Anyone else seeing lagging on Anveo Direct with just one server (residential use)? I rebooted the PBX without success. No recent changes in firewall or router (QOS settings stable, SIP helpers disabled, etc).
anveodirect 50.22.101.14 Yes Yes 5060 LAGGED (62 ms)
Other AD servers appear okay:
anveodirect_2 67.212.84.21 Yes Yes 5060 OK (26 ms)
All other providers okay as well (not showing as LAGGED).
I don't want to use one of my last remaining free tickets. For now I have bypassed this server, using the others, because calls sent to this server fail.
Anyone else? Any ideas on how I might address it? TIA
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[Anveo] [Anveo Direct] LAGGED - only server 50.22.101.14
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[Anveo] Call flow
Does anybody on Anveo has a sample of how I set up "call forward no answer" and "call forward busy" ???? Can't figure this out.
Amnon
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video supported sip provider
I have a Grandstream GDS 3710 video door system. Doorbell button calls a sip phone (android with grandstream wave app installed). Calls complete, audio works fine. No video.
I now know, voip.ms (sip provider) does not support video calls.:(
Any suggestions? I see "Broadsoft" on the grandstream wav. app. Not sure if that will allow me to create new did numbers on broadsoft. Assuming Broadsoft can be used in this manner. Looks like a big Cisco corp. Suggestions to which sip provider to use with the GDS 3710? Thx
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Lan based VOIP/SIP for home (NO INTERNET)
I'm looking into making a no internet based VOIP for my home.
I have icecoldapps servers unlimited pro for android.
I got sip software to connect to android based server but can't seem to dial out to other devices.
What am I doing wrong?
Can anyone provide me a step by step guide?
Thanks in advance
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Cisco ATA 187 CUCM bypass?
I have a Cisco ATA 187 ATA that I want to use for SIP. I believe you can anyway. From my research it requires Cisco Unified Communications Manager to work. I was wondering if anyone knows of a bypass to make it work? If not i'll probably write off the $5 I spent on it and call it a day.
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Impossible to port number?
I'd like to port a number to a VOIP company but all the ones I've tried to "check availability" with show that they are unable to port.
I used this website: https://www.telcodata.us
to enter the phone number info and it only shows one company as servicing this area (it is a rural area). The company listed is the local phone company that I want to port out of.
Since this is the only company that services this phone number, does that mean that there is no way of porting it out? Thanks.
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VOIPo down again
I know, I sound like a broken vinyl record. Both 618-632 #### and 618-624 #### lines are down and I guess they were down yesterday too but I did not follow-up on wife's complaint yesterday. Anyway today incoming rings after calling party's call already went to voicemail. Outbound calls do not connect but sorta ring the calling part with one ring but no connections. Courtesy ringback generally does not work.
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SIP to SIP Dialing
If I make a SIP to SIP call from my account at sip2sip.info to 100@subdomain.sipdomain.com (which is not connected with sip2sip.info), the call connects. If I make that same call with my account on anveo, I get a call "forbidden call" error. If I make the call with my account on voip.ms, I get an immediate "failed to establish call" error. My sip2sip.info account is free. My anveo account is free. My voip.ms call is a paid account. I really don't understand SIP, but it seems to me that a SIP to SIP call on the same computer, on the same LAN, on the same ISP should be possible from almost any provider to almost any provider. (All of the services advertise free SIP to SIP calls, but maybe that means only on their networks.) Can someone explain the SIP landscape to me?
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[General] Call rates to EU countries
I am not sure how far this is a result of an action or inaction by government authorities in EU countries, the fact is that the costs call termination in those countries for calls originated outside of EU/EEA skyrocketed. With the result that long distance calls to some destination became as much as 1200% more expensive, getting to the level of early nineties.
In some cases, it only affects calls to mobile numbers, in some countries both cell phone numbers and land lines.
Following EU countries do NOT participate(yet) in this foolishness: United Kingdom, Ireland, Germany, Denmark, Hungary, Slovakia, Poland, Romania.
