Review of voip.ms | DSLReports.com, ISP Information

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Review of voip.ms (voip)

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Six Month Rating

Reviews:

241 reviews (215 good) (5 bad)

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Review by xraminx

member for 5 years, 245 visits, last login: 90 days ago
updated 200 days ago

Oakville,ON
Contract price not specified.
"Cheap. Flexible."
"Tech support is pathetic."
"Don't count on their tech support"

Web-site:

Ease of Installation:

Call Quality:

Reliability:

Tech Support:

Value for money:

(ratings match consensus)

I bought a DID number from them and tried to set up DID based inbound routing on my PBX.

Turns out their servers do no include the DID number in the SIP request making it close to impossible to set up DID based inbound routes on your PBX. (Edit: it does, you need to set the SIP URI properly).

Their tech support is not knowledgeable. The first response I got to my detailed question was like the tech support was brushing me off, hoping I would resolve the issue myself. (And I resolved the issue myself at the end.)

Their support system is ridiculous: I gathered some valuable SIP debug information from the Asterisk log and pasted it in the support ticket but when I hit the "Submit" button, some crucial information was stripped from the log I pasted, rendering it useless.

Use their service if you think you can solve the problems yourself. Don't count on their tech support. You are on your own.

One serious issue is that it takes a long time for an incoming call to get connected. They have to fix this. The person calling your DID number will have to wait a long time before they hear the ring.

Comments:

Mango www.toao.net join:2008-12-25 Alberta kudos:11	So... ...did you remember to set "Device type" to "IP PBX Server, Asterisk or Softswitch"?
	permalink · 2012-10-29 17:00:25 · ·

xraminx

join:2008-04-26
Oakville, ON

Reviews:

·voip.ms

Re: So...

Yes I did. sip debug shows the TO: field is missing the DID number:

To: <sip:my_pbx_ip_address>

It is supposed to look like:

To: <sip:my_did_number@my_pbx_ip_address>

permalink · 2012-10-29 18:11:59 ·

Mango www.toao.net join:2008-12-25 Alberta kudos:11	Re: So... Testing it and will post back shortly.
	permalink · 2012-10-29 18:15:15 · ·

Mango
www.toao.net

join:2008-12-25
Alberta
kudos:11

Reviews:

·Anveo

·Shaw

·AcroVoice

·Callcentric

·callwithus

·voip.ms

·FreePhoneLine

·TELUS

I set up a sub account with password authentication, configured my Asterisk server with its credentials, registered, routed a DID to it, and it worked. I didn't test IP authentication because I don't have a server on port 5060.

Another way to accomplish what you want is to set up a SIP URI (from the DID Numbers menu). Use the syntax {DID}@my_pbx_ip_address. Note that {DID} is literally {DID} and will be replaced with the DID number that your caller has dialed. Then you route your DID to the SIP URI you created.

Let us know if either of those scenarios works.

permalink · 2012-10-29 18:24:41 · Print ·

xraminx join:2008-04-26 Oakville, ON 2 edits	Re: So... Thanks Looks like I found the solution 3 minutes later than you
	permalink · 2012-10-29 18:34:41 · ·

Mango
www.toao.net

join:2008-12-25
Alberta
kudos:11

Reviews:

·Anveo

·Shaw

·AcroVoice

·Callcentric

·callwithus

·voip.ms

·FreePhoneLine

·TELUS

Re: So...

I'm glad to hear your setup is working now. I can't address your issues with their technical support, however you might want to consider revising your other ratings, particularly "Ease of Installation", since the solution to your problem is clearly documented.

I notice you've rated Call Quality and Reliability at 50%. If you are having audio quality issues and downtime, please let us know if you would like help troubleshooting it.

permalink · 2012-10-29 18:51:56 ·

xraminx join:2008-04-26 Oakville, ON	Re: So... I updated the review.
	permalink · 2012-10-29 19:45:24 · ·

Mango www.toao.net join:2008-12-25 Alberta kudos:11	Re: So... Thank you! And feel free to let us know if you have any other questions we may be able to answer. Have a great day!
	permalink · 2012-10-29 19:49:59 · ·

xraminx

join:2008-04-26
Oakville, ON

Reviews:

·voip.ms

Okay I figured out what the problem is and the DID based routing works now.

