Hello List, I have seen that when ever asterisk gets a SIP INFO request from a SIP channel it generates the requested DTMF tone and writes to the destination channel also it forwards the SIP INFO message. As I am very new to this domain, it is really confusing me. Why not asterisk writes only the tone and can avoid forwarding of SIP INFO? I know I may be wrongly interpreted the things, Can somebody please explain me the scenario, if possible?
Thanks Rajib ________________________________ From: Deka, Rajib IN MAA SL Sent: Monday, August 29, 2011 3:34 PM To: '[email protected]' Subject: Asterisk is delaying DTMF (SIP INFO) relay in MeetMe Hello List, 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. Regards, Rajib Siemens Ltd.
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