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.
I think the AsteriskNow part of this is irrelevant, otherwise this would have been the wrong forum.
You need to provide sip set debug on, verbose level traces, to see what is to blame for the delays.
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.
Wireshark is OK for the initial diagnosis, but, if you ever have to submit a bug report on Asterisk, for this, you will need the sip set debug output.
right…if we are looking at sip messages wireshark can be quite nice.
Either way its kludgy and time consuming if you have some odd issue going on.