We are trying to determine how to build out an IVR system we are working on. The system needs to be able to handle probably at most 5-10 concurrent calls at peak times. Other times we are just looking for a reliable service. For incoming calls we've been using BroadVoice VOIP and before that VoicePulse VOIP. VoicePulse's IAX service started dropping DTMF inputs that we were processing soon after launch and after a few months of reliable service from BoradVoice SIP, we are experiencing problems catching the digits (simple 6 digit numbers) that people are inputting. The question: Is it unrealistic to think that an all network solution (meaning calls VOIP in to the machine and out from the machine) for this kind of load is doable? Or, would it be better to get POTS phone lines involved with a hardware solution like: http://www.digium.com/index.php?menu=product_detail&category=hardware&product=TDM400P (or a better product)? If the network approach is possible why are we having so much problems just capturing the digits people are pressing? The machine is connected directly to a T1 that has at least 900kb up and down available at any time and is 3GHz with decent specs. What is the problem here? (Asterisk 1.0.9) Thanks for any input, Michael
Michael, Doing an All-Network setup is completely doable but there are many factors to consider. First of all, I didn't see any mention of how many connections it takes before Asterisk starts having difficulty with DTMF. You mentioned that the computer is directly connected to a T1, is it the only computer using the T1 or are there others? Also what kind of network is it? Do you have a good SLA? What kind of packet loss do you experience on average? What is your ping time to the Broadvoice proxy that you're using? Are you using any kind of QoS? Remember that Broadvoice only uses G.711u/a so with RTP + UDP + IP overhead you're looking at ~85kbit/s so at around 9-10 concurrent calls you're going to be pushing it a bit with 900Kbit available bandwidth. You might try turning the SIP RelaxDTMF setting on, that may help, also if you don't have and are not planning on getting any Zaptel hardware, consider using Ztdummy or ZapRTC as an RTP timing source. I know that on the wiki it says that they are really only useful for MoH or MeetME but I've found it to help greatly with audio quality and Asterisk's DTMF detection. YMMV. Good Luck! -Chris
I'm also using Broadvoice and was having a lot of problems with DTMF. I 'm in Ft. Lauderdale, FL and I was (inadvertently) using the dca proxy. When I changed it to use the Miami proxy, my DTMF tones started to work reliably I had done some digging and found various posts on the internet where others using Broadvoice had similar problems and changing proxies seemed to have resolved the issue. You may want to give that a shot and see if it helps. On 10/4/05, Michael Stearne <mstearne@entermix.com> wrote:> We are trying to determine how to build out an IVR system we are > working on. The system needs to be able to handle probably at most > 5-10 concurrent calls at peak times. Other times we are just looking > for a reliable service. For incoming calls we've been using > BroadVoice VOIP and before that VoicePulse VOIP. VoicePulse's IAX > service started dropping DTMF inputs that we were processing soon > after launch and after a few months of reliable service from > BoradVoice SIP, we are experiencing problems catching the digits > (simple 6 digit numbers) that people are inputting. > > The question: Is it unrealistic to think that an all network solution > (meaning calls VOIP in to the machine and out from the machine) for > this kind of load is doable? Or, would it be better to get POTS phone > lines involved with a hardware solution like: > http://www.digium.com/index.php?menu=product_detail&category=hardware&product=TDM400P > (or a better product)? > > If the network approach is possible why are we having so much problems > just capturing the digits people are pressing? The machine is > connected directly to a T1 that has at least 900kb up and down > available at any time and is 3GHz with decent specs. > > What is the problem here? > > (Asterisk 1.0.9) > > Thanks for any input, > Michael > _______________________________________________ > --Bandwidth and Colocation sponsored by Easynews.com -- > > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >-- --- Derek M. A. Lee-Wo Email: (Home) derek@leewo.net (Work) dleewo@voicerite.com Fax: (US) 413-826-0641 (UK) 08701 338414 Family Portal: http://www.LeeWo.net Personal Blog: http://www.DereksPerspective.com "Those who will not risk cannot win" - John Paul Jones