voip.ms discussion

[brookz]

Anyone having issues calling toll free #'s today?

Getting this on my asterisk box back from voip.ms
"Dial failed for some reason with DIALSTATUS = CONGESTION and HANGUPCAUSE = 21"

[Jake]

I have two simple questions before I decide to change providers.

CNAM Caller name Lookup - 0.8 cent per query

If I get many calls from the same land line will I be billed each time for Caller name lookup?
If I don't enable CNAM will I still see the number?

I read earlier that cell phones can pose a problem with CNAM. I am not too worried about that though.

[notsure]

Yes and Yes.

And, VoIP.ms is great.

[Mango]

Hey Jake. If you receive lots of calls from a particular number, it may be useful for you to add the number to your VoIP.ms phone book. In that case, you will see whatever name you've entered in the phone book and there will be no fee.

Also, just for the record, even with CNAM disabled you will still see some names, because CNAM is sometimes free in Canada, depending on the phone company of the person calling you. (If you have CNAM turned on, these names will still be free.)

Hope you enjoy VoIP.ms

m.

[Jake]

I currently use them for Long Distance and I am very happy. I cringe when I think of the phone bills I got in the past.

Thanks for the phone book tip. I will do that.

[pjreid]

So I've been using voip.ms now for about a month and a half, and more or less happy with it--certainly the advanced features and reduced cost are appreciated.

I have a Cisco SPA-2102 router/adapter which I have hooked up in bridge mode in front of my WiFi router. I've configured it according to various tips etc that I've found here and elsewhere. My internet provider is Rogers, I have the High Speed Express service.

I'm having intermittent problems--sometimes you pick up the phone, get a dial tone and dial a number, and it never connects (it's not a dialling plan problem as most times it works just fine). Other times you place a call and it connects, but one side or the other can't hear the other person; sometimes this is only temporary, other times you have to hang up and reconnect the call. It happens for both incoming and outgoing calls.

I've set things up to use Value routing for outgoing calls, would this cause any of the issues?

My wife uses our home phone quite heavily (according to the CDR report, nearly 40 hours of talk time this month--most of which would be hers), and isn't pleased with these problems, so I definitely want to try to fix things up.

Any tips you can offer would be appreciated.

Edit: I'm using the toronto voip.ms servers, if that makes a difference

[Telex]

I reviewed 139 pages of posts and made a crib sheet of the topics that were important to me (dial plans, CNAM usage, how to use sub-accounts, IVR configurations, SIP URIs, etc). I now need to put all of that great information into practice!

FWIW, I'm a new voip.ms customer using an Obihai Obi110. My router has no QoS control so I'll be replacing it shortly with an ASUS device. So far I've only been testing the service and still have a PSTN line (Primus). However, as soon as I add e911 to my voip.ms DID, I'll be cancelling the PSTN line. I won't be porting my existing number and will simply supply friends and family with the new voip.ms DID. I will probably leave it unregistered.

The voip.ms wiki is a great resource but I've found this forum to be exceptional for discovering ways to put all of voip.ms's services to good use. A special thanks to Mango and FunkyTown for contributing untold hours of your time!

[RamKat]

Since I am in the process of changing from a DSL ISP to a Cable ISP, I expect to switch my ATA between the two routers and voip.ms will therefore detect that my account will be jumping between ISPs, as I will be working on issues with the Cable setup and switching back to the now stable DSL setup when I am not. I am aware that my ATA will be assigned different local LAN IP addresses by the respective routers. Is there anything else that I should be aware of in this process? (such as voip.ms potentially blocking my account etc).

PS My apologies if this issue has been discussed before but a search did not reveal anything along these lines.

[Techluvr]

Originally Posted by pjreid

So I've been using voip.ms now for about a month and a half, and more or less happy with it--certainly the advanced features and reduced cost are appreciated.

I have a Cisco SPA-2102 router/adapter which I have hooked up in bridge mode in front of my WiFi router. I've configured it according to various tips etc that I've found here and elsewhere. My internet provider is Rogers, I have the High Speed Express service.

