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[General] Two-line ATA > are different SIP ports req'd?

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Hi, I have a two-line ATA[1]. Recently I had an intermittent problem where the caller would not hear ringing. After the configured number of seconds, the voicemail greeting would kick in. I could call back and I would hear ringing when I dialed. When no ringing was heard by the caller, the phone connected to the ATA would not ring either. Two changes were suggested by the VoIP provider: Set NAT Keep Alive Enabled: to Yes - this was set to No on the DID line. Oops. Set NAT Mapping Enable: to Yes They also suggested to change the SIP port from the default 5060 to 5080. The problem seems to be resolved. As I like to understand what happened, I asked them what could have been the fix: the port number change or enabling NAT Keep Alive Enabled: They said the latter. As I think about it, shouldn't NAT'ing handle IP/port numbers? Each line on the ATA was configured to a different VoIP server. Each was using port 5060. Thoughts? Thx! -pablo References [1] - Cisco SPA122

[Anveo] Setting Up New Account; Am I Missing Something?

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Signed up for an Anveo account with intent to port a single ring.to number over before the latter closes. I'm very new to paid VOIP services, and only knew of ring.to from cellular-service-oriented websites. After signing up, it appears the only way to pay is through PayPal links where the minimum payment is $15, despite the services I'm intending to sign up for being one phone number/DID and the required E911 service that it seems must be setup before I can do anything with the account. And I can't even seem to do that without adding funds to the account, which leads to the PayPal links. Cyclical cycle here. So, am I missing something here; like an order button where I sign up for the number port and E911, the system sends me to a billing information screen where I enter my info., the payment processes and I'm good to go? Or is the only way use Anveo through PayPal? Please note #1: Yes, I realize there's a difference between Anveo Retail and Anveo Direct. I'm trying to sign up for Anveo Retail. Please note #2: Yes, I have only one number however I'm not interested in talkatone.

Increasing spam cals via GV

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Lately I'm getting increased spam calls to my Google Voice numbers. I resorted to setting up an IVR in my PBX. It was slightly painful getting IVR to successfully answer calls, but now it does. In a nutshell, I'm forwarding all calls to two free CallCentric DID's from two Google Voice numbers (yes I already checked, Google Voice won't let me use same CC DID to forward both numbers), which then forwards to virtual extension on PBX, which in turn makes IVR pick up the call. Forwarding calls from Google Voice directly to IVR was met with no ringing on PBX, but ringing on remote end and GV voicemail eventually picking up the call. So far it seems to work well. I followed instructions in this thread: https://community.freepbx.org/t/google-voice-to-did-to-extension-ok-but-if-the-call-ends-on-ivr-google-voice-voicemail/10963/5 Credit to Cocooningweb for this.

Any Alternative to CallCentric?

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Any Alternative to CallCentric free number wise? I need a did that accepts incoming calls only, outgoing not needed. today as an example I only got 1 call out of the 40 that called today. Most of the time users get a CallCentric error message. This has been an on going thing for 2 years now. I would use Google Voice BUT calls pick up the instant it starts ringing instead of only when voicemail picks up by phone system. Thanks. I'm up for paying a couple of dollars a month. But I'm done with call centric.

[General] obi202-FPL-ph2 reserved for alarm system, How to configure

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Hi! I am currently using only one provider (FreePhoneLine). ph is connected to my home phone ph2 is connected exclusively to the alarm panel (CID format) I added the following to Star code profile A 5149041234,alarm1 mode,set($Noji1,200),set($Noec1,1),set($Cdm1,3),set($Lbdt,1),call(5149041234) If setting ITSP Profile A/General/DTMF Method to InBand then alarm testing is ok also voice test (1-416-342-9562) is ok but using star code (*98 on FPL) get me onto voicemail correctly but does not recognize any touchtone commands menu (3 or *) So I am very near the perfect solution. Any ideas ? Note that DTMF Method=Auto works perfectly for voice test and voicemail inquiries but not with alarm testing. One solution maybe to have 2 profiles (one for ph with Auto) (one for ph2 with InBand) but I really don't know how to configure this solution. other solution is correcting the problem with recognition of touch tone commands in FPL's voicemail control. Every hints will be appreciated. Thanks and have a nice day ItiBi

"Service Offline Alert -- OBi Device (xxxxxxx) is Not Registered"

