Been using Obis for years and have set up multiple gateways in addition to the sp(x) but just recently had one of the "Hey, how does this work?" moments.
Since the gateways only make use of 4 fields to make the call, AccessNumber, DigitMap, AuthUserID and AuthPassword, the only other option I've ever added was op=sn; we've always prefixed the accessnumber with sp(x).
I suspect the gw pulls the settings from you sp(x) you prefix. But seems different providers have different setting requirement: i.e. keep alives, ports, etc. So, one size doesn't fit all, but it seems to work 90% of the time based on 4 fields.
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Clarification - how do Obi Gateways get its other settings?
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Wag the dog
I don't know if you all noticed. While megacorp was distracting you all by trying to make you think they are being hurt by a couple guys in a garage. They just removed another major independent buying choice. They really don't give a *** about that software you aren't paying them for. They answer to investors NOT YOU. So the only thing they probably care about is a new company making hardware. Well where are they going to sell it. I doubt e4 will have anything non megacorp now. How long before voipsupply starts shifting. I would guess if you call e4 or voipsupply you couldn't buy hardware from those other guys. Seriously quit playing the game
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News Item: FTC Warns 19 VoIP Service Providers about robocalling
FTC Warns 19 VoIP Service Providers That ‘Assisting and Facilitating’ Illegal Telemarketing or Robocalling is Against the Law
The 19 service providers were not named:
https://www.ftc.gov/news-events/press-releases/2020/01/ftc-warns-19-voip-service-providers-assisting-facilitating
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[General] Frontier FIos -> IPComms SIP -> Grandstream GVX3350 = FAIL?? Why?
Greetings,
I have an account through ipcomms and I can plug the user info into a software sip dialer and dial all day long no problem0000, but when I plug it into my phone I just get a busy signal when I dial. I have Grandstream on chat with me trying to work it through for several hours yesterday to no avail.
I have read that Frontier can be dicey with voip other than their own solution, but if that were the problem then I wouldnt think the softdialer would work either.
I have udp 5060, 20000-60000, and I believe a few others opened inbound - nothing being blocked outbound. On the IPCOMMS side of things, I do see the device go REGISTERED - So I know we are at least getting that far.
Any suggestions?
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Link: This App Automatically Cancels and Sues Robocallers
I'm sure not everyone will want to do this but for those truly motivated to make life hell for illegal robocallers, you may like this:
This App Automatically Cancels and Sues Robocallers
On a related note, just this morning calls from several numbers came into the system I help with, and they were "car warranty" robocalls, but all came from different numbers. But in checking telcodata I found all those numbers were in thousands blocks owned by "Peerless Networks", a company that has managed to garner some not-so-favorable BBB reviews. It would be interesting to know what percentage of robocalls originate from customers of that company (which I had never heard of before until today).
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Need a FAX server solution
I have had good luck with Elastix and Hylafax on voip.ms g.711 codec. I just retired my in house server and moved my IVR to voip.ms.
Now I just need in-house fax solution.
RonR's Raspberry Pi solution will work (anybody using it in a small business?) but, is there another simple stand-alone solution just for the fax instead of installing Asterisk if there is no need for IVR? Winprint-Hylafax and fax to PDF worked great on Elastix and I would like to have a similar solution. I cannot use voip.ms fax service for privacy concerns. Yes, voip data can be tapped too but it is reasonable best effort if I have a in-house fax recipient server vs. incoming faxes sitting on outside server.
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[PBX] 911 Changes for February - Kari's Law
Kari’s Law - Dial 911 Without a Prefix
RAY BAUM’S Act - ensure that “dispatchable location” is conveyed with 911 calls
“Dispatchable location” is defined as “the street address of the calling party, and additional information such as room number, floor number, or similar information
Howdy,
I got an email from bulkvs about Kari's Law.. I don't remember seeing it discussed before, and even the email didn't have information about the law or requirements of, so creating a thread for it..
Basically as I now understand it, calls to 911 now have to go through without needing to dial 9 (or any other prefix) for an 'outside line', for the US anyways.
I believe most of us already have dial-plans in place for this, but just as a reminder..
One of the places I used to work had an extension 911 in the building, and it was getting calls that were supposed to go to 911 (manufacturing plant). Their solution was to simply change the extension number.. Argh..
I don't know if the Ray Baum's Act part is part of the requirements for February or not, this seems like it will need 911 cooperation to achieve.
