so I had a provider that was not sending me a proper DID or at least it wasn’t showing up properly for my asterisk system, I found this article, but after I added the stuff for extensions_custom.conf and then added the stuff for the trunk, the calls route properly, which is good, but I loose audio? nothing!!! Please help!!!
I dont think you lose audio due to your dial plan configuration, Most of the audio issues are associated to misconfiguration on the NAT parameters in your sip.conf file. As you make references to extensions_custom.conf it seems you are using FreePBX so those settings are adjusted through the GUI, which is not supported here
oh ok. thanks for that info though.