From the promo:
"Paying to port your numbers? Get paid instead. Get $1 free credit for every number you port to Telnyx during Black Friday week only. Hurry, offer ends soon!"
https://telnyx.com/products/number-porting
US numbers are $1/mo. each. + $0.0075/min. geographic or $0.015/min. toll-free.
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Telnyx "beyond free" porting Black Friday
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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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[PBX] Trouble setting up PBX extension on remote Obi200
Hello:
I have a friend in Florida who has an Obi200 which has voip.ms working fine on SP1.
I have a friend here in Quebec who has an Obi200 which has voip.ms working fine on SP1 *** AND *** an extension from my “rentpbx.com” PBX working fine on SP2.
Try as I might, I cannot get an extension from my PBX to register on my friends Obi200 in Florida: “Register Failed: No Response From Server”
Could this problem be caused by something like Fail2Ban or is it an Obi setting?
Thanks for any suggestions,
Rob.
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Clicking red Submit button in FreePBX now causes PJSIP failure
Curious whether anyone else has seen this - running FreePBX 14.0.13.4 with Asterisk 13.28.0 on a system running Debian 3.16.72-1 (x86_64) and now for some reason any time I make a change in the FreePBX GUI and click the red button to submit the changes, I get messages like this in the CLI:
[2019-09-01 15:31:41] ERROR[10207]: res_pjsip.c:4058 endpt_send_request: Error 171060 'Unsupported transport (PJSIP_EUNSUPTRANSPORT)' sending NOTIFY request to endpoint [extension number].
I searched on "PJSIP_EUNSUPTRANSPORT" and the only solution I found was to reboot the system, which does work but is rather inconvenient. Has anyone else run into this and if so, did you perchance find a better solution?
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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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[CallCentric] CallCentric discontinues iNum services/support on 11/30/19
Just received an e-mail from CallCentric annoucing the discontinuation of iNum services and support as of November 30, 2019. An FAQ webpage https://www.callcentric.com/faq/49#539 has more details.
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[Anveo] Cyber Monday: Sock Rocket (Starter) Package as low as $2.49
The emailed deal | Starter Package - http://anveo.com/consumer/service.asp
Get the new, limited time offer, subscription package before the 'Sock Rocket' reaches 'The Lagrange Point' on 12/02/2019 11:59 PM EST, after which the package will no longer be reachable!
The 'Sock Rocket' Subscription Package includes all of the features in the Starter Subscription Package, but for only $2.49/month if paying annually or $3.49/month if paying month to month.
Note: Only users with active subscription (at the moment of the email announcement) to our Free subscription package are eligible for the 'Sock Rocket' Subscription Package. To change subscription package please login into your Anveo account dashboard and click CHANGE plan link.
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[PBX] Hosted to self-hosted - what did you choose?
If you moved from a hosted PBX solution like 8x8, Jive, OnSIP, or something similar, to self-hosted (whether in the cloud or on-prem) - what did you choose and why? How many phones/users did you migrate?
(It's easy to find "success stories" / white papers about people who moved from PBX to a hosted solution, but not the opposite.)
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Old/Quiet VoIP companies we haven't heard much from lately.
Thinking of some companies that were well known but have been pretty quiet lately at least 'round here:
1) AxVoice, which really has spiffed up their website compared to a few years back. Not really sure who runs them these days, but interesting offerings.
2) ViaTalk, around a long time, based in upstate NY. The founders of Voipo had come from ViaTalk around 2006, but ViaTalk still operates.
3) 1-VoIP, solid folks based in Oklahoma.
4) VoicePulse, based in NJ, around a long time, has shifted away from residential to pretty much be a business and SIP trunking provider.
-----
I'm curious if folks have any recent experience/comments with these or other companies that don't make much splash here.
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[General] Small Business VOIP - Callcentric/voip.ms/anveo vs FreePBX/etc
I'm planning a switch for my small business from the current POTS lines to VOIP. No matter what we do it will save a lot of money as the POTS lines are expensive. We will have around 10 physical phones but only a couple of them are very active, so I don't really want to pay the $20+/mo/phone cost for a fully hosted VOIP service when some of the phones will hardly do any calling. About 2000 minutes/mo total in incoming 1-800 calls and just a little in outgoing.
We need some sort of PBX, but really not very many features. A basic IVR with a couple of queues, music on hold, extensions and voicemail.
I'm looking for opinions on going with a VOIP provider that has built-in IVR support to do what we need to do vs having PBX hardware on site.
The VOIP providers that seem to be popular, and have the PBX features we need, are Callcentric, voip.ms, and anveo. It looks like you can set up queuing and IVR and voicemail with them without needing any on-site hardware besides the phones.
Otherwise we could run FreePBX, FusionPBX, or 3CX on-site on a small computer and hook that into any VOIP provider.
I do tech stuff for a living, so I know I could set up FreePBX or whatever, but learning and keeping up-to-date on such software for a one company project isn't necessarily the best use of my time unless there is an advantage to doing so.
I'm not necessarily looking for the cheapest possible cost, as any of these solutions will be much cheaper than our current POTS lines. My top priority is reliability and getting something setup that "just works" without me having to always fiddle with it.
So my questions are:
- Is going with one of these providers with built-in IVR like Callcentric, voip.ms, or anveo, going to satisfy my desire that it just works and works reliably? If so, are there opinions on which provider is the most reliable? (or is there one I missed?)
- Is there some other reason to go with an in-house PBX like FreePBX beyond the additional features they offer? Would I find the performance or reliability better if hosted on-site?
Thanks for the help and opinions!
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[Anveo] Does Anveo Direct support tls?
I recently was able to get tls for Flowroute up and running on Asterisk and was wonderring if AD supports it. Couldn't find any reference on the web. I have funds with Anveo Direct I would like to use.
