn't see an option, but conference works.
You can set call forwarding at three different levels:
  1. VoIP provider level
  2. FreePBX level
  3. Hardware/Software level E.g VoIP telephone or softphone.
Usually you would only set the “PSTN Failover” option at the VoIP provider level. If your PBX was to go down for any reason, all inbound calls to any of your three DID’s would be diverted to the telephone number you entered under the “PSTN Failover” option within your VoIP providers control panel.

To set any call forwarding options at the FreePBX level, download and install the “Call Forward” module. Once installed, you can set call forwarding by dialling star codes followed by the number you want to divert the call to. E.g. - To divert all calls to telephone number 1234, dial *721234 from the phone/softphone you want the diversion set on.

List of FreePBX Call Divert star codes:
Screen Shot 2017-05-17 at 10.03.24.png
Screen Shot 2017-05-17 at 10.03.24.png (123.74 KiB) Viewed 226 times

To set call forwarding at a hardware/software level, well this differs from device to device and softphone to softphone. Watch this video to learn how to set call forwarding on a Snom 300:


newone wrote: Tue 16th May 2017, 23:12What offers do the UK providers (good SLA,uptime,what else?) have and what should I look for (50 users)?
Make sure your DID's are protected and that your VoIP provider will allow you to port your numbers out if you so choose.

https://www.itspa.org.uk/wp-content/upl ... P_v4.1.pdf
https://www.itspa.org.uk/wp-content/upl ... actise.pdf

The last thing you want is to be stuck with a VoIP provider that's unreliable and expensive because you can't move your number to another provider. Or worse, lose your business numbers because the VoIP provider goes bust!

All my DID's are registered with VoIP providers whom are registered members of ITSPA: https://www.itspa.org.uk
newone wrote: Tue 16th May 2017, 23:12What should I look out for in my setup and what are the best practices to follow?
Security, security and more security...

Don't just setup your PBX and leave it sitting there for months or worse, years. Keep it up do date and apply any updates and patches regularly.
newone wrote: Tue 16th May 2017, 23:12Mixture of hard and soft-phones any tips
Snom telephones are robust and reliable.