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[Voip.ms] Hacked Voip.ms accounts
Anyone getting their sub accounts hacked and getting high volumes of calls made in the space of a few minutes? I have gotten hit several times this year and each time it costed me a few hundred dollars with no recourse from Voip.ms. The only reply I keep getting is to change my password even when I point out that I have changed my password and the same account was hacked again.
Shouldn’t there be a simple method of stopping calls when the account goes zero, why allow the account to go -250.00+ before stopping new calls?
I don’t seem to be getting anywhere with getting a solution so looking for ideas on another provider that can do reseller sub-accounts that can accept payments directly. Before Voip.ms I was with net2phone but setting up different rate tables and managing customer payments was a nightmare.
Voip.ms has an ok model but my biggest challenge is separate sub-account security settings as even with just that I am sure I could limit my customer breaches.
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[PBX] FreePBX for the Raspberry Pi
The included script (install) and archive (install.tar.gz) will build
FreePBX 13, 14, or 15 plus Asterisk 13, 15, 16, or 17 on a Raspberry Pi.
iptables, dnsmasq, and exim4 are also installed.
Installation takes approximately 35 minutes to complete on a Raspberry Pi 4B.
Download the latest Raspbian image:
https://downloads.raspberrypi.org/raspbian_lite_latest
Write the image to an 8 GB or larger SD card. To accomplish this, I recommend Etcher or imageUSB:
https://etcher.io/ or http://osforensics.com/downloads/imageusb.zip
Create an empty file named ssh in the /boot/ directory (type NUL > ssh).
Connect the Raspberry Pi to your LAN using an Ethernet cable.
Insert the SD card and power up the Raspberry Pi.
Copy install and install.tar.gz to the /home/pi directory. To accomplish this, I recommend WinSCP:
https://winscp.net/eng/download.php
Using an SSH client, log in using pi:raspberry. To accomplish this, I recommend PuTTY:
http://www.chiark.greenend.org.uk/~sgtatham/putty/download.html
Make the install script executable:
$ chmod +x install
Run the install script:
$ sudo ./install
When prompted:
Set pi user password
Set root user password
Select FreePBX version
Select Asterisk version
Answer Edge option
Answer IPv6 option ('No' recommended)
Review selections
Set Hostname (Item 2 / N1 - Hostname: FreePBX)
Set Localisation Options - Locale (Item 4 / I1)
Set Localisation Options - Timezone (Item 4 / I2 - in US, use America, not US)
Expand Filesystem (Item 7 / A1)
Finish / Reboot Now: No
The Raspberry Pi will reboot.
Log in as root.
If desired, enable PuTTY logging when prompted.
The system will be updated and then reboot.
Log in as root.
If desired, enable PuTTY logging when prompted.
Confirm install.
Installation will proceed unattended and then reboot.
Log in as root.
Installation will complete.
GVSIP
=====
To use Google Voice SIP trunks, Asterisk 17 MUST be used.
Configure FreePBX settings as follows (FreePBX 14 illustrated):
Settings -> Advanced Settings -> Dialplan and Operational
SIP Channel Driver = both
Settings -> Asterisk SIP Settings -> General SIP Settings tab -> Media Transport Settings
STUN Server Address = stun.l.google.com:19302
Settings -> Asterisk SIP Settings -> Chan SIP Settings tab -> Advanced General Settings
Bind Port = 5160
Settings -> Asterisk SIP Settings -> Chan SIP Settings tab -> Advanced General Settings
TLS Bind Port = 5161
Settings -> Asterisk SIP Settings -> Chan PJSIP Settings tab -> tls
tls - 0.0.0.0 - All = Yes
Settings -> Asterisk SIP Settings -> Chan PJSIP Settings tab -> 0.0.0.0 (udp)
Port to Listen On = 5060
Settings -> Asterisk SIP Settings -> Chan PJSIP Settings tab -> 0.0.0.0 (tls)
Port to Listen On = 5061
If any changes are necessary, reboot after all changes have been submitted/applied and recheck everything.
Running:
asterisk -rx "module show like pj"
should display around 48 loaded modules with all but around 2 of them displaying a status of "Running".
Install Certificate Manager module (if not already installed).
