Is it allowed? From my understanding it should be with http://www.callcentric.com/dids/service_provider_unlimited but just making sure?
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[CallCentric] Auto Dailer with CC
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[General] Softphone Suggestions?
Hi everyone! :) I've read a few threads from searching around, but can't find anything conclusive. I want to use a softphone on my laptop and desktop for my VoIP.MS service. I was using the free 3CX version, but it turns out 3CX has deprecated the software and it isn't supported. Also, my laptop has a 4K screen and the softphone is one for ants at that resolution since 3CX doesn't scale.
I know there is bria.. but that's not free. And I tried Linphone, never ever again. Anyone have any suggestions? (Yes, I've read the softphone setup guide on VoIP.MS' website to get some ideas... many of them seem discontinued or not updated for a long time)
Cheers for the help!
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[Anveo] Porting A Canadian TN With USA Account Address
Has anyone run into issues porting a Canadian telephone number associated with a USA account address?
The Setup:
I have two numbers out of Vancouver (BC) that I am trying to port from Flowroute to Anveo Direct. Flowroute, the original provider, uses my USA based address for verification purposes if they are unable to reach me in time for verification. Usually the manual verification window is four business hours except in cases where the port is "expedited" by the carrier.
What Happened:
Anveo's LNP department rejected my original port request with my USA account address due to "[Anveo] cannot use US address for the porting process of Canadian numbers". Flowroute suggested that I use a dummy address in Canada and manually approve the transfer when it came in. I decided to try the dummy address route because there is a four business hour window for approving transfer requests with Flowroute. The updated dummy address port request was submitted to the Anveo exactly one week ago. There was no communications from Anveo after submitting the update form other than that they were working on the request.
That transfer request hit Flowroute today and had an "expedite" flag on it which cut the approval time down from 4 hours to 1 hour. I was unaware that the port request was going to be coming in today so I was not watching email very closely. This reduced approval window caused the port request to be rejected by Flowroute because I did not get the approval email until 30 minutes after the approval request expired (or 90 minutes after the request came in from the carrier). I immediately emailed Flowroute but there was nothing they could do due to the expedited processes being used.
I am now waiting on a support ticket to Anveo Direct to hopefully sort this whole thing out. The annoying part of this story is that this rejection delay, coupled with the Thanksgiving holiday, will likely cost me another month of service ($2.50) with the previous provider, Flowroute.
Any suggestions on how to resolve this issue and get the number ported over?
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[Voip.ms] Running multiple sub-accounts from the same IP address
If this has already been asked, I'll apologize in advance.
I have an ATA and a softphone logged in from the same address, I can receive calls on both, but only make calls from the ATA, both are using toronto6.voip.ms I'd like to keep using toronto6, because I like the internal extension feature.
Is it possible to keep it this way, or do I need to change the server for the softphone? (I've also tried changing the port number on the softphone but I haven't figured out which port numbers I need to be using)
Cheers,
EQ
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setiing up voip.ms in OBitalk
I just created a voip.ms account and got an email from them with a SIP username and password.
In OBitalk I added my device and clicked on SP1 to configure it. On the following page I chose voip.ms and that brought me to the voip.ms configuration template.
There's a few things in this template that are a little unclear to me.
First, "Use this service for Emergency 911 Calls" checkbox. Should this be checked?
Second, "Service Provider Proxy Server". Atlanta 1, GA is there by default. Should I pick the closest location to where I live?
Third, username and password. Both fields were prepopulated when the page displayed. Username had my e-mail address and password was hidden but I presume these are the username and password for my OBitalk account? Is this where I should enter my SIP username and password from voip.ms?
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[General] Looking for Cheap call rates to India
Hello,
I have been using localphone since long time for my calling, but wondering if there is any other reliable provider with cheap rates.
Thank you
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[Anveo] Unable to receive calls, but able to dial out
Hello,
I am using a Thomson modem with a Cisco SPA122 ATA. I am able to make calls without issues, but unable to receive ANY calls.
The device is connected and online, and nothing has been changed.
I am wondering if anyone has an idea to what I can do to check / fix the issue? Thanks
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[Unlock] Unlocking the BasicTalk ATA
Important NOTE: a better unlocking method has been posted later in this thread. Please see this post for a permanent unlock developed by uid://1479488. My soft unlock may help in some cases if the ATA has "called home".
I have some good news for those of you looking for an inexpensive ATA.
