Darren Nickerson
2004-May-03 21:35 UTC
[Asterisk-Users] RFD: With echo and other distortion, can ulaw/alaw line quality ever be good enough for faxing?
Folks, I've been following recent discussions regarding echo and echo training with much interest, since it's a problem we've never been able to eliminate here. We're facing two challenges presently, and they may be related (or not): a) Cisco 7960s in the office here echo back to our staff, but the customer hears decent sound. b) We've yet to be able to pass faxes through asterisk, despite trying a number of approaches A bit about our setup: Two analog lines come into the office, into the FXO card of an CAC ADIT 600. These are 'connected' to the TDM interface of the ADIT, and go out to Asterisk via 2 channels of a a 24-channel fxs_ks line into a port of a TE405P. We have two other ports on the TE405P occupied by digital fax boards, connected to asterisk via AT&T 5ESS PRI. And of course there's the 7960 handsets, connected directly to a switch at 100MBps (fdx). Challenge a) ========In terms of looking into a), we started with: http://asterisk.sohoskyway.net/Asterisk_Doc/current/docs-html/x939.html Clearly some of the details are outdated, since the echo canceller options live in zconfig.h now, not a Makefile, but it's a decent starting point. Which gives rise to my first question - which echo canceler should we be using? We have Steve, Steve 2, and Mark 1-3, with an aggressive option with 3 that could make calls scratchy ;-) Is there any conventional wisdom here? Next question ... we have control over gain at the zaptel level, and also at the ADIT level apparently. Which should we use when trying to adjust things with ztmonitor? Next question ... it's not clear to me what the target is with ztmonitor. Are we shooting for tx and rx levels that are balanced. and about 50% of the scale at their max energy? I found that when I called a voicemail service and listened to the auto-attendant that the RX meter of ztmonitor was about 50%, but I found the TX meter fluctuated wildly when I spoke. I could reduce my voice level slightly and it would be very low energy, but with a slight increase in volume it was pegged. The settings I made seemed to do nothing to change this (I stop asterisk, unload and reload all zaptel modules, and restart astertisk between each test). Is this really supposed to work? I've managed to get the connection extremely distorted with some settings, but have yet to make a change that improved the quality or removed the echo. Challenge b) ======= In some ways, this is the most worrying problem for us, since we would love to be able to pass faxes through asterisk reliably, and from the traffic on this list and the fine efforts of Mr. Underwood, it seems many others would too. People have spoken here of the nearly immediate echo cancelation on pure TDM circuits, and so I would have thought we'd escape the echo problems above in either of the following setups: i) Plain ole' HP fax machine plugged into TDM400 FXS card ii) Brooktrout/Eicon fax board connected to TE405P In practice we're rarely able to train with the remote fax machine when dialing to an outside line through asterisk and the ADIT 600. If we do manage to get connected and the connection supports ECM error correction, we can see lots of retransmitted frames which again points to very poor line quality. In contrast, by connecting directly to the analog lines we can send faxes all day with the trusty little HP ... ie: the lines aren't inherently bad. I guess what I'm looking for here is a sanity check ... are we trying to push the limits here, or should this stuff be working much better than this? We're willing to invest some time getting faxing to work, but I'd hate to ask people here to dedicate their time and energy to something that's never really going to work well due to limitations in Asterisk/zaptel technology. -Darren -- Darren Nickerson Senior Sales & Support Engineer iFax Solutions, Inc. www.ifax.com darren.nickerson@ifax.com +1.215.438.4638 +1.215.243.8335 (fax
Darren Nickerson
2004-May-05 12:35 UTC
[Asterisk-Users] RFD: With echo and other distortion, can ulaw/alaw line quality ever be good enough for faxing?
