Asterisk 16.1 This statement appears in the features.conf doc: Note that the DTMF features listed below only work when two channels have answered and are bridged together. They can not be used while the remote party is ringing or in progress. If ..

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I use this dial paln: [o2-in] exten => o2,1,Answer exten => o2,n,Playback(hello-world) exten => o2,n,Ringing exten => o2,n,Dial(SIP/10&SIP/20&Local/s@no-op,25,rt) exten => o2,n,Playtones(425/1000,0/4000) exten => o2,n,Wait(30) exten => o2,n,Hangu..

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, Using Asterisk 16.1.1, with PJSIP, Im asked to build a SIP trunk to a system that uses exclusively tel: uri on inbound and outbound calls. I could not find documentation or sample config about tel:uri. Is this doable? If not possible with PJSIP,..

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