I have a Vonage virtual number for a couple of friends in Great Britain which works great. When my internet goes down, all of my Vonage calls are forwarded to my T-mobile cell phone. My connection is wired; no WiFi at all.
Does anyone know if my UK friends would be charged for making an international cell call when they call me on my UK virtual number using their landline when my internet is out and they ring through on my mobile phone, please ?
I've called Vonage; they don't know. I have tried getting an answer from the UK landline provider (Virgin Telephone) -- still waiting (and waiting ...!) to hear.
(FWIW, when I tried calling UK landline using the Vonage extension app on my cell phone with NO wiFi, my friend was charged the equivalent of $40 USD .... for a 11 minute call. Needless to say, I'm worried they would get charged with this virtual number/forwarding).
Thanks very much in advance.
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[Other] Vonage virtual number question
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[Voip.ms] Calling queues setup with CSIPSimple
Hi,
Just signed up for VOIP.ms to use for my small business and so far it's been great except for the fact that I can't figure out the calling queue setup. I installed CSIPSimple on my Android phone and setup my subaccount there which can make outbound calls using the app. But when customers call the number and go to the calling queue, it doesn't ring my phone at all. I dial into the queue using my subaccount and it says I'm already logged on. But when callers dial into the queue, I'm unavailable. I asked VOIP.ms tech support about this and they said it has to be on the same server. The problem is I am on server 8 for the calling queue DID but I can only select servers 2,3,4 on CSIPSimple for my subaccount. What is the issue here? Why can't I select the same servers or why aren't all the options there?
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[Voip.ms] Using VPN with voip.ms
Anyone know how to use voip.ms with a vpn service? My phones lose service and I can't even login to voip.ms website with vpn service turned on.
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[Anveo] Email to SMS
Ok, I'm having a blonde moment : how can I send emails to SMS (my Anveo number).
From what I read, the feature is supported but email to 1234567890@anveo.com fails.
What am missing?
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Voxbeam offering 10 free DIDs
The promo came by email; they granted permission to repost it in this forum: "We are giving away 10 free On-net USA or UK numbers. For US, Boston and New York are available. For UK, London 0208 / Manchester or Sheffield are available."
You can get up to 10 numbers in one city, or mix and match across the available locations. The service is completely free, with no setup or monthly charges and no charge for incoming calls. Each DID is hard-limited to two channels; a 10,000 minute monthly limit applies to the entire offer.
The promotion is not automated -- send an email to dids@voxbeam.com mentioning "FREE USA and UK DIDs" with your specific request. In my case, I asked for a 1-646-740-XXXX, a 1-617-372-XXXX and a 44-208-XXX-XXXX. Of course, you can only get numbers in blocks assigned to Voxbeam and cannot pick specific numbers. You also need to provide the address where they will initially send calls, e.g. myserver.mydomain.com:5060, though you can change this later from the customer portal. They provisioned the DIDs within a few hours and they all work fine.
I don't know whether an unfunded account (you get a small credit at signup) is eligible, but suspect that if you ask nicely your request will be granted.
If appears that you can do failover via DNS SRV (not tested) and you can of course do manual failover by changing the destination host on the portal or via DNS. You can specify multiple destinations on the portal, but none of the options are useful for failover, unless the servers are always 'hot'.
There is no incoming CNAM and I don't know whether outbound CNAM is available, or whether a charge applies.
Blocked caller IDs do get though, though you will need to look in both P-Asserted-Identity and Remote-Party-ID for the calling number.
RDNIS is supported but my attempt at testing it failed for reasons unrelated to the DID.
No SMS, though they may allow a split port.
Offer expires 12/31/2016, though you should probably enter your request well before the holidays.
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Callcentric vs VoIPO vs F9
Hi Guys,
Thinking about moving to Voipo.
I like idea of unlimited local incoming outgoing ....
How is the quality, of Voipo ... is this reliable dependable company.
Or some other options ?
