I have Obi-202 connected to a regular phone, everything works fine.
What I would like to do is to run SIP client on my Windows desktop at the same time so that if someone calls me it would ring both the phone and SIP client at the same time and I would be able to pickup either.
Is this something that Anveo supports - 2 devices registered and active at the same time? If not - is there a way to still get it to work somehow using the Call Flow?
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[Anveo] Anveo - using Obi 202 and SIP client on PC at the same time
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[Asterisk] ipv4 Asterisk and ipv6 cellphone clients
Happy Easter! Hope everyone is staying safe!
Please consider this predicament:
Some of my Asterisk users are on IOS and android cellphones. Recently, a few of those have started getting IPV6 addresses from their carriers.
On this side me and my ISP are not "ipv6 aware" yet. This leads to issues:
[2020-04-11 10:45:17] ERROR[15303]: netsock2.c:303 ast_sockaddr_resolve: getaddrinfo(" <<some ipv6 address>> ", "41730", ...): Address family for hostname not supported
I am sure there must be others who have gone this route and experienced similar hurdle before - every time I post a question here in this forum I gain significant insight. Please share your thoughts on this situation.
Thanks!
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Has anyone tried using ipsum as a way to fend off attackers?
I was reading another forum and noticed a reference to ipsum, and looking at the instructions on the linked page, it seems relatively easy to install and use, particularly if you already use iptables. It got me to wondering if using that would help stop some of the attempted breakins from sip scanners. I've never had one be successful yet, but I figure might as well be proactive and try to block the ones I can. A fixed blacklist/whitelist is impractical because some of my users are not at fixed ip addresses, although I do have other ways to lock down the system, and I was just looking at this a a bit of added protection. I just wondered if anyone else might be using this and if so, do you have any installation or usage tips?
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Unlocking a Primus SPA122
Hi all,
I've got an old spa122 that I want to activate with a new provider but it's currently lock to primus. Anyone here know if there's a way to unlock it? I know a couple year back there was a user who could do that on the forum but can't find him anymore.
What I don't like is that I paid the ata from primus and I should have the right to do what I want with it after I leave primus.
Thank
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[Voip.ms] Pulling CDR Entries via VoIP.ms API fails
Good Sunday.
In order to programmatically pull CDR entries from my VoIP.ms accounts, I configured and uploaded to my website the "07.CDR.php" example REST for PHP script available here: https://voip.ms/api/v1/API.zip (requires logging-in).
Even though the script expressly includes "method=getCDR", invoking the script with curl or using a web browser results in a "missing_method" error message.
I have reviewed all relevant API documents here: https://voip.ms/m/apidocs.php (requires logging-in).
The documents define missing_method as: missing_method Method must be provided when using the REST/JSON API
As far as I can tell, the script includes all required functions and parameters. Unfortunately, VoIP.ms disclaims support, as follows:
Please note that we do NOT provide support with your programming language. The only support provided by our staff is for bugs, documentation errors, documentation missing or other questions regarding the functionality of the API. Support regarding the API will be addressed via the ticketing system only. Our programmers do not have access to the Live Chat.
(Emphasis added)
Has anyone who cares to share successfully pulled CDR entries with the REST/JSON API?
Thank you.
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[Unlock] New Grandstream ATA used by Vonage [Unlocked]
Looks like Vonage uses a new Grandstream ATA for their service, model HT802.
It's Grandstream newest generation of ATAs, with 2 ports.
Currently available at Walmart and BestBuy for $10.
If unlockable, this could be a nice, inexpensive 2 port ATA. Interesting that they are using a MicroUSB port for power.
I don't have a way of getting any of these soon to test as I am located in Canada. However, I am sure there are similarities between unlocking those and the BasicTalk HT701 (though probably they will not accept Mackey's firmware without some modifications).
Edit: seems like some new members are jumping into unlocking these adapters without going through the thread and reading all the comments. The proper unlocking method, tested by many already is posted here.
