I configured an asterisk server to receive calls from one sip trunk and then dial out through another (my VoIP provider). Both trunks are configured with dtmf mode SIP INFO. The thing is: When I complete a call and send DTMFs, Asterisk Server always ignores the first dtmf i send, it answers a 200 OK to the endpoint but do not forward the signal to the other call leg. From the second DTMF on, it answers 200 OK and forward the SIP INFO to the other leg normaly. Have you guys ever seen this? I did the same deploy on a lab environmet and got the same results.
don't know why this solved the problem, but it did. So I'll share with everyone, in case someone have this same problem. I commented the line "disallow=all" on the trunk configuration. I kept using the same codec I was using before (G711a), but for some reason the INFOs only started working properly when I did this change.