voip.ms direct dial to another account?said by gsar :
Yes, assuming you're using an ATA or softphone that allows it, you should be able to dial (either directly on a softphone, via a speeddial setting in your ATA, or a dialplan entry) a SIP URI that looks like NNNNNNX@sip.us1.voip.ms where NNNNNNX is your 6-digit account number at voip.ms followed by the subaccount extension number.
For example, if you have a Linksys/Sipura ATA, the following dialplan will let you call the subaccount 9 of account 123456 at voip.ms by just dialling #1 on the phone keypad.
(<#1:firstname.lastname@example.org>S0|...your existing dialplan...)
Thanks, that worked, sort of. I am seeing occasional problems with voip.ms where one or the other end hears nothing. When I use that dial plan to call my voip.ms sub account from callcentric, the voip.ms phone rings, but neither end hears anything.
If I use the callcentric quick dial to dial email@example.com, both ends hear each other.
So while I like the voip.ms flexibility, I'm still leery about their quality.
External SIP URIs
said by MartinM:Are you still around MM?
It will be per main account and sub account.
Main account is always 6 digits. If your account is 100000, the URI would be firstname.lastname@example.org.
What will be important is using the right server in the URI. If your device is registered with sip.us2.voip.ms, email@example.com would work, firstname.lastname@example.org would not.
For subaccounts, it's always Your main account, underscore and an alphanumeric name. In this example, we'll assume the subaccount is 100000_martin.
SIP URI would be email@example.com
For internal communications, we already allow subaccounts to call each others by the use of internal extension.
For example, you can creat extension 1. Extensions are prefixed with 10, so you would dial 101 to reach that subaccount. It doesn't matter if another customer has 101 has well, we use account code on dialing to not mix extensions.
I can't get the SIP URI for my main account to function - and have been told by support that I need a virtual DID for this. A sub-account, however, does have an external sip uri?
cross posted to voip tech chat forum
Incoming caller ID number are wrong Hello,
I am not technicly inclined, just regular user of different VOIP services. What bother me with VOIP.ms is that I got incoming caller ID number all wrong.
I have decided to keep my cell phone number and transferred it to VOIP.ms. I got two more cell phones one for US, one for Canada, few phone numbers I use for business, they are with three VOIP providers with different features and pricing. Yet, I want to be reached by my old number, so I opened account with VOIP.ms and now I transfer all calls to one of my phones.
Everything is good except one detail. I am getting caller ID on incoming calls completely wrong. No right numbers. So frustrating!
My intention was to use this set up for business as well. Now I am thinking of shopping around for something more predictable and reliable.
Anyone has had such a problem?
VOIP.ms does not deny the existence of problem, yet they did not fix it since my first call about it in March 2010. They simply do not know what is going on, I guess.
Open ticket, close ticket. Try this, try that.... I am not software tester, nor quality engineer.
Any help? And I do not have such issue with other providers.
Woodland Hills, CA
Re: * features take two? I've never seen such a list, either. I have been assuming that if any of those vertical feature codes work for you, it is not necessarily because VoIP.ms offers that feature assigned to that code, it is because that code is being implemented by your ATA and/or PBX.
For example, you might be able to dial *72 and a number to turn on Call Forwarding, but not because VoIP.ms responded to the *72 code, rather, your ATA remembers the *72 instruction and when you get a call from VoIP.ms, your ATA responds with call forwarding information, and VoIP.ms then handles the call per those instructions.
Another example: You dial *78 for Do Not Disturb. VoIP.ms does not receive that code, your ATA does. When VoIP.ms sends you a call, your ATA responds with the Do Not Disturb info, and VoIP.ms acts accordingly, acting as if your end is busy, etc.
I may be wrong on this... But take a look at the manual for your ATA or PBX and try the feature codes it lists, and see if they work in conjunction with your VoIP.ms DID.
| Edit: I just started an account. Here are the features:|
- Caller ID
- Voice Mail (from what I understand, you can have as many as you want)
- Digital Receptionist (basically a menu when people call you)
- Calling queues (if you're on the line and get another call, they're on hold with little music!)
- Call forward
- Ring groups (I think this is good if you're not always at the same place, two or more numbers can ring no matter where people call?)
- Call-back (you call the number, hang up, the number calls you back with a dial tone. Good if you have My5 and a cell phone, you pay Voip.ms rates)
- DISA (Direct Inward System Access). Here's the description from the site: "Similar to the call-back, The DISA (Direct Inward System Access) let you dial numbers via your VOIP Account from Inbound Calls. The main difference is that a DISA doesn't "call you back". It gives you directly the access to dial a phone number of your choice, similar to what a Calling Card service does. For example, you can create a DISA Entry, and configure one of your DID Number routing to point to a DISA. When you call, you will be prompted a PIN number and the telephone number you want to dial to. You can also use DISA from IVR's (Digital Receptionist), Time Conditions and CallerID Filtering."
