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Asterisk Tutorial 52 – Asterisk CDR Part 2

Introducing Asterisk CDR Part 2

Welcome to the part two on Introducing Asterisk CDR (Call Detail Records). Following on from last time, when we had a brief introduction to Call Detail Records, it is time to get more in-depth.

Understanding Asterisk CDR Record Field Mapping

At first glance, Call Detail Records can appear can daunting as the recorded details can overwhelm you with information.

For example, take a look at the screenshot below as a newbie and you are not really going to be able to make head nor tail of it.

Thankfully, CDR records are not as complicated as they first seem. Look again at the image above and you will notice some information such as both mine and Mathias’ name. You will also notice some time stamps. However, things get a touch more complex when deciphering the rest of the dataset.

Thankfully, Asterisk themselves have a very good wiki article explaining what’s what. When you compare the list below (taken from the Asterisk Wiki) with the image above, you can start to gain a more comprehensive understanding of Asterisk CDR field variables.

  • ${CDR(clid)} Caller ID
  • ${CDR(src)} Source
  • ${CDR(dst)} Destination
  • ${CDR(dcontext)} Destination context
  • ${CDR(channel)} Channel name
  • ${CDR(dstchannel)} Destination channel
  • ${CDR(lastapp)} Last app executed
  • ${CDR(lastdata)} Last app’s arguments
  • ${CDR(start)} Time the call started.
  • ${CDR(answer)} Time the call was answered.
  • ${CDR(end)} Time the call ended.
  • ${CDR(duration)} Duration of the call.
  • ${CDR(billsec)} Duration of the call once it was answered.
  • ${CDR(disposition)} ANSWERED, NO ANSWER, BUSY
  • ${CDR(amaflags)} DOCUMENTATION, BILL, IGNORE etc
  • ${CDR(accountcode)} The channel’s account code.
  • ${CDR(uniqueid)} The channel’s unique id.
  • ${CDR(userfield)} The channels uses specified field.

Adding Custom Fields to Asterisk CDR Records

If you wish to add a specific label under certain circumstances to your CDR records, you can do so by modifying your dialplan in the extensions.conf and modifying the syntax of the desired extension using the following syntax:

same => n, Set(CDR(userfield)=custom field)

Which will then appear similar to below:

Once your extensions.conf changes have been saved and you have reloaded your dialplan the new syntax will be active and will appear in the Call Detail Records as shown below:

While the topic of Asterisk CDR records can go into much more detail, it is often only required highly bespoke Business Models. Therefore that is it for our Asterisk CDR records.

Join us next time when we will be back with more on Asterisk.

More Info

pascom are the developers of the software-based mobydick VoIP phone system for business. mobydick offers businesses with a viable alternative to proprietary PBX solutions. mobydick combines all the flexibility of Asterisk in one easy to install, manage and use solution.

Packed full of Unified Communication tools and PBX functions, mobydick is the fully featured Open Standards phone system to meet today’s communication needs.

For more on our mobydick phone system and to arrange a free personalised demo, give us a call on +49 991 29691 200 / +44 203 1379 964 or contact us via our website. Alternatively, take mobydick for a test spin with our free community download and find out how your business can benefit.

Until next time – Happy VoIPing!

Previous Tutorial

Source: http://blog.pascom.net/voip-guys/asterisk-tutorial-52-asterisk-cdr-part-2/

Publish Date: June 15, 2016 5:00 AM


mobydick Enters Canadian VoIP Market with Smarttel Communications

pascom Announce Smarttel Communications mobydick Reseller Partnership

[Deggendorf, Germany & Abbotsford, Canada| 10th June 2016] pascom Netzwerktechnik GmbH, developer of the software-based mobydick VoIP phone system for business, announced Canadian telecommunications solution provider Smarttel Communications as an approved mobydick Reseller.

Based in Abbotsforf, BC Canada, Smarttel Communications deliver customer orientated Business Communications solutions and are experienced Asterisk solutions providers. Thanks to adding the mobydick VoIP phone system to their solutions portfolio, SmartTel Communications will now be able to provide their customers with an easy to manage and maintain hybrid IP PBX solution as well as providing them with a fully functional Unified Communications solutions thanks to the newly developed mobydick Cloud Stack.

Mathias Pasquay, CEO and founder pascom Netzwerktechnik GmbH, on announcing the new partnership.

“We are delighted to welcome Smarttel to the mobydick partner family. Not only are they experienced in delivering Asterisk PBX solutions, but also carry a complimentary product range including approved SIP trunking solutions which will serve to ensure that our mutual customers are able benefit from a top of the range Unified Communications solution and excellent customer service. Furthermore, that Smarttel are also joining our new Cloud Solutions Partnership scheme, they will be well positioned to offer a complete package across the British Columbia area and further afield in Canada.”

About Smarttel Communications

Smarttel Communications have been active in the IT & telecoms industries for nearly 10 years. Initially starting out as network consultants before expanding their offering to include cost effective Business Communication solutions based on Asterisk, the worlds leading open source IP telephony software. Currently serving customers in the Vancouver and Lower Mainland Area, Smarttel focus on providing their customers with excellent customer service with a local touch.

For more information about Smarttel, please visit smarttel.ca

About pascom – communication without borders

Founded in 1997, pascom are the developers of the next-generation mobydick phone system software and are at the forefront of revolutionising how businesses communicate. mobydick provides businesses with a robust Unified Communications solution to face up to today’s business communication challenges. Through facilitating and enhancing collaboration and increasing mobility, business are able to realise dramatic productivity increases whilst benefiting from unparalleled scalability and cost efficiencies.

For more information about pascom and mobydick, please visit www.pascom.net

Source: http://blog.pascom.net/pascom-news/smarttel-mobydick-reseller/

Publish Date: June 14, 2016 5:00 AM


Asterisk Tutorial 51 – Asterisk Call Detail Records Part 1

Introducing Asterisk and Asterisk Call Detail Records

Following on from last week’s Asterisk Logger tutorial, we thought it would be a good idea to take a look at Asterisk Call Detail Records, more commonly known as CDR records. In part one of our introduction to Call Detail Records, Mathias provides us with a quick glimpse into what CDR records are, why they are useful and where and how to start with configuring your Asterisk CDR.

