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Help_Please
Anon
2011-Jun-27 10:30 pm
Concerned With Call Log | |
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| Adam4 Premium Member join:2011-05-30 Woodland Hills, CA |
Adam4
Premium Member
2011-Jun-28 2:24 pm
Re: Concerned With Call LogThose call records are "crack attempts" trying to access your ATA, trying to find a resource they can steal calls from. I know this from direct experience, as in the past, users on my system have gotten these "calls" from "asterisk".
The crackers are looking for VOIP servers. Your ATA is a VOIP client, but the crackers don't know that. All they know is that they see a VOIP protocol answering to the standard port like 5060.
But, because your ATA is a client and not a server, there is nothing on your client that the crackers can access and/or steal. For you, these crack attempts are safe and if anything, just a nuisance.
If the crack attempts become incessant and/or annoy you, you could try to change the port your ATA uses from 5060 to something else. I'm not sure, though, how this would affect how VoIP.ms interacts with you. You could try it experimentally and if VoIP.ms stops working, change it back. You could also try putting your ATA behind a firewall. | |
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Help_Please
Anon
2011-Jun-28 8:14 pm
Re: Concerned With Call LogThanks for the reply.
I do indeed have it behind a firewall (With ports forwarded) and I am running the latest firmware on the ATA.
The existence of the "attempts" is annoying, but am comforted by the fact that these entries aren't a result of a misconfiguration on my part.
I Guess I will keep a eye on it and hopefully they cease in the future.
Thank you. | |
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Adam4
Premium Member
2011-Jun-28 9:13 pm
Re: Concerned With Call LogThey may not cease, but it will probably be sporadic. If you could see the logs of the network activity, you'd probably see it's a different IP address every day, as one tries, fails, moves on, and then another comes in and tries... | |
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| | MZB join:2010-11-25 Dunrobin, ON |
to Adam4
said by Adam4:If the crack attempts become incessant and/or annoy you, you could try to change the port your ATA uses from 5060 to something else. I'm not sure, though, how this would affect how VoIP.ms interacts with you. Just to note that there are no problems in using a port other than 5060 when communicating with voip.ms. (And it is probably a good idea to use a different port). The only difference is in IP calling - when caller will have to explicitly specify the port number in the URI. | |
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| | | Adam4 Premium Member join:2011-05-30 Woodland Hills, CA |
Adam4
Premium Member
2011-Jul-4 2:25 pm
Re: Concerned With Call Logsaid by MZB:there are no problems in using a port other than 5060 when communicating with voip.ms. Good to know! Thanks! I didn't know for sure as I don't use VoIP.ms on an ATA, I have VoIP.ms connected to my Asterisk box. | |
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matryx
Member
2011-Jul-12 6:28 pm
long distance with cellphone..I was wondering if there is a way to use my cellphone to use voip.ms long distance. Not sure how everything work but when I had "Call Select" I called a local access number and then I dialed the outgoing number. Just wondering if I can do something similar to this with voip.ms | |
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| DanTou join:2006-10-14 Quebec, QC |
DanTou
Member
2011-Jul-13 6:51 pm
Re: long distance with cellphone..Exactly the same.
You setup voip.ms to filter on your cell number.
With CallerID Filtering, you direct calls from your cell to DISA. When you call home, you willl be able to dial any number.
Small improvement: if you want to be also able to call home from your cell phone...
With CallerID Filtering, you direct calls from your cell to IVR. Configure IVR this way, for example, 1 to ring home, 2 for DISA, 3 for your parents, ... | |
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| MZB join:2010-11-25 Dunrobin, ON |
to matryx
... and keep in mind plans which allow you free calls to/from a selected number. Your DID could be that number. | |
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DanTou
Member
2011-Jul-15 6:18 pm
Re: long distance with cellphone..In my case, it is. $8.15/month unlimited incoming/outgoing on my cell. | |
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| Dee Bee Premium Member join:2005-05-08 St Catharines, ON 1 edit |
to matryx
I also use my voip.ms account (in my case I use my second DID for this purpose) to call long distance on my cell since I don't have a LD plan and have it set up for DISA as DanTou indicated.
I found it works ok, the only issue being a slight degradation on the voice quality of the conversation either on my end or the other party's end.
