Hello everyone. I currently maintain one small asterisk sip-trunc. It works great so far but i still have this one little problem. Sometimes when a caller dials our number we can here the call getting in, but as soon as someone accepts the call, the other side only hears the occupied signal. Sometimes this also happens when dialing out. I don’t really understand what the error is telling me and when searching the only solution i found was adding
allow=all to all endpoints to allow all codecs. How can codec negotiation fail if i allow all codecs? Can someone help me here to at least demystify the problem?
Thanks in advance and have a nice day.