Also EEA countries Norway, Iceland and Switzerland do not. No concerns here.
ALL other EU countries do .
So far, most VoIP providers did not find any answer to this and simply jacked up the rates for US customers.
Having an EU country DID and setting it as caller ID won't help by itself with most providers.
Some providers offer a partial relief if you have an account in UK pounds or Euro currency ( Localphone )
Rebtel provides decent rates to a few of these destinations but is generally on the expensive side.
There is a way with Twillio by having their DID to set it up for lower rates, i do not have the details, the blogger "stewart " once mentioned it.
The cheapest and most convenient solution , while mainlining a high audio quality is ZADARMA. You will need an EU DID, but you do not have to use theirs, you can use one you have with another provider. Prices are good and so is the call quality and reliability. Their web site provides instructions how to set up your ATA or SIP phone, although their dial plans can use some improvement. You can also use them on your cell phone through their own app or other VoIP apps like Zoiper or Linphone. Some cell phones seem to work with them only in wi fi mode. The only disadvantage of Zadarma is that there is no calling card feature available.
Other possibility is to use cell phone apps like Signal, telegram, IMO etc for calls to other cell phones, something what increasing number of people does, although the quality is hit and miss.
For private calls, having DIDs in these countries makes sense, for a minimal amount of money you have excellent call quality and with private calls you can always ask people to call back.
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Voip.ms - Distinctive Ring
Hello-
Long time, no Post.
I'm trying to set up Distinctive ring for the few numbers that call that I want to answer the phone. Mainly just family and friends.
I have been searching the VOIP.MS wiki (which is awesome by the way) but don't see anything that addresses setting up incoming number for a distinctive ring?
Any advice/guidance on if this is possible?
I use the OBI200 with VOIP.MS.
Thanks for your help!!
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Periodic PSA: BYOD VOIP is a very low margin business
Just a PSA: No one is getting filthy rich selling consumer VOIP, especially at smaller scales.
Ask yourself: What sort of expenses are baked into the 1 penny per minute termination rates an Anveo or voip.ms or Callcentric? Let's start a hypothetical VOIP provider! How do we keep the denziens here with service and from complaining?
1. Infrastructure: Just ask VOIPoTim, infrastructure is hard. When implemented from scratch today, it would be obvious to have a datacenter failover solution. You need redundancy! Want to prevent a hurricane from bringing down your service like Callcentric and Sandy? You need another datacenter! Whatever you thought it would cost, multiply it by 3. Of course, you'll need to pay extra for DDoS mitigation expenses, which you'll find out quickly from the first Ukranian who DDoSes you. You need support engineers on-call 24-7! You need to pay for bandwidth!
2. Software: You might notice that most VOIP providers are using relatively ancient server-side technologies like Anveo (ASP - wtf this is 2019), voip.ms (PHP), or a2billing. You don't want this. Your customers deserve a shiny modern experience! Now, you have to integrate that frontend with backend management and billing, all while keeping everything secure. Keep in mind, POTS VOIP is dying, so the backend integrations are going to be janky and more costly.
3. Support: There is a subset of "expensive" customers. When you're selling termination for a penny a minute, or DIDs for a buck a month, it only takes a few percent of expensive customers to bring you into the red. But even the most competent customer might run into issues configuring their SPA-122 with your service. Of course, you also want full-time US-based employees paid a working wage, with health insurance! Can't afford support? Set up some forums and let the blind lead the blind.
4. Legal costs: The regulatory environment in the US and Canada is a joke. You'll have to pay the initial and ongoing bills for the lawyers. If you get any papers wrong, you'll be paying more. Better have the lawyers draft up that language for binding arbitration, though!
5. Carrier costs: Not too expensive really
6. Profit: Whatever is left over, probably none
Of course I would complain to a McDonald's manager if they screwed up my order more than once. But I wouldn't be a repeat customer.