Run: mv /root/obihai.* /etc/asterisk/keys/
Run: chown asterisk. /etc/asterisk/keys/obihai*
Click: Admin -> Certificate Management -> Import Locally
Settings -> Asterisk SIP Settings -> Chan PJSIP Settings tab -> TLS/SSL/SRTP Settings
Certificate Manager = obihai
Configure gvsip.dat for your Google Voice account(s). If you have more than one Google Voice account, copy
the five [gvsip1] sections to [gvsip2], [gvsip3], etc. Then edit each of the five [gvsipN] groups as follows:
Change (3 places):
NNNNNNNNNN to {10-digit Google Voice number}
Update:
refresh_token={Google Voice Refresh Token}
oauth_clientid={Google Voice Client ID}
oauth_secret={Google Voice Client Secret}
contact_header_params=obn={Google Voice SIP Name}
Upon completion, copy gvsip.dat to /etc/asterisk/pjsip_custom_post.conf:
cp gvsip.dat /etc/asterisk/pjsip_custom_post.conf
For each Google Voice account, create a Custom Trunk as follows:
Connectivity -> Trunks -> Add Trunk -> Add Custom Trunk - General tab
Outbound CallerID = <+{10-digit Google Voice number}+>
Connectivity -> Trunks -> Add Trunk -> Add Custom Trunk - General tab
CID Options = Force Trunk CID
Connectivity -> Trunks -> Add Trunk -> Add Custom Trunk - custom Settings tab
Custom Dial String = PJSIP/+$OUTNUM$@gvsipN (Replace 'gvsipN' with the [gvsipN] group number from gvsip.dat)
Upon completion of GVSIP configuration, run: fwconsole restart
Utility scripts included in /root:
install-opus
============
Install OPUS Codec
abn / dbn / ebn / ibn / qbn
===========================
Add / Delete / Export / Import / Query Blacklist Number
add-fcc-blacklist / del-fcc-blacklist
=====================================
Add / Delete FCC Blacklist
exclusions.fcc
==============
Numbers to Exclude from FCC Blacklist
ipt-add / ipt-del / ipt-chk / ipt-dsp
=====================================
Add / Delete / Check / Display iptables Entries
cell-phone-presence-bt / cell-phone-presence-obi
================================================
Cell Phone Presence Detection
pbx-backup / pbx-restore
========================
Backup / Restore PBX Configuration
image-backup / image-check / image-compare / image-set-ptuuid / image-shrink / image-mount
==========================================================================================
Backup / Check / Compare / Set PTUUID / Shrink / Mount an Image of the System SD Card
upgrade
=======
Upgrade / Update Linux
asterisk-upg-to-15
==================
Upgrade Asterisk 13/14 to Asterisk 15
asterisk-upg-to-16
==================
Upgrade Asterisk 13/14/15 to Asterisk 16
asterisk-upgrade
================
Upgrade Asterisk
set-timezone
============
Set System and PHP Time Zone
regen-ssh-keys
==============
Regenerate SSH Keys
clear-cache / clear-logs
========================
Clear Cache / Logs
install-nut
===========
Install Network UPS Tools
remove-nut
==========
Remove Network UPS Tools
install-zram
============
Install ZRAM swap file
remove-zram
===========
Remove ZRAM swap file
install-fax
===========
Install Hylafax Server
add-fax-extension
=================
Add Hylafax Extension
del-fax-extension
=================
Delete Hylafax Extension
purge-fax
=========
Purge HylaFAX Server
HylaFAX fax server
==================
1. Execute install-fax: ./install-fax
2. Execute add-fax-extension: ./add-fax-extension
Multiple fax exntsions may be added to support simultaneous sending and/or receiving of faxes.
SendFax
=======
SendFax is a program to send a fax file from Windows to a HylaFAX fax server.
No installation is required and no changes are made to your system.
Supported file tpyes are pdf, ps, tif, and tiff.
A cover page can be generated and prepended to outgoing faxes.
Leaving 'File to Send' empty will send only a cover page.
To configure, click Edit -> Options:
IP Address: (the IP address of your HylaFAX server)
Port Number: (the port number of your HylaFax server, normally 4559)
Username: (your username on your HylaFAX server, normally root)
Password: (your password on your HylaFAX server, normally blank)
Email Address: (the email address to deliver notifications to)
Notifications: (notification types to be sent)
Page Chop: (which pages to chop trailing whitespace from)
Threshold: (minimum trailing whitespace (in.) before chopping is used)
Modem: (which modem to use for outgoing faxes, normally blank)
Cover Folder: (folder to save cover page information in)
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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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Not able to log in to my Callcentric account.
I'm having problems with my connection to Callcentric. It just continues to timeout and keeps saying to try again. Not having any issues with other sites. Anyone else having this problem?
Thanks....
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[General] Home user lookin to replace Vonage. I have an OBI200.
