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Old 2010-07-08, 10:48 AM   #1066
TheVikking
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Quote:
Originally Posted by thatdaveguy View Post
Tomato WRT54GS
PAP2T

You guys were so helpful before, you guys fixed my connection uptime entirely. But now I need to solve my call quality problems.

About half the time when I call people they say they can barely hear me. Either I'm robotic, cutting in and out, or whatever, it's a huge problem.

However, I've never once had a problem hearing other people.

I've tried QoS on my router which didn't help. I'm very careful not to use my connection at all when I'm using my VOIP anyway.

I've tried Toronto Seattle and Montreal which all produced the same results.

Any ideas?

I'm Glad to see i'm not the only one with these issues. I have been with voip.ms for almost a year now with all premium routing and I love some of the features they offer but during aprox 1 out of every 3 calls I make or receive, the person on the other end complains about not being able to hear me at some point in the call. Sometimes it clears itself up and other times I am forced to call back.

I have had 3 different routers (DD-WRT, Trendnet and now a DLINK DIR-825) and It's always the same problem. I have tried many of the tweaks suggested on this forum and others but the problem still persists.

My internet connection is 10\1 Meg bussiness (Videotron) and I'm not using bittorrent or anything, sometime my PC is even turned off when I have issues, meaning that the PAP2 is the only device using the connection!

I also used to have issues with my PAP2 (5.1.6) loosing it's registration with voip.ms every 10-15 days (seemingly at random). I would have to switch servers from MTL to TO before it would register again. Finally, after adjusting the some of the registration time settings away from the ones suggested by voip.ms it does not seem to be an issue anymore (40 days and counting)

So I guess this can serve as a heads up for anyone else experiencing the same issues, it may be that the real issue is on voip.ms's side.
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Old 2010-07-08, 12:41 PM   #1067
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I also had problems with voip.ms losing registration every 1-2 days, never had such problems with vbuzzer with the same equipment.

Mango's suggestion about changing the UDP keep alive interval seems to be working so far on my WRT54G (tomato) and SPA3102.

However, I still have the same problem at my inlaw's place with voip.ms. They have PAP2v2 (Vonage CYT unlocked) and Netgear WNR834B (stock firmware). They had no problems with vbuzzer before, only when I converted them to voip.ms.

Unfortunately I cannot find UDP timeout or UDP Keep Alive parameters on PAP2 and Netgear.
Is there a solution for this setup?

So I agree there is something different with voip.ms. Anybody know what it is?

Thank you!
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Old 2010-07-08, 03:36 PM   #1068
reeso
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Hi guys!

After 24 hrs I'm able to use my voip.ms telephone. The problem was with my ATA (Linksys spa2102) that I bought from CanadaComputers on Merivale. Really annoying to go there again since I live in QC. But I'm glad it's working.
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Old 2010-07-08, 03:51 PM   #1069
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I had issues with my SPA2102 losing registration after a few days and discovered it was my DLINK DIR-825 router causing the issue. A bought a WRT610N and never lose registration now.

Another thing to check is your speed and jitter:

http://speedtest.net
http://pingtest.net

These two sites will give you a status of your connection.
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Old 2010-07-08, 04:02 PM   #1070
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I agree if I replace both ATA and Router it will probably work better. But the question remains why does the same hardware work with vbuzzer but not with voip.ms?

What's different with voip.ms? Who is sending whom the UDP keep alive, and how does it time out?
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Old 2010-07-08, 04:06 PM   #1071
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Are you using the same codec for both?

Do you have any ports forwarded on your router?
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Old 2010-07-08, 04:09 PM   #1072
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Yes, same codecs.

No, no ports forwarded.
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Old 2010-07-08, 04:14 PM   #1073
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There must be something else different then. Voip is not the same across companies. Each voip provider has different settings to be made. It's quite possible that one of your settings is incorrect. Mango has a list of settings for voip.ms. Have you followed his recommendations.
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Old 2010-07-08, 05:58 PM   #1074
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Hey all,

I thought of sharing my success story about voip.ms and QoS (timely I guess).

I have being using the service for 6 months now, and I noticed that every time I was downloading something, the PAP2T would unregister from voip.ms. Obviously a bandwidth issue.

I have a Linksys router that has a QoS (quality of service) feature. I set it up so that the port where the PAP2T was connected to had the highest priority, and nothing happened. I would start downloading a linux distro or anything big, anything that would take more than a minute to download, and the adapter would drop from voip.ms

Long story short: a gaming router I was looking at was almost $200 in Canada, too much for what it provided. So I assembled a Astaro Security Gateway UTM firewall with the newer Intel D501MO motherboard (two Atom cores, 64 bit, no fans!), 2 GB of RAM and an old sata drive in a mini-itx case all for $260 including shipping; set up VoIP on it, set up bandwidth up & down guaranteed to the VoIP adapter, and voila! It works! Even better, as an added bonus, the Astaro appliance also provides web & mail security, two antivirus, etc. to my home network, stuff that a simple gaming router wouldn’t provide.