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Geo DNS SRV with Cisco SPA112
I have SIP servers in New York and Las Vegas handling calls from Cisco SPA112's all over the US. I currently use DNS SRV records with equal priority for the two servers to provide basic load balancing and redundancy. I would like to reduce latency and have the Cisco devices register to the closest server. I want to figure out whether or not I need to choose a DNS provider that supports GEO DNS.
Does anyone know if the Cisco devices already contain the logic to register to the closest server (without GEO DNS enabled) or do they require 'help' from a service supporting GEO DNS to determine the closest server?
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[Voip.ms] Guide Cisco 7941 getting started
Guide to getting VOIP.ms working on Cisco 7941
After buying my 7941 on ebay, I was driven mad trying to get this phone working on voip.ms. The guides on the internet were incomplete or using other services. I figured I would give my knowledge out there to all that want a cheap voip phone. (these go for ~$15 on ebay right now)
Step 1) When you get your phone, it most likely will have firmware on it for Cisco's call manager. This is not what we want. We want the SIP firmware to connect to voip.ms. Pay attention here because you cant use the latest firmware, the phone wont register with it. To get the firmware, go to: https://software.cisco.com/download/home/280083379/type/282074288/release/8.5(2)SR1
The firmware on that exact page 8.5(2)SR1 is what you will want. The exact file you want is the one that says "Phone Firmware Files Only - Compatible UCM" (the bottom one) Do not download the top one its not needed. In order to get this file, you will need to register an account with cisco. This account is free, but you must provide actual information as cisco's servers will look for falsafied info. If you get an email after you register saying your account must be verified, dont panic, i had that too and its a small waiting game for it to activate.
Step 2) Now that you have your firmware, You will need a TFTP server to upload it to the phone. This step is going to be cut into two parts. You will follow one or the other depending on your situation with your phone. If you can access the settings menu in your phone (settings button above the volume control) press it and then type * * # on the phone's keypad and go to Network Configuration >> IPv4 Configuration >> make sure DHCP is enabled and then scroll down to Alternate TFTP set that to yes. If you CAN'T set it, go to Step 2B. If you CAN set it, continue with step 2A below and skip 2B.
Step2A) This step assumes you can set "Alternate TFTP" to "Yes" under Network Config >> IPv4 Config. Scroll down to TFTP Server 1 (its in the same settings menu as the "Alternate TFTP") press edit and set the IP address you will be using for the TFTP server (a windows desktop computer is fine.) To enter the periods in the IP address, use the * button. Now on the desktop computer you secified in the phone, you will need a TFTP server. I recommend SolarWinds TFTP server. (https://www.solarwinds.com/free-tools/free-tftp-server) Once installed create a folder on your desktop named cisco and unzip the files from the zip we downloaded earlier into the folder. Put the XMLDefault.cnf.xml file i have for you in the folder with the files. Now in SolarWinds, hit file and configure and you should see start and stop buttons. Press stop if its started and at the botton fo the window hit browse and select your cisco folder on your desktop. Now hit start at the top and close the window. Now, on your phone press settings then * * # * *. The phone should reboot. It will boot normally and then reboot itself again. Don't press anything it should reboot into an upgrading screen and begin installing the firmware you downloaded.
Step2B) This step assumes you cannot set the TFTP server in the settings menu or you cannot access the settings menu at all for whatever reason. For you, there is a youtube video that explains how to get the firmware installed. https://www.youtube.com/watch?v=OTdeb1YMtsI Make you you use the firmware we downloaded in step 1!! After you follow these steps and get the proper firmware on the phone press the settings button then press * * # on the keypad and then go to Network Configuration >> IPv4 Configuration >> Alternate TFTP. Set that to yes then Settings button >> Network Configuration >> IPv4 Configuration >> TFTP Server1 and set this to an ip your will use for the tftp server (can be a windows desktop computer). (use * key as the decimal point)
Step 3) Alright! we got our firmware installed and we are almost done! Now time to get this puppy configured. I have included an XML file named "SEPMAC.cnf.xml" Go to settings on your phone and then network configuration. Your phone's MAC address should be shown as the 3rd setting down. in the downloaded XML file (SEPMAC.cnf.xml) replace the MAC portion of the filename with this MAC address. Example: SEP00670B84A05.cnf.xml. Next, right click the file and select edit (or open it in notepad or your favorite ACII editor). You'll wanna use the find tool in the editor and replace all instances of the following with your information. ##VOIPMS## - Replace with your voip.ms server of choice. ##LABEL## - this is the label that appears on the phone screen beside the line. You can put whatever you want here. ##VOIPMSUSER## - this is your voip.ms username number. ##VOIPNAME## name you want to be sent to VOIP.ms. Not sure what this is for but you can safely put your name such as John here. ##VOIPMSPASS## - your voip.ms password. Be sure to set this in your account settings. Save the file in your cisco folder.
