voip.ms discussion

[hopetindall]

As you using premium routing?

Caller ID delivery isn't guaranteed otherwise.

[skycan]

Isn't DID by default is premium. Incoming calls are automatically charged as premium.

[nbegin]

Originally Posted by flyboy320

I have had VOIP.MS for over a year now, but the last couple of weeks I am finding that 1 out of every 2-3 calls is dropping or cutting out after the other party answers. It rings as normal, and then as soon as the other person answers I can hear them for about 1-2 seconds, and then I can no longer hear the other person, there is no busy signal or dial tone, just silence. I am using value routing and it has also happened when someone calls me as well. I have also noticed this on other servers as well (currently on Toronto2, but have experienced it on Chicago as well). Happens local, long distance, and toll free numbers.

Any ideas where I might try to trouble shoot? So far it hasn't happened on my softphone (x-lite) on my laptop when at home, so it seems to point to my PAP2T setup.

I've been having the same problem for almost a year now, primarily on long-distance calls. I raised the issue with support and they wanted me to dismantle my network and just put the PAP2T directly connected to the modem (not going to happen). I haven't really tried to resolve it seriously yet because I don't make a lot of calls. I would like to get it fixed and now that others are having the same problem I poke around with is some more.

[skycan]

Caller ID on DID is working again. Maybe VOIP.ms did something.

[new-guy]

I have just set up my PAP2 a couple weeks ago on voip.ms
This thread is extremely long, but i have covered the first 20 pages or so

Anyways, i noticed my testing had call take about 30 seconds to get placed, which i take is related to my dial plan.
So i have come up with the following, but before i go ahead and put it in, i was hoping for some double checking:
I am in 519 are code, and we make call to 502 country code, so:

(x11|*xx.|0|519xxxxxxxS0|1[2-9]xx[2-9]xxxxxxS0|011502xxxxxxxxS0|011[2-9]x.)

my decoding:
x11 - allow 911, 411, etc
*xx. - allow feature calling
0 - allow call to operator... does anyone do this anymore? ;-)
519xxxxxxxS0 - 519 area code number dial right away
1[2-9]xx[2-9]xxxxxxS0 - NA long distance
011502xxxxxxxxS0 - intl calls to 502 country dial out (they use 8 digit #s)
011[2-9]x. - any other intl calls allowed

Am i leaving anything important out?

[RamKat]

Someone mentioned to me that it is possible to setup your VOIP/ATA account so when you call your VOIP DID from your cellphone, you could have the option to effectively make a call from your cellphone though your VOIP account, which would be a cost saving for long distance calls. Would this be a feature of the VOIP.ms account or the ATA? If this is possible, are there any pointers on how to perform this setup.

[wysiwyg]

RamKat,

That is correct. All you need to do is obtain a SIP client for your smartphone device. Something like SIP Droid for Android or Bria, Media 5, amongst many others for iPhone. Some will also have all the settingsd already there for VOIP.MS, so you just need to enter your username and password. Be sure to create a new Sub-account in your profile for your SIP app connection.

Some things to consider:

1. It is recommended to use the G.729 codec for GSM connections. It is an add-on on all apps.
2. Ensure your GSM data plan allows voip. Some prohibit it (mostly in Europe)
2. When using Wi-FI, ensure you are in a quiet environment and the speed is sufficient. This is often a challenge for public/open hotspots.

[gzink]

I think you want to look at DISA and set up an option to call out and have it come up as an option when your numbers call your DID.
I have it set so if any of our mobile numbers call, the "receptionist" IVR? kicks in and gives us a choice to press a key for DISA and then we can enter a pin and dial out wherever, if my memory is correct. Been a while.

[RamKat]

Thank you wysiwyg, I will certainly give that a try when I finally give in and get a smartphone, at this point I am still way too cheap for that.


gzink, thank you for the DISA keyword, that is just what I needed to get going(I can google therefore I am). After reading a bit about it the other option would be to use callback. I like Mango's idea of an iNUM earlier in this thread. I will certainly try a few options.


EDIT I set my account up using the DISA feature and it is working like a charm - thanks again gzink

[RamKat]

Here is another question to the gurus. Sometimes when I answer the caller wants to leave a message for one of the family members. Instead of me taking the note (and forget about it) is it possible to forward the call to our voicemail system?

[Spreader]

Originally Posted by flyboy320 View Post
I have had VOIP.MS for over a year now, but the last couple of weeks I am finding that 1 out of every 2-3 calls is dropping or cutting out after the other party answers. It rings as normal, and then as soon as the other person answers I can hear them for about 1-2 seconds, and then I can no longer hear the other person, there is no busy signal or dial tone, just silence. I am using value routing and it has also happened when someone calls me as well. I have also noticed this on other servers as well (currently on Toronto2, but have experienced it on Chicago as well). Happens local, long distance, and toll free numbers.

I came on here to ask about the same thing.

Only on inbound calls. Today heard person for a few seconds then cut out. Alot, ALOT of the time I pick up and nothing at all, just a dead connection.

Found this out calling the wife and third time she could hear me. Tested by calling from cell to Voip.ms account and confirmed the problem.

Currently on Toronto2 as well!

[grog]

Originally Posted by Spreader

Only on inbound calls. Today heard person for a few seconds then cut out. Alot, ALOT of the time I pick up and nothing at all, just a dead connection.

I'm having the exact same problem too. Started sometime in the last few weeks. A similar problem I've noticed is that if I dial the number on my cordless phone and then press talk it often fails. Instead I have to hit talk first and wait for the dial tone to kick in, then it dials fine.

Answering calls is the most frustrating though. I'd say this now fails something like 70-80% of the time.

[yoder]

Some of these one-way audio problems have been noted on the voip.ms homepage. I called in today, and they suggested it was a DID vendor issue.
They seemed to be pretty confident about that...

Later that evening I called into Teksavvy and had a similar problem with them not being able to hear me...

My DID Vendor is for Orillia, and I understand Teksavvy is based in Chatham.

Anyone know how to figure out who the DID vendor's are? (CLECs as i understand?)

[hopetindall]

I'm getting 1 way audio on my Toronto and Mississauga DIDs too.

For incoming calls - audio goes into voip.ms; doesn't come back to PSTN.

VOIP says it's a DID vendor issue - but don't give any time to clear.

;-(


Still broken at 22:15 EST 2011-12-08

[grog]

FYI, there's a new update on the VOIP.ms page that says the DID problem is now fixed:

Update Dec 9th, 6:00 AM EST

Our provider informed us that the work scheduled to restore service was completed on their end at 6am: They informed us that their switch vendor has assured that it was completed successfully and that audio issues will no longer be experienced.

Their testing confirms that one-way-audio issue is no longer reproducible. Please re-test and confirm that normal call flow is restored for your SIP circuit(s) that were affected.

We are currently testing many numbers to make sure there are no more issues.

I did a few quick tests calling from my cell phone and so far it seems to be working fine. Hopefully this is now resolved.