I've just got my hands yesterday on a couple of BasicTalk ATAs (I've had my eyes on them for a few months but I live in Canada and don't go to US that often) and I put together a small tutorial for unlocking them.
The ATA is a Grandstream HT701 with a customized firmware.
I posted it on my website at http://voipfan.net/unlock/ht701bt.php
I will leave the access open to everyone for a couple months then make it available to registered users (like my other unlocking tutorials).
Enjoy and if you run into any trouble please post here.
--
Providers (through asterisk): voip.ms, freephoneline, smartcall.ro, ipcomms, callcentric. Hardware: Vonage VDV21, Moto VT2x42, Linksys SPA series, Grandstream HT series, Panasonic KX-TGP5x0
http://www.voipfan.net
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Best place to host a DID Trunk?
I need some advice with low budget on this. I currently tried running it on a dedicated old Core2Duo laptop with FreePBX which was okay but not great lag wise despite having a decent low ping to the trunk with both callcentric or DIDLogic from my home ISP. Well I tried it on a VPS server that has well more than enough BW to handle things 1Gbps port speed and still when getting calls its AWFUL at times. Mostly the lag is very minimum but I need to reduce it just a tad more to be comfortable.
Basically I want somewhere thats free or cheap (can get a new DID/number if needed) so I can have the best experience call wise, Only need incoming and Music on Hold I can choose (instead of basic ring)
My setup, Caller Calls My Cell #, They're forwarded to the DID which goes to my PBX, and then rings my SIP app.
Thanks. I've made a few posts here about it in the past.
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[Equipment] Neat Obihai "trick" for trunk prefix
Audience: Any OBi user needing to prepend a prefix, tech prefix, password, etc. to every phone number sent out on a trunk. Useful with Flowroute, Anveo Direct, Sip Broker toll free, etc.
A few days ago I configured a slot on my OBi202 to send toll free calls to Flowroute using IP authentication. Using Flowroute with IP authentication requires a "tech prefix" to be prepended to every phone number sent to Flowroute.
While it is possible to prepend the tech prefix in the SPx DigitMap, there's a better way: apply the tech prefix in the phone port OutboundCallRoute. Here's what the OutboundCallRoute addition for Flowroute in SP2 looks like:
{(Msp2):sp2(>12345678*$2)}
Here '12345678*' is the Flowroute tech prefix and '$2' is the number being called.
NOTE: The '>' is optional but I think it improves clarity.
The "trick" here is the catenation of the prefix and '$2'. The Obihai Device Administration Guide of September 2013 (latest version) says that the allowed syntax for the target (the element following '>') is:
target = number to call OR $2
but it's more flexible than that, as I found out.
Tested with Firmware 3.0.1 (Build: 4581).
Notes for implementing toll free to Flowroute
--------------------------------------------------------
1. The ITSP Profile B (for SP2) DigitMap is as follows:
(18(00|44|55|66|77|88)xxxxxxx|<1>8(00|44|55|66|77|88)xxxxxxx)
2. The additional phone port OutboundCallRoute rule above must be inserted prior to any rule for calling NANP numbers, since rules are checked left to right and the first matching rule is used.
3. The end of the phone port DigitMap should look like this (for SP2, addition in bold):
|(Msp2)|(Mpli))
4. In SP2 SIP Credentials:
- set URI to your desired outbound caller ID number
- leave everything else blank
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"RTSP" ALG?
Was poking around my router today and noticed that RTSP ALG was on. Does this impact voip from my SPA112 or Polycom phone using Callcentric, voip.ms, etc?
Also, I wanted to check that SIP ALG was off but it seems I don't even have SIP ALG in this router. Always thought I did have it and I had it turned off. I guess I remembered incorrectly. ;)
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[General] copy of CWU announcement
Dear Customers,
To support calls to PSTN number in countries with 9 digits long phone
numbers we have to change the rule to dial another user of our system.
To call another user you have to dial 000 followed by the 9 digits
username, i.e. 000123456789. The old format (without 000 prefix) is
currently accepted, but will be disabled on February 1st 2017.
---
CallWithUs.com team
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Need recommendations for switching from POTS to VOIP (Business)
So, currently we have two POTS lines (AT&T) connected to a Grandstream U6104 PBX and I would like to switch over to a VOIP provider.
Problem is everyone seems more interested in selling a hosted PBX solution where I would end up paying per extension (currently have 5) rather than something I can plug into my Grandstream PBX.