Step 4) Download the dialplan.xml I provided. No need to edit it. Place it in the cisco folder.
Step 5) reboot your phone (settings button then * * # * *) With luck, your phone should reboot and download the xml files in the cisco folder and register with voip.ms. Congrats!
BONUS INFO!
Follow this URL to learn how to install custom backgrounds and other cool things with the phone. https://www.whizzy.org/2017/02/cisco-7941-asterisk-and-sip/#flash
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Getting a DID (phone #) from Localphone might be slow
I've obtained phone numbers (DID) from various providers and the process is generally pretty quick.
Usually it's right away, or if it's going to take a day or two days for a particular DID, the provider will let you know up front.
Trying to get a DID from Localphone, it said numbers were available for the US city I wanted, so I added funds to my balance and then I clicked to pay for the number.
AFTER clicking, a notice came up which said something like "It may take up to 14 days to get your number. It will probably be a lot less but be advised...."
This was not a special number or anything like that, and as said, it was in a city they said was available.
So my gripes with Localphone:
1) How on earth can it take 14 days (even "up to" 14 days) in an American city they said was available? How exactly are they obtaining these numbers if that be the case? Does it involve waiting for someone to pass away?
2) The "up to 14 day warning" did not appear until AFTER clicking to purchase the number.
Now, they DO allow you to "cancel" afterwards which I did after an hour or so.
Yes, I admit I am an impatient person, but I didn't want to have to wait "X" number of days which could have been 2 or 6 or 14.
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Localphone has good service and good prices.
It's the lack of upfront notice that I object to.
Again, this was for a new number, NOT a porting.
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[Voip.ms] VoIP.ms New Caller ID Regulations
In line with all the recent rulings in favor of protecting and empowering consumers, VoIP.ms will start enforcing valid Caller IDs as per the NANPA and ITU-T E.164 standards as of January 1st, 2020. This regulation will apply for all customers including reseller’s clients.
Considering that VoIP.ms already has call filtering solutions, enforcing Caller IDs is simply the cherry on the cake! In fact, we found value in joining all the other phone providers in this massive industry shift in order to ensure that consumers benefit from a minimum level of protection against nuisance calls.
Therefore, calls displaying the following Caller ID information will be blocked by our platform:
International calls with Caller ID containing less than 7 digits and greater than 15 digits;
North America calls with Caller ID not containing exactly 10 digits (exception will be made for 7 digits for 310-xxx numbers);
North America calls with Caller ID containing 10 digits, but with an unassigned NPA (first 3 digits of the number);
North America calls with Caller ID containing 10 digits, but where an unassignable NXX is used (i.e. the second block of 3 digits where the first digit is either zero (0) or one (1)).
Compliance with these standards will ensure proper call termination and reduce potential connection issues, thus resulting in a great call completion rate.
If you are passing your Caller ID from VoIP.ms customer portal, please review your Caller ID number to make sure it complies with these standards. In order to do so, head over to your portal, click “Main Menu”, “Account Settings”, then the “General” tab, edit the “Caller ID number” field and hit save. You can also do this for your Sub Accounts by going to "Sub Accounts" menu, then click on "Manage Sub Accounts".