Has anyone else noticed this and is there any tweaks that may impove the call quality or is this a function of the call going through so many different parties i.e. cell provider-voip provider-landline provider and then the reverse order again? | |
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| DanTou join:2006-10-14 Quebec, QC |
to matryx
Have you used Premium rate or Value rate?
I have tried both and there is a big difference in quality. Also, Premium rate garantees 100% CallerID accuracy (important if you use CallerID filtering).
Premium rate is only for outgoing calls. Incoming calls are always $0.01 per minute and quality depends on the caller's phone/provider/settings.
I find the difference in price worth it. My bill is probably les than $2 a month higher. | |
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| | Dee Bee Premium Member join:2005-05-08 St Catharines, ON |
Dee Bee
Premium Member
2011-Jul-17 6:35 pm
Re: long distance with cellphone..I have premium routing for both DIDs. The distortion isn't very much, when I ask the other party if there is any garbling of my voice they say there is a slight amount but that they can still easily understand me. I find the same sometimes with the other party I am calling. I guess with all the parties the call has to go through: mobile provider-voip provider-landine provider and then back again and the multiple analog to digital and vice versa conversion the quality is not really that bad. Even with a bit of distortion it sure beats paying 40 cents per minute long distance to my cell company. When I call from my home phone through my ATA the call quality is excellent both ways. | |
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Anonymous Caller ID AsteriskHello, I have been using Voip.ms for about 1month 1/2 and since about 4 days I have been receiving some sort of spam. I contacted voip.ms and was refered to this: » wiki.voip.ms/article/FAQ ··· er_ID.3FThis is exactly my problem. I am using a grandstream 286 as my ata and dont wanna disable anonymous calls since alot of legit persons call using anonymous. I can receive these calls 5-6 times a day and of course there is no one on the line. I opened ports 5060 and 10k trought 20k also on my router but that was because I had a random problem where people would stop hearing me talk but I would continue hearing them and opening these ports seemed to have solved the problem. I know that blocking those ports would probably solve this sort of spam. Each time I receive this anonymous call this is what is written in my router log: [LAN access from remote] from 117.120.5.229:40080 to 192.168.1.107:5060, Monday, July 18,2011 06:54:33 [LAN access from remote] from 218.94.82.36:14265 to 192.168.1.107:5060, Monday, July 18,2011 05:22:09 Anyone got this problem and have a different solution. Not alot of support from Voip.ms chat. | |
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| MZB join:2010-11-25 Dunrobin, ON |
MZB
Member
2011-Jul-18 3:32 pm
Re: Anonymous Caller ID AsteriskAssuming the calls aren't coming in from voip.ms (ie. they do not appear on your CDR), then Voip.ms isn't at fault - so you can't really expect them help.
I don't know anything about the Grandstream box, but here are a few suggestions: •Change your ATA to a non-standard port (i.e. not 5060) so hackers will be less likely to find you. •If there is an option on the Grandstream to restrict incoming calls to only the IP address you are registered with, switch it on. (This prevents calls from other than voip.ms being accepted) •Place the ATA behind the firewall, forward the SIP and RTP ports to the ATA, and use firewall rules to only accept packets from the voip.ms server you register with.
The option on an SPA2102 is called "Restrict Source IP" - don't know if Grandstream supports this feature.
Try changing the port first: it may be all you need. | |
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Re: Anonymous Caller ID AsteriskI found this option in my ATA, would that work for youre advice number 2?
Allow incoming SIP messages from SIP proxy only
If set to Yes, the device will ignore any SIP message that does not come from the IP address (Source IP in the IP header) that it is registered to. Default is No | |
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MZB
Member
2011-Jul-19 4:17 pm
Re: Anonymous Caller ID Asterisk(... without reading the manual) I think so. Note that if you are expecting SIP calls from other than voip.ms this will eliminate them too. (You probably aren't). Recommend the SIP port change too. You can get an idea of the number of attempts to probe a port by looking at the SANS Internet Storm Centre. Compare Port 5060 » isc.sans.org/port.html?port=5060with port 5061 » isc.sans.org/port.html?port=5061So unless you need to give out a SIP URI to lots of people, a little security by obscurity is a worthwhile precaution. | |
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craigeb78
Anon
2011-Jul-20 4:02 pm
Overall Good Experience - Just not this timeFirst, Voip.ms quality, service, cost, etc. have been great. I use a sip trunk to a freepbx box for 2 DID's.
