[ckoral]

I have been using Vonage for about 5 years now. I recently started looking for an alternative solution. Can anyone offer a tried and true simple route to replace vonage?

requirements:

-number portability - nice, but not necessary
-voice mail - wav files (or I can use grand central)
-standard home phones have dial tone in house
-call waiting
-call id
-USA calling
-don't care about 911
-need a phone numberj
-vonage like quality

I pay after taxes about 19 bucks for my vonage 500 min plan. I am happy with it, but I would really like to get it down to less than 10 bucks a month (or even less).

I tried skype. I bought a philips voip841. I thought the rj-11 jack on that box would distribute a dial tone to my standard home phones. It did not, it was for combining telco company and voip. I learned this after I purchased.

A buddy of mine loaned me a PAP2 unlocked linksys. I don't know if that is a good box to start with, and I will have to buy my own anyway.

What I am looking for is a review or recommendation from someone who has similar needs.

Please let me know.

[boatman]

If you have a Grand Central number then you are in a good position to "roll your own" VoIP phone service.


1. Configure the PAP2 using the details here but don't get your Gizmo5 account just yet because you may want to choose your Gizmo5 user name to be the same as your VoIP.ms user name.


2. Get your Voip.ms account (you don't have to fund the VoIP.ms account until you want to make calls). I recommend VoIP.ms because of the very reasonable cost, 1 cent/min to USA, and they have an excellent customer interface with all the bells and whistles any VoIP geek could wish for. This includes:

- Static IP Authentication (so VoIP.ms can share same line with Gizmo)

- Option to set outbound caller ID to whatever you want the other party to see.

- Choice of normal or premium call routes. The high quality route is only about 25% more expensive than normal route, so 1.25 cents for the premium US calls.

- Multiple SIP server locations, four in USA, pick the closest one for lower voice latency. Closest one to me is Los Angeles with under 45 millisecond ping time.

- Email notice when dollar balance drops below a level set by the user.


3. If you don't already have a Gizmo5.com number, go there now and get one. If you choose your Gizmo5 user name to be the same as your VoIP.ms SIP account name, then you can use standard SIP authentication at both places. If you want your Gizmo5 user name to be something different that is OK too because you can just use SIP Static IP authentication at VoIP.ms (but only if your IP address does not change often, as SIP Static IP authentication requires that you keep your current IP address in your web account at VoIP.ms).


4. Once you have your Gizmo5 account, forward your Grand Central number to your Gizmo5 1747xxxxxxx number.


Often it's conveinient to use just one line on the PAP2 for both inbound and outbound calls, therefore your VoIP.ms and gizmo5.com SIP accounts will share the same line and you can use a single line phone. To do this you need to have a special dial plan, see below.

================================================

This dial plan will route outbound PSTN calls to VoIP.ms while allowing the ATA to register and receive calls from the Gizmo5 network (including inbound Grand Central calls). You will need to use the SIP Static IP authentication feature at VoIP.ms or make your Gizmo5 user ID and password identical to those at VoIP.ms.

To place a PSTN call through VoIP.ms just dial normally. To place a PSTN or other type of call through Gizmo5 begin dialing with #1. To reach another Gizmo5 user just dial their 1747xxxxxxx number, you can also direct dial to Gizmo5 411, 611, and 1222xxxxxxx conference system.


Note: Press # as an enter key after dialing any number.


Toll free numbers beginning with:
8 are sent to fwd.pulver.com (dialing example: 8004267378)
18 are sent to sip.tollfreegateway.com (dialing example: 18004267378)

Toll free calls can also be sent to:
SipBroker - dial beginning with #0 (dialing example: #0 18004267378)
Gizmo5 - dial beginning with #1 (dialing example: #1 18004267378)
VoIPmich.com - dial beginning with #2 (dialing example: #2 18004267378)
sip-happens.com - dial beginning with #3 (dialing example: #3 18004267378)


Before installing this dial plan you must do the following:

1. Find the line S0<911:0012085551212@us3a.VoIP.ms>| and change 12085551212 to the number of the emergency dispatch center in your area.

Note that VoIP.ms has SIP proxies in several locations in North America and one in the UK. You should use the closest one based on ping time. us3a.VoIP.ms is in Los Angeles. Test ping time to all their proxies before selecting one. Once you have selected the closest proxy make the changes in the dial plan below.

