js
Flashlight Enthusiast
I did the **** thing, typed in my password using the letters to numbers translation on the keypad. It worked. I enabled WAN access. But I still can't log in as admin. Only as user.
more importantly,
. . . the sound quality is now . . .
EXCELLENT!
It could be the firmware upgrade, or it could be that I set DMTF to "AVT" and strict holdoff and a value of 70, just as the firmware notes suggested (if using AVT, anyway).
Is there a problem with using "AVT" mode as opposed to "Auto"?
It's funny, in my phone bill for this month I got a notification of rate increases. HAH! Not for me, you *******s!
Good stuff JS, we keep finding new commands! Try this:
723646# [plus password# if set], then press 1# (enable admin access)
For the record, this is my configuration:
CABLE
- cable modem
WAN
- Apple Time Capsule (video server, nat/dhcp on)
LAN
- 2102 (nat/dhcp off)
I was wondering what settings I should fill in to have it setup as you do above. As things are now I have:
I assume I plug an ethernet cable into the lan port of the timecapsule and in to the wan port of the spa-2102. I assume that I don't have to worry about the PPPoE settings on the spa-2102 and that I'll need to set wan to DHCP and make sure that the DHCP server on the spa-2102 is disabled.
Are there any other settings i shoudl know about? Do I need to worry about setting QoS on my router or forwarding ports on the router and/or timcapsule?
As I mentioned above timecapsule has nat/dhcp on; serves 10.0.x.x addresses to two laptops. I simply plugged a spare LAN port on the timecapsule into the WAN port on the SPA-2102. Turned it on and everything worked!
I'm happy that everything went so smoothly but still a bit curious. According to the phone *codes the SPA-2102 has an ip address of 10.0.1.3 (or 10.0.0.3? can't quite remember at the moment...) but I can't access the web-based setup page if I put this address into my browser.
he did that VoIP test (not the speakeasy one, but the more comprehensive one) and these were his results:
QoS was 1 percent (!)
jitter was 1635.1 mS
packet loss was 16.9 percent.
Supposedly his download speed is 11000 or so kbps, as measured using speakeasy, but his download speed was 1.65 mbps using the VoIP measure, and his upload was 305 kbps.
I could NOT frigging believe how bad these numbers were. This can't be right. Either his router is hosed, or he has a bad cable, or his cable modem (because he has high speed cable internet) is hosed, or his ISP is way off their game.
Am I right in being simply flabbergasted by these numbers? I mean, a 17 percent packet loss? That's outrageous! And a 1.6 second jitter? WTF?
I'm curious how you found the thread (this guide), was it some kind of Google search?
What you need to do is turn off this security option. Fire up this phone code:
7932# [plus password# if set], then press 1# (enable access via WAN port)
..enabling WAN side access (telling the box that both sides of the router are on a trusted network) and you should be able to get in.
More soon JS, tonight Lego's took priority:
https://www.candlepowerforums.com/threads/235256
DTMF Tx Mode
DTMF Detection Tx Mode is available for SIP information and AVT . Options are: Strict or Normal. The default is Strict for which the following are true:
•A DTMF digit requires an extra hold time after detection.
•The DTMF level threshold is raised to -20 dBm.
•The minimum and maximum duration thresholds are:
• strict mode for AVT: 70 ms
• normal mode for AVT: 40 ms
• strict mode for SIP info: 90 ms
• normal mode for SIP info: 50 ms
The ATA Admin Guide also mentions another feature which I don't find on my SPA3102:
DTMF Tx Strict Hold Off Time:
This parameter is in effect only when "DTMF Tx Mode" is set to "strict," and when"DTMF Tx Method" is set to out-of-band; i.e. either AVT or SIP-INFO. If a user inadvertently sets the value to less than the default value, the system checks and reverts to the default value. There is no max limit on what the user can set of this parameter. A larger value will reduce the chance of talk-off (beeping) during conversation, at the expense of reduced performance of dtmf detection, which is needed for interactive voice response systems (IVR).
Default is 90ms.
Also, why should the second preferred codex be 729a, and not, say 726-40? I'm trying this out, with the third codex being 726-20. 711u is 64kbs encoding, high quality, and 726 is high quality as well but doesn't put as much bandwidth strain on things, depending on the kbps rate you chose. 729a is certainly easy on bandwidth, but it outright sucks at sound quality.
On another note, it seems to me that your guide should also say to set QoS Dsq to "TBF" = total bucket filter, right? Tom's Hardware reviewed the SPA2000, with an earlier firmware version, and he found significant improvement with TBF enabled.
Second, I also suspect that unless you chose QoS QDisc = "TBF" (instead of "NONE"), you simply won't have QoS working for you. It has to have some method to shape the upstream packets, and "NONE" is, as it says, no method at all = no shaping at all. But, I'm just guessing on this, and am still in the process of testing it out in real world use.
I don't think that your recommendation for how to set the QoS maximum uplink kbps number is correct. According to this qos tutorial the number should be set at 90 to 95 percent of your lowest uplink speed. Now, this is the case if that figure, that setting, is used by QoS to properly fill the pipeline with packets. It needs to know how much bandwidth it has to fill in order to properly shape the upstream traffic. Some routers automatically and dynamically figure this out, but it seems that reports are that they don't do this well, and setting it manually to 95 percent of your lowest uplink speed is best.
On the other hand, if that figure in the 2102 is only limiting the input from the ethernet port to the internet port, and leaving all the rest of the bandwidth free for the VoIP to use, then 80 percent is the number to use--or rather you'd really want to figure out how much kbps you need for your particular codex, I would think, and subtract that amount, plus a margin, from your uplink speed.
But, my experience thus far makes me suspect that lowering the QoS uplink speed does not free up more uplink bandwidth for the 2102 to use. It didn't help at all, in fact.
when I lowered the uplink number, and tested during a phone call, the uplink speed my computer saw was approximately that uplink parameter minus a 200 or so kbps. So, I more than suspect that this paramter is TOTAL LAN to WAN throughput, including the 2102 VoIP throughput. So, you want it set to the lowest uplink speed you are likely to see, minus 5 percent or so as a safety margin.
Also, what is your DTMF holdoff method? "strict"? And what is your strict holdoff time?
Next, DTMF is "Dual Tone Multi Frequency" (or something like that), better known as "Touch Tone" (as opposed to Pulse--i.e. old rotary phones). So, it is all about getting the numbers you press on your keypad to do what you want them to do. AVT (also known as RFC2833) is an out-of-band method. IN-BAND, is, of course, an inband method. This would only work with codex 711u, as 729 doesn't have enough fidelity to transmit these tones well. Callcentric suggests setting the DTMF Tx method to "Auto", although I have had no issues with it set to AVT, either. I currently have mine on Auto. However, if you do have auto set, make sure to have a "yes" for AVT and in-band process settings. (I think those are the ones--those are the defaults, in any case). AVT is preferred overseas, if I'm remembering my recent reading correctly.
In any case, I'm not sure how these settings relate to voice sound quality. I'm sure a beep or two could get generated in the middle of a call, maybe, but other than that, I'm still researching this issue.I found this and thought it was worth posting:
Main point is that if you set mode to strict, make sure the strict hold off time isn't below the minimum listed for the DTMF Tx Method you might use. If you have Auto set, chose the greater of AVT or SIP.