I still recommend reading through all the posts to learn from other members experience before starting anything.
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Polycom OBI200 + Callcentric + Double NAT
Hi folks:
I've got a less than ideal network situation in which I'm trying to use a Polycom OBi200 device with Callcentric and I'm having some issues and am looking for suggestions. My elderly mother just went into assisted living and they don't provide phone service for free, but they do have free wifi for the residents (well, actually the cost of the wifi is obviously baked into the rent, but you know what I mean). I installed an Ubiquity ER-X router in her apartment with a TP-Link AC750 used as a wifi to ethernet bridge in order to present the wifi to the ER-X's WAN port as the internet connection. I used OpenVPN on the ER-X to establish an outgoing VPN connection to my home so that I can access the devices behind the router, which include a scanner and a printer. Lastly, I installed a Ubiquity AP connected to a LAN port on the ER-X to provide a local secure wifi network to her apartment which I can connect to my home network via the VPN.
I bought the wifi dongle for the Polycom OBi200, which is connected to the local secure wifi, as the phone needed to be connected at a location other than where the router is situated. This means that the Obi200 is now double natted. The problem I'm having is that mom can make outbound calls just fine all the time, but inbound calls intermittently fail, and when they do Callcentric reports that the extension isn't registered. When I check the status of the Obi200 from the Ubiquity wifi AP controller app it shows connected, and I can successfully ping it from the ER-X router console.
My understanding is that SIP communicates via UDP and my suspicion is that the UDP connection has timed out either in my router or in the upstream assisted living facility's router, which is causing Callcentric to mark the extension as down, but as soon as an outgoing call is made everything comes back up for a while. I turned on X_KeepAliveEnable and set the X_KeepAliveExpires value to 120 per the Callcentric instructions.
As I'm not sure about the full implications of these settings I wanted to find out if I could simply set the KeepAliveExpires down to something like 15 seconds (I'm assuming that the units are seconds and not minutes) in an effort to keep this from happening. There is also a SIP conntracking module setting in the ER-X router that I can tinker with as was suggested in some threads, but I'm not sure what that does exactly either.
I will say that the call quality is quite good when it's working, so I don't think the issue is that the facility's wifi is being overloaded, and also it's a new facility and only about a quarter full at this point, so there aren't near as many people there yet as the system was hopefully designed to handle.
Comments and suggestions welcome.
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Any Ooma Core (original Ooma Hub) users left here?
The Ooma Core service level is what came with the original Ooma Hub and is the truly free one (later Ooma offerings charged monthly taxes/fees).
Mine has been going strong for 11 years, although I don't use it much now ever since I lost incoming CNAM (caller ID w/ name) at some point - now only the ph# shows and not the name. Free incoming CNAM was one of the perks of Ooma Core (later Ooma offerings you had to subscribe to Premier service to get it).
Just curious if anyone else has been hanging onto the Ooma core service, and whether you still have incoming CNAM or not?
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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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[General] SIP protocol and NAT...
I just stumbled across this VoIP KB that might interest others:
https://wiki.4psa.com/display/KB/SIP+protocol+and+NAT+problems
OE
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Question on CallerID spoofing - Call log
Hi,
Not sure if this is the place to ask, but here goes.
Yesterday, my phone rang, and when I looked at the CallerID, it was my name and number displayed. I didn't answer, but instead, decided to log in to Vonage to see what the "Recent Calls" showed. (On calls where they populate the name with a long string, I don't see the number on my handset, but the Vonage log will show a number.)
If it had only displayed the incoming call, I would have just chalked it up to spoofing. What surprised me was that there were two entries for the call. The incoming, along with an outgoing, showing the same time/length, leading me to wonder if they were able to initiate the call from my own device (VDV23-VD).
Customer Service was more frustrating than helpful, with the guy putting me on Hold, then coming back and asking the same question of "Did I actually see two recorded entries in my log" (which I had explained was the very reason for my call). When he put me on Hold for the 3rd time, I hung up. (His initial suggestion was to use the Selective Call Blocking to block my own number.)
