Asterisk Tutorial 46 – SIP Provider Outbound Calls - pascom - ContactCenterWorld.com Blog
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:
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.
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.
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.
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: April 20, 2016 5:00 AM
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