Thanks for share this information because recently I'masds use VoIP phone system so face this type of problem and not remember how to set up VoIP system. So now I'm easily install VoIP account.
Hi
I know this may be a simple question, but I'm having trouble getting a definitive answer.
If i use VOIP MS and get, for example, a Cisco Linksys PAP2T, can i then just plug my existing home phones into the PAP2T?
I have a regular Panasonic Dect 6.0. It has a base that plugs into a pots wall jack, with 5 other wireless phones throughout the house.
Will the functionality be the same as it is with pots?
IE: If someone dieals the VOIP MS number it will ring on all devices?
Any of the 6 phones can dial out?
Yes. That's exactly what I have except my phones are VTech. There is a bit to learn and you would probably have a system a little less reliable that what you had (with the emphasis on "a little"). About once a year or so, I have to reboot my router or PAP2T to get it working again.
On the other hand, you should get extra features that you didn't have before like having your voicemails emailed to you.
Thanks Phils
Any comments on the Cisco Linksys PAP2T?
If there is a better device, I don't mind spending the money.
This is all coming about because I'm going crazy with all the damn marketing calls and scams.
I read on DSL reports that you can use VOIP MS as a way of eliminating these calls.
The idea being, you direct incoming calls to an answering service.
It has a "white list" for numbers you know are good (family). If the incoming call matches the white list, the call is put through.
Any number not matching the list goes to a prompt. Press "1" to have the call go forward or it's automatically disconnected.
Anyone else do this?
Currently I am using voip.ms and knock on wood so far the service is excellent. I have cable internet 20/10 connection. I am using Cisco SPA122 in bridge mode connected behind the router. I have two voip services, Voip.ms for US and Canada and Callcentric outbound only, for Europe. On the router I forwarded the ports used by voip services. For Callcentric I connected dedicated Panasonic cordless phone. To voip.ms port I connected the entire house wiring.
If you want to port your number to voip.ms do it as per there instructions. DO NOT call Bell in advance. Porting takes a week.
Voip.ms has many servers. It is not true that closes is better. Ping and tracert few and see which one has the best ping and fewest hops. I am in Montreal. For me Toronto, not Montreal server works best.
Those are not technical articles, but loose discussion. Most people here are not experts. The reader is encouraged to read through several posts and makeup his mind.
You have to understand that people come here from different backgrounds and with different knowledge and budgets.
Currently I have a crappy router, but I have what I have and I have to make it work. Perhaps it is not ideal, but it is what it is.
I have read your blog on restricted cone NAT router, and perhaps your right. More reading is needed. If you are more experienced on the subject than educate us, but don't knock us down.
Is this what you were talking about. No I'm not, I just followed his advice.
I thought you were knocking my posts.
Being new to VOIP I take peaces of information wherever I can, and post what works for me.
I don't agree with his recommendation of port forwarding, unless things won't work any other way. SIP scanners are extremely prevalent - operated by hackers looking for unsecured VoIP equipment they can use for personal gain or criminal activity. If your VoIP equipment rejects such calls you may never know you're getting scanned. The attacker however does know your equipment is there, and will keep attempting to crack it for an indeterminate amount of time. If on the other hand your equipment is behind a "restricted cone NAT" router with no port forwarding or DMZ, there will be no indication that your VoIP hardware even exists. I would never say that something is completely "unhackable", but that would be as close to unhackable as you could get.
If a user's router is such that port forwarding is absolutely necessary, then a high SIP port (X_UserAgentPort if you use an OBi ATA) between 20000 and 65535 should be used. Most SIP scanners scan around port 5060. While using a high SIP port does not make you invulnerable to the scanners, it makes it less likely they will find you.
If you do remove your port forwarding, the primary issue to watch for is that you're still able to receive incoming calls. Whether or not the SIP port is forwarded will not affect, for example, audio quality.
I have SIP port on line 1 set at 5060 and line 2 set at 5061. If I turn off virtual server for port 5060 on the router, I can not call in.
On ATA I have Restrict Source IP turned on for both lines. From the documentation: "Restrict Source IP : Permit/prohibit the SPA device from accepting SIP packets from anywhere other than the registered SIP proxy."
From the above statement is the device restricted to only the voip service provider?
Also in the ATA device I have EXT SIP Port field. From the documentation:
"EXT SIP Port : If you want or know the specific UDP port you want the SPA to advertise as the public address port, then fill in the value in this field."
I think this is equivalent to your X_UserAgentPort. So in this field should I put in any number between 20000 and 65535?
As for the router I think it's time to look for a new one. Any suggestions for a reliable budget router?
I am not clear on what EXT SIP Port is for, and the administration guide unhelpfully describes it as "The external SIP port number". I have never needed to use this configuration option.
In any case, you can change the SIP port from its current 5060/5061 to a random number between 20000 and 65535. Since your router requires the virtual server to be set up, you would need to update this with the new port number you choose.
If you do decide to get a new router, I recommend anything you can install Tomato firmware on. Tomato is an excellent firmware that performs particularly well with VoIP. I've been using Asus hardware (specifically the RT-N16) for several years now and am pleased with it - but there are many brands that are compatible.
Anyone here setup an Obi 200 with voip.ms? I have already ported by home phone number from Bell, but now I can't get this setup. The Obi is plugged into my router. I tried the instructions above, the ones at voip.ms. Now when I dial out, I get an error saying my call is rejected due to error 603. And if I try to do an inbound call, I just get a busy signal. Anyone have an easy walkthrough for the Obi and voip.ms?
EDIT1: Dialing out works. But dialing in just gives a busy signal. Baby steps..... Any advice?
Some of Obi's ATA products are very similar to Linksys ATAs so following the Linksys setup (discussed near the beginning of this thread) or searching for similar issues on Linksys ATAs may provide a solution.
I first explained the issue to Obi, and they said to check with voip.ms. Voip.ms told me to check the server I was using. Kinda weird, but I could seem to get the Obi to stick with the same server setup with voip.ms. Anyway, after a bunch of retries, it stuck, and I am up and running. Blocked ONE telemarketer number....a 647 number....and already no more calls at night. Woohoo!
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