What Are Call Detail Records?

On the surface, CDR records do not seem to overly complex or sometimes even necessary. However, for some scenarios CDR is simply a must particularly when it comes to account billing for example.

In a nutshell Call Detail Records provide phone system administrators with access to historical data sets such as call time and date, duration and number. This information can then be used in order to develop a selection of phone system functions.

Call Detail Records Use Cases

Firstly, the CDR data set can be used to create a call journal for phone system users. Such journals can contain information such as caller ID, call destination, call time and date as well as call duration as well as whether the call was answered or not.

The second and more complex purpose of Call Detail Records is for call billing. Such use case scenarios can include inbound call centers, legal practices and any other environment where call billing is essential. This is where is gets complicated as we start talking about billing seconds, when to start billing, calculating talk time and so on.

Configuring Asterisk Call Detail Records

The simplest starting point is to start with the cdr.conf and write your CDR datasets to a .csv file. To access the cdr,conf, enter your Asterisk console as root and use the command:

/etc/asterisk# vi cdr.conf

As always, this will open the configuration file and docmentation which can be used as a reference point.

Once you have made your required configurations, the next step is to review the csv file. This can be done using the command:

/etc/asterisk# tail -f /var/log/asterisk/cdr-csv/Master.csv

It is important to understand that per default, Asterisk rights the CDR line after the hangup in the dialplan. This of course can be modified to force the CDR line to be added as soon as the call is answered.

That’s it for our introduction to Call Detail Records, next time around, we will get in to a bit more detail regarding Asterisk CDR with some Tips & Tricks.

More Info

pascom are the developers of the software-based mobydick VoIP phone system for business. mobydick offers businesses with a viable alternative to proprietary PBX solutions. mobydick combines all the flexibility of Asterisk in one easy to install, manage and use solution.

Packed full of Unified Communication tools and PBX functions, mobydick is the fully featured Open Standards phone system to meet today’s communication needs.

For more on our mobydick phone system and to arrange a free personalised demo, give us a call on +49 991 29691 200 / +44 203 1379 964 or contact us via our website. Alternatively, take mobydick for a test spin with our free community download and find out how your business can benefit.

Until next time – Happy VoIPing!

Previous Tutorial

Source: http://blog.pascom.net/voip-guys/asterisk-tutorial-51-asterisk-call-detail-records-introduction/

Publish Date: June 8, 2016 5:00 AM


pascom Welcomes mobydick Reseller SoftBCom

pascom Announce New mobydick Reseller SoftBCom

[Deggendorf & Berlin Germany | 06th June 2016] pascom Netzwerktechnik GmbH, developer of the software-based mobydick VoIP phone system for business, announced SoftBCom Berlin GmbH as an approved mobydick Reseller.

 

New mobydick Reseller SoftBCom, Berlin based Systems Integrator

Based in Berlin, SoftBCom adopt an added value approach to customer care, delivering bespoke customer orientated, enterprise grade solutions at optimum pricing. Through SoftBCom adding the mobydick phone system to their offering, their customers are now able to benefit from a complete Unified Communications platform that perfectly compliments SoftBCom’s exisiting telecommunications and Call Center product range. As a result, SoftBCom’s customers are able to enjoy not only additional productivity, mobility and cost saving benefits but also benefit from seamless integration into their existing infrastructures.

Vladimir K. Dudchenko, CEO & Co-Founder SoftBCom Berlin GmbH, said of the new agreement. 

“We have tested pascom’s mobydick product set, and were very impressed by its exquisite functionality. Moreover, the fact that mobydick’s development direction fits very well with our clients’ demands, we are delighted to become a reseller of mobydick. We are sure that mobydick will become an important and core component of our integrated solution offerings.”

Mathias Pasquay, CEO and founder pascom Netzwerktechnik GmbH, on announcing the new partnership.

“Having SoftBCom on board is great news for pascom. Not only do they carry a perfect compliment of Telecommunication and Call Center solutions, their customer orientated approach and focus on providing a complete solution will ensure the highest levels of customer care. Furthermore, thanks to SoftBCom’s international customer base, our partnership will help further increase our presence in the global market.”

About SoftBCom Berlin

SoftBCom Berlin is an IT service company specialising in communication system integration, focusing in particular on the delivery of contact center solutions ranging from 2 – 5000 Agents. SoftBCom offers a complete service package including the implementation, fine tuning and ongoing maintenance of highly complex solutions. Furthemore, SoftBCom provide additional integration services in ensuring interconnectivity between various business critical applications.

For more information about SoftBCom Berlin, please visit www.softbcom.de

About pascom – communication without borders

Founded in 1997, pascom are the developers of the next-generation mobydick phone system software and are at the forefront of revolutionising how businesses communicate. mobydick provides businesses with a robust Unified Communications solution to face up to today’s business communication challenges. Through facilitating and enhancing collaboration and increasing mobility, business are able to realise dramatic productivity increases whilst benefiting from unparalleled scalability and cost efficiencies.

For more information about pascom and mobydick, please visit www.pascom.net

Source: http://blog.pascom.net/pascom-news/pascom-mobydick-reseller-softbcom/

Publish Date: June 6, 2016 5:00 AM


Asterisk Tutorial 50 – Asterisk Logger

Introducing Asterisk and The Asterisk Logger

Welcome to the 50th episode of Introducing Asterisk, the step by step guide to Asterisk phone systems, and to celebrate we’re kicking off a new topic by taking a look at the Asterisk Logger.

What is The Asterisk Logger

In the past we have used Wireshark for debugging our SIP protocols. However, not everything speaks SIP – what then? Until now, we have always used the Asterisk CLI when testing and debugging our system. By which I mean we made some changes, set the verbosity level to 3 (asterisk -rvvv), reloaded the dialplan, made a call and then analysed the CLI output.