So, I was just wondering if anyone had seen this before and/or am I worrying about nothing?
Thanks in advance!
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How does gv911.com compare to voip.ms?
With Verizon FIOS as our home ISP, I have successfully ported our two original copper telephone land line numbers into GoogleVoice, and our physical phones are connected through two little ObiHai boxes to GoogleVoice. The two numbers are two separate GoogleVoice accounts with two separate Gmail email names. (The two Gmail names were set up only for the Google Voice use - they are in the form of [telnumber][firstnameinitial]@gmail.com. We will never be giving those Gmail addresses to anyone.)
Now, I would like to add E911 to at least one of these phone lines, and (if no extra cost) both, at the lowest monthly cost. (If there are any taxes and fees, we are in Manhattan on the lower upper west side.)
So, in your experience, how does gv911.com compare to voip.ms, for both reliability and cost? And please also compare Anveo.
Please give me your thoughts first for a single GoogleVoice-ObiHai line. And then please let me know whether there is any way to make the service work on both our lines but for no extra charge.
Thanks!
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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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Use your free Voxbeam DID with Callcentric Free Voicmail
FREE IS A VERY GOOD PRICE!
I have seen at least one other post where someone wanted to do this. They never came up with a solution.
After moving my asterisk from a VPS machine on the web to a Raspberry Pi 3 at home it seemed more desirable to do it this way.
I get free DIDs from Voxbeam but no voice mail.
I get free Voice mail from Callcentric but have no DIDs there
I do not want Voicemail on My asterisk (Raspberry Pi 3) as I run several apps on my Pi 3 and prefer to not have any recordings there and keep CPU overhead as low as possible.
It took me some head banging and a while to figure this out. I messed with it a few months back but I guess I was not thinking straight.
You will need
Voxbeam Account
Voxbeam DID
Callcentric Account
Register to Callcentric account
Voicemail activated on the Callcentric extension you plan to use.
At first I tried directing a Voxbeam DID to 1777XXXXXXX@in.callcentric.com. That DOES NOT work. Either Voxbeam blocks it , or they do not support DNS SRV records.
So as a work-around after some head banging, I tried sending the call to *4621777XXXXXXX@sipbroker.com in the voxbeam portal, and that DOES work!
Calls will go from Voxbeam to Callcentric by way of SIPBroker but I am getting good low latency and high quality.
I think you could even use cal treatments oif you wanted to.
I have *97 mapped in my asterisk to dial into the Callcentric Voice Mail on the callcentric trunk.
One shortcoming of the Callcentric Voicemail however is that it appears you can not disable the "please leave a message after the tone" from playing after the personal greeting which really sucks for multi-lingual greetings.
It is also worth noting that after years of problems with inbounds on Callcentric, the new version of the configuration they have posted for asterisk 14 is working 100% on my asterisk 13 on Raspberry Pi. https://www.callcentric.com/support/device/asterisk/14
I know I am not the only one that battled with inbounds from Callcentric. I also from time to time saw problems with inbounds from Voxbeam to asterisk even when my asterisk was on a fixed public IP . With this configuration running for several days it seems rock solid.
BTW I have a fistfull of DIDs from South Boston , NY State (917) , and London (208) that are not being used if someone needs a freebie. PM Me , first come, first served.
Mark
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[Voip.ms] Voip.ms - set toll-free call routing on a per-did basis?
Unless I just don't see this in the account portal - when setting the toll-free out-bound routing option (ie value vs premium) for toll-free calling it seems to be a setting that is applied account-wide vs on a per-did basis (if you have an account with more than 1 DID like I have). I have a fax and credit-card terminal connected to a secondary ATA line with its own DID and was thinking that maybe I'd have better performance for those devices when they dial a toll-free number if I had routing set as premium. For my other DID's I wouldn't need premium routing when calling out (voice) on those.