- Time Condition (I think this is to direct a call to a DID depending on the time?)
- Caller ID filtering, one of the very interesting feature. If some telemarketer keeps on calling (like newspaper) block their number and they'll get a message saying your number is not in use.
- Phone Book, you can speed dial using *75## and I think it changes the name of the person who's calling on your Caller ID if you want to. For example John Smith could be replaced by "Grandpa Smith".
- e911 ($1.50/month per DID). This is to make sure your call to 911 gets to the right place and they get your address. This is the only feature that costs extra.
There are a few other features and I have no idea what they are for. I think all the Star (*) features are enabled in your ATA depending on the model.
I should get my ATA by mail within 2 weeks I hope, I can't wait to try that.
| |Dee BeePremium
North York, ON
Re: Please help me get my number back! Have you tried calling your Voip.ms number from a cell phone or other landline?
I found this site very good for setting up an ata: »www.toao.net/25-linksys-ata-conf ··· guration
It is by Mango, whom posts quite often in the Voip Tech Chat forum here at dslr:
»VOIP Tech Chat
I am sure if you post in that forum someone will attempt to help. MartinM might be able to steer you in the right direction since he is associated with Voip.ms
Have you logged at ticket with Voip.ms online?
| |N9MDToo busy to chatPremiumReviews:
Boca Raton, FL
| |said by TryingVoip:This issue ... porting back to a RBOC (PSTN, real phone company) ... has been discussed in the VOIP Tech Forum ... and has nothing whatsoever to do with your problem. It is almost impossible at this time to port VoIP or Cellular numbers to PSTN accounts, even if the number originally came from PSTN: »Re: Porting a VOIP to Landline (issue with translations)
After reading a thread, titled "Nightmare Experience porting number back to Verizon...", I am getting the impression that there is a lot that can go wrong in porting numbers.
said by TryingVoip:Your comments define the problem for us. Vonage has an internal Routing Table for its own customers, as do all VoIP providers. This means that calls from one Vonage customer to another are handled within the Vonage server system ... i.e., these calls do not go out over the internet for termination via nationwide Routing Tables as do calls among or between PSTN, Cellular, Beeper, VoIP providers.
All calls from one US Vonage and one Canadian Vonage number (the only Vonage numbers tried) get this message: "The number you have dialed is not in service; please check the number or try again".
Such calls do not seem to reach the voip.ms server. They do not get to fail-over as I have set my fail-over to direct to my voice-mail. But, they do not get my voice-mail message. There is no record of these calls in my voip.ms Call Detail Record.
When a call from Vonage fails, logging into voip.ms shows that my voip.ms adapter is registered. Calls originating from other than Vonage work fine most of the time.
Calls from my voip.ms to Vonage, and others, connect properly.
Do not blame Voip.ms. You said it yourself: "Calls originating from other than Vonage work fine most of the time." That's because your phone number is properly registered with the "external" (non-Vonage and non-Voip.ms) Routing Tables. It is Vonage that is causing your problem! They have not removed your number from their internal Routing Table ... so anyone calling your number from a Vonage account will be directed internally to a "cancelled" Vonage account.
You must demand that Vonage remove the number from their Routing Table.
By the way, as you will see from my Voip.ms review, it is an excellent provider in all respects.
said by BingoRingo:Incorrect speculation! For example, many have transferred a PSTN number to a Cellular provider and then to a VoIP provider within days. There is no "blocking" of multiple portings ... but I repeat that a Voip to PSTN port, even if the number was originally from the PSTN company, is practically impossible.
I'm just speculating here, but maybe for technical or practical reasons you cannot move the number twice within a certain time frame?
»[Other] From Vonage to Google Voice in under 96 hours
VOIP.ms pour les nuls/for dummies Suite à la création de plusieurs "threads" sur le forum Vidéotron portant sur le voip, et surtout voip.ms, on va essayer de se rassembler ici pour répondre aux questions, échanger des trucs et configs, et parler des nouveautés de voip.ms. Essayons un "thread" bilingue mais tout le monde qui n'est pas avec Vidéotron est évidemment le bienvenu!
Aussi, pour que ce soit plus facile pour tout le monde, il serait préférable de partir un nouveau post dans ce forum (et éviter de faire un post de 200 pages comme sur DigitalHome.ca) pour poser des nouvelles questions, et quelqu'un qui écrit un tutoriel en réponse pourrait aussi venir le poster ici.
Following multiple threads on voip and voip.ms on the Videotron forum, we'll try to gather here to share our experiences, answer quetsions, share new tricks and tips and talk about new voip.ms options/features. Let's have a bilingual thread but everyone outside Videotron is welcome of course!
If we get many questions, it might be easier for each question to have its own thread, to avoid a 200-page thread like on DigitalHome (for those who have seen it...!). But if someone feels like making a tutorial to answer one of those question, why not post it here also?