That’s fine for testing purposes when first setting up your Asterisk phone system. But good luck using this method in a productive system – you will need the reading skills of superman. Moreover, the question of how to debug your dialplan or historical data still remains? And this where the Asterisk Logger (logger.conf) comes into play.

In a nutshell, the Asterisk Logger is a logging mechanism which can write logs to files or to the syslog system (i.e. the Asterisk Console). In order to configure the Asterisk Logger, you will need to access logger.conf as the root asterisk user using the following command:

vi etc/asterisk/logger.conf Where vi equals your choosen Linux Text Editor.

Using The Asterisk Logger

Scroll down to the [Logfiles] section within the logger.conf configurations file. Under Logfiles, you will find a number of options (or levels as they are referred to by Asterisk) which can be included in the log file, as shown below:

These options are not all necessary, but it is good to know they exist should you need more information in your log files. Pay particular attention to the verbose option as it also provides you with the option to set the verbosity level.

It is also worth noting the syntax format is also explained in the above screenshot. When configuring your Asterisk Logger, you will need to use the following format:

filename => level options (i.e. notice,warning,error)

This is illustrated clearly by the screenshot below which was captured further on in the logger.conf configurations file.

It is also important to note that the grey lines are actually your configurable options which can be modified to match your needs, as highlighted below:

The filename "console" relates to the Asterisk console and the filename "messages" relates to an actual log file. Note that if the settings were left as above, the verbosity level would be set to whatever default level you have configured in the Asterisk Startup configurations. Therefore, for debug purposes we would suggest setting the verbosity level for the "messages" log file as explained below:

messages => notice,warning,error,verbose(3)

Our recommendation would to set a verbosity level of 3 as this will ensure that dialplan elements will also be logged and can therefore be analysed later when debugging historical issues.

Save your changes and that’s it you’re down – well almost. Don’t forget to log back into the Asterisk CLI and reload the Asterisk Logger using the command below:

logger reload

Now you are done! However, in the video above we also explain a few additional options such as Logger Rotate and so on – so worth watching it in full!

Mathias Top Tip

First up, when configuring the Asterisk Logger, we wouldn’t recommend adding a fixed verbosity level to the “console” log file. Mainly as this can always be changed using the verbosity switches as we have done in every tutorial before now.

Next up, when you are first configuring your system it could be a good idea to set the default verbosity level in the Asterisk startup configurations. However, one setup and you have ironed out the crinkles, then in order to save hard disk storage space, we would recommend setting it back to 0 or the equivalent of -r.

In order to do this, you will need to modify the verbose level in the etc/init.d/asterisk.conf to match you desired levels of verbosity.

More Info

pascom are the developers of the software-based mobydick VoIP phone system for business. mobydick offers businesses with a viable alternative to proprietary PBX solutions. mobydick combines all the flexibility of Asterisk in one easy to install, manage and use solution.

Packed full of Unified Communication tools and PBX functions, mobydick is the fully featured Open Standards phone system to meet today’s communication needs.

For more on our mobydick phone system and to arrange a free personalised demo, give us a call on +49 991 29691 200 / +44 203 1379 964 or drop us a line via our website. Alternatively, take mobydick for a test spin with our free community download and find out how it can benefit your business communications.

Until next time – Happy VoIPing!

Previous Tutorial

Source: http://blog.pascom.net/voip-guys/asterisk-tutorial-50-asterisk-logger/

Publish Date: June 1, 2016 5:00 AM


Asterisk Tutorial 49 – NAT & NAT Tables Explained

Introducing Asterisk and NAT Tables

Building on from last week’s Introduction to Network Address Translation (NAT), it is time to take a look at NAT Tables, NAT & SIP and how we can ensure that our NAT table does not forget our SIP connection account details.

How NAT Affects SIP Connections

Before moving forward, it may be a good idea to refresh your SIP knowledge in order to understand how SIP Registers work.

That said, what you really need to know and understand about NAT is that without it we wouldn’t be able to make and receive calls. That is why having a good understanding of NAT is essential when configuring your Asterisk phone system and registering your SIP providers.

The good news is that most Providers use NAT themselves to ensure that packet routing is done correctly. Meaning, even if the information in the SIP header contains your private IP address, providers generally use NAT to make sure that they send the packets to the right place.

Source NAT vs Destination NAT

As mentioned last time, there are two distinct types of NAT; Source NAT and Destination NAT. This begs the question – which type of NAT do we need, Source NAT, Destination NAT or do we need them both?

For starters, you will definitely require Source NAT. Without it, you will not be able to establish an internet connection as you local IP address will not be routed and your SIP registration will fail. That makes Source NAT an essential element of our telephony platform because as we know an internet connection is rather essential for VoIP telephony.

What about Destination NAT? By using source NAT to conduct their SIP registry an entry in the NAT table is then created. Some people think that as the entry only stays in the table for a short period of time, they need destination NAT in order to allow your SIP provider to route incoming telephony to their phone system. While this may the case in some cases, most providers do things a bit differently.

Destination NAT is in principal a sound methodology. However, it does have its drawbacks when working with SIP providers. If you use Destination NAT, there are risks involved as you will need to open your SIP ports and forward them to your internal ports. In itself, this could be okay if you only accept calls from the carriers IP address or host name.

But what about larger providers, they may have multiple IP addresses or constantly changing host names as a result of load balancing – what then? Some people are then tempted to open their ports to all incoming SIP requests and this is very dangerous as it will open your system to brut force attacks – so don’t do it.

Mathias Top Tip

If you have to use Destination NAT for whatever reason, then please restrict to the IP range of your provider as this will ensure the incoming SIP requests from your provider will be accepted whilst rejecting other perhaps malicious requests.

Thankfully most providers use an alternative method of keeping your NAT table open to ensure continued service and that is by forcing the NAT table to remember the SIP account credentials. In other words, always have an entry in the NAT table that contains the relevant information.

