I am facing a couple different problems with DTMF on Asterisk (G.729 codec and dtmfmode=rfc2833) as follows: 1) I come into Asterisk using a Voicepulse DID over IAX. I use DTMF for authenticating the user and collecting the destination phone number, which works without problems. However, when I connect the second leg of the call, via SIP to my service provider (Level-3) to terminate a PSTN call, I face a DTMF problem. e.g if I call into my voice mail and am prompted to enter my password, it is not recognized. Asterisk is bridging the RTP at all times and does not use reinvite. Not sure of the problem is at the IAX end or SIP end. Using ethereal, for every key pressed I see a total of 4 DTMF events being sent from Asterisk to level-3. 1 with "End of Event" set to "False" and Event Duration of 0, and 3 more each with "End of Event" set to "True" and Event Duration set to 800. The Volume for all the 4 DTMF events is set at 10. However, this is not received at the PSTN end. I am pretty sure this is not a Level-3 problem and have discussed it with them. Is the total of 4 DTMF events for every key pressed correct? Are the DTMF durations of 0 and 800 correct? If the volume setting of 10 too low? How do I fix this problem in Asterisk? 2) When I come over SIP from Level-3, I use Asterisk to play voice prompts and collect DTMF digits for authentication (using Authenticate() dialplan command) and final destination number (using "background"). The DTMF digits are correctly interpreted by Asterisk. However, immediately after playing the respective voice prompts, Asterisk abruptly stops sending Level-3 any RTP packets, until all the DTMF digits are received. This period of no RTP messages from Asterisk can last up to 10 seconds, based on how fast the user enters the DTMF digits. Level-3 does not like this and flags it as an error as their media gateway complains of an NRS error, which is indicative of no RTP packets from Asterisk during an active session for a prolonged period (several seconds). How do I tell Asterisk to continue sending RTP packets, while waiting on the DTMF input from the user? I do not know if silence suppression is relevant to this but is currently disabled as Level-3.. Thanks.. scm -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050217/49667f60/attachment.htm