I have an asteriskNow 1.7 server on a public IP. Calls placed from soft client to soft client or soft client to PSTN SIP trunk take 4-6 extra seconds to cut the audio through after call supervision is completed. Anyone know how to cut the audio through immediately?
This is especially bad with iPhone soft clients, but it happens with every client I’ve tested, including a couple Polycom desk phones. Maybe we are getting a SIP 183 message, a SIP 200 message and then 4-6 seconds later the audio starts flowing.
wireshark baby, get down to those SIP messages and verify your initial thought is right or not?
when you say an extra 4-6 seconds…what are you comparing them to? You ahve to have some benchmark obviously…the possibilities here are limitless really.