Folks, The silence was deafening ... I had a few private replies but overall I'd have to conclude that most people on this list aren't interested in faxing thru Asterisk. You're all probably jazzed about VoIP and fax is forgotten for now ;-) We have learned the answer to _one_ of the questions I asked below - the Adit channel bank's gain controls can be usefully applied to adjust each of the incoming telephone lines independently, and by combining the Adit's gain setting with Asterisk's, fun and good things can be made to happen ;-) Another interesting observation is that echo can be seen visually when using ztmonitor. Muting one side of the call (the RX side), and speaking from my handset I can see my voice appear stongly on the TX side, and then I can see the echo appearing on the RX side at an attenuated level, but it follows my voice perfectly ;-) I spend half a day talking to myself, and I have to say, for the first time in ages I was running out of things to say! :-) In other interesting news, updating zapel from CVS seems to have made things significantly better. I had CVS from about 4 weeks ago previously, so perhaps something was amiss. We can actually get faxes transmitting now, and although they fail about 30% of the time, this is definitely progress. Another data point, is that MARK3 echo cancellation is audibly worse under our particular conditions. We're using MARK2 with the aggressive option enabled, and while occasional flares of distorted (but attenuated) echo can be heard, it's much easier to have conversations with customers! I'm disappointed nobody commented on the echo cancelation options in the source... I can only assume nobody actually knows the origins of any of them any more, and the other ones are mostly anachronisms that are no longer useful, and MARK2 is the clear winner. Unfortunately, the news from Digium's technical support is not good. Here's what they had to say in response to our inquiry about faxing: "Faxing in asterisk is only at varied levels of support and is definitely not at a production level. However, if you wish to work on this we wouldn't mind at all :-). The source code is of course free to your disposal." I'm not sure I understand that this support rep is saying to me ... we've had much success faxing from one PRI to another with them both connected to Asterisk via a TE400P. Asterisk doesn't even break a sweat. We're rarely getting the theoretical maximum of 33,600 that we can get with a T1 loopback though, so something's slightly amiss, but it's only when we try to go out to the PSTN that things start to fall apart badly. -Darren -- Darren Nickerson Senior Sales & Support Engineer iFax Solutions, Inc. www.ifax.com darren.nickerson@ifax.com +1.215.438.4638 +1.215.243.8335 (fax) ----- Original Message ----- From: "Darren Nickerson" <darren.nickerson@ifax.com> To: <asterisk-users@lists.digium.com> Sent: Tuesday, May 04, 2004 12:35 AM Subject: [Asterisk-Users] RFD: With echo and other distortion, can ulaw/alaw line quality ever be good enough for faxing?> Folks, > > I've been following recent discussions regarding echo and echo trainingwith> much interest, since it's a problem we've never been able to eliminatehere.> > We're facing two challenges presently, and they may be related (or not): > > a) Cisco 7960s in the office here echo back to our staff, but the customer > hears decent sound. > b) We've yet to be able to pass faxes through asterisk, despite trying a > number of approaches > > A bit about our setup: > > Two analog lines come into the office, into the FXO card of an CAC ADIT600.> These are 'connected' to the TDM interface of the ADIT, and go out to > Asterisk via 2 channels of a a 24-channel fxs_ks line into a port of a > TE405P. We have two other ports on the TE405P occupied by digital fax > boards, connected to asterisk via AT&T 5ESS PRI. And of course there's the > 7960 handsets, connected directly to a switch at 100MBps (fdx). > > Challenge a) > ========> In terms of looking into a), we started with: > > http://asterisk.sohoskyway.net/Asterisk_Doc/current/docs-html/x939.html > > Clearly some of the details are outdated, since the echo canceller options > live in zconfig.h now, not a Makefile, but it's a decent starting point. > Which gives rise to my first question - which echo canceler should we be > using? We have Steve, Steve 2, and Mark 1-3, with an aggressive optionwith> 3 that could make calls scratchy ;-) > > Is there any conventional wisdom here? > > Next question ... we have control over gain at the zaptel level, and alsoat> the ADIT level apparently. Which should we use when trying to adjustthings> with ztmonitor? > > Next question ... it's not clear to me what the target is with ztmonitor. > Are we shooting for tx and rx levels that are balanced. and about 50% ofthe> scale at their max energy? I found that when I called a voicemail service > and listened to the auto-attendant that the RX meter of ztmonitor wasabout> 50%, but I found the