Once you have registered your Asterisk phone system to your SIP provider, most carriers are able to keep the NAT table entry from expiring by sending a so called “ping” request every 30 seconds or so, to which your system will simply respond with a 404 answer.

As the entry expiry time in the NAT table is about 1 minute, by sending a request every 30 seconds, carriers are able to ensure the SIP credentials are stored in the NAT table, thus allowing both incoming and outbound telephony without needing Destination NAT or port forwarding.

This can be seen when using set sip debug in your Asterisk CLI as every so often a new request will come in and be displayed on the CLI output.

More Info

pascom are the developers of the mobydick phone system that businesses love. Based on Asterisk, mobydick provides businesses a flexible, fully featured Open Standards phone system to meet today’s communication needs.

Why not take mobydick for a test spin with our free community download and find out how it can support you and your business communications.

For more on our mobydick phone system and to arrange a free personalised demo, give us a call on +49 991 29691 200 / +44 203 1379 964 or drop us a line via our website.

Until next time – Happy VoIPing!

Previous Tutorial

Source: http://blog.pascom.net/asterisk-tutorials/asterisk-nat-tables/

Publish Date: May 25, 2016 5:00 AM


Kamailio World 2016 – Asterisk Queues TechTalk

Taking Asterisk Queues to the Next Level with Lua Scripts

Last week saw our very own Mathias Pasquay (pascom CEO, VoIP Guy and Asterisk enthusiast) and Thomas Weber (pascom CTO and now honorary VoIP Guy) delivering a techtalk on Asterisk Queues and the challenges faced when implementing Advanced Call Routing, Skills Based Routing & Fair Queueing at Kamailio World 2016 in Berlin.

The Problem with Asterisk Queues

You know the day when a prospect customer comes to you and says “I need a highly customised solution for my callcenter involving advanced call routing etc etc”? Some companies might sit there and think that’s too bespoke, we can’t do it natively and so on. Well we didn’t, but it did leave us with a bit of a problem, namely how to deliver Skills Based Routing and Fair Queueing using Asterisk’s app_queue.

Asterisk’s app_queue is actually pretty good. It works well, includes a good selection of call strategies and so on – but and there always is one. In the case of app_queue, it unfortunately does not always fit everyone’s business logic and it is not easily adaptable to new business cases.

The Solution

As with most things, if you are looking for an answer it is best to speak to those in the know and listen to what they have to say. Options that we considered before settling on our solution included implementing queueing solutions using the Asterisk AMI, ARI or Dialplan.

However, this can often be overly complex as well as cost ineffective. Thankfully, we came up with a simpler, more effective option to achieve your business goals…..using Lua scripts.

Lua Scripts

The Lua scripting language is known as a fast interpreter with a small footprint, which is one of the reasons why it is provided as an embedded option in many RTC server applications, including Kamailio and Asterisk.

For more on Next Generation Asterisk Call Queueing, watch the guys in action during Kamailio World in the video above.

More Info

pascom are the developers of the mobydick phone system that businesses love. Based on Asterisk, mobydick provides businesses a flexible, fully featured Open Standards phone system to meet today’s communication needs.

Why not take mobydick for a test spin with our free community download and find out how it can support you and your business communications.

For more on our mobydick phone system and to arrange a free personalised demo, give us a call on +49 991 29691 200 / +44 203 1379 964 or drop us a line via our website.

Until next time – Happy VoIPing!

Source: http://blog.pascom.net/voip-guys/kamailio-world-asterisk-queues-techtalk/

Publish Date: May 24, 2016 5:00 AM


Asterisk Tutorial 48 – Introducing Network Address Translation

Introducing Asterisk and Network Address Translation

Over the past couple of weeks we have received a large number of requests on our YouTube channel, asking us to do a tutorial on Network Address Translation (NAT for short) and NAT tables. Which is why, in this the first in a series of Introducing Asterisk Tutorials on the subject, Mathias gets to grips with explaining NAT and helping us understand what the technology is, what it does, how it works and why we need it.

Moreover, as we referred a lot to NAT and NAT tables during our Introducing SIP tutorials, we felt it was only fair to put together a more detailed NAT explanation. What’s more, this tutorial provided Mathias with the perfect opportunity to show off his artistic talents – so sit back, relax and admire.

What is Network Address Translation

As the name Network Address Translation suggests, NAT is the method used to translate a private network IP address into a public IP address.  your internal company LAN network in order to access a public network, such as the internet. In an ideal scenario, we would not actually need NAT. However, as many private networks use the same internal IP address structure, the question then becomes how to identify and route access requests and responses correctly within the public domain.

Source vs Destination NAT

Network Address Translation can be performed in both directions. This functionality is known as Source NAT and Destination NAT.

In essence, Source NAT is the translation of an internal private network IP address into a public address and is used when internal users access external networks such as the internet. Using Source NAT is the most common usage and it ensures that the private network address is remains masked, i.e. hidden behind the public IP address.

Destination NAT is effectively the opposite to Source NAT in that this methodology is utilised to translate a public IP (destination) address to a private address. This technique is often used to map a single public IP address to several private IP addresses, i.e. to enable access to a web based service such as a gaming server etc.

Source NAT and Destination NAT are considered to be the two main types of NAT. However, there are other methodologies, but in the main they can be categorised into either Source or Destination NAT.

More Info

pascom are the developers of the mobydick phone system that businesses love. Based on Asterisk, mobydick provides businesses a flexible, fully featured Open Standards phone system to meet today’s communication needs.

Why not take mobydick for a test spin with our free community download and find out how it can support you and your business communications.

For more on our mobydick phone system and to arrange a free personalised demo, give us a call on +49 991 29691 200 / +44 203 1379 964 or drop us a line via our website.

Until next time – Happy VoIPing!