Thx
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download & sort Future Nine call records
1. Are there easy ways to download Future-Nine call records in formats allowing by column sorting & totaling - of incoming & outgoing minutes in spreadsheets, etc? Entering 1 mo time spans, mo. by mo. in their online records UI is too lengthy.
2. I'm trying to check monthly total minutes, over a couple years to see if PAYG or Bare Essentials would be cheaper than current America Free (@ $13.50).
3. The only very basic description of plans I find is http://www.future-nine.com/plans-extended.html. Are there other PAYG or Bare Essentials details I should know, aside from the general TOS Ver 1.3 (03/01/2008)?
4. Can't find info on charges / fees for switching from America Free to PAYG or Bare Essentials, if any.
If I'd keep my current DID # (@ $5 / mo on PAYG) - and only other DID fee to switch plans is PAYG $5 setup?
5. Bare Essentials - don't find discussion of costs when go over 250 free min.
6. F-9's online call cost simulator gave differing results for US calls - same route. So, not sure of cost / min on PAYG, or Bare Essentials - if over 250 free min.
* Once, I entered a US number (Simulator ID'd as US) - it gave $.01 for gray & .02 for white - US calls, same #.
Another try - a different #, it showed $.01 for both gray & white routes - US calls.
** For US calls, do prices ever vary for all gray route (or all white) depending on area code? The simulator seemed to show that.
Thanks.
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[General] Nomorobo
Nomorobo says it basically only works with certain providers. They say VOIP providers. But since they work via a simultaneous ring setup, wouldn't they work with any provider that supports simultaneous ring?
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[Unlock] italkbb spa122 password
I want to know if there is anyway to get the password for the italkbb spa122 ata because apparently, italkbb has started encrypting the xml files so now it is no longer possible to eavesdrop on the traffic using wireshark.
Thanks
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Comcast's newish Flash-free speedtest
Comcast's Flash-free speedtest page, which isn't widely known:
http://speedtestbeta.xfinity.com
It was created in August 2016. Their Flash-based site at...
http://speedtest.xfinity.com
...is having trouble today with Flash v.24, so on that page,
they're suggesting the Flash-free version.
This is one of the speedtests that I use to stress an Internet
connection, to see how well a router's QOS for VOIP stands up.
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Trying to reach Sam@Circlenet
I am trying to reach Sam about a refund. Emailed him it will soon be 3 weeks ago. Thought he might still be reading this board. Feel free to PM me or email me if you recognize the nickname.
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[Asterisk] Asterisk 14 CLI
I built a new FreePBX 13 / Asterisk 14.2.1 system and everything works except debug output from the Asterisk CLI. Running asterisk -rvvvvv connects and reports the proper pid. Commands in the CLI work properly and report the expected details. When making and receiving calls, however, there is no dialplan trace output. 'sip set debug peer peer-name' reports the correct ip address but also produces no output.
Has anyone else experienced this and have a solution?
Thanks.
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[Asterisk] OAuth 2.0 Support for Asterisk 13 or Asterisk 14
The attached package should provide OAuth 2.0 supoort to any Asterisk 13 or Asterisk 14 system:
1. Extract oauth2.tar.gz to the /root directory.
2. Edit oauth2.creds. The first line must contain your Oauth 2.0 Client ID and the second line must contain your OAuth 2.0 Client Secret.
3. Make oauth2 executable: chmod +x oauth2
4. Execute oauth2: ./oauth2
5. Use your OAuth 2.0 refresh token(s) as the Password(s) in FreePBX's Motif module (or as the secret(s) in xmpp.conf of plain Asterisk).
6. After clicking Apply Config in FreePBX, wait 15 seconds after the Reloading dialog box disappears before doing anything else.
