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Asterisk Tutorial 45 – SIP Provider Inbound Call Rules

Introducing Asterisk & SIP Provider Inbound Call Rules

Welcome back to the VoIP Guys and Introducing Asterisk. Before we get started with adding Inbound Call Rules for our SIP provider, we have an announcement to make. Due to the Easter break, we will not be on air for a couple of weeks, but rest assured we will be back in a few of weeks with more exciting and entertaining Asterisk phone system tutorials. We know that many of you will be disappointed and we apologise for any inconvenience caused, however sometimes we need a break too!

That said, time to set course and sail full steam ahead with configuring our Asterisk phone system to enable inbound calls from our SIP provider, which means that we will finally get round to making that call that we talked about last time as Mathias guides you through the last necessary configuration steps including adding inbound call rules in our dialplan.

Adding Inbound Call Rules

To add inbound call rules to allow incoming calls via your SIP provider, you will need to add a new context within the extensions.conf which will look similar to that below:

[provider] exten => _X.,1,Goto(phones,100,1)

The above configuration using the regular expression _X., will route all incoming calls to the extension defined in your phones context – in this case the extension 100. Of course, this is not the perfect solution, as you will likely have more than one extension in which case you would route the calls for each number to the relevant extension. In a productive Asterisk phone system and you are routing all incoming calls to one extension, then that extension would normally belong to a Queue or an IVR menu.

Once you’ve finished adding your provider context within the dialplan, try calling the number. If you’ve followed the steps from today’s and the last few tutorials – it probably won’t work. The reason for this is how the Asterisk authentication process works.

When receiving the SIP invite, the information contained includes the phone number and when we configured our SIP provider peer and the call rules above, we named them “provider” which is not contained within the SIP invite. Therefore, Asterisk tries to match the information received in the invite to what we have configured and it can’t. This results in the calling being rejected as Asterisk is not able to authorise it.

To rectify this, there are a few possibilities. The first option is to rename all our provider peers to match the telephone number. Obviously, this is okay with one or two numbers, but if you have more numbers, this option becomes time consuming and complicated. There is a much simpler and more effective option and that is Mathias’ Top Tip.

Mathias’ Top Tip

When authorising incoming calls, it is possible to force Asterisk to ignore the invited username in the authentication process and only check the host name and port. To do this, we need to go to our sip.conf and add one line to our provider peer as follows:

insecure=invite

This is will result in your Provider Peer configuration looking similar to below:

Finally, as we have said before, always check the audio quality before going live. Bad audio quality really does have a negative impact on your company image and how professional you are, so make sure it is not just okay, but good if not excellent.

That’s it for this week. Next time, we will get started on the more complex configuration to allow us to make outbound calls via our SIP provider.

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-sip-provider-inbound-call-rules/

Publish Date: March 30, 2016 5:00 AM


mobydick 7.12 Live & Uncut

Coming Soon: mobydick phone system version 7.12

April 7th is a big day for us at pascom, for it is the day that will see mobydick 7.12 being released. Yes we’ve had releases before (quite a few actually), but none have been quite like this one. Yes we’ve added new features and so on before, but none have been quite like what’s coming in mobydick 7.12.

The best part of it is, we are inviting you to join us for a Free, Live & Uncut Release Keynote as we unveil the latest mobydick phone system version, which as you may have guessed is packed to the rafters with new features and improved functionality.

Throughout the live broadcast, we will showcasing what’s new with mobydick 7.12 as well as introducing the new mobydick Cloud Stack and Session Border Controller,  highlighting more interoperability as well as demonstrating a few usability and functionality improvements.

How to Register

To register, all you need to do is visit our website and fill out four little boxes on our event registration page.  Once signed up, we will send you some additional event information including how and where to watch.

Register now and join us on April 7th at 14.00 CEST and we look forward to presenting mobydick 7.12 to you.

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/voip-for-business/mobydick-7-12/

Publish Date: March 24, 2016 5:00 AM


Asterisk Tutorial 44 – SIP Provider Peers

How To Add SIP Provider Peers in Asterisk

Welcome back to Introducing Asterisk from the VoIP Guys and apologies for not being on air last week – it’s CeBIT’s fault! However, we are back and have got another tutorial for you all. Having registered our Asterisk server with our SIP account, we are now ready to get started with SIP Provider Peers.

SIP Provider Peers differ slightly from the SIP phone peers which we configured earlier in the series in that when we configure our host name, we will not use the dynamic settings as we did with our phones. Plus we will also need to configure some additional settings.

