I started a new project of integrate FreePBX with Microsoft Teams, for that, i need a SBC for deal with the TLS. So, first i started with DSipRouter, it is a easy tool to use, i manage to integrate my PBX with DSipRouter really easy in a schenario that i set my softphone to register in the IP of DSipRouter and DSip routes the register to my PBX. In that case DSip dont even try to register for itself, it only route the register, and thats what i want. But i got stucked with the TLS, and after seen a news that the new version of dsiprouter has a paid tool for MS Teams, i now think its impossible to do it without paying for.
So i moved on for Opensips, and i got the opposite schenario, i manage to deal with the TLS, but i can’t make the OpenSips integrate with the FreePBX. Most of the material i found online, is to use the extensions registered in OpenSips, and the integration with FreePBX (Asterisk) is made only for deal with voicemail, IVR, and other medias that OpenSips can’t deal. But my goal here is just use OpenSips for treat the TLS packages coming from MS Teams, all the registration must be done by my PBX! Is possible to configure the OpenSips to just receive and route the register same as i can do with DsipRouter?
If anyone could help me with that sharing some infos and docs i’ll appreciate.
Hmmm… I saw that asterisk support TLS over SIP, but most of the tutorials i found ask for a SBC like Kamailio/OpenSips for do the TLS part with MS Teams since Asterisk couldn’t, so i never give a chance to try only with Asterisk dealing with TLS.
You can set Opensips as a proxy, or you can modify Asterisk as described in that thread. If you’re doing anything production-level and want Microsoft’s support, stop your current effort and buy a Microsoft certified SBC appliance.
To do this with opensips, run it on a different set of ports so that it does not sit in front of Asterisk for everything, but only for your Teams traffic. You’ll have to edit opensips’ routing script too to basically proxy traffic received from Asterisk to MS, and traffic from MS to Asterisk, and throw away anything else it happens to get. (You can do all this easily from the routing script without involving any database or even the dialog module)
btw, I did exactly this, had the Teams interop working with opensips, and then abandoned it for a better solution.
Maybe the Asterisk project would accept a patch to handle the few peculiarities of the Microsoft interop requirements so that you could connect directly without custom code work or the extra SIP proxy.