Thanks to some of the other threads on this forum, I decided to try and set up my own voip, and signed up with voip.ms. I managed to set up a PAP2T in about 20 minutes, and, amazingly, I was calling across the country immediately. So far, voice quality is excellent. We'll see how it goes over a period of time and I will give an update in six months or so. Assuming all goes well, it begs the question --'Why doesn't everyone do it this way"?
A user in another forum is reporting that voip.ms has duplicated the dialing delay issue. Apparently it's limited to value routing and a temporary fix is to switch to premium routing.
you don't have to switch to premium. You can switch to another server. In my case, I left it as value, but switched to the New York server from the Toronto server. I had the dialing delay issue on both the Toronto and Montreal servers.
One thing I noticed is that when I'm making an outgoing call (value or premium) after dialing the #, there's about 10-13sec pause before I hear any ringing or when I get response when trynig to do echo or 811 test.. anyone else experiencing this??
I just set up a voip.ms account and a new PAP2T-NA, and noticed that ringback wasn't happening until after a quite long delay of 10 seconds or so. This may or may not be the exact same issue as above.
I fixed it here, by going in the Regional/Advanced settings on the PAP2T and setting Interdigit Long Timer to 0 (was 10).
This probably breaks overseas dialing or something (?)
For some reason the PAP2T doesn't ship with a very good dial plan. In fact, lots of its default settings don't make much sense. For example, you will also likely want to set your RTP Packet Size to 0.02 instead of the default 0.03.
Mmm.. that looks good, though I'm rather rusty at dial-plans right now. Will that work with 2-digit asterisk prefix numbers *xx (as opposed to *xxx) too?
Also, I'm in area code 613 , so I suppose there must be a simple way to add 7-digit dialing within 613 to that plan (and have the ATA generate the 613 prefix for me) ?
you will also likely want to set your RTP Packet Size to 0.02 instead of the default 0.03.
Mmm.. that looks good, though I'm rather rusty at dial-plans right now. Will that work with 2-digit asterisk prefix numbers *xx (as opposed to *xxx) too?
Here's some bedtime reading for you, and anyone else who likes tinkering with this stuff This is the new revised Cisco manual and it is VERY useful. (It's also good for other devices such as the 2102 and 3102.)
mlord said:
[e911] seems to be the one sticky point with other providers (eg. voip.ms ?).
VoIP.ms allowed me to order e911. However, I have (fortunately) not yet had a chance to use it. Calling 911 just for testing is very frowned-upon here.
Also, I'm in area code 613 , so I suppose there must be a simple way to add 7-digit dialing within 613 to that plan (and have the ATA generate the 613 prefix for me) ?
( *xxx. | [4689]11S0 | 0 | <:613>[2-9]xxxxxxS0 | 1[2-9]xx[2-9]xxxxxxS0 | 011xxxxxx. | 1900xxxxxxx! | xxxxx. )
- Add 613 to any 7-digit number
- To make calls outside your area code, you must dial 1 first
What's with the xxxxx. at the end of your dial plan?
Here's what I've got now, to allow 7-digit dialing within 613, and both 10 and 11 digit dialing to any area code (including 613):
S:10, ( <:613>[2-9][2-9]xxxxxS0 | 1900! | 900! | 1.[2-9]xx[2-9]xxxxxxS0 | [34689]11S0 | 011xxxxxx.S5 | *xxx.S2)
Odd thing: For some reason, when I try to shorten the 1-900 stuff to simply 1.900! (rather than 1900!|900! it just gives me an immediate reorder tone when I go off hook.
xxxxx. means four or more digits - basically the "almost anything goes" plan as this doesn't match * or # - perhaps I could do [*#x][*#x][*#x][*#x][*#x]. to have a TRUE "anything goes" plan.
For some reason, the period does not perform as expected when it is in the middle of a disallowed sequence - the device apparently ignores any remaining digits in the sequence. I do not know why. Looking at your post though I just realized my sequence blocked 1900 but not 900. Doh!
A couple of things to keep in mind: you may want to change the <:613>[2-9][2-9]xxxxxS0 sequence to <:613>[2-9]xxxxxxS0. This way, the second digit in the telephone number may be 0 or 1.
Does 10-digit dialing work with your plan? I would think the unit would dial out after the first seven digits, if the first was not 1. You may want to remove S0 at the end of the seven-digit sequence to give the caller three seconds to dial an 8th digit.
No, I don't want it preventing me from doing 10-digit dialing, which is what would happen then. Eg. 41612345678 would get treated as a 7-digit number (6134161234) if I did that. Ditto for 819 area code calls across the river here in Gatineau QC.
VoIP.ms allowed me to order e911. However, I have (fortunately) not yet had a chance to use it. Calling 911 just for testing is very frowned-upon here.
Heh, yeah. But voip.ms does have 811 for testing your 911 information delivery. Nice feature, that.