My one DID though, I have noticed people report getting busy signals, no matter when they dial.
I opened up a ticket, and recieved a response stating they were able to duplicate the problem, but wanted 2-3 samples to investigate. So I really only have 1 or 2, so I sent them the info for the 2 that I have. And they sent back an email stating the same exact thing, "Please forward us 2-3 examples to help the carrier investigate".
Finally, I have another calling from a different area with the same problem. So I forward this to voip.ms along with the first 2 that were already sent. And this is what I get back:
In this case we need a few samples with the source, you can call your number from 3 different providers.
Provide source (provider and number), date, time and destination (DID number called) from these calls to investigate.
We require 2-3 different samples, it's very important to continue with the investigation with the carrier.
I appreciate the time you take sending these emails.
Let us know the requested information as soon as possible so we may better assist you with resolving the issue.
Seriously, are they not getting the full details of my emails or what? At this point, I'm wondering if there is a language barrier, or even if I sent them 80 examples, they wouldn't help.
Frustated at this point, as I'm unsure what business I'm losing. | |
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Re: Overall Good Experience - Just not this timeI doubt the language barrier is present as these folks speak English. I heard that some DIDs had troubles, and you might have to get a new one (yes it sucks) but I think what they mean is 3 providers (like AT&T, Verizon and Comcast telephone, even Skype, for example). Maybe the samples you sent were from the same provider?
What about calls from voip.ms? Try calling yourself from a sub-account using a softphone (like Blink for example) and see what you get. | |
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Xperts
Anon
2011-Jul-22 7:39 am
voip.ms limits the account without any reasonI had been using voip.ms for almost 3 years now and been paying them about $1000 per month. But of recent I have been having big issues like service unavailable to Uk and Canada routes. Tried the premium route with same result. The support staff tell ms me that your channels limit has been dropped down to 3 from 15 without giving me any reason why is it so? Looking for another voip provider so I can say a final goodbye to voip.ms. | |
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Re: voip.ms limits the account without any reasonHave you tried asking them to raise it back to 15? I would think if they deny your request they would give you a reason.
Can you speculate on reasons that they might not be making money on your account? | |
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Re: voip.ms limits the account without any reasonsaid by ArgMeMatey:Can you speculate on reasons that they might not be making money on your account? Seems very strange. Voip.MS is strictly PAYG for outbound calls.... But the OP is from Pakistan. If he was spending so much money each month, maybe he was reselling the service, or tunneling, or doing something else the company didn't like. A company does NOT get rid of a good customer for no reason! | |
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to Xperts
Xperts were you using this service to make telemarketing calls? | |
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Re: voip.ms limits the account without any reasonsaid by BingoRingo:Xperts were you using this service to make telemarketing calls? At $1,000 a month it's either that, or a LOT of girlfriends!! | |
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1 recommendation |
Re: voip.ms limits the account without any reasonLet's assume he used the value route and only called North America, that means he uses 192,307 minutes/month.
Divide this by 15 channels, it's 12,820 minutes/month per line, or almost 214 hours. Let's divide this by 30 days it means each line uses more than 7 hours A DAY including weekends.
Either our little friend is lying about his monthly bill or he's one hell of a telemarketer (most likely a scammer), which if it is the case, I'm more than happy to see that voip.ms doesn't mind cutting their profit to get rid of its unethical customers.
EDIT: I don't think Xperts will come back here | |
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Re: voip.ms limits the account without any reasonAgree with all of that, and kudos to Voip.MS | |
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Xperts
Anon
2011-Aug-1 8:44 am
Re: voip.ms limits the account without any reasonInteresting to see people do reply on posts here.
First of all, yes I am reselling their service via their reseller portal. If they provide a reseller portal, they want me to resell don't they?
second, before opening an account I mentioned specifically "Can I send Call Center Traffic?" The sales agent asked would it be dialer traffic. I said no and it will be manual dialing. As I said it was home users as well as manual dialing call centers.
I have asked them to raise the limit or give me a reason, as to why the blockage. They don't tell me its because of my call center traffic, just respond as "Our Account Management department denied your request."
Anyway, if there is any good voip termination company out there with a reseller portal, and able to handle call center traffic, please respond.
Thank you all for your comments. | |
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| MZB join:2010-11-25 Dunrobin, ON |
to Xperts
Sounds unfair now you've explained your situation a bit more.