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  Just got back from trip and see this in my gmail for my new Obihai 202 twice - on Mar 15 and Mar 17:quote:admin@obitalk.com Mar 17 (4 days ago) to me Service Offline Alert -- OBi Device (xxxxxxxxx) is Not Registered Your voice service is not registering. This could be due to a temporary service disruption or an incorrect login credential. In most cases, reconfiguring the voice service with the proper login credential will resolve the issue. Check out the OBiTALK Self-Help Guide for assistance troubleshooting your OBi device. admin@obitalk.com Mar 17 (4 days ago) to me Service Online Alert -- OBi Device (xxxxxxxxx) is Registered Your voice service has reconnected. You will be able to make and receive calls now. No action is required.The above "reconnection" was 15 minutes after the "Alert". Note that this is the same issue as in an earlier post here: https://www.dslreports.com/forum/r31178809-Equipment-Obi-device-service-alert# What do you think?

[General] Vonage / Motorola VT2142

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I have one of these sitting around so I thought I would unlock it and then use Simonics Google Voice Gateway for my brother. I started reading old forums and could not find any info on whether that is possible and really questioned it when I read this: said by Toro years ago: One important note: the VT2142 accepts numeric SIP usernames. The only other characters that are accepted in the SIP username are the lowercase characters a,b,c and d. On the other hand, the Betamax services use alphanumeric usernames to login to the web site, and the same username is also used for the SIP settings. So when you create your Betamax account, make sure you use only the characters mentioned above and numbers.That seems like a serious restriction however, my brother does not yet have an account with Simonics (nor do I) so maybe follow the advice given for Betamax users? Any info is appreciated. Thanks.

$20 Orange Pi has onboard 8GB eMMC flash,would RaspPi PBX's run?

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I was just reading a small blurb about the new "Orange Pi PC Plus Quad Core Development Board" which supposedly will sell for $20 and will include 8GB eMMC flash memory on the board itself. Since I'm not that familiar with memory types I just wondered if eMMC memory is more reliable than SD cards. The problem with running a small PBX on a Raspberry Pi is that if you don't take extreme measures to minimize SD card writes, in particular by disabling most of the logging performed by Asterisk and FreePBX, the SD cards have a tendency to die prematurely. I have an original Raspberry Pi and SD card that's been working for about three years now but I have turned down the logging to ignore all but critical entries. I would just wonder if eMMC memory is any more robust. I think 8GB would be enough to run a small PBX with just a few extensions. The larger problem would be whether any existing software would run on an Orange Pi. I suppose it's just different enough from a Raspberry Pi that if you tried to install RasPBX, or the XiVO build for the Raspberry Pi, it would probably crash and burn. But it sure would be nice to be able to build an inexpensive, fast, and hopefully reliable PBX that would not depend on an SD card for storage. This is the article I was reading, and for the record I have nothing whatsoever to do with this company that makes these, in fact I have never even seen an actual Orange Pi: http://www.cnx-software.com/2016/06/09/orange-pi-pc-plus-quad-core-development-board-with-1gb-ram-8gb-emmc-flash-sells-for-20/ -- I refuse to believe that corporations are people until Texas executes one.

best OBI configuration for second home-office, residential line?

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I'm a home-office/residential customer who currently has one POTS line and one VOIP line using a Obi200 (which turned out to be easier to set up than I thought, thanks to instructions on the Callcentric web site). The AT&T reseller I use for the POTS line is ending its agreement with AT&T, and I choose not to deal with AT&T directly because I have found them to be dishonest and deceptive. So, I want to replace the POTS line with a second VOIP line. My questions are: 1. Is there an advantage to replacing the Obi200 with an Obi202 versus just buying a second Obi200? 2. I'm currently using all the ethernet outlets on my DSL router. Can I add an ethernet switch and run the two Obis through that? (I just want to be sure that the switch doesn't add a complication.) I don't want to direct incoming calls to the former landline DID to the existing VOIP line. I want to keep them separate. Thanks in advance for advice from the knowledgeable folks here.

Mobile Voip provider for calling from/to Japan (good or bad)

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Here in Japan the cost of calling mobile phones is prohibitive, so am looking at alternatives to Skype or Google Hangouts (which are reasonable for Landlines but quite high for mobile 090 and 080 numbers. So far, most of the Hopi providers have equally high prices for calling Japan Mobiles. (I am using an Iphone so would need a Voip app available on the iOS store that works with that provider). By chance I found a thread on one of these, VoipVoip, and it turns out, after checking, that their prices are in the area of 4 US cents a minute even for the mobile numbers 090 and 080, which typically cost 9-15 cents minute. http://www.dslreports.com/forum/r22952830-Thoughts-on-VoIPVoIP In the above thread they also mention both Phone Power and Call Centric as having user friendly websites, but on checking I found that their rates were similarly cheap (less than 3 cents I think for mobile numbers in Japan. My questions: 1) Anyone have any comparisons or comments about any/all of these companies' services? 2) It appears with VoipVoip and Call Centric that you do not have to buy an unlimited plan. Phone Power is a bit vague on that. Again, I want this mainly to use within Japan to avoid both local phone companies and SmartPhone, which is about 16 cents a minute, expensive but half of the phone company rates. And I would use it mainly on my iPhone. Thanks in advance, paul

Best ATA for the Money ?