I've read on a different forum that they do indeed provide E/911 for us Canadian customers too -- kinda makes sense, since they were started in Canada, but their sign-up page for E911 only mentions the USA.
Yep. They set up E911 on my 613 DID in under 2 business days. and I'm cutting the wires from Bell later this week.
There just doesn't seem to be a downside to these guys, apart from their very short amount of time in business (only since 2007). Well, that and requiring that payments use a foreign currency.
I would LOVE to use VoIP at the office. Except...VoIP.ms' technical support is very very slow. Cases in point:
First DID assignment: less than one hour.
Second DID assignment: six weeks three days.
First and second e911 activation: one week.
Number port: Nine weeks two days, including at least five hours of downtime. (The port was complete six days after the FOC.)
Serious outgoing call quality issue: 20 hours.
I've emailed them twice to ask why my number port took so long after the FOC to complete, and why there was five hours of downtime. No response, which makes me very suspicious that there was some human error involved. This is really disappointing because the call quality is better than analog, you can practically configure the service to do card tricks, and who else has seven servers?
I get that they're a budget-priced service, but I wish there were like "Value" and "Premium" levels of support like there is for call routing or something
You might want to change the 1900! | 900! string in your dial plan to 19xx! | 9xx! to block other variations on the 1-900 theme e.g. the "976" services available in Canada; http://www.crtc.gc.ca/Eng/INFO_SHT/t1001.htm
oops, you're right, Mango, it would block 905 and other legit area codes and the 2nd part would also block 911 (I completely overlooked that, since I'm not in the habit of 10-digit dialing for LD).
I guess mlord could consider adding 1976x.! | 976x.! then...
Note that it also includes a special entry for preprogrammed voice mail access from the DECT handsets we use, replacing 12345 with the actual voip.ms voice mailbox number.
Speaking of which.. the Vtech DS3111-x DECT 6.0 handsets are the perfect complement to voip.ms (or similar providers): large, visible voicemail buttons/indicators set them apart from competing handsets, and no confusing local answering machines. Pretty amazing functionality when combined with the PAP2T and voip.ms.
All of the voip.ms calling features just work without fuss on them. I'm still amazed by it all!
Sapstar, are you Canadian? If you use one of the US servers and attempt to call a Canadian toll-free number, it often won't work. Try the ca1 or ca2 server instead.
If that doesn't help, what are the symptoms of the problem? Do you get a reorder (fast busy) tone? A message that says the number is not available in your calling area? Or...?
Not sure if it's my dialplan. Here's what I use
(1[2-9]xx[2-9]xxxxxxS0|[2-9]xx[2-9]xxxxxxS0|<*98:09712341>S0|*xxS0|*xx.|[346]11|0|00|4xxxx|<911:14504734686>S0|xxxxxxxxxxxx.)
Even my softphone doesn't want to send the calls...
And to answer the other questions I forgot to specify before I posted... Yes, I'm from Canada, Montreal to be precise. I'm connected on CA1. Thanks to all of you, I could configure my pap2 in the correct way and am now happy with my new voip phone (and provider). A few weeks ago, I had no clue how digital telephony worked. Learned by trial-and-error (many errors !)
thanks again for helping the clueless like me !
J.P.
Edit: just tried it on CA2: same results... but on US4, it works! So now I'm wondering; is it me or Voip.ms ???
The problem is most likely with the company that "owns" the toll-free number. If they are a USA company, with a USA toll-free number, then the number will not work in other countries (eg. Canada) unless they pay for that added service here.
Switching to a USA voip.ms server will route the call from within the USA, thereby permitting USA-only toll-free numbers to work.
It really seems to be an issue with CA1 and CA2, for both Canadian toll free and US ones.
I re-routed to US4 (NY) and everything is OK, both CAN & US works from there. I sent a little note to support about that, just in case.
Playing with all of that, I also found out that their best-effort-routes (055) don't seem to work either. Maybe they're working on upgrading without telling their customers, again.
I know that voip.ms will provide callerID blocking, but is there a way to block those annyoing 866 incoming calls? I get the same number calling maybe a few times each evening.
Also I need some self educating on all things VOIP. A google search leads to wikipedia crap I wish to avoid. Any good self-educating websites?
I know that voip.ms will provide callerID blocking, but is there a way to block those annyoing 866 incoming calls? I get the same number calling maybe a few times each evening.
If you have a PAP2T for your VoIP connection, then you can program that device to deal with the annoying incoming call based upon its CID number.
This feature is under the User 1 and User 2 tabs, and allows setting automatic call-forward for specific incoming CID patterns. EDIT: just leave the forwarding number field blank, and the PAP2T will ignore/not answer the call. Or forward them back to themselves, so they don't tie up the line while being ignored.
I love it! If this idea takes off, we should have a lot of awfully confused telemarketers calling themselves all day
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