TX meter fluctuated wildly when I spoke. I couldreduce> my voice level slightly and it would be very low energy, but with a slight > increase in volume it was pegged. The settings I made seemed to do nothing > to change this (I stop asterisk, unload and reload all zaptel modules, and > restart astertisk between each test). > > Is this really supposed to work? I've managed to get the connection > extremely distorted with some settings, but have yet to make a change that > improved the quality or removed the echo. > > Challenge b) > =======> > In some ways, this is the most worrying problem for us, since we wouldlove> to be able to pass faxes through asterisk reliably, and from the trafficon> this list and the fine efforts of Mr. Underwood, it seems many otherswould> too. People have spoken here of the nearly immediate echo cancelation on > pure TDM circuits, and so I would have thought we'd escape the echoproblems> above in either of the following setups: > > i) Plain ole' HP fax machine plugged into TDM400 FXS card > ii) Brooktrout/Eicon fax board connected to TE405P > > In practice we're rarely able to train with the remote fax machine when > dialing to an outside line through asterisk and the ADIT 600. If we do > manage to get connected and the connection supports ECM error correction,we> can see lots of retransmitted frames which again points to very poor line > quality. In contrast, by connecting directly to the analog lines we cansend> faxes all day with the trusty little HP ... ie: the lines aren'tinherently> bad. > > I guess what I'm looking for here is a sanity check ... are we trying to > push the limits here, or should this stuff be working much better thanthis?> We're willing to invest some time getting faxing to work, but I'd hate to > ask people here to dedicate their time and energy to something that'snever> really going to work well due to limitations in Asterisk/zapteltechnology.> > -Darren > > -- > Darren Nickerson > Senior Sales & Support Engineer > iFax Solutions, Inc. www.ifax.com > darren.nickerson@ifax.com > +1.215.438.4638 > +1.215.243.8335 (fax > > _______________________________________________ > 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 >
Stephen Karrington
2004-May-26 13:32 UTC
[Asterisk-Users] IAX Worldwide Termination Service
Hello, We are proud to officially release our worldwide IAX termination service. We offer IAX termination from any Asterisk server or IAX softphone. With your own Asterisk server our system allows you to send as many simultaneous calls as you want for termination using only ONE Diamond account. This works well for call centers, and medium to large PBX deployments. We have a really good reseller program for all the Asterisk consultants out there. This is described on the "opportunity" page at our www.diamondcard.us site. Account signups can be done online with a valid credit card, e-gold.com or goldmoney.com account. We have complete up to the second online call traffic, billing statement and instant recharge features. There are no monthly or signup fees. You prepay for your minutes and thats it. You can get started for as little as $5.00. We have a lot of other value added services that are available with your account but I need to be brief here. Our site is at www.diamondcard.us. For online support or sales questions you can reach us on Yahoo messenger instant chat at dreamtime_net@yahoo.com or ICQ 36768098. Please let me know if you have any questions. Sincerely, Stephen Karrington Dreamtime. net Inc. www.diamondcard.us Corporate Office 101 California Street, 22nd Floor San Francisco, CA 94111-5802 Voice - 877-203-9308 Fax - 310-943-2606 Dreamtime is your global choice for worldwide communication services, viral marketing software and direct sales channel automation.
Stephen Karrington
2004-May-27 12:03 UTC
[Asterisk-Users] IAX Worldwide Termination Service
===8<==============Original message text==============On Thu, 27 May 2004, Chris Sullivan wrote:> Your rates page still has no useful information. Can you guys at least > put some rates up somewhere?They have their rates available at http://www.diamondcard.us/exec/voip-rep-acc-type?secRel=/secondary/corporate&priRel=/templates&secId=corporate It is not user-friendly, but their rates are posted. I prefer the rates published in plain text, so I can copy/paste into my spreadsheet and compare which one offers the best rate for a selected country... Also, avoid visiting their page with a browser other than exploder... their javascript is also not mozzila-friendly... -- Hi Hermann, We will get around to doing another type of price list where its all in one sheet. The link above is not complete. That could be a reason for your error. Please tell me what kind of error you are getting and what browser and version number do you have? Thanks. S _______________________________________________ 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 ===8<===========End of original message text===========