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Which is the best ATA with the latest, fastest and greatest CPU/chipset for the money ? I know many with suggest the older Ciscos and Obihai, but they are using older chipsets that are like 2-3 generations old compared to the Grandstream 8xx series.

Grandstream HT802 vs HT812

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Alright ATA pros chime in here. So from what I gather the only difference between the HT802 and the HT812 is the extra LAN port. The extra LAN port would be advantageous if there is only one Ethernet wall jack in an office or absence of a switch in an office to connect other devices to the internet or is there more ? The HT802 is currently the go to ATA for many of the larger VoIP providers, I guess a saving of $5 or more is a huge savings for them if they sell hundreds and thousands of these ATAs.

[Voip.ms] So VoIP.ms let my numbers get ported and stolen away from me...

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Yeah, so that really happened. Apparently someone generated (Photoshopped maybe?) an invoice and ported my numbers away. No email, phone call, or any contact from VoIP.ms. They simply state that it was "authorized" because the information matched mine. My last VoIP provider at least emailed me and asked me if I authorized the port before they ported the DIDs to VoIP.ms With that logic I can port the number for the president of the United States as long as I give a generated photoshopped invoice of 1600 Pennsylvania Avenue. What is really troubling is someone has the exact DIDs I have as well as my name, address, etc...

[Equipment] RPi 0 and Rpi 0 W on sale US $3.14 @MicroCenter

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I just posted here about the subject. You can grab them from your local MicroCenter store before the offer expires and/or out of stocks. I am sure you can use the Rpi 0 as a JTAG tool and probably Rpi 0 W as a PBX, etc.

[General] LIVE CSPAN3/Radio FCC/FTC Forum: Combating Robocalls...

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3 hours, 0 minute The Federal Communications Commission and the Federal Trade Commission host a forum to highlight agency collaboration in combating robocalls and protect consumers. https://www.c-span.org/video/?442990-1/fcc-ftc-host-forum-combating-robocalls You can probably catch it afterward on podcast. OE

Hotel VOIP change over. Consultants? List?

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I posted @ weeks ago re: @60 unit Hotel primed for a total VOIP change over from 60's copper system. I've installed excellent Internet to units and plenty to building, staff, desk, etc. Anywhere to look for experts that will consult on this? List somewhere? Hotel Owners want to save money and I'd like to make money. I'll be glad to elaborate further here and in detail via email Thx

Which VOIP providers support sending RDNIS info?

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im learning callcentric does not. which do? TIA

48 Hour Battery Backup with Modem and PAP2T?

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I have an AT&T Pace 5268N VDSL combo router (using wifi) and a PAP2T for a VoIP system. I want to run them off of a UPS so that the phone stays up when power is out. Would it be possible to keep them up for at least 48 hours with a residential UPS? Does anyone have recommended UPS models? Sorry I'm a bit new to this and finding the answer has been a little more challenging than I expected.

[VOIPo.COM] Calls to Taiwan won't go through directly

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I have VoipO. I can't call Taiwan directly from my ATA. I have to use remote access number. The last time I had a successful direct call was 3/8. Yesterday 3/23 it was still not working. It's been over 2 weeks now. I received a response from support 3/18 after power recycled my ATA: Quote Thank you for response. Our developers are working with our upstream carriers on some routing issues, my apologies. There is no ETR at this time, but they are working actively to resolve this matter promptly. I'm truly sorry for the inconvenience and we should have this feature working again as quickly as possible. Thank you. End Quote Nothing changed since then. I don't mind using remote access number sometimes. I also don't expect 100% up time. But this is getting old after more than 2 weeks. Does anyone have a similar situation where support simply gave up? Thanks.

[Anveo] Does anyone know how to setup a Grandstream HT701 on Anveo Retail

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Hello: I can get it to "register" at Anveo Retail but not to work. I have done a reset back to factory settings. Can anyone point me to a tutorial or can anyone share which settings need to be adjusted? Thanks, Rob.
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