The SIP URI that you forward your DID to, must look like:

1{DID}@my_pbx_ip_address

But I had set it to

my_pbx_ip_address

One serious issue is that it takes a long time for an incoming call to get connected. They have to fix this. The person calling your DID number will have to wait a long time before they hear the ring.

Nevertheless, as I mentioned before, you cannot count on their tech support. You are on your own.

permalink · 2012-10-29 18:27:30 · Print ·

MartinM VoIP.ms Premium,VIP join:2008-07-21	Re: So... I do not agree that you can't count on our technical support. There's a lot of documentation in our wiki, and "my_pbx_ip_address" is clearly something you need some technical knowledge. Let me know if there's anything else we can do to help you get you started with your first PBX. -- Martin - VoIP.ms
	permalink · 2012-10-29 20:33:09 · ·

xraminx join:2008-04-26 Oakville, ON Reviews: ·voip.ms 1 edit	Re: So... MartinM, you are entitled to your opinion. I wrote about what I actually experienced. Tech support was useless. They have access to all my settings on their servers. I also sent them detailed logs from my Asterisk CLI. They could not find the problem. Here is the issue number. Go check it out for yourself: Ticket #37355 BTW, it's not my first PBX.
	permalink · 2012-10-29 20:45:59 · ·

MartinM
VoIP.ms
Premium,VIP
join:2008-07-21

Re: So...

I will take a loot and try my best to help you, from a quick look at the ticket is that you tell the technician that he used a crappy canned response and that our ticketing system trunks parts of your logs. That is normal, I would recommend perhaps attaching the log in a text file, as is the standard in the industry, and perhaps, refrain from using juvenile language.

Regards,
--
Martin - VoIP.ms

permalink · 2012-10-29 20:50:22 ·

xraminx

join:2008-04-26
Oakville, ON

Reviews:

·voip.ms

Re: So...

Yes. I did say the first answer was a crappy canned response because it was. And I feel insulted to get a useless response to a detailed question I asked.

If you do not consider the first response a crappy canned response then you have no idea what an acceptable customer service is like.

And the problem is resolved by myself so you don't need to look further.

I suggest you improve your tech support instead of defending it.

permalink · 2012-10-29 20:55:21 ·

MartinM VoIP.ms Premium,VIP join:2008-07-21	Re: So... At the end of it, it is working and we had nothing to fix, I understand your frustration in not getting an immediate answer in fixing your error. For that, I deeply apologise and I will make sure the technician who attended your case gets interviewed about your case. I'm happy that the service is working as intended, Best Regards, -- Martin - VoIP.ms
	permalink · 2012-10-29 21:26:21 · ·

xraminx join:2008-04-26 Oakville, ON Reviews: ·voip.ms	Re: So... And I appreciate you looking into this. What you need to fix is your customer service. Looking forward to using your service and buying more DID's from you. If in the future I use your customer support and I am satisfied, I will certainly change my review. Regards Ramin
	permalink · 2012-10-29 21:30:56 · ·

MartinM
VoIP.ms
Premium,VIP
join:2008-07-21

Re: So...

Glad to hear that.

Make sure you include your traces in an attached text filedocument (SIP Traces contain HTML QUOTES). The best format is a .pcap file that can be parsed properly, like wireshark, if you don't know how to do that you can copy the asterisk output but as mentionned, in a text file.

Also, always make sure your account and subaccounts are set as PBX and not ATA Device, as documented in our wiki, if you want to have the DID included in the invite string.

Regards,
--
Martin - VoIP.ms

permalink · 2012-10-31 12:56:05 · Print ·

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