I can no longer afford Vonage. I have an OBI200 and want to switch to a cheap but reliable provider. I'm a very light phone user so I don't need a ton of minutes. Most if not all of my calls will be in the US. I also have a BYOD tracfone cell as a backup. I want caller ID with name. I also want to port my Vonage number to the new service.
1. How long does it take to port a number and start using the new service?
2. I feel like I'm going to miss my vonage device and an IP phone may have the features I like. Will any of the cheap used or refurb IP phones on ebay work with the service? Any suggestions? I like to be a able to redial a number (my normal phone saves all button presses like the phone number and prompts where Vonage "Call Logs" just saves or can dial the received, dialed and missed numbers) and check voicemail.
3. It looks like CallCentric has free porting right now. Do they have caller ID with name?
4. Any way to get caller name with Anveo without having to upgrade to a higher tiered service?
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OBi Phone dial plan question
I have an OBi1032 running 5.1.11 (Build: 4858EX.1311-olisom5b).
Recently I added another SP and was confused by the dial plan behaviour. I have a programmable key configured as Line Monitor for SP2. When I press it, I see "Enter Number" on the display followed by the name I assigned to SP2.
Then I dial a number and it follows the default dial plan and routes the call via SP1.
Is there a way to make a programmable key only route calls via a specific SP?
Thanks.
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Talkatone Revoked Ported In Number
I use to have RingTo to park my number and about two years ago ported my number to Talkatone. I have been pretty good with making a call here and there but they finally got me, I did not see their warning email cluttered in junk email and they revoked my ported in number . I realized about two weeks after the fact. I just emailed their support but I doubt they will do anything. It is kind of BS that they do this, I understand you get what you pay for - in hindsight I should have just paid $25 to port to GV but then I would have to use a mobile carrier as a middle ground. Surprised there are not more FCC rules that prevent this - they should give you more than one email warning the day before
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[Voip.ms] Still waiting for promised MMS service at VOIP.MS
Regarding Bandwidth.com MMS on VOIP.MS's DID, I posted this on July 7, 2019:
www.dslreports.com/forum/r32441498-Voip-ms-MMS-on-VOIP-MS-s-Bandwidth-DIDs
...and user Steve VoIPms replied in that thread on July 9, 2019:
"Well we have good news for you, it's on our schedule already and
will be implemented pretty soon. We had to push it back a few weeks
due to other more important releases."
Five months later, nothing. I'm in the beta program, so I think that
I'd know if they had rolled it out for testing.
(In general, VOIP.MS's public announcements about timelines for new
features aren't reliable, and this announcement fits that pattern. I
like the company, but they have some quirks, and this is one.)
Has any of you heard further about when MMS might be implemented
at VOIP.MS?
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[Equipment] Arris T25 - Xfinity call waiting id
Hi folks, I'm writing on behalf of a friend. She purchased and installed an Arris T25 Docsis 3.1 voice modem so she wouldn't have to pay Comcast $14/month rental any more. She followed the steps to get Comcast to recognize the T25's MAC address, etc., without an issue.
Everything is working great, *except* caller ID when she's on the phone. She'll hear the tone, bit there is no caller id information on the handset display. Caller ID appears on the TV, and displays on the handset while it isn't in use, but not during a call-waiting situation.
Any suggestions? Is there a tweak that can be made in the T25's configuration? Or, does the Evil Empire need to get involved?
Thanks!
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Polycom speakerphone quality
A friend needs to replace his desk phone. He uses speakerphone almost exclusively. Doubletalk performance is important; he often has trouble interrupting the other party, being soft spoken and sitting about two feet from the mic.
I've been looking at the Polycom VVX series. The simplest/cheapest model (VVX 300) has all the functionality he needs, but he would consider a more expensive one if it sounded clearer to him and/or the remote party.
Can someone with experience with this series please comment on whether the higher end phones sound any better? Also, if you have both Polycom and another brand that sounds better, please post details.
Thanks.
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[Equipment] Yealink T21PE2 - Goodwill Find - Need Desk Stand
My local Goodwill store had three Yealink T21PE2 phones on sale for $4.99 each. I quickly grabbed them without noticing they did not include the desk stand. Does anyone have surplus stands available? If not, I can purchase them for about $9 from several online sellers.
The part number seems to be STAND-T21.
Thank you.
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[Anveo] using GS wave softphone on IOS outside the US
Hi,
I have Anveo as my home service via OBI... I travel outside the US often but I have a question... as far as I can see this GS wave can be installed and configured to work under iPhone, if I use it outside the US (connect via wifi) would it count as long distance form that country to the US of it would be like if I was in the US?
Just wondering, thanks!
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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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