Asterisk Tutorial 44 – SIP Provider Peers - pascom - ContactCenterWorld.com Blog
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:
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”
Now we can start configuring our peer as follows:
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.
The context in the dialplan where the call resolution should start.
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.
The password required by your SIP Provider for inbound / outbound telephony authentication. Can be found in your Provider’s web portal (GUI),
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.
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.
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 23, 2016 5:00 AM
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