You really should try and get an explanation from voip.ms as to the reason. It could be limited capacity somewhere, a request from one of their suppliers, a maximum risk they want to take on a single customer, maximum risk they are willing to accept if an account drops to zero funds, or complaints about telemarketing calls, or ...
I don't see any channel limit on my account except on DIDs (maximum simultaneous incoming calls), and then it varies with the DID. If the problem is at that level, it may be an issue with the local telecom provider they deal with.
Did you try a support ticket / live chat / telephone? Might be an answer the first level support people are unaware of. | |
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Xperts
Anon
2011-Aug-7 11:22 am
Re: voip.ms limits the account without any reasonyes I have tried to in support chat multiple times and opened ticket but the response was..
"We cannot raise you channel limits for now, you are welcome to try again..".. .
Funny isn't it. they can just tel me we can wont be able to raise your channel limit at all. The issue is at their UK route most, bearable on North America route.
I keep my balance threshold at $50. I lost quite a few clients because of this, as I said in my first post I am researching for other voip providers with similar rates.
uptil now I have found group3 (per sec billing and much lower value and gold rate, Gafachi giving per sec billing on usa route and per min rate lot less then voip.ms. Sippath is another option (their canada route is down mostly but is good on europe route) will be posting more if I find any.
My termination requirement is europe, north america and australia and I am looking for about 50 plus concurrent calls. kindly post if someone can help me in this regard.
Thank you. | |
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to Xperts
If you're a reseller maybe one or more of your customers did something against voip.ms' policies? | |
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bradmajors
Anon
2011-Jul-22 4:14 pm
switch back to AT&T landline?If I transfer my AT&T land line telephone number (and service) to voip.ms, would there be a problem if I later decided to transfer it back to AT&T?
I have been using voip.ms for all outgoing calls for a month without problem. I am just wondering if I can change my mind. | |
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Re: switch back to AT&T landline?There should not be a problem, but keep in mind that AT&T (and Verizon) want their landline (POTS) business DEAD. Here: » arstechnica.com/telecom/ ··· when.arsSo why would you want to switch back to AT&T? ------------------------------ Now, if you want to "test" Voip.MS for inbound calls, why not just get an inbound number (DID) from them? They are very cheap. If later on you transfer (port) your AT&T number, you can just cancel the DID you obtained for testing, or just keep both! | |
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| Dee Bee Premium Member join:2005-05-08 St Catharines, ON |
to bradmajors
I did what PX Eliezer suggested when I was looking for a new phone provider to replace Bell Canada. Before I made the switch I set up an account with Voip.ms and got a DID for 98 cents a month. I tried it for a few months, setting up my ATA and making tweaks here and there until I was satisfied that it was a service capable of replacing POTS for my needs. During the time I was trying out the voip service I kept my Bell service. When I felt I was ready to make the switch I kept the original DID and also had my existing phone number ported over to become a second DID. I use the original DID for experimenting and have it set up so that I can call it from my cell and basically get mobile long distance at a tiny fraction of the mobile providers price. I found this way to be less stressful since I didn't have to worry about having all the equipment and the voip service setup right away as soon as a port over would happen. I could take my time and read (many helpful people here with great posts) and tinker and learn and still use the POTS until I made the switch over. | |
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Cisco SPA3102 or Obi110 for VOIP.MS?Hello, We're going to subscribe to VOIP.MS. Just wondering which of these two adapters, Cisco SPA3102 or Obi110, is the better choice. I would appreciate your suggestions. | |
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Re: Cisco SPA3102 or Obi110 for VOIP.MS?I use a PAP2T so I cannot really do anything for you, but the Linksys ATAs are very popular and it's easier to setup if you look online as there are many tutorials, strong community support. | |
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to taytong888
said by taytong888:Hello, We're going to subscribe to VOIP.MS. Just wondering which of these two adapters, Cisco SPA3102 or Obi110, is the better choice. I am using both the SPA-3100 and the Obi110 with voip.ms. As you may know, the Obi110 is easier to set up, although it seems to give you access to "expert" settings if you're inclined. I don't know about the 3102, but the 3100 is much more complicated to set up. The 3102 has been out for years and there is virtually no manufacturer support, although there is a lot of peer support. The Obi110 is so simple, I haven't needed any support from anybody. If I were setting up from scratch today, the Obi110 would be my choice. | |
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lagz51 join:2011-03-23 Carignan, QC |
lagz51
Member
2011-Jul-29 8:05 pm
Quelqu'un à une idée??Salut à tous!
