Greetings to All. Previously we have an E1 line that is connected to FreePBX and various trunk links that are working fine for long time, but recently we have deployed new SIP line from our service...
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Hi, rsmithuk (Richard Smith) ; GSnover (GSnover) ; Great @sorvani ! This is what I was looking for. I will try and let you know how it goes. I have now created the pjsip trunk and added the IP addresses on the match/permit field. I have created an outbound route to test outbound calls using this trunk and I was able to make calls successfully. Now how do I make sure that inbound calls use this new trunk I created? I want to make sure inbound calls work before deleting my previous chansip trunk.
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Can anyone recommend a SIP provider that supports G.722 codec? I’ve checked with my current providers, they only do G.711 and G.729
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