If you are passing your Caller ID from your own system such as a PBX or a switch, you only need to make sure it follows the guidelines aforementioned.
On top of this, the good news is that we have a few features in our roadmap to allow you to add an extra layer of protection against nuisance calls.
For more information, check out our Wiki article right here or directly contact our support.
https://wiki.voip.ms/article/Caller_ID
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[PBX] New E911 Requirements Coming in Feb. 2020
If you are a VoIP provider, you may wish to review the FCC's new E911 rules that take effect on Feb. 16, 2020. Incredible PBX 2020 deployments will be compliant in late December, 2019.
https://www.wileyrein.com/newsroom-articles-FCC-Adopts-New-E911-Rules.html
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Obihai OBi200/202/302 + OBi1022/1032/1062 + OBi50x firmware mods
So I want to add the ability to configure these devices for GV using oauth without obitalk, similar to the changes for the obi100 (and add an ssh server, for grins).
I think I have the MD5s in the firmware file worked out (its the same "Goodbye! Reboot Now" garbage as the 100), and I see where the oauth refresh token code is, so it should be pretty straightforward unless there is code signing that I missed.
The only hiccup is... I don't actually have an OBi20x :-(
Anyone have one of these devices that wants to be a guinea pig? You should definitely have a way to SPI the flash back *when* i brick the thing the first couple tries...
[or if someone has one sitting in the closet, you could just send it to me. ill name the fw after you :-)]
EDIT: speaking of flash, its supposed to have a w25x128 on board, but is it the SOIC package or some BGA madness?
QUICK SUMMARY:
Custom firmware made for all obi devices, thanks to the help of generous hardware donations and bold testers.
See obifirmware.com to download latest.
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ObiHAI Obi100/Obi110 Firmware Mod Discussion
So many of us have the Obi1xx series devices that recently stopped connecting to google servers due to a certificate update. This thread is intended to discuss the possibility of modifying the firmware to update the certificate and let these devices work with Google Voice again.
These devices are based on a MIPS-X processor similar to the Sipura ATAs and there is not a lot of tools/docs out there about them except for a Yahoo Group mostly related to DVD player chipsets. The venerable DogFace05 who was an expert with these types of devices once posted that he was able to extract this firmware sucessfully. Not sure if he is still around. Anyone else familiar with this architecture?
It seems that the place to start looking is the end of the firmware update file which contains some kind of table. Then there seems to be a loader section which presumably decompresses one or more other sections and loads them to RAM before executing the firmware.
So the questions are:
Can we extract, modify, and repack the firmware and create proper checksums/signatures?
Where is the certificate stored and in what format?
Can we drop in a new certificate without messing up other things (e.g. if the length of the certificate has changed) or do we need to move the certificate and patch the code pointing to it?
Is updating the certificate enough or is the codebase missing support that is necessary (e.g. if key length has changed)?
Anyone who wants to participate please post your thoughts.
Thanks
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CWU: New DIDs not available in 2020
Is there any reading between the lines to be done here? Just got this from CallWithUs:
Dear Customers,
Effective January 1st 2020 you will not be able to purchase new DIDs.
We will continue support of the existing DIDs and third party DIDs
forwarded to our service. If your DID will be automatically removed
because of insufficient account balance on the DID due date, you will
not be able to purchase the DID again and restore the incoming
service. You can set a low account balance reminder in "user settings"
menu in customer portal.
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CallWithUs.com team
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Canadian LECs to start blocking calls with bad Caller IDs
Not sure if this has been posted before, but since the deadline is approaching I thought it worth raising some visibility here.
The CRTC has mandated universal call blocking of invalid numbers that must be in place all carriers no later than December 19, 2019.
This means calls will be rejected if they contain an obviously incorrect Caller ID. I foresee many users who haven't polished their setups will see their calls blocked, especially if they haven't configured an outbound number or if they're trying to pass an extension number to the PSTN as their Caller ID.
https://crtc.gc.ca/eng/archive/2018/2018-484.htm
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