Hello everyone, Having recently gotten Broadvoice inbound DTMF to work (thanks Greg)...I am now running into a frustrating problem...when a call comes in to the BV number via a cell phone (tested with 3 different cell phones; albeit all on T-Mobile) the beginning of the IVR welcome audio is cut off. A call placed via a landline phone over the PSTN to the BV number does not exhibit the problem.
In googling for answers, I came across the recommendation to issue an Answer followed by a Wait of 2 or 3 seconds to let the SIP settle down before playing back audio. I do indeed have this in place; in fact I have been using that advice for quite a while now whether answering SIP or not.... I decided to play with the Wait times to see what would happen and this is what I found: Initially I was using "Wait,3" - in this configuration whether by landline or cell phone, * would answer without any sort of ring indication at all - just silence from the time you finish dialling until asterisk starts to play some audio. I kept increasing Wait by 1 and testing with no difference either in the lack of ring indication, or the audio cutoff when using a cellphone until I reached "Wait,7". At this point, there is still no ring indication, but the audio cutoff when calling via cellphone is fixed. Increasing to "Wait,8" gives exactly 1 ring, then * answers and the audio is perfect calling from cell phone...each additional second of wait time above 8 gives additional ring indication, and perfect audio... 1) Should asterisk really take 7 seconds of wait for SIP to "settle" and not cut off audio? 2) Why would a call from a landline phone have no cutoff problems, even with a "Wait,3"? 3) I expected that there would either be no ringing indication at all, or that it would start immediately (or soon) after dialing. Why is there no ring indication unless you "Answer" then "Wait,8"? 4) If there is no ring indication prior to * answering, and I have no "Ringing" command configured in * anywhere in my extensions.conf, where is this ringing coming from? This has been a bit rambling, I apologize...any feedback greatly appreciated. Marty _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