Mon set-up étant fait, tout allait bien jusqu'à ce que je me rende compte qu'il y avait un petit problème. Je vous explique.
J'ai configuré un IVR pour mon numéro principal à la maison. 3 options lorsqu'il n'y a pas de réponse: 1 boite vocale maison, 2 mon cell et 3 le cell de ma blonde.
Tout fonctionne ... le problème est que lorsque quelqu'un transfert vers l'option 2 ou 3 (donc veut parler à moi ou ma blonde) et qu'il n'y a pas de réponse sur nos cellulaire, et bien c'est la boite vocale de la maison (option 1) qui embarque et non celle de nos cellulaires...!??
Des idées?? Merci! | |
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| DanTou join:2006-10-14 Quebec, QC |
DanTou
Member
2011-Jul-30 10:52 am
Re: Quelqu'un à une idée??Dans DID Settings, augmente le "Dial Time Out in seconds".
Je pense que ça s'applique à tout, même IVR. | |
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lagz51
Member
2011-Jul-31 7:06 pm
Oui t'as raison ca s'applique.. je vais essayer mais je comprends pas pourquoi une fois transféré sur un des deux cell IVR a toujours un incidence .. je fais quelques tests pour vérifier.
L'affaire cest que je veux pas trop l'augmenter car c'est IVR qui répond après x temps pour ensuite rediriger les appels sur les cell ou BV. Si j'augmente trop le temps, je pense que le monde attendrons même plus pour avoir les options... | |
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New Rogers Service causing JitterI have recently upgraded my Rogers Internet to Extreme with the Cisco DPC 3825 Gateway modem (set to bridge mode). Since upgrading I have notice poor call quality with voip.ms (increased jitter but no packet loss). I am located in the east end of Ottawa and I am connecting to the toronto2.voip.ms. I use premium routing but I do not use QoS on my new Linksys E3200 router.
Has anyone else noticed increased call problems with voip.ms on Rogers Internet over the past month?
I am wondering if the new Cisco 3825 is causing the problem. | |
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mau108Mau Premium Member join:2001-10-07 Thornhill, ON |
mau108
Premium Member
2011-Aug-15 12:18 am
Outbound calls show number as 15142285030Outbound calls show number as 15142285030 any idea why?
My phone number is a 416 number but users that are receiving my call are getting the 514 number displayed. I am using a PAP2-T adapter.
Outbound settings in VOIP.ms panel show my number as display number. Inbound calls work fine they get to my phone when users dial our 416 number. | |
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E911 and anonymous caller idHi,
I have my caller id setup as "0000000000" for anonymous caller id. I'm wondering if there's any way to override it to set the correct caller id number when dialing 911. I know that I could setup a speed dial entry for 911 and then override it from there, but I feel it would kind of miss the point of 911 since someone would have to know to use the *75 speed dial code instead of dialing 911...Any clean way to do this with voip.ms ? | |
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anon800
Anon
2011-Sep-25 2:28 pm
Are Incoming "Hunt Groups" supported by voip.ms?voip.ms looks fantastic, but I don't see anything about it handling multiple channels for multiple incoming phone calls. For example, directing simultaneous incoming calls to one phone number and spreading those calls over multiple channels (lines), just as happens with a traditional ILEC T1 PRI connected to a traditional PBX.
Some people refer to this functionality as call hunting or rollover.
Is this part and parcel of what voip.ms provides?
I'm looking at potentially moving about 96 phone lines from a traditional PBX to SIP.
Is voip.ms truly structured, refined and reliably stable to be relied on by business enterprises?
Thanks for your experienced comments. | |
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Memorex
Anon
2011-Sep-27 11:06 am
DISA/Callback from a payphone (via voip.ms).Is it possible ?Hi,
I was wondering if I it's possible to setup my VOIP.MS account to be able to use DISA (or Callback) feature from a payphone. Thanks! | |
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canadadave
Anon
2011-Sep-27 11:09 am
VOIP.msI have used 5 different VOIP services, VOIP.ms is the best service, their Dallas POP is 7 hops from Southwestern Ontario, amazing.