Adding SIP Provider Peers

As mentioned above, the process is similar to how we added our SIP phone peers way back in tutorial 5, but with some differences. In order to get started, you will need to enter the Asterisk console (as root) and call up the sip.conf using the following command:

vi /etc/asterisk/sip.conf

Once in the sip.conf, you can then add your peer just as we did for our phones. Assign a peer name that will help you identify it more easily in the future, particularly if you intend to connect multiple providers. In our case, we named our SIP provider peer”[provider]“.

Now we can start configuring our peer as follows:

type=friend

Normally, a client will register with a peer. Using the type=friend allows authentication between peers and clients and vice versa, meaning we only need one peer instead of having to configure two.

context=provider

The context in the dialplan where the call resolution should start.

allow=ulaw, alaw

The voice / audio codecs supported by your SIP provider. ulaw and alaw are pretty much the standard here. For more on Audio Codecs, please refer to Asterisk Tutorial 36.

secret=password

The password required by your SIP Provider for inbound / outbound telephony authentication. Can be found in your Provider’s web portal (GUI),

host=sip.providername.com

The host name for your SIP provider. As we know the host name / IP address where we registered our Asterisk system to our Provider in the last tutorial, we can use this information as opposed to the dynamic setting we used for our phone peers.

nat=force_report,comedia

Very much dependent on your network setup. We will get into more detail regarding NAT and NAT tables in a future tutorial.

Dynamic vs Fixed Host Names

When configuring phones, we set the host to dynamic as unless we have configured them with a fixed IP address, chances are we will not know the IP address of the phones themselves. With the case of a SIP provider, it is the otherway round.

The provider will supply us with a host name / IP address where we register our Asterisk system meaning the host should be set to this name / address.

From the providers point of view, they do not know our IP address and so they will configure their system as dynamic.

Mathias’ Top Tip

Never trust the context you add for your provider, as everyone on the outside of the system can call the number and therefore access this context. Therefore, you should use the context to ensure whether the number is correctly formatted and whether or not you actually own the number before switching contexts and routing the calls to a context that you do trust, i.e your internal phones etc.

Another best practice is to configure a context for each provider as this will support you later on when adding numbers to provider contexts as having separate contexts will provide you with an easier to manage overview.

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-44-sip-provider-peers/

Publish Date: March 23, 2016 5:00 AM


Flowroute mobydick Interoperability Partnership

pascom Finalise mobydick Flowroute Interoperability Partnership

[Deggendorf, Germany & Seattle, USA | 14. March, 2016] pascom Netzwerktechnik GmbH & Co. KG, developer of the next-generation mobydick phone system, today announced successful interoperability certification with Flowroute, the leading provider of advanced communication services. The certification verifies compatibility and document configuration recommendations for pascom’s mobydick VoIP phone system used in conjunction with Flowroute’s communications services. Certification sees the US based VoIP provider joining pascom’s expanding global network of certified Technology Partners.

The successful certification ensures the full operation of all mobydick phone system’s features with Flowroute’s SIP trunking service to deliver cost-effective, reliable calls for mobydick customers while allowing them to maintain control of their entire communications solution. Thanks to the technical collaboration between both companies, as well as ongoing testing of the newly developed SIP provider template, customers can be assured of high and reliable call quality.

In addition to ensuring reliable and high call quality, the new template enables mobydick phone system administrators to quickly and easily connect Flowroute to their mobydick phone system with just a few mouse clicks by selecting the Flowroute SIP Trunking template from within the Gateway options menu of the mobydick Commander management portal. Doing so Flowroute services will automatically be configured for both inbound and outbound call rules.

Dan Nordale, Flowroute Chief Marketing Officer, said of the new partnership;
“Flowroute customers expect the highest performance from their communications technologies and we appreciate pascom’s drive to clear this high bar. Our customers push the envelope with communications and we have found pascom to thrive in these situations, so we know this formalization of our partnership is an important step for our joint customers. pascom’s growing global customers base depend on flawless communications and we are eager for them to experience the Flowroute difference” 

Mathias Pasquay, pascom Founder & Chief Executive Officer, stated of the new interoperability;
“Our new interoperability partnership is great news for our mobydick customers who will now be able to more easily gain maximum benefit from a combined mobydick Flowroute VoIP solution. Further to this, welcoming Flowroute to the mobydick family again demonstrates our commitment to supporting our customers worldwide in the delivery of top quality, hassle free communications.”

About Flowroute

Flowroute is the leading provider of communication services for cloud-based companies. Flowroute gives developers and enterprises carrier-quality services with unprecedented performance, transparency and control to add voice and messaging capabilities into their apps and services to create unique user experiences.

Flowroute is privately held and headquartered in Seattle, WA. For more information: www.flowroute.com.

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/mobydick-flowroute-interoperability/

Publish Date: March 14, 2016 5:00 AM

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