2. Change all occurrences of 208 to your area code.

3. Put the entire dial plan on one line by removing the carriage returns after each | character.

L:60,S:60,(*xxS3|
<:*1>800[2-9]xxxxxx<:@fwd.pulver.com>|
<:*1>866[2-9]xxxxxx<:@fwd.pulver.com>|
<:*1>877[2-9]xxxxxx<:@fwd.pulver.com>|
<:*1>888[2-9]xxxxxx<:@fwd.pulver.com>|
<:164164>1800[2-9]xxxxxx<:@sip.tollfreegateway.com>|
<:164164>1866[2-9]xxxxxx<:@sip.tollfreegateway.com>|
<:164164>1877[2-9]xxxxxx<:@sip.tollfreegateway.com>|
<:164164>1888[2-9]xxxxxx<:@sip.tollfreegateway.com>|
[46]11<:@proxy01.sipphone.com>|
S0<911:0012085551212@us3a.voip.ms>|
1747xxxxxxx<:@proxy01.sipphone.com>|
1222xxxxxxx<:@proxy01.sipphone.com>|
<#0,:>[x*][x*].<:@sipbroker.com>|
<#1,:>[x*][x*].<:@proxy01.sipphone.com>|
<#2,:>[x*][x*].<:@tf.voipmich.com>|
<#3,:>[x*][x*].<:@tollfree.sip-happens.com>|
<#9,:>[x*][x*].<:@us3a.voip.ms>|
<:00>1[2-9]xx[2-9]xxxxxx<:@us3a.voip.ms>|
<:001>[2-9]xx[2-9]xxxxxx<:@us3a.voip.ms>|
<:001208>[2-9]xxxxxx<:@us3a.voip.ms>|
00xxxxxxx.<:@us3a.voip.ms>|
<011:00>xxxxxxx.<:@us3a.voip.ms>|
[x*][x*].)

[ckoral]

So I need to register with dyndns.org or something similar for the static IP part?

[boatman]

No, the ATA will be registered on the Gizmo5 network (proxy01.sipphone.com) for incoming calls, including calls from Grand Central. Static IP Authentication is only needed for outbound calls via voip.ms, and even then only if your Gizmo name does not match your voip.ms name.

[gsar]

Thanks to boatman for the nice summary, but there is a hitch in making it work without IP authentiation. Gizmo userids apparently *have* to start with a letter, but voip.ms userids are all numbers.

Anyone know of a way to get gizmo usernames that start with numbers?

[zulphie]

Hi,
Vonage is toooo expensive. I just start using following voip service. They have very good quality at very cheap rates.
www.786tel.com

Good Luck

[ckoral]

us & canada is 20 bucks a month. I fail to see how that is any cheaper than vonage. Then on top of it you must purchase a phone number for 9 bucks.
Am I missing something? Is it 20 bucks a year instead of a month?

[boatman]

You won't find the best price in one of those Vonage-like package deals which include inbound phone number (DID) and some amount of outbound calls for one monthly fee.

If you want the lowest cost, you will have to buy your inbound number separately from your outbound minutes. Inbound phone numbers start at $2.95/mo at callcentric.com, also check les.net. You should expect to pay about 1 cent/min for outbound calls to USA, less to Canada.

[gsar]

Here is an alternative to boatman's recipe that has been working well for me so far: instead of getting a Gizmo account, go for a Voxalot (free) account. They allow you to choose the userid (all numbers) and if you get lucky you will be able to get the same userid as the voip.ms one.

However, even picking the same userid is not necessary with voxalot. Just add voip.ms as one of the outbound providers (voip.ms officially support voxalot) and route outbound calls other than toll-free through voip.ms or other providers that you can add to voxalot over time as you discover better providers.

If you bought a DID with voip.ms you can point it at your voxalot account in the voip.ms customer portal using the SIP URI yourvoxalotaccount@us.voxalot.com.

Combine that with an IPKall or similar DID that points at your voxalot SIP URI, and you've got a multiple-inbound-and-outbound-providers-on-one-line solution that Just Works, and has all the useful frills such as voicemail. Definitely much better than the various call forwarding/static-ish IP address hacks that are out there for combining providers, and simpler than running your own Asterisk. And you can do it all using a very basic single-line ATA, SIP phone, and/or soft phone!