In the past I have used Broadvoice, Magicjack, 8X8, BudPhone, Acanac, this is the best service.
All start out good, over time, the above services become poor voice quality. | |
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DanTou join:2006-10-14 Quebec, QC |
DanTou
Member
2011-Oct-9 2:26 pm
Port number from city not in listCan we port a DID from a city not listed on voip.ms site?
Will that DID still be tied to this city? In other words, would that DID still be local for people calling from this city? | |
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VOIP.ms Incoming Call ProblemI recently ported my parents pots line to voip.ms. The port completed without issue however, I have noted that when an incoming call is made from one of the towns local providers the caller gets a "this number is no longer in service" message. The best I can tell all other incoming/outgoing calls work fine.
I contacted voip.ms whose response was that the provider needs to update their routes. I contacted the provider (South Central Rural Telephone Co) and was advised:
"The VOIP service (Level 3) you have chosen to provide your parents service with has chosen not to get local interconnection with SCRTC to handle local calling service. We have addressed this issue with Level 3 in the past and they have chosen not to consider this. SCRTC has no way to complete the local call to that number. "
Is this something voip.ms could correct by switching to a different provider (I was under the impression that they broker their service out to wholesalers)?
Would another voip company allow this to be done?
Please note:
I also use voip.ms on the same exchange and don't have this problem however I originally ported my pots number to an AT&T GoPhone in an effort to port to Google Voice. I ported from AT&T to voip.ms after finding out that the number was not elegible for Google Voice port.
Any direction from someone knowledgeable on these issues would be greatly appreciated. | |
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How to choose the best voip.ms server?I am using voip.ms over Rogers from Ottawa. I currently use the Montreal2 server but have some occasional jitter issues. My ping times are fastest to the Chicago server and the Tracert to Chicago has the fewest hops (8 vs 11). From the Montreal server when I call some 800# I get routed to the french IVR service and cannot get out. If I switch to the Chicago server will it be assumed I am calling from the US? How do you select the right voip.ms server? Ping speed, fewest hops, keep it local? | |
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omasse join:2004-12-21 Montreal, QC |
omasse
Member
2011-Dec-5 8:14 pm
Delays in voice transmissionI'm trying voip.ms on a PAP2T in parallel with another VOIP landline from Videtron I've had for years. I use the Premium routing, disabled silence suppression, use g.711u, picked a server which has a low ping, and my router has QOS which gives right-of-way to the PAP2T. In other words, I think I've done everything right. The echo test gives me a small delay, but it's in the milliseconds.
However, the tests I've done with other local landline owners give me the impression of calling long distance across the Atlantic in the 80s. Sound quality is okay but there is a delay, conversations are not fluid, and I have to repeat myself. That's not good enough and I don't think I'll be able to switch in these conditions.
Any suggestions?
Thank you | |
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1 recommendation |
Re: Delays in voice transmissionYou may want to see this, done by Voip.MS power user Mango: » www.toao.net/25-linksys- ··· gurationNote specifically: optimal RTP Packet Size setting is 0.02 (that is, 0.020) As well as the other suggestions there.... | |
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| | omasse join:2004-12-21 Montreal, QC |
omasse
Member
2011-Dec-6 9:23 pm
Re: Delays in voice transmissionThanks! I think the RTP packet size nailed it. I actually toned it down further to 0.010, quality is more important than bandwidth to me. It's a little late to call a relative to test with someone else but I'll be sure to try it tomorrow. | |
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npuser
Member
2011-Dec-29 8:26 pm
voip.ms domains - dummping godaddyVoip.ms - is there any talk internally to dump godaddy in protest and move the domains to different registrar? GoDaddy is staunch supporter of SOPA - if SOPA goes through your main site and wiki with any mentions of any unauthorized configuration examples become your liability and domains may be shutdown for mere publication of config samples for good. | |
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adminsly
Anon
2012-Jan-1 1:48 am
what is the best to replace cell plan?hi, my plan with telus will end in january and i wish to save money by going with voip.ms , i know i need to port my number .
what else do i need ? i just need to replace telus by voip.ms for my cell phone. can my number be store on voip.ms server , so i dont need a voip router? or the voip router is a must? for the cell wich provider is the best at the best price , probably only data plan ?
thanks for your in and info !!!
sly | |
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