Category Archives: VOIP


FOP2 with FreePBX overview

FOP2 makes a great addition to FreePBX, especially if you use your phone system in a sales environment. FOP2 is a web based panel for managing live calls on your PBX.

I’m going to write a few posts going over some of the features, and this post is designed to give you an overview of what FOP2 is and how it works.

My test phone system has 3 extensions linked to a SysAdminMan hosted FreePBX system –

3 x FreePBX extensions

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Insecure home routers when using SIP

This post looks at reports from last year which I must admit had passed me by. They show how using a SIP device with a vulnerable router could leave you seriously exposed to VOIP fraud calls.

The reports focus on the BT Home Hub 3, but now that I’ve read it’s possible with one router, I have concerns that others could be affected.

When you have a SIP phone at home (or in the office) this is what you would expect to happen –

  • phone connects to external Asterisk server or VOIP provider on port 5060
  • the firewall opens the required ports allowing the reply from the external Asterisk server or VOIP provider

What actually happens on the HH3 (at least the firmware in the reports, this could have been resolved in later firmware) is this –

  • phone connects to external Asterisk server or VOIP provider on port 5060
  • the firewall opens the required ports allowing ANY EXTERNAL IP to connect to the phone

The difference is fairly subtle, but the result is not. This means that while your phone (SIP device) is switched on and connected to a remote Asterisk server or call provider, any SIP scanning against your public IP will get forwarded to the phone.

If the dial plan on the phone allows calls to be placed, then those calls would be completed. This could result in expensive VOIP fraud calls.

What should you do?

I recommend that everyone run a SIP scan on their public IP (this is your home/office IP) to ensure that no SIP devices respond. If they do and you are a SysAdminMan customer then please open a support ticket to discuss this more.

You can run a SIP scan from this site –

If you find any affected routers that are not the BT HH3 please post a comment.

You can find more info here –

Thanks, Matt


Setting up a trunk to a SIP call provider in Asterisk can be a pain. Even for me – and I do it a lot!

When we set up a SIP call provider in Asterisk (a SIP trunk) and send calls to them our Asterisk server will send the call provider a SIP INVITE. Their system could respond in many different ways – decline the call, ask for user info, silently drop the call, process the call ….

Sometimes the call provider will send back information in the SIP reply as to why the call has failed. Maybe you have no credit, maybe the codec is not supported, maybe you have the number in the wrong format … but often they will not, they will just not process the call and send back a generic response.

At this point you need some help from the call provider. You need them to tell you why the call is not completing correctly.

Recently I was helping a customer set up a trunk to didlogic. His Asterisk server was sending the call to didlogic, but their system was responding with a fairly generic reply, indicating the SIP credentials were incorrect.

They seemed either unable or unwilling to investigate why this was. Even after being provided with SIP traces they showed no interest, just replying with generic suggestions that were no help.

It turned out that creating a new SIP account in the didlogic web portal resolved the issue. The old SIP credentials didn’t work, the new SIP credentials did.

I get asked a lot for call provider recommendations, and this is the only experience I have with didlogic, but based on it I couldn’t recommend them.

If you are having issues sending SIP calls to them (and getting just s SIP 407 back) maybe try creating a new set of SIP credentials in the didlogic portal.

Update 30/5/2014

I had a response to this post from DIDLOGIC which sounds encouraging. This was the reply –

Hello Matt,
Thanks very much for taking the time to detail these issues. It has been a couple of months and we have made our maximum effort to get a handle on the situation with authentication errors related to various flavors of Asterisk.
We apologize for causing your such inconvenience. Please be advised that there is now much more scope for troubleshooting these rare occurrences, as we have invested heavily in improving our support and customer service.

In case you ever run into such difficulty again, please send a short message to [email protected] and CC the engineer assigned to your account. All system builders and VoIP consultants integrating our SIP trunking solutions for their clients get a dedicated point of contact and such trivial issues are dealt with immediately.

Thanks again for trying our services.

Using a GoIP with A2Billing for outbound calls

This is a quick guide to using a GoIP GSM gateway with A2Billing as an outbound trunk. It doesn’t go through setting up rate cards and customers in A2Billing, guides for that are already available on this site.

First some general info and tips.

This guide was written using the following. The thing most likely to cause issues is a SIP ALG in your router. NAT issues are a pain to track down!

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New GoIP (GSM to SIP gateway)

For quite a while now I’ve had lots of enquires from customers wanting to use a GoIP with A2Billing on their SysAdminMan VPS. So I thought I’d order a test one and write some documentation!

A GoIP is a small box that accepts a full size GSM SIM. It can then take Asterisk/SIP calls and send them via a mobile phone provider, or pass incoming mobile calls to Asterisk.

I ordered from this e-bay seller on 23/11/13 – and the GoIP was delivered today, 10 days later.


It was delivered with a 2 pin (rather than UK) plug which is no great surprise. So you’ll need a power adapter for it. Alternatively you could use a different power adapter, which is 12v 500mA output.

A couple of things to consider if you are thinking about trying a GSM/SIP gateway –

  • You are likely breaking your mobile operators terms of service. If you abuse it be prepared to be disconnected!
  • You will not be able to pass a caller ID to the GSM network. The person being called will see either the SIM card caller ID, or no caller ID

Understanding latency and packet loss with mtr

If you have call quality issues with VOIP then one of the first things to check is if there is any packet loss or unexpected high latency on the network connections. A great tool for this is called ‘mtr’.

However, I often see some misunderstanding of the reports produced by mtr.

