![]() Realm= (your space URI)įromdomain= (your space URI)Įncryption=yes (enables media encryption) Secret=password (your SIP endpoint password) Host= (your space URI)ĭefaultuser=freepbx (your SIP endpoint name)įromuser=freepbx (your SIP endpoint name) Type=peer (sets up for IP authentication) In the Peer Details box, add the following lines: Trunk Name: freepbx (the name you gave the SIP endpoint) On the Sip Settings tab and the Outgoing sub-tab, enter the following: Outbound CallerID: 333 (whatever DID you purchased) ![]() Trunk Name: SignalWire (Can be anything you want) With the above in place, having this SignalWire SIP Trunk added and working on FreePBX is done in just a few easy steps.įind Trunks under the Connectivity menu, then add a SIP (chan_sip) trunk, and on the General tab enter the following: Disable the Encryption option for AEAD_AES_256_GCM_8 as Asterisk is not currently supported and will cause a warning on the Asterisk Command Line (though calls would still work).There are a couple of settings we need to change on the SIP Endpoint we have created in SignalWire to get good interworking with FreePBX: Editing the Signalwire SIP Endpoint for chan_sip FreePBX If you don't have a SIP endpoint set up already, the first step is to create a SIP Endpoint and connect it to a phone number. IF you still prefer to move forward with chan_sip, here are the instructions! Creating a SIP Endpoint Further, there are limitations in chan_sip when it comes to receiving inbound calls from SignalWire, and moving to PJSIP is strongly advised - as discussed in this guide. The chan_sip module in Asterisk has been deprecated since Asterisk 17 and will receive no further development effort.
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