Previous Tutorial

Source: http://blog.pascom.net/voip-guys/network-address-translation/

Publish Date: May 19, 2016 5:00 AM


mobydick voipGATE Interoperability

pascom Finalise mobydick voipGATE Interoperability Partnership

[Deggendorf, Germany & Leudelange, Luxembourg | 10. May, 2016] pascom Netzwerktechnik GmbH & Co. KG, developer of the next-generation mobydick phone system, today announced the successful completion of interoperability testing voipGATE, a global VoIP provider based in Luxembourg.

Thanks to the successful completion of interoperability testing, mobydick customers can be assured that all mobydick phone system features are fully operable with voipGATE’s SIP trunking services whilst ensuring they continue to benefit from cost-effective, high and reliable call quality. Moreover, the completion of interoperability and compatibility testing has resulted in a newly developer mobydick SIP provider database template, providing mobydick phone system administrators access to voipGATE’s extensive range of services in over 60 countries.

In addition to ensuring full functionality, the new template enables mobydick phone system administrators to quickly and easily connect  to their mobydick phone system with just a few mouse clicks by selecting the voipGATE SIP Trunking template from within the Gateway options menu of the mobydick Commander management portal. Doing so will automatically connect and configure voipGATE inbound and outbound telephony services, ensuring customers are up and calling inside a matter of minutes.

Mathias Pasquay, pascom Founder & Chief Executive Officer, stated of the new interoperability;
“By welcoming voipGATE to our ever growing family of interop partners, mutual customers will benefit greatly in terms of ease of setup as well as being able to rest assured that mobydick’s advanced feature set is fully compatible. Furthermore, voipGATE’s extensive presence in over 60 countries will benefit any mobydick phone system customer operating across national borders by providing access to a single high quality provider.”

About voipGATE

Founded in 2004, voipGATE operates locally and worldwide as a leading operator in Voice over IP, by offering telecommunication services and local phone numbers in more than 60 countries, allowing customers to establish a virtual presence in their key markets. voipGATE has strong partnerships with key PBX and mobile device vendors to ensure and offer its customers both flexibility and ease of use.

For more information about voipGATE, please visit voipgate.com

About pascom – communication without borders

Founded in 1997, pascom are the developers of the next-generation mobydick phone system software and are at the forefront of revolutionising how businesses communicate. mobydick provides businesses with a robust Unified Communications solution to face up to today’s business communication challenges by facilitating and enhancing collaboration and increasing productivity and mobility whilst benefiting from unparalleled scalability and cost efficiencies.

For more information about pascom and mobydick, please visit www.pascom.net

Source: http://blog.pascom.net/pascom-news/mobydick-voipgate-interoperability/

Publish Date: May 17, 2016 5:00 AM


Asterisk Tutorial 47 – SIP Provider Caller ID

Introducing Asterisk and Configuring your Caller ID

We are back again and this time we are finishing of our Introduction to SIP providers. As we promised at the end of the last episode, this tutorial focuses on how to configure your Caller ID in your Asterisk Dialplan.

That of course means handing over to Mathias, our resident Asterisk expert, and letting him guide you through the sometimes complicated world of Caller IDs.

The Problem with Caller ID

Sadly, as is often the case with these things, there is no global standard when it comes to Caller ID formats and telephony providers.

Some providers configure the caller ID for you, others don’t. Some providers only require you to set the originating number, while others require the whole number plus country code. Some providers us “+” to signify the international dialling code while others use “00”. Some providers use a random number if you haven’t configured your caller ID instead of sending the call with a Withheld (or private or anonymous…..) number.

In theory, you could just do nothing and see what happens, but the outcome is very much a case of put luck and is therefore highly unrecommended. Especially, as doing so does not convey professionalism to the call recipient.

Then of course there are other scenarios. You could have a number block from a specific provider and the provider may choose the first number in the block as your caller ID. You may want all your outbound telephony to transmit the Caller ID of your sales / customer service team as opposed to individual extension numbers.

All of this begs the question; “what do we need to do in order make our caller ID transmission more consistent and therefore more professional?”

Thankfully the answer is relatively simple.

Mathias’ Top Tip

As we have already established, sadly there is no uniform Caller ID format, so the question is how do we find out which format our provider uses? You could of course ask google and read through any setup documentation your chosen provider has available.

However, documentation is not always up to date and you may get conflicting results from a google search. Just as well then that there is an easier way. Most VoIP providers use the same Caller ID format for both inbound and outbound calls.

Therefore, by simply making an inbound call to your system whilst have the sip debug running you should be able to ascertain which format your provider uses.

Configuring your Caller ID

To configure your outbound caller ID, simply follow the steps below:

Step 1 – SIP Set Debug

Enter the Asterisk Command Line Interface (CLI) and enable the sip set debug via the following command:

sip set debug peer provider where provider = your provider peer name

Step 2 – Make an Inbound Call

As you should have already configured your Asterisk phone system to accept inbound calls via your provider (see previous tutorial), you will now be able to make an inbound call and use the CLI debug output in the first INVITE to determine the Caller ID format used by your provider as shown below:

Step 3 – Update Your Dialplan

To configure your outbound call rules to show your desired caller ID, simply modify your outbound call rule in the extensions.conf by adding a new first line including the CALLERID variable as shown below. To do this you will need to be logged into your Asterisk system as the root user.

exten => _X.,1,Set(CALLERID(num)=+16463439077)

In your setup, you will need to enter the your desired caller ID number as opposed to our Manhattan number and the resulting syntax should appear similar to that below.

Step 4 – Test Your Configuration

Before testing your configuration, you will need to reload the dialplan. To do this you will need go back to the Asterisk CLI using the command asterisk -rvvvv where the “v” characters reflects the level of verbosity you wish to have and then enter the command:

dialplan reload

Don’t forget to the disable the SIP debug as you will hopefully require any debug information. The next thing to do is to make an outbound call from your Asterisk PBX to a CallerID enabled phone and check to see whether your desired configurations have worked correctly.

Further Caller ID Considerations

While the above setup is relatively straight forward, but of there is more to the Caller ID function that is more complicated than the above setup. For example, you may have multiple numbers registered to your provider and therefore may require multiple outbound call rules with varying Caller IDs.

