Wazo 20.13, how trunk works now?

Hello Wazo user!

The 20.13 release which will arrive very soon will have a major change in terms of SIP management. Indeed, Wazo-Platform has been using the PJSIP channel for several versions in place of the obsolete original SIP channel in Asterisk. This migration was started during the release of Wazo-Platform a year ago now. It was made to go in transition to PJSIP and we had simply made a connector which translated the configuration of the chan_sip to the chan_pjsip. It was overall fairly transparent to the admin person. On the other hand, it was a good puzzle and we had a number of settings that were a lot more complex.

For more than 6 iterations in a row (one iteration = 3 weeks) we have undertaken to add new APIs to natively manage PJSIP only and remove any occurrence of chan_sip in Wazo. This API has finally been finalized, including the migration and a new management interface in Wazo UI to configure it. But we didn’t just do that, we also took the opportunity to add a new functionality to directly manage templates in the system. These templates allow you to make a configuration that will be replicated to all the users or SIP trunk that you want. In addition, this new template system is able to manage the inheritance of other templates.

Since this is a major change in how it works, I will give you a global view of how it now works, hoping that it will help you more easily take over the management of SIP trunks in Wazo Platform.

You will also see that there have been some organizational changes in Wazo UI to help you.

First go to the trunks management page which is now in the Call Management menu.

Then Add or Edit a SIP trunk.

We can see that now there are new tabs and different forms.

The first tab (trunk) allows you to define a label, a name, the transport (UDP or TCP) for example, but also templates and the context to define where the call will be routed when entering.

The second tab (Aor) is the AOR (Address Of Record) and where we will define the address of the SIP server. There are also a whole bunch of other options available which are those provided by Asterisk. We now have in each of the menus a list of all the configuration keys available for configuration. You just have to choose what you want. Documentation for these values ​​is available on the official Asterisk wiki website. For example for endpoint it is managed by the res_pjsip.so module and the documentation is here: https://wiki.asterisk.org/wiki/display/AST/Asterisk+17+Configuration_res_pjsip

The third tab (Authentication) is to manage the incoming authentication of your trunk.

The fourth tab (Endpoint) concerns the configuration of your endpoint. For example rewriting your From field with from_user or from_domain.

The fifth tab (Identify) allows you to manage the identification on an incoming SIP message, or more generally an incoming call. You can match by IP or by header.

The sixth tab (Registration) concerns the registration of your Wazo Platform to a SIP provider for example. The two important values ​​are client_uri and server_uri. The values ​​are now in the form of a SIP URL. For example sip:username@provider:port

The seventh tab (Registration Outbound Auth) will be for the authentication of your registration (REGISTER).

And finally the last (Outbound Auth) will be the authentication for your outgoing calls.

I hope this has been clear to you and that you can quickly get used to this new way of working which offers a lot more power than before. The APIs are obviously on the same operation and are used by Wazo UI.

Looking forward to reading you.