What platforms will you be using softphones on? Windows, OSX, Android etc.
newone wrote: Tue 16th May 2017, 23:12What about redundancy? Can i have 2 analog lines as backup to the sip trunk?
Simply put, yes. You can add a couple of OBi110's into the mix but be aware, analog lines have there limitations.
newone wrote: Tue 16th May 2017, 23:12Should I think about buying the specialized hardware or is it ok if I run the setup in a VM (Type 1 Hypervizor) or bare metal. I guess intel I7, ssd and 16 gb ram will do for 20 recorded calls at the same time and all system requirements?
No specialised hardware is required. Your I7, ssd and 16GB ram will do the job just fine!
newone wrote: Tue 16th May 2017, 23:12My setup should be like: 3 numbers each one IVR or queue-extension, voice recording and possibility to redirect the call to non recording extension (can the caller be sent back to the recorded call? how?). I know it's allot but at this point I need any advice I can so please can you help a poor lost sheep
Simply transfer the call to the extension or group that has call recording enabled. For the record, you can initiate call recording on an extension that isn't set to record all calls by pressing a button or two on your phone, you don't have to transfer a call to an extension that is set to record all calls.
User avatar
By newone
#4592
Thank you WelshPaul first for the post editing (you are a master!) Second for shedding light on all the things I didn't understand. Now :-D :
Hi WelshPaul and thanks for the fast reply! This is amazing ...
Of course, this assumes that your VoIP provider passes the DID through to FreePBX.
I think the provider (sipgate.co.uk) doesn't, I already tried setting the DID at the extension level, it creates an inbound route linked to that extension, when I call the assigned number it goes to the first IVR set on the original inbound route wich has no DID set (ANY). With DID set on both, none of them work anymore ... number not in service .. log said:
Code: Select all
[2017-05-17 14:55:11] VERBOSE[7393][C-0000001e] pbx.c: Executing [445601067xxx@from-trunk:1] Set("SIP/sipgate-00000018", "__FROM_DID=445601067xxx") in new stack
[2017-05-17 14:55:11] VERBOSE[7393][C-0000001e] pbx.c: Executing [445601067xxx@from-trunk:2] NoOp("SIP/sipgate-00000018", "Received an unknown call with DID set to 445601067xxx") in new stack
[2017-05-17 14:55:11] VERBOSE[7393][C-0000001e] pbx.c: Executing [445601067xxx@from-trunk:3] Goto("SIP/sipgate-00000018", "s,a2") in new stack
[2017-05-17 14:55:11] VERBOSE[7393][C-0000001e] pbx_builtins.c: Goto (from-trunk,s,2)
[2017-05-17 14:55:11] VERBOSE[7393][C-0000001e] pbx.c: Executing [s@from-trunk:2] Answer("SIP/sipgate-00000018", "") in new stack
[2017-05-17 14:55:12] VERBOSE[7393][C-0000001e] pbx.c: Executing [s@from-trunk:3] Log("SIP/sipgate-00000018", "WARNING,Friendly Scanner from 217.xx.xx.xxx;branch=z9ertw0eee.766asdfsafdsadfsadf5f54fc31b57ad.0") in new stack
[2017-05-17 14:55:12] WARNING[7393][C-0000001e] Ext. s: Friendly Scanner from 217.xx.x.151;branch=z9hGerteeee.76sadfasdfsdfac31b57ad.0
[2017-05-17 14:55:12] VERBOSE[7393][C-0000001e] pbx.c: Executing [s@from-trunk:4] Wait("SIP/sipgate-00000018", "2") in new stack
[2017-05-17 14:55:14] VERBOSE[7393][C-0000001e] pbx.c: Executing [s@from-trunk:5] Playback("SIP/sipgate-00000018", "ss-noservice") in new stack
[2017-05-17 14:55:14] VERBOSE[7393][C-0000001e] file.c: <SIP/sipgate-00000018> Playing 'ss-noservice.ulaw' (language 'en_GB')
[2017-05-17 14:55:15] VERBOSE[7393][C-0000001e] pbx.c: Executing [h@from-trunk:1] Macro("SIP/sipgate-00000018", "hangupcall,") in new stack
[2017-05-17 14:55:15] VERBOSE[7393][C-0000001e] pbx.c: Executing [s@macro-hangupcall:1] GotoIf("SIP/sipgate-00000018", "1?theend") in new stack
[2017-05-17 14:55:15] VERBOSE[7393][C-0000001e] pbx_builtins.c: Goto (macro-hangupcall,s,3)
[2017-05-17 14:55:15] VERBOSE[7393][C-0000001e] pbx.c: Executing [s@macro-hangupcall:3] ExecIf("SIP/sipgate-00000018", "0?Set(CDR(recordingfile)=)") in new stack
[2017-05-17 14:55:15] VERBOSE[7393][C-0000001e] pbx.c: Executing [s@macro-hangupcall:4] Hangup("SIP/sipgate-00000018", "") in new stack
So I guess it's a normal behaviour for my current setup. I will need to clear this up with the providers before I go into production mode.
If you have 50 users, you want 50 extensions.
I will have that and probably more, arranged into call groups and with different settings. This was the first thing I tried as I will have at least 3 call groups (and it worked :million: ).
Concurrent calls?
Thanks for that ... didn't know you can make as many calls as you like on just one line .
You can set call forwarding at three different levels:
I will do call frw at my PBX leve ... will try this in about 5 minutes ;-)
Make sure your DID's are protected
You are just brilliant ... Didn't know anything about ITSPA .. but will read all the documentation.
Security, security and more security...
This is the only thing I know 100% how to deal with.. but in terms of freepbx/asterisk (not centos/rhel) are there any special security tips ?
What platforms will you be using softphones on?
Windows workstations and in some cases extensions will ring on the android/ios mobile phone to. I am using Xphone for Windows and Zoiper for Android testing at the moment. But I am looking for the most user-friendly solution.
you don't have to transfer a call to an extension that is set to record all calls.
It's a trade requirement for compliance. This is the main reason why I prefer doing this by myself and not renting a cloud based PBX from the providers. Plus I like a good challenge once in a while and this is just what I needed (and I learn new stuff to which is awesome).