I have stress tested it downloading a bunch of podcasts, a linux distro and being on a Skype video call all at the same time, to max out my upstream and downstream bandwidth, and the PAP2T doesn’t unregister anymore. The Astaro box reserves/guarantees a certain amount of bandwidth to my VoIP equipment. Sweet.

The Astaro box running on the Intel D510MO doesn’t break a sweat, CPU usage less than 1%.

I plugged a meter type thing to my whole gear that is 24/7 running and it burns less than a dollar a month. If anybody is interested, I can repeat the exercise and report back.

Highly recommended; no babysitting required once set up (gear: UPS, cable modem, Astaro box, switch, wireless router and the PAP2T ATA).

Overkill? Maybe, but I am pretty happy.

Cheers

BTW, thanks to all on this thread, Mango and FunkyTown specially. I went through the whole 50+ pages (at the time) before dropping Rogers and switching to voip.ms
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Old 2010-07-09, 06:51 PM   #1075
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I noticed that VoIP.ms have changed their description on inbound calls to "Inbound DID Flat Rate". It used to be "Unlimited Inbound DID". I just noticed this new one today. Yesterday it wasn't stating this.
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Old 2010-07-09, 07:44 PM   #1076
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Haven't had a chance to read through the thread yet, will do later tonight but does anyone have problems in the Ottawa area using the Montreal server?. I find it's really unstable compared to the Toronto server.

I tested the ping and hops and this is what I got

13 hops 174.137.63.202 (Toronto)
ping 28ms

8 hops 67.205.74.164 (Montreal)
ping 27ms

Anyone in the Ottawa-Gatineau area have similar statistics?
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Old 2010-07-09, 07:58 PM   #1077
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I'm in Gatineau. This is what I get for Montreal:

Route to montreal.voip.ms (67.205.74.164), 30 hops max
*1 *192.168.0.1 *7 ms *2 ms *2 ms
*2 *64.230.200.62 *11 ms *11 ms *10 ms
*3 *64.230.38.133 *11 ms *10 ms *19 ms
*4 *64.230.170.174 *158 ms *41 ms *15 ms
*5 *216.6.115.29 *45 ms *13 ms *12 ms
*6 *216.6.115.54 *13 ms *16 ms *17 ms
*7 *206.82.135.30 *51 ms *15 ms *19 ms
*8 *67.205.74.164 *14 ms *17 ms *14 ms

*--- end ---

This is what I get for Toronto:

Route to toronto.voip.ms (174.137.63.202), 30 hops max
*1 *192.168.0.1 *2 ms *99 ms *27 ms
*2 *64.230.200.62 *19 ms *14 ms *16 ms
*3 *64.230.38.129 *14 ms *80 ms *11 ms
*4 *64.230.138.126 *25 ms *18 ms *20 ms
*5 *216.191.190.157 *99 ms *17 ms *16 ms
*6 *199.212.169.174 *294 ms *204 ms *204 ms
*7 *216.191.57.178 *205 ms *35 ms *24 ms
*8 *24.102.5.13 *21 ms *21 ms *75 ms
*9 *24.102.5.18 *21 ms *66 ms *21 ms
10 *174.137.63.202 *73 ms *22 ms *21 ms

*--- end ---

My ISP is Bell. This really depends a lot on your ISP as I'm sure Rogers and Videotron would have different results.
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Old 2010-07-12, 01:44 PM   #1078
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What's everyone using in the "callerID number filtering" screen to filter telemarketers?

I was thinking of using something like these (I don't want to use Anonymous CallerID number filter in case people with cell phones call as some cell phones don't have a CNAM record).

- All 800 prefix
- All 888 prefix
- All 877 prefix
- All 866 prefix

Would these rules filter most telemarketers?
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Old 2010-07-12, 04:28 PM   #1079
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It would filter toll free callers which aren't necessarily tekemarketers. The power company, gas, insurance companies, etc often use toll free numbers. I get many telemarketer calls from the 514 (Montreal) area code. I usually filter the first 8 digits of the 10-digit number. That usually fixes the situation. It doesn't take long to get a great list built which means no more telemarketer calls.
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Old 2010-07-12, 08:06 PM   #1080
recneps77
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Telemarketers can also spoof to any number they like.

I've also received a few (legitimate) calls that came from internal extensions at two different organizations - both had spoofed number to prevent you from calling them directly, so it's easily done.
(and it wasn't the main switchboard number, it was like 555-5555, lol)

Someone had the right idea.. I think it was in this thread(?)
Have a quick message "If you're a real person, press 1 now"
By the time their autodialer connects them, you can send them to voicemail or just disconnect. Meanwhile, those who pressed '1' get connected to your phone to ring, or to voicemail, forwarding, whatever you choose.
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