We are using ‘F’ parameter in meetme Dialplan application to broadcast SIP INFO (1 and 0) as DTMF tone to all the participants.
The DTMF configuration for all the connected SIP clients is SIP INFO.
The problem we are seeing, asterisk is taking some time to broadcast the SIP INFO message to all the participants from the time of its appearance. The time latency varies from 1.5 sec to 6 sec. We have activated the highest debug and verbose level but we are not able to track down the problem. Please help us out to overcome this problem as 6 sec latency is not acceptable in real-time scenarios. Also if possible let us know (technically), whether it is a know issue in asterisk.