Thanks again Paul for everything and sorry for the messy posts but it's been a while since I've posted on forums
User avatar
By WelshPaul
#4593
Are you using Sipgate basic, Sipgate team or Sipgate trunking for testing?

If the DID isn't being passed then either register three different accounts, trunks or create three extensions at sipgate and and apply a DID to each account/trunk/extension. Once done, create a trunk for each account/trunk/extension and instead of entering a DID under the inbound route > DID Number, enter the SIP account user ID. I have to do it this way myself with Voipfone and Voicehost as they too don't pass the DID along (or at least they never used to, i haven't checked to see if it's changed).
newone wrote: Wed 17th May 2017, 15:59
Security, security and more security...
This is the only thing I know 100% how to deal with.. but in terms of freepbx/asterisk (not centos/rhel) are there any special security tips?
Sorry, I never advise on securing servers. I don't want to be held accountable for any hacked PBX's that have run up thousands of pounds worth of calls. :omg:
newone wrote: Wed 17th May 2017, 15:59
What platforms will you be using softphones on?
Windows workstations and in some cases extensions will ring on the android/ios mobile phone to. I am using Xphone for Windows and Zoiper for Android testing at the moment. But I am looking for the most user-friendly solution.
I use Zoiper personally although as to which is more user friendly is subjective.
newone wrote: Wed 17th May 2017, 15:59
you don't have to transfer a call to an extension that is set to record all calls.
It's a trade requirement for compliance. This is the main reason why I prefer doing this by myself and not renting a cloud based PBX from the providers. Plus I like a good challenge once in a while and this is just what I needed (and I learn new stuff to which is awesome).
Sorry what I should have wrote is that you don't have to transfer a call back to an extension that is set to record all calls by default because you can simply enable call recording on the extension that isn't configured to record calls by default by pressing a button on the phone or softphone.

Come to think about it, You can also set "Extension Recording Override" in General Settings. This overrides the individual extension "Record Always" setting. Doing it this may means that you can simply stop recording the call and start again without transferring the call anywhere to begin with or does your trade requirement for compliance not allow this?

EDIT: Oops, I clicked submit a little to prematurely so if you scratching your head thinking... Hmm I'm sure that wasn't there a minute ago it's because I edited my own post. :laugh:
User avatar
By newone
#4596
Are you using Sipgate basic, Sipgate team or Sipgate trunking for testing?
I am using the "free sipgate trunking trial" account. I can call any of the 3 numbers available and it works. I just didn't manage to make it work the way I want it to(trial and error got me so far). Basically my setup will look like this: Trunk provider - asterisk PBX(freepbx on LAN office network). I would like 3 numbers or more configured on the trunk one for each department, each with either IVR or queue or group and call recording on one of the numbers (attached to on group).
I will have a timed condition after 5 pm to redirect all calls to another country (maybe setup another pbx there) for after-hours support and same as here I will need to record all calls to that group (preferably on the UK pbx or I will make a script to backup on my servers).

On the security side I get it ... it's only as good as the user/admin... once I will have my production server up and running I will have time to tweak and hopefully make a "perfect" setup for my case.
User avatar
By WelshPaul
#4597
You do realise that Sipgate trunking can only handle 5 incoming calls at any one time? and that's if you subscribe to their plus package at £19.95 a month. You only get the first DID free, you will be paying an additional £1.95 each for the other two phone numbers. If you used Voipfone, you could have unlimited inbound and outbound calls (well as much as your hardware/broadband can handle) and your monthly charge could be as low as £6 plus VAT. There are no contracts, no hidden charges and they are a registered member of ITSPA too.

