Asterisk Tutorial 45 – SIP Provider Inbound Call Rules - pascom - ContactCenterWorld.com Blog
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:
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.
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!
Publish Date: March 30, 2016 5:00 AM
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