Another aspect to consider is the CLIP no screening function which allows you to transmit any number as the caller ID when making outbound calls. To test whether CLIP no screening is enabled or not, simply set any caller ID and follow the steps above and check whether the number is transmitted correctly. If not then CLIP no screening has not been enabled and you will need to contact your provider.

In addition, it could be the case that setting the caller ID did not work as sometimes you need to manipulate the Caller ID in the SIP header elsewhere using a different method. However, the good news, is that the above process is the most common.

More Info

pascom are the developers of the mobydick phone system that businesses love. Based on Asterisk, mobydick provides businesses a flexible, fully featured Open Standards phone system to meet today’s communication needs.

Why not take mobydick for a test spin with our free community download and find out how it can support you and your business communications.

For more on our mobydick phone system and to arrange a free personalised demo, give us a call on +49 991 29691 200 / +44 203 1379 964 or drop us a line via our website.

Until next time – Happy VoIPing!

Previous Tutorial

Source: http://blog.pascom.net/asterisk-tutorials/asterisk-tutorial-47-sip-provider-caller-id/

Publish Date: May 11, 2016 5:00 AM


Vector Technologies mobydick Certified

Vector Technologies becomes first mobydick certified partner in Middle East

Burhan Kamal of Vector Technologies receives the official mobydick certification from James Barton of pascom during CeBIT 2016 in Hannover.

Dubai-based IT and connectivity solutions company Vector Technologies, has become the first mobydick certified value-added partner in the Middle East. The ceremonial announcement was made in Hanover, Germany at CeBIT, the global IT event for digital business.

The certification was extended as Vector Technologies completed the rigorous training required by pascom, the developer of mobydick. As an approved value-added reseller for the business telephone system, Vector Technologies also offers pascom a way to penetrate the UAE market and wider GCC region.

“We’re delighted to receive this official certification and look forward to continue working closely with pascom as one of our principal suppliers,” said Burhan Kamal, managing partner of Vector Technologies. “mobydick is a telephone system with wonderful functionality and the UAE’s history in embracing superior technology makes this a great fit.”

James Barton, marketing head of pascom, concurred with the statement of future cooperation. “It’s wonderful to have a certified mobydick partner in a region with such an appreciation of advanced technology. Vector Technologies has supported us with localisations and on-the-ground market knowledge, hence adding value to our offering.”

Both Vector Technologies and pascom were attending CeBIT. Since its establishment last year, Vector Technologies continues to grow in size and scope of services, having recently also added cyber security services. For its part, pascom was showcasing mobydick Cloud Stack, the company’s latest product development which is being released in April.

About pascom – communication without borders

Founded in 1997, pascom is a Linux IT systems integrator with over 15 years worth of experience delivering tailor-made IP telecommunications and network infrastructures solutions and thanks to their Asterisk based mobydick phone system software, pascom have grown in to a market leader in the IP telecommunications across the DACH region and beyond.

mobydick – communications without borders – is a software based, open standards platform which delivers powerful, innovative business communications solutions. Based on Asterisk, the mobydick phone system provides businesses with a scalable and flexible alternative to proprietary IP PBX solutions whilst simultaneously allowing companies to reduce telecommunication costs, increase mobility and boost productivity.

Delivering a whole host of unified communications tools, mobydick’s platform-independent, user-friendly admin interface, provides users and IT admins alike with an enhanced user-friendly experience and significantly decreases IT administration further boosting productivity, optimising workflows and increasing efficiencies.

For more information about pascom and mobydick, please visit www.pascom.net.

About Vector Technologies

Vector Technologies Co LLC provides IT and electronics connectivity solutions for consumers and businesses in the United Arab Emirates, with strategic plans in place to grow further in the Middle East. Clients benefit from a free, no obligation setup assessment in clear language void of technical jargon, supplemented with quality customer service and suggested IT solutions.

Businesses of all sizes have come to appreciate the tailor-made IT services Vector Technologies offers, including solutions for storage, networks, servers, data backup, security, IP cameras, voice systems and web presence

For more information about Vector Technologies please visit www.vtec.ae

Source: http://blog.pascom.net/pascom-news/vector-technologies-mobydick-certified/

Publish Date: May 9, 2016 5:00 AM


Kamailio World 2016

pascom Announce Kamailio World 2016 Sponsorship

[Deggendorf, Germany | 03. May, 2016] pascom Netzwerktechnik GmbH & Co. KG, developer of the next-generation mobydick phone system software, today announce their participation at Kamailio World 2016 (18-20th May) in Berlin as a Silver Sponsor.

pascom’s involvement with Kamailio World stems from more than just being an event sponsor. During Kamailio World 2015, Mathias Pasquay, pascom CEO and Co-founder, spoke on the topic of deploying Kamailio when moving an Exisiting Asterisk based IP PBX to the Cloud without any functionality loss – which has been a core aspect of the company’s product development over the past 12 months.

Fast forward to 2016 and Mr Pasquay will once again take to the stage to discuss another major product development focus – leveraging the power of Lua Scripting to not only enhance Asterisk phone system call queueing but also its practical applications across the Kamailio and Asterisk Open Source communities.

Mathias Pasquay, pascom CEO and co-founder, on Kamailio World:

“As Kamailio World 2015 was such a success and as pascom integrates a whole manner of Open Source solutions in the development of mobydick, being at Kamailio World provides an opportunity to give back to the community as well as gaining an insight to the latest innovations stemming from the community. Moreover, as the 2016 event celebrates 15 years of innovation – I simply couldn’t miss it.”

The 2016 event focuses on realtime communication technologies and how their constant innovation not only challenges the imagination but also paves the way for the future of realtime communication.