Use this link (https://www.voipfone.co.uk/signup.php?free=113560) to sign up and you can try out their services for free for 30 days.

You don't have to use them, just putting it out there as you asked about concurrent call limits in your original post.

In the meantime, I will take a look at Sipgate trunking and post back with more information on how to route those inbound calls...
User avatar
By newone
#4598
Yes... for compliance reasons I need to have a red phone extension where payments would be processed. sales/customer service (recorded) transfers the call to payment processing(red phone/not recording) and then call gets transferred back to cs/sales for followup/other reasons. I theory I know this can be done but I'm a newbie at voip planning and concept.
Another question would be if it's worth it to use specialized equipment like Cisco, Polycom, Avaya?
I only used sipgate because it was the first link on google for free test pbx trunk with uk numbers ... I will have to talk to the vendors before I can make a decision. I will choose the best offer and SLA because I'll need help from them and yes voipfone has some of the best reviews.
User avatar
By WelshPaul
#4599
newone wrote: Wed 17th May 2017, 18:19 Yes... for compliance reasons I need to have a red phone extension where payments would be processed. sales/customer service (recorded) transfers the call to payment processing(red phone/not recording) and then call gets transferred back to cs/sales for followup/other reasons. I theory I know this can be done but I'm a newbie at voip planning and concept.
Then I would simply recommend transferring the call from extension A (sales/customer service (recorded)) to extension B (red phone extension) and once payment has been processed transfer the call back to extension A.
newone wrote: Wed 17th May 2017, 18:19 Another question would be if it's worth it to use specialized equipment like Cisco, Polycom, Avaya?
I only used sipgate because it was the first link on google for free test pbx trunk with uk numbers ... I will have to talk to the vendors before I can make a decision. I will choose the best offer and SLA because I'll need help from them and yes voipfone has some of the best reviews.
To be honest, what you require is pretty simple and straight forward.

Three DID's pointing at three different IVR menu's with a total of 50 Extensions.

Create some groups, Group 600 for example can be used for sales/customer service (recorded), add some of the 50 extensions you created to it. If you have 20 or so customer service reps then add 20 or so extensions (e.g extensions 200 - 220) to group 600. Group 601 could be technical support, group 602 payments etc...

Trunk with DID 1 > Inbound Route > IVR Menu 1 > Option 1 (sales/customer service (recorded)) > Group 600 > All extensions in this group ring and whoever is free answers the call. All calls will be recorded!

Trunk with DID 1 > Inbound Route > IVR Menu 1 > Option 2 (technical support) > Group 601 > All extensions in this group ring as above.

Trunk with DID 1 > Inbound Route > IVR Menu 1 > Option 3 (payments (not recorded)) > Group 602 > All extensions in this group ring as above. Whoever is free answers the call and takes payment, calls are not recorded.

Customer service can either transfer a call to group 602 which will ring all extensions in this group that are free or they can transfer the call directly to one of the extensions within this group by entering that extension number and payment can be taken. Once complete, transfer the call back to customer service department by transferring the call back to group 600 or they can transfer it back to the original customer service agent by entering their extension number. E.g extension 200

I have to wait until Sipgate approve my trial before I can test this further with their service.
User avatar
By WelshPaul
#4605
Well I still haven't had my Sipgate trunking account setup, maybe because I used an outlook email address to register? :laugh:

Anyway, make sure you setup and configure your sipgate trunk as per this guide: https://teamhelp.sipgate.co.uk/hc/en-gb ... -trunking-

It would be very odd for Sipgate to offer multiple DID's on a single trunk with no way for a PBX to identify on which DID the call originated in order to route them. What number format did you enter into the “DID Number” of the inbound route? If you haven't done so already, try entering it in the E.164 format - E.g. 442031234567. If this doesn't work for you, drop Sipgate an email and explain to them that you're trying to achieve the following but it doesn't appear to work:
  • DID 1 > Sipgate Trunk > Inbound Route 1 > Extension 200
  • DID 2 > Sipgate Trunk > Inbound Route 2 > Extension 201
  • DID 3 > Sipgate Trunk > Inbound Route 3 > Extension 202
User avatar
By newone
#4607
WelshPaul wrote:Well I still haven't had my Sipgate trunking account setup, maybe because I used an outlook email address to register? :laugh:
I know ... it takes a while to activate the account. I managed to finish my PoC, colleagues are happy with the "setup" and now it's time to move to the service providers and main connections :-/ .