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If you don't already have an OAuth 2.0 Client ID, Client Secret, and refresh token(s):
1. Go to Google Developer Console: https://console.developers.google.com/project
2. Log in with your Google Voice username/password
3. Click CREATE PROJECT
4. Enter a Project name
5. Click Create
6. In the left pane, click Credentials
7. Click OAuth consent screen
8. Enter a Product name shown to users
9. Click Save
10. Click Create credentials
11. Click OAuth client ID
12. Select Web application
13. Enter a Name
14. Enter https://developers.google.com/oauthplayground at Authorized redirect URIs
15. Click Create
16. Record Client ID and Client secret
17. Go to https://developers.google.com/oauthplayground
18. Click the gear icon
19. Check Use your own OAuth credentials
20. Enter OAuth Client ID and OAuth Client secret
21. Click Close
22. Enter https://www.googleapis.com/auth/googletalk at Input your own scopes
23. Click Authorize API
24. Click Allow
25. Click Exchange authorization code for tokens
26. Reopen Step 2
27. Record Refresh token
To create a refresh token for additional Google Voice accounts, log out, log in to the desired account, and go to step 17.
.
Credit to Ryan Tilton, dziny, carlb8, phonesimon, and others for the original res_xmpp.c modifications.
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What secret sauce makes some VoIP over WiFi drop less?
A few people in my home want to start doing VoIP over WiFi. I know a lot about WiFi , so I won't have major issues with roaming, channels, interference, loss, etc. so no need to discuss these points.
I want to know specifically what secret sauces are needed to make VoIP play well with networks that are occasionally 'bumpy'. Things like Skype, Google Hangouts, etc. I think have some sort of logic that make them more tolerant in dealing with less-than-ideal network conditions - so that they won't drop the call if your WiFi stops receiving packets for 2 seconds.
We had all sorts of issues at work with a certain provider that allegedly used plain-ish SIP w/ 7.11 codec. Calls would drop every 5 to 15 minutes over wifi and I never found out why, because the wifi connection itself wasn't THAT horrible. We were using Bria.
tl-dr; Are some some parameters to be tweaked either at the client software level or the voip-provider level to make voip more resilient in bad networks? So it doesn't just 'drop' calls at the first sign of trouble? How effective are these best practices versus proprietary implementations like skype designed for bad networks?
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voip setup question
disclaimer: total newbie here!
My current set up:
cable internet --> cable modem --> linksys router --> wired computer + 3 WiFi devices
pots line --> base station + an additional handset
modem, router and wired computer are all located in the bedroom. There's a phone jack in the bedroom but on the opposite wall. Base station is in the living room and the other handset is in the kitchen.
My idea was to get Obi202 and hook it up like this:
cable internet --> cable modem --> obi in bridge mode --> linksys router
Ideally, I would like to keep both handsets where they are now. What do you guys suggest?
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Blocked call routes Digit maps Obi
(!1(24[26]|26[48]|284|345|441|473|649|664|721|758|767|784|8[024]9|86[89]|[89]76|900)xxxxxxx|1xxxxxxxxxx)
The other day, a family member managed to call Nassau, Bahamas with their fat fingers. Instead of dialing NYC 212, they dialed 242 which was surprising since I thought we had 242 blocked in my digit maps above. Luckily, they immediately hung up for just a 10 cents charge for the call.
Does the 2nd set (1xxxxxxxxxx) negate the first set and allow all 11 digit calls to go thru?
Anyone have a good digit maps that blocks such 3 digit non 011 international calls?
Afraid to test by trial and error with digit maps since a wrong parentheses somewhere can allow a call to go through resulting in charges!
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[Equipment] OBis on sale
From my inbox...
OE
Amazon 12/12 to 12/16, 12/14 to 12/15 & 12/18 to 12/19*
- From 12/12 to 12/16, get $20 off all OBi IP Phones (OBi1062, OBi1032 and OBi1022).
- From 12/14 to 12/15, Amazon will feature the OBi200 for $37.99.
- From 12/18 to 12/19, the OBi202 will be on sale for $49.99.
Newegg 12/16 to 12/17 & 12/20 to 12/21*
- From 12/16 to 12/17, Newegg will feature the OBi202 for $49.99.