Are there any companies that I can use with my Grandstream and have an unlimited plan? Switching over to VOIP I can go down to one "line" (sorry if I'm using mixing up the terms here), because it can handle multiple channels?
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[General] Call forwarding service for Online Contests?
Hello everyone. My wife has a sweepstakes and contest hobby that has provided us with many great prizes over the years. Everything from gift cards to an all expense paid trip to Sydney, Aus. The only downside has been that our home phone number has been completely overwhelmed by robocalls and telemarketers.
Over the last year or so we have migrated away from using the home number and only use our cell numbers for contests. What I would like to do is port my old google voice number to a service like ANVEO. Then that number will call our home phone (with a new number) and my wife's cell at the same time. She is a stay at home mom and prefers the home phone due to its 4 handsets conveniently placed all over the house.
I know that is what google voice offers, but I'm intrigued by all the features of a service like ANVEO and the fact that businesses rely on it. I also like the price structure of it (dirt cheap). I know there are other providers to consider, but is Anveo a good option for us?
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Callcentric to Voip.ms vis sip uri issue
So I have a DID with Callcentric that I have a call treatment set on that sends all call to one of my voip.ms DID's via did@washington.voip.ms.
Lately I have been having a lot of issues where when I place a call to the CC DID from my Verizon cell phone, I get a long period of "dialing" then a Verizon Wireless error messages. I don;t think it is a Verizon issue as I have had several friends tell me they tried to call and the dialing went nowhere.
Anyone else have a setup similar to voip.ms on the Washington server and seen an issue like this?
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[Future9] voicemail to email
I've been having problems with email notifications of email with Future-Nine.
I used to get emails with a .wav attachment of the voicemail. It quit sometime last summer (I think). I've tried opening a couple support tickets, but I've not heard anything.
The service quality is otherwise pretty good, but its also pretty annoying not to know when there is voicemail waiting.
Does this work for anyone else?
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[Future9] Is Future9 still viable?
So I have an account with them and still use them. But I was just on their system to send an account consolidation request and noticed this about their account system, - Release : 22 January 2008.
If that is the last time that UI and respective components were updated, it kind of worries me.
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[General] VoSP recommendation
From time2time, I see folks here recommended CallCentric (CC) as the choice of VoSP to port out from an existing VoSP to. Also, I noticed some requests to port out a current number to Google Voice (GV). Personally, I prefer the later merely for free calls and the call quality is not bad, AFAICT (based on my experience). But, what I don't know is if anyone out here has a success story to port a Comcast landline to Google Voice and can share the experience here. The reason I ask is because I have a friend (old couple) who will be migrating from an existing Comcast to a new AT&T fiber connection and would like to recommend them to port out their existing number to either CC or GV. TBH, I prefer to recommend but have no experience with porting out an existing line to GV. Please feel free to chime in and I will be back in a day or so.
Currently, Comcast charges them US $200+ / month for a triple play service, i.e. 100 Mbps broadband Internet, cable TV, and two phone lines, while AT&T to offer a 2 Gbps (full duplex) broadband Internet, a DirectTV, and no phones, for US $130 / month.
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[Other] Has anyone experience with this compay: http://choosemcc.com/?
My wife and in-laws are Russian an cal mostly Moscow lanfdlnes but sometimes cell phones. I found the MCC flyer in on of the local Russian groceries stores. They bill out of a PO Box in Manhattan. I originally set it up with an ATA at our old house but during the move lost the settings. Of all the services we've tried (VOIP.MS and Callcentric), MCC has the best voice quality and minimal drops. Of course here are busy times when the volume of calls is overwhelming and noting gets trough.
Everyone in my family is moving to soft phones (CSIP Simple) and I've tried to find servers address and passwords to place SIP calls no to available. Repeated emails to various departments at MCC, like customer service and technical questions are never answered. Telephone calls are uselsess unless you speak Russian or other eastern European languages,which I don't. Trying to translate through my wife is difficult to say the least.
I don't think the problem is on my end as a I run a home run to the router and domestic and other international calls go through oK.
Google hangouts worked superbly when my wife and daughter were in Moscow last summer. Even the video wasn't too choppy but it is beyond my 78 year old mother-in-law's capabilities. She doesn't even have internet
Any one hae have any suggestions?
Bob
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eMTA vs. sMTA
Can anyone point to some information that explains the difference between these options?
I am trying understand the difference between using my cable company eMTA for phone and how that device works with my phone system as opposed to a third party solution like Vonage.
Thanks.
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