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[Future9] Is Future Nine Down?
1) Phone isn't registering
2) Can't login to "My Account" at https://www.future-nine.com/A2BCustomer_UI/ . Get the message "Connection failed".
https://www.future-nine.com/A2BCustomer_UI/
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Bell Canada Application to block Wangiri fraud calls
Saw this discussed in French in the DSLR Canadian Chat French forum.
BellCanada asked the CRTC (their FCC) to block this type of fraud call:
Bell Canada et al. explained that Wangiri fraud calls are made mostly to Canadian mobile wireless service subscribers. The calls appear to originate from overseas. The telephone receiving the call is allowed to ring only once or twice, in the hope that the recipient will be intrigued enough to call back the number appearing on their call display.
When they do, the payment structure for the processing of the overseas call results in the fraudulent actors receiving a payment. A huge volume of such calls are performed....
For instance, between 16 and 20 March 2020 alone, Bell Canada et al. recorded more than six million Wangiri fraud calls on their networks.
https://crtc.gc.ca/eng/archive/2020/2020-125.htm
So we all know this scam, but I didn't know the word [Wangiri] which in Japanese means something like "one and cut".
Good to be aware of.
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BellCanada's request to the CRTC to block these calls was supported by their competitor Rogers, but was partly opposed by the Internet Society Canada Chapter (ISCC).
Don't know what their beef is.
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need some advice on a fax solution
have a satellite medical office (hipaa laws) that is going trying to get some government work (possible security concerns). we need to have a fairly stable setup in place and working before we bid.
we have a fiber internet connection that we have our vpn and voip on.
owner is sending a hardly used lexmark mx810dfe over there this weekend from one of his other offices.
at&t is a joke/no go for an analog fax line to install over there, especially if we dont get any work out of it.
i saw the other thread for a fax server solution but wasnt sure if it was bad form to hijack that thread asking for my own advice but my difference is that i have an actual fax machine to use.
volume could actually be pretty heavy on some days, 100's of pages to and from many entities 24 hours/day.
does any of the voip providers provide a rather rock solid T.38 setup where I could use an ATA adapter? if so how much do i need to worry about failed incoming/outgoing?
or is there a nice all in one fax server that i can setup/buy to to hook up to the fax machine/network?
i know services like faxage but with that volume are we throwing away money each month for something that can be done in-house?
also worried about with services like faxage, i need to be able to have employees be able to walk up to fax machine, put a paper in the feeder, type in a number, and hit send. I cant pre-setup the many numbers that we will faxing to b/c they will have new multiple fax numbers we need to send to daily/weekly.
TIA
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RCA DECT cordless VOIP phones questions
I have been looking for a cordless voip phone and ran into these
https://www.ebay.com/itm/Brand-New-RCA-IP160S-VoIP-Business-Telephone-System-Service/192789977352?hash=item2ce32d9908:g:OpcAAOSwywRaOAsR
the one above mentions service, as well as the desk phone below, but not all do .
I am wondering if some of these are locked to a provider and others are unlocked as this document
https://www.manualslib.com/manual/574482/Rca-Ip160.html refers to programming SIP settings.
What really blew my mind was that the system offers cordless desk phones as well.
https://www.ebay.com/itm/RCA-IP070S-Cordless-Accessory-Deskphone-For-RCA-VoIP-Office-System-IP160S-IP170S/324140558552?hash=item4b784868d8:g:mHIAAOSwmmxW6tUa
does any one know?
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[Anveo] Transfer to conference call
I'm looking for a way to setup an Anveo call flow that can transfer someone to a third-party conference call number and automatically dial the conference # after connecting. The Transfer call control doesn't seem to support continuing the call flow to be able to use the Send Keys control, for example. And it doesn't seem to support using commas in the phone number to indicate pauses like you can do on a cell phone.
The only hacky way I think is to use an audio file of dialing the necessary digits and add that to the "Call Announcement" in the Transfer control.
Any other ideas?
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