- From 12/20 to 12/21, the OBi200 will be on sale for $37.99.
Newegg.ca 12/23 to 12/27, 12/28 to 1/1, 12/15 & 1/3*
- For our Canadian customers, on 12/15 and from 12/28 to 1/1, Newegg.ca will feature the OBi202 on sale at C$69.99 + free shipping.
- From 12/23 to 12/27 and on 1/3, the OBi200 will be on sale for C$49.99 + free shipping.
* WARNING: The Amazon and/or Newegg promotions happening or not is dependent on several factors outside of Obihais direct control. At the time of writing and distribution of this newsletter, to the best of our knowledge, the aforementioned promotions are scheduled to occur on the dates indicated. We apologize if there are any inconsistencies with actual promotion dates, including the occurrence of the promotion, in the first place.
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[Voip.ms] Widespread glitch today for inbound at VOIP.MS
Two of my clients called to tell me that they can't
receive inbound calls today on their VOIP.MS DIDs.
These two clients both use dallas.voip.ms as the
DID point of presence. On their list of DIDs, the
point of presence says "dallas-old" in the last
column, and no point of presence is chosen on
the list below. The solution was for me to go into
each account and choose dallas.voip.ms and then
save it.
I called VOIP.MS, and they're aware of the glitch.
(Nothing on their website yet, however.)
It occurred this morning during some sort of
server migration, which they aborted when saw
the problem. But by that time, apparently many
DIDs had been switched to the non-existent POP,
"dallas-old". VOIP.MS told me that they can't fix this
for all users (at least not immediately), so I'm fixing
it for my clients, one by one.
I don't know whether other VOIP.MS POP cities are
affected. I know that not all are affected. I suggest
that you check your VOIP.MS accounts, especially
if they use the Dallas or Dallas2 POPs.
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Best way to port mobile number to VoIP with SMS?
I want to port my existing mobile phone number to a VoIP provider but also need reliable sms support: I currently use VoIP.ms for most of my numbers but I find its sms support to be really hit or miss - sometimes texts don't seem to be received at all or there's a long delivery delay.
Any suggestions for another provider (or providers, if there's a way to split the voice and text parts to separate companies)?
Fyi this is a Canadian #.
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[General] Help transitioning from Vonage to Obi202
I've been using Vonage since they first launched, but realized I'm probably due to switch to something more flexible using an Obi202. I've poured over the forums and think I have things figured out, but just wanted to check if any of you are kind enough to give some input.
My use case - located in Canada:
1) Need Canadian inbound number to receive calls
2) Need US inbound number to receive calls
3) Need outbound calling where a toll free number will recognize me as calling from within Canada
4) Need outbound calling where a toll free number will recognize me as calling from within the States
5) Need to be able to send an occasional outbound fax
6) Calling pattern: Calls to Canada & USA (roughly 80 minutes) and Japan (roughly 30 minutes)
7) Want voicemail which can send me an e-mail when a voicemail arrives
8) Want ability to ring the Obi202 device home physical lines as well as simultaneously ring a Canadian mobile number
I'm thinking of purchasing an Obi202 and getting the following setup on it:
1) Callcentric Dirtcheap Canadian number ($2.95)
2) Callcentric Free States inbound number in NY State
3) Callcentric North America Basic ($1.95)
A few questions:
1) Is this the best setup for my calling patterns/needs?
2) I believe I will be able to specify if the outbound call should show as US or Canadian - does that work when calling something like 1-800-Go-Fedex to get the "right" country's customer service?
3) Would it make sense to mix any of the Anveo products into this for some of the pattern?
4) Am I right that the total monthly would be $4.90 (covering up to 120 minutes of Canada & US calling), unlimited inbound, plus 83 cents for the 30 minutes of Japan calling?
5) If I pick up on the mobile phone when it simultaneously rings on an inbound call, I assume that is part of the 120 minutes included, and if I go over it is the same per minute cost?
Thank you so much for your guidance!
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