I was going to write a guide to mtr but there is a really good one already that you can find here –

Two of the most important things the post highlights can be summarised like this …

Packet loss

This packet loss is OK …

[email protected]:~# mtr --report
HOST: ducklington               Loss%   Snt   Last   Avg  Best  Wrst StDev
  1.                  0.0%    10    0.3   0.6   0.3   1.2   0.3
  2.                50.0%    10    0.4   1.0   0.4   6.1   1.8
  3.                0.0%    10    0.8   2.7   0.8  19.0   5.7
  4.        0.0%    10    6.7   6.8   6.7   6.9   0.1
  5.                  0.0%    10    7.2   8.3   7.1  16.4   2.9
  6.                0.0%    10   39.1  39.4  39.1  39.7   0.2
  7.                 0.0%    10   39.6  40.4  39.4  46.9   2.3
  8.          0.0%    10   39.6  40.5  39.5  46.7   2.2

This packet loss is BAD …

[email protected]:~# mtr --report
HOST: localhost                   Loss%   Snt   Last   Avg  Best  Wrst StDev
  1.                   0.0%    10    0.3   0.6   0.3   1.2   0.3
  2.                  0.0%    10    0.4   1.0   0.4   6.1   1.8
  3.                60.0%    10    0.8   2.7   0.8  19.0   5.7
  4.        60.0%    10    6.7   6.8   6.7   6.9   0.1
  5.                  50.0%    10    7.2   8.3   7.1  16.4   2.9
  6.                40.0%    10   39.1  39.4  39.1  39.7   0.2
  7.                 40.0%    10   39.6  40.4  39.4  46.9   2.3
  8.          40.0%    10   39.6  40.5  39.5  46.7   2.2


If you’re expecting a ping/latency of around 40ms this is OK (even with a 254ms latency along the route) …

[email protected]:~# mtr --report
HOST: localhost                   Loss%   Snt   Last   Avg  Best  Wrst StDev
  1.                  0.0%    10    0.3   0.6   0.3   1.2   0.3
  2.                 0.0%    10    0.4   1.0   0.4   6.1   1.8
  3.                0.0%    10    0.8   2.7   0.8  19.0   5.7
  4.        0.0%    10    6.7   6.8   6.7   6.9   0.1
  5.                  0.0%    10  254.2 250.3 230.1 263.4   2.9
  6.                0.0%    10   39.1  39.4  39.1  39.7   0.2
  7.                 0.0%    10   39.6  40.4  39.4  46.9   2.3
  8.          0.0%    10   39.6  40.5  39.5  46.7   2.2

If you’re expecting latency of a lot less than 400ms then this report is BAD …

[email protected]:~# mtr --report
HOST: localhost                   Loss%   Snt   Last   Avg  Best  Wrst StDev
  1.                  0.0%    10    0.3   0.6   0.3   1.2   0.3
  2.                 0.0%    10    0.4   1.0   0.4   6.1   1.8
  3.                0.0%    10    0.8   2.7   0.8  19.0   5.7
  4.        0.0%    10  388.0 360.4 342.1 396.7   0.2
  5.                  0.0%    10  390.6 360.4 342.1 396.7   0.2
  6.                0.0%    10  391.6 360.4 342.1 396.7   0.4
  7.                 0.0%    10  391.8 360.4 342.1 396.7   2.1
  8.          0.0%    10  392.0 360.4 342.1 396.7   1.2

Go read the guide to find out more and read a description of the traces above –

Using Android with FreePBX – CSipSimple extension

I’ve used a few different Android SIP clients as extensions on FreePBX and my current favourite is CSipSimple

Installation and setup is straight forward. There are several built in configuration profiles for call providers, or you can choose advanced and enter your FreePBX server details to use CSipSimple as a FreePBX extension.

CSipSimple account setupCSipSimple add accountCSipSimple Registered Account

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What to consider when chosing an ITSP

Firstly what is an ITSP? As described in our glossary, “ITSP:  Acronym for Internet Telephony Service Provider. This is a company that allows calls to and from ‘normal’ telephone numbers (PSTN)”.  Here’s a few things to think about.

Call plans – Plans maybe a cost for each call, bundled minutes, or plans for domestic / international usage. Providers offer the ability to phone free between customers in their community, is this something you’ll need?

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New or novice users my find this basic glossary useful when using their Asterisk VOIP, a few telephone industry acronyms

CDR: Call Data Records or Call Detail Records.  These are logs of calls that have passed through the phone system.

GUI: Usually pronounced “gooey”,  Acronym for Graphical User Interface, the graphical administration software used to manipulate the phone system

Inbound call: This is a call from a regular telephone number to your PBX. Also referred to as Origination

ITSP:  Acronym for Internet Telephony Service Provider. This is a company that allows calls to and from ‘normal’ telephone numbers (PSTN)

NGN: Acronym for Non Geographic Numbers, such as 0845 in the UK

Outbound call: This is a call from your VOIP PBX to a regular telephone number. Also referred to as Termination

PBX: This stands for Private Branch Exchange, it is an old fashioned term for a phone system

PSTN: Public Switched Telephone Network. This is a term used to described the ‘normal’ telephone network. Non VOIP landline and mobile/cell telephone numbers.

SIP: A communication protocol for phones, a language to make phones and phone systems, talk to each other

Trunk: A link to another phone system or call provider. For example you would have a ‘trunk’ to your call provider (ITSP). Your system would send VOIP calls down the trunk to your call provider, they would send the call to it’s destination (a landline or mobile)