For more information and ticket registration, please visit www.kamailioworld.com

About Kamailio

Kamailio® (successor of former OpenSER and SER) is an Open Source SIP Server released under GPL, able to handle thousands of call setups per second. Kamailio can be used to build large platforms for VoIP and realtime communications – presence, WebRTC, Instant messaging and other applications.  It can also easily be applied to  scaling up SIP-to-PSTN gateways, PBX systems or media servers like Asterisk™, FreeSWITCH™ or SEMS.

Among the powerful features: asynchronous TCP, UDP and SCTP, secure communication via TLS for VoIP (voice, video, text); WebSocket support for WebRTC; IPv4 and IPv6; SIMPLE instant messaging and presence with embedded XCAP server and MSRP relay; asynchronous operations; IMS extensions; ENUM; DID and least cost routing; load balancing; routing fail-over; accounting, authentication and authorization; support for many backend systems such as MySQL, Postgres, Oracle, Radius, LDAP, Redis, Cassandra, MongoDB, Memcached; Json and XMLRPC control interface, SNMP monitoring.

About pascom – communication without borders

pascom are the developers of the next-generation mobydick phone system software and are at the forefront of revolutionising how businesses communicate. mobydick provides businesses with a Unified Communications solution to face up to today’s business communication challenges and is renowned as a robust, productivity and collaboration enhancing communications platforms that delivers unparalleled scalability and cost efficiency.

For more information about pascom and mobydick, please visit www.pascom.net

Source: http://blog.pascom.net/pascom-news/pascom-kamailio-world-2016/

Publish Date: May 4, 2016 5:00 AM


Asterisk Tutorial 46 – SIP Provider Outbound Calls

Introducing Asterisk SIP Provider Outbound Calls

We are finally back! With so many “goings-on” over the last few weeks, finding the time to put this tutorial together really has been a struggle – so thank you all for your patience. Now to the task at hand – SIP Provider Outbound Calls.

Last time around, we finished up with configuring our Asterisk phone system to accept incoming calls from our SIP provider. That of course means that it is time to delve into what we need to configure our Asterisk system to make outbound calls via our SIP provider – which means we finally make a real life call to the outside world!

SIP Provider Outbound Call Dialplan Configuration

In order to configure our test system to make an external outbound call, we first need to modify our dialplan in order to enable it to make outbound calls before moving on to editing our SIP provider peer.

The problem here is that when making calls via a provider is that the provider essentially represents the entire world which begs the question – how do we configure our dialplan so that we can call any desired number using our provider?

Thankfully the answer is quite straight forward. In essence, you need to take the “outgoing” context we created way back in tutorial 11 and alter it a little bit to reflect your SIP provider peer and add an Asterisk Variable (see tutorial 13).

Before moving forward, it is assumed that you have configured your dialplan contexts as we did in our earlier tutorials, namely that you have added the relevant syntax in the [phones] context. Once done, the [outgoing] dialplan context should look similar to:

Before editing:

exten =>_X.,1,Dial(SIP/outside)

After editing:

exten =>_X.,1,Dial(SIP/provider/${EXTEN})

To give you an idea what this means, the new configuration basically tells our system to make the call using SIP via the peer “provider” and to store the number in the variable.

If you try making an outbound call now, chances are that your provider will reject it as it will not be able to authenticate the SIP peer making the call. It is for this reason, that the next step involves modifying our provider peer a touch in order to circumnavigate this problem.

Modifying your SIP Provider Peer

The problem with our current configuration is that our provider is firstly unable to authenticate our users (SIP peers) and secondly our local domain (IP address). Therefore, we need to add some further syntax to our provider peer (in sip.conf) which will ensure that we can get round this issue.

Step 1

The first step is to add the fromdomain syntax to our provider peer in order to enable the provider to authenticate the sender domain, as follows:

fromdomain= insert provider domain e.g. sip.flowroute.com

This will ensure that as opposed to transmitting the local domain, our external outbound telephony will set the from, i.e. sender, information to reflect your provider.

Step 2

Now we have resolved the problem with domain authentication, we need to be able to make sure that our provider can also authenticate our phone system users. To do this we can add another parameter called defaultuser and add the corresponding user details (Tech Prefix) from your SIP provider as shown below:

defaultuser= insert user details / techprefix e.g. 1234567

Mathias’ Top Tip

Having followed the above, it is now time test your configuration to ensure that everything works correctly. Unlike us, it is advisable to always check the call audio quality to ensure that there are no issues with the audio. We didn’t, as for the purposes of this tutorial, there is no need to incur the cost of an international phone call, however for productive systems it is of course considered a best practice to do so.

More Info

pascom are the developers of the mobydick phone system. Being based on Asterisk, mobydick provides businesses with an easy to install, manage and use Open Standards phone system.

Why not take mobydick for a test spin with our free community download and discover how it can support your business communications.

Want more info about mobydick or would like a free personalised demo? Give us a call on +49 991 29691 200 / +44 203 1379 964 or drop us a line via our website.

Until next time – Happy VoIPing!

Previous Tutorial

Source: http://blog.pascom.net/voip-guys/asterisk-tutorial-46-sip-provider-outbound-calls/

Publish Date: April 20, 2016 5:00 AM


CeBIT 2016 Review

CeBIT 2016 – Our Most Successful Yet

[18.04.2016 | Deggendorf] CeBIT 2016 may be a few weeks behind us now but the interest generated and the success of our 2016 exhibition is very much ongoing.  Figures released by Deutsche Messe AG indicate that attendees engaging in business talks is continuing to rise. Furthermore, international attendance remains on par with a quarter of visitors heralding from shores further afield, confirming CeBIT is still the premiere IT exhibition despite growing competition from other international shows.

Introducing the mobydick Phone System

The 2016 show saw pascom exhibiting alongside IP phone developer Auerswald GmbH as their 2016 CeBIT Cooperations Partner. The change in stand location proved timely in opening new avenues for reaching a previously unexplored domestic market segment whilst maintaining our presence at the centre of the Unified Communications world that is Hall 13.