I'm having intermittent problems--sometimes you pick up the phone, get a dial tone and dial a number, and it never connects (it's not a dialling plan problem as most times it works just fine). Other times you place a call and it connects, but one side or the other can't hear the other person; sometimes this is only temporary, other times you have to hang up and reconnect the call. It happens for both incoming and outgoing calls.

I've set things up to use Value routing for outgoing calls, would this cause any of the issues?

My wife uses our home phone quite heavily (according to the CDR report, nearly 40 hours of talk time this month--most of which would be hers), and isn't pleased with these problems, so I definitely want to try to fix things up.

Any tips you can offer would be appreciated.

Edit: I'm using the toronto voip.ms servers, if that makes a difference

pjreid:

We are having the same problems. My wife is also annoyed at the inability to dial out calls, and having the audio drop out during a call. I'm using a cable internet connection with a 10/.512 speed connection; voip shouldn't be having any problems.

I'm glad to see that I'm not the only one experiencing problems. Well, no I'm not glad you're having problems, but it indicates the problem is more widespread than just my connection or setup. Have you contacted voip.ms to see if they have any suggestions? Now that I know it's not just my connection, I'll do that too.

[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.

[mojonba]

Im trying to set up a way to quick dial my sister. She has a voip.ms subaccount with no DID. Voip.ms has an extension feature when you dial 1 + the extension no. but it only works when both account are connected to the same voip.ms server. Due to location we are connected to different servers (chicago & tampa). As a workaround, I thought I could dial my sub account SIP URI which according to voip.ms is XXXXXXY@chicago.voip.ms where XXXXXX is my account no. and Y is the extension no. Such extension works fine whenever I am dialing from another provider but it doesnot work when dialing from within voip.ms. Any ideas on how can I call a subaccount connected to a different server?

[RamKat]

Originally Posted by RamKat

Since I am in the process of changing from a DSL ISP to a Cable ISP, I expect to switch my ATA between the two routers and voip.ms will therefore detect that my account will be jumping between ISPs, as I will be working on issues with the Cable setup and switching back to the now stable DSL setup when I am not. I am aware that my ATA will be assigned different local LAN IP addresses by the respective routers. Is there anything else that I should be aware of in this process? (such as voip.ms potentially blocking my account etc).

PS My apologies if this issue has been discussed before but a search did not reveal anything along these lines.

To answer my own question above. I switched the ATAs RJ45 from the DSL router to the Cable ISP router (running DD-WRT) in the basement and when I got back to the main level the phone was back up and running. I have not had to switch back since.

[lawman]

Originally Posted by mojonba

Im trying to set up a way to quick dial my sister. She has a voip.ms subaccount with no DID. Voip.ms has an extension feature when you dial 1 + the extension no. but it only works when both account are connected to the same voip.ms server. Due to location we are connected to different servers (chicago & tampa). As a workaround, I thought I could dial my sub account SIP URI which according to voip.ms is XXXXXXY@chicago.voip.ms where XXXXXX is my account no. and Y is the extension no. Such extension works fine whenever I am dialing from another provider but it doesnot work when dialing from within voip.ms. Any ideas on how can I call a subaccount connected to a different server?

mojonba,

Did you use the phonebook in the voip.ms account online? And set up the speed dial option for your sister's external SIP URI address? That way you can quick-dial her from your phone to her using the alias. I don't think it should matter which server she's registered on. If this works, you just have to set it up for her vice versa.

[mojonba]

mojonba,

Did you use the phonebook in the voip.ms account online? And set up the speed dial option for your sister's external SIP URI address? That way you can quick-dial her from your phone to her using the alias. I don't think it should matter which server she's registered on. If this works, you just have to set it up for her vice versa.

I did, all I got was a busy tone.

[skycan]

I have been using VOIP.ms for a couple months. Recently I noticed incoming call to a DID number from a Rogers cell phone shows up as unavailable. Calls from same cell phone to another cell phone callerID shows up correctly so does not look like a Rogers problem.

I checked my calls detail record, it looks like it was working fine at least until 10/20 and first problem showed up on 10/30.

Any idea if this is a VOIP.ms problem, should I raise a ticket ?

Thanks