I have FreePBX Server and TalkSwitch IP PBX (TS). I created a SIP Trunk in Asterisk server connecting to TS.
The registration is successful, I can make outgoing calls and receive incoming calls without any problems.
I enabled the IVR into this SIP truck so that the caller can enter the desired extension.
The problem, SIP trunk does not recognize the dial tone from analog phones or PSTN connected to TS while it is working fine with IP Phones connected to TS.