Previously reticent System Integrators who had been hesitant to consider software based VoIP solutions are now reconsidering. The change in thinking has been brought about in part due to the upcoming 2018 ISDN cut off in Germany combined with experiencing increasing customer pressure stemming from businesses increasing becoming more aware of the more advanced feature sets, increased flexibility and scalability benefits that software solutions, such as mobydick, are known to deliver.

Our Most Successful CeBIT To Date

Throughout the 2016 show, it was apparent that not only stand visitor numbers were up, so too was the quality and level of interest in mobydick phone system solutions.

Thomas Schmidt, pascom Director of Sales speaking on the final day; 

“As a first impression, considering the volume and quality of visitors throughout the week being of such a high calibre, it is with an assured confidence that I predict CeBIT 2016 to be quite possibly our most successful CeBIT to date.”

Once again, our participation at CeBIT 2016 provided the perfect platform from which to showcase our mobydick Unified Communications solution to a wide variety of domestic and international potential customers and channel partners.

Mathias Pasquay, pascom CEO & Co-Founder speaking during the event; 

“Over the previous four years, our presence at CeBIT has played a crucial role in expanding our geographical reach both nationally and internationally and this trend looks set to continue during this year’s [2016] show.” 

About pascom – communication without borders

pascom are the developers of the next-generation mobydick phone system software and are at the forefront of revolutionising how businesses communicate. mobydick provides businesses with a Unified Communications solution to face up to today’s business communication challenges and is renowned as a robust, productivity and collaboration enhancing communications platforms that delivers unparalleled scalability and cost efficiency.

For more information about pascom and mobydick, please visit www.pascom.net

Source: http://blog.pascom.net/pascom-news/pascom-cebit-2016-review/

Publish Date: April 18, 2016 5:00 AM


mobydick 7.12 Release – Highlights

What’s New in mobydick 7.12

Cloud Telephony

mobydick 7.12 delivers a significant number of “under the hood” additions, modifications and improvements. This also includes mobydick 7.12 being the first mobydick phone solution that is cloud compatible and provides the foundations on which future mobydick phone system versions will be further enhanced. To this end, the development of mobydick 7.12 ran in parallel to achieving the first milestone of the mobydick Cloud Stack which will be fully functional with mobydick 7.13 next quarter.

Auerswald Support

mobydick 7.12 sees the addition of Autoprovisioning and Interoperability support for Auerswald GmbH’s superb range of Android based IP Phones, meaning end users are now able to benefit from the ease of use and familiarity of their Android Smartphones, directly on their desktop without losing any of the functionality that mobydick interoperability delivers.
For mobydick phone system administrators, the new support also delivers significant time saving and ease of device commissioning benefits. Thanks to the transition from the previous tabular device configuration settings to a XML menu editor, device family management in mobydick 7.12 has been greatly simplified, enabling admins to simply copy and paste the settings and customising them to meet specific device / user requirements. This transition will be available for Snom, Yealink & Aastra (Mitel) phones in mobydick 7.13.

Outbound Trunk Selector within the mobydick client

With mobydick 7.12, it is now possible for mobydick end users to select the phone number (meaning trunk) they use when making calls from a drop down list. Enabling this function has numerous benefits including allowing end users to decide on the most appropriate number without needing the corresponding “In-Prefix” associated to each trunk. What’s more, this new functionality also reduces the need for mobydick phone system administrators having to configure multiple least cost routing call rules when multiple trunks are in operation and when used appropriately can produced significant cost savings which will make the finance team happy.

Copy/Paste Licences

mobydick 7.12 brings about a change in the methodology behind how licences are issued and kept up-to-date thanks to migrating away from the old md-lic packets to a mobydick licence key that can be copied and pasted directly eliminating the need for packet downloads and uploads when managing licences. In addition, should a mobydick licence become invalid due to new user values or software maintenance contract extensions for example, mobydick will revert back to the community version. This will provide mobydick system administrators to access the commander where they will be able to use the newly integrated “Check for licence updates” tool or paste the new licence key directly into the licence overview section of mobydick, meaning they can get the mobydick phone system operational again with the minimum of fuss and in no time at all.

Database Backups

The mobydick Database backup tool has been greatly improved and now automatically includes music on hold and customised prompts as well as providing mobydick phone system administrators with more flexibility in what is included by providing options to include recordings, faxes and voicemails.

Conference Rooms

The mobydick Conference Rooms no longer use the MeetMe application, but rather have been migrated to the newer more stable Confbridge. MeetMe has been deprecated and also lead to problems and stability issues within the conferencing system, whereas Confbridge delivers greater stability. However, this does mean that for the time being the little used dynamic Conferencing functionality as been removed until the Confbridge integrate has been expanded to support dynamic conferencing.

New Debian System & Asterisk Certification

The mobydick phone system has been upgraded to Debian 7 (Wheezy). Wheezy is approximately 25 – 30% quicker than Debian 6, which in turn provides additional benefits in terms of the number of users the various mobydick appliances can support. This also applies to virtualised instances thanks to Wheezy providing an optimised platform for vmWare and Hyper-V virtualisation solutions.

Furthermore, upgrading to Wheezy ensures that the mobydick Linux platform will continue to be supported until May 2018 as well as ensuring continuing security updates for the system can be implemented as does the upgrade to asterisk 11.6-cert 12 which delivers a greater spectrum of security features.

Further Modifications

md-cmd 7.12.00

  • Support for junghanns cards removed
  • PHP-Version updated to 5.6
  • conf_bridge-based conference rooms integrated
  • dynamic conference rooms removed

md-updater 1.12.00

  • Added support for copy-paste licenses
  • Update blocker installed, if a junghanns card is in operation

ex-mobydick 2.11.00

  • JRE 8u74 integrated
  • Kernel 3.16 for 64bit-Versionen integrated
  • Kernel 3.2 for 32bit-Versionen integriert

For the full release notes, please refer to our Wiki site:

wiki.pascom.net

Source: http://blog.pascom.net/pascom-news/mobydick-7-12-release-highlights/

Publish Date: April 7, 2016 5:00 AM

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