How does this work ? I haven't researched anything similar until now. I was under the false impression that everything will go trough our internet connection. But now I need a dedicated 24 channel T1 and an active stand-by line.
For 99.9% QoS (SLA guaranteed) 24 channel T1 line which providers are best in the UK?
What about a backup solution or active stand-by?
I guess my setup will look something like this: voice provider/ISP - Firewall - Switch - Sangoma200g - Phones.
Is a Sangoma 200g gateway fit for a job like this? ( the reason I chose sangoma/freepbx is that it has the crm integration module and I don't need to have the plugin developed by a third party or the crm team)

I am so confused right now ... when I accepted the challenge I thought I have the big picture but now I can see that this is much more complicated than I initially thought. Can you please help me understand how the connections from the T1 line to my phones work ? In my head it looked like this : ISP - Firewall - Switch - Sangoma - Phones
User avatar
By WelshPaul
#4608
newone wrote: Fri 19th May 2017, 10:18 How does this work ? I haven't researched anything similar until now. I was under the false impression that everything will go trough our internet connection.
Well you can use your existing internet connection (if it's fast enough), I don't recommend it though!
Get a separate fibre broadband connection and use that purely for VoIP. ;-)
newone wrote: Fri 19th May 2017, 10:18But now I need a dedicated 24 channel T1 and an active stand-by line.
No you don't need a dedicated 24 channel T1 at all, who told you that?

A fibre broadband connection with the following speeds :
  • Estimated Download Speed: 32.5 - 44.7 Mbps
  • Estimated Upload Speed: 6.1 - 8.9 Mbps
Will be capable of:
  • Estimated Simultaneous SD Calls: 222
  • Estimated Simultaneous HD Calls: 89
I believe Virgin Media Business broadband is capable of delivering speeds of up to 350MB and no telephone line is required.
Virgin Media's Service Level Agreement (SLA): https://www.virginmediabusiness.co.uk/p ... nd-SLA.pdf

If Virgin Media isn't available in your area, get a second phone line installed and have a dedicated VoIP fibre broadband connection put on that line. Broadband speeds can be pretty dire over an BT Openreach line so make sure you check what speeds that line will be capable before committing to anything. Even better, if fibre to the premises is available in your area, get that!
newone wrote: Fri 19th May 2017, 10:18For 99.9% QoS (SLA guaranteed) 24 channel T1 line which providers are best in the UK?
Again, this is subjective. Anyway, like I stated above, you don't need a T1.
Sign up to a business fibre broadband package that offers a suitable (SLA).
newone wrote: Fri 19th May 2017, 10:18What about a backup solution or active stand-by?
You could always have 2x business fibre broadband connections for use with your VoIP service and use a dual wan capable router so that if broadband A goes down, broadband B takes over.
newone wrote: Fri 19th May 2017, 10:18Is a Sangoma 200g gateway fit for a job like this? ( the reason I chose sangoma/freepbx is that it has the crm integration module and I don't need to have the plugin developed by a third party or the crm team)
Again, this is subjective. I don't have any experience using Sangoma hardware myself so I am unable to advise.
Having looked at the spec sheet, the first thing I noticed was this:
  • Fixed configuration of 60 VoIP calls
Do you plan on exceeding this limit?
newone wrote: Fri 19th May 2017, 10:18Can you please help me understand how the connections from the T1 line to my phones work? In my head it looked like this : ISP - Firewall - Switch - Sangoma - Phones
I have zero experience with T1 lines but yea, ISP - Firewall - Switch - Sangoma - Phones looks right to me too.
System advice?

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