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[Other] Experience with Telnyx?
I've been looking at providers that support tls and SRTP. I found an old post that gives some info on Telnyz. They have a pretty good rate for outbound and are located in my state. They say they carry much of the traffic on their own fiber network. Wondering if anyone has experience with them. Good? Bad?
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Transfer Call Externally With Original Caller ID
How can a call be transferred to an external (PSTN) number, after you had already answered the call and spoken to the caller, so that the party you transfer the call to sees the original caller's number as the Caller ID?
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[Other] Ooma is rolling out a new wireless home phone
Ooma is rolling out a new wireless home phone service that states: "Perfect for homes without internet". It is called the Ooma Phone Genie. The unit looks like a telo but with a separate device looks like it might be an antenna. The description states: "Genie is a home phone and internet service that uses a wireless LTE network" It comes with voicemail, caller-ID, call-waiting, 911 calling but it doesn't say caller id name. all this for $11.99/mo plus taxes and fees .
I can't tell by the map what LTE network they are using. I think this is a smart move for Ooma this wireless home phone technology is probably more reliable for elderly people than VOIP is. I think this service is VOIP. Ooma is using VOLTE here is a link https://www.ooma.com/home-phone-service/phone-genie/?offer=TELO&purchase_code=SBRD-TELO&xutm_source=SEM&xutm_campaign=SEM-1047495514&xutm_medium=SEM-Google&xutm_term=Telo&_vsrefdom=Google-SEM&om_phone=866-575-5585&keyword=Phone-Genie&adid=345697436145
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Ooma Questions
We currently have our VoIP service from AT&T as part of a bundle with Internet 1000 and U-verse U200. I'd really like to ditch the phone and TV service (replacing it with YouTube TV). I remember hearing about Ooma way back in 2009 or so but never signed up. We still use the landline quite a bit even though both of my parents now have cell phones as well. How does the call quality compare to service with AT&T VoIP? Also, I read nightmares about cancelling the service. This especially concerns me seeing as my parents want to keep that number. Are there any thoughts on this?
Thank you!
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[General] Cable Company Phone service
Does anybody here have any experience with cable company phone service? I need to know how reliable it is. This service would be for my parents. My dad is 86 and my mom is 82.
I got them cellphones and added them on my family plan with cricket wireless they each have unlimited talk text and data but they just won’t use the cellphones at all. They are very stubborn and they use their landline all the time. It is like they are still living in the 1970s and I can not get them to change. They still have a wall phone in the kitchen. My dad retired in 1991 and is on a fixed income. The problem is Centurylink is charging them an obscene amount for their phone service. Their monthly phone bill is $107.00 a month that is a bundle price that includes phone + internet (crappy slow dsl) the phone service is:
Home Phone Unlimited $45.00
DSL Internet $29.99
Taxes $32.00
I contacted the local cable company (Wave Broadband) and they said they could give them 100mbs internet plus phone bundle for $49.99 + aprox $7.00 in taxes price for 1 year promo. The salesman said the phone would work in the event of a power outage if you you purchase a battery and install it on the provided embedded Multimedia Terminal Adapter (eMTA) I am trying to convince my dad but I am wondering how reliable cable voip is. The centurylink landline is very reliable in the 70s they buried all the cables underground so they are not effected by storms.
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[Anveo] Made a Mistake
Since my primary phone is an Obihai device, I made the mistake of porting my VOIPo number to Anveo using the special Ohihai pricing. The package is ideal and reasonable for this low use number, but it turns out that I cannot access the number with a SIP client at my office. Only one device is allowed per number on that plan. I didn't realize this ridiculous limitation when I set up the account, paid for it, and ported the number. Unfortunately I have to port back to VOIPo or find a new provider. I'm just offering this information for any other non-savvy SIP users like me who might get caught off guard.
The account would be great from anyone who uses a Obihai device and then connects the analog tip and ring to a single set or a key system. However, the limitation of one device per number somewhat defeats the purpose of SIP, in my view.
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[CallCentric] freepbx failing Forbidden
after a conversation yesterday i attempted a brand new freepbx install with a free call centric account.
it works fine with my obi, so there must some configuration issue.
any help or pointers would be great, im not strong on freepbx.
[2019-12-04 14:55:59] VERBOSE[26684][C-00000005] pbx.c: Executing [s@func-apply-sipheaders:13] Return("SIP/callcentric-00000004", "") in new stack
[2019-12-04 14:55:59] VERBOSE[26684][C-00000005] app_stack.c: Spawn extension (from-pstn-toheader, 917771234567, 1) exited non-zero on 'SIP/callcentric-00000004'
[2019-12-04 14:55:59] VERBOSE[26684][C-00000005] app_stack.c: SIP/callcentric-00000004 Internal Gosub(func-apply-sipheaders,s,1(1)) complete GOSUB_RETVAL=
[2019-12-04 14:55:59] VERBOSE[26684][C-00000005] app_dial.c: Called SIP/callcentric/17771234567
[2019-12-04 14:55:59] WARNING[2351][C-00000005] chan_sip.c: Received response: "Forbidden" from ';tag=as6ab3da9b'
[2019-12-04 14:55:59] VERBOSE[26684][C-00000005] app_dial.c: Everyone is busy/congested at this time (1:0/0/1)
[2019-12-04 14:55:59] VERBOSE[26684][C-00000005] pbx.c: Executing [s@macro-dialout-trunk:26] NoOp("PJSIP/1000-00000004", "Dial failed for some reason with DIALSTATUS = CHANUNAVAIL and HANGUPCAUSE = 21") in new stack
Current Asterisk Version: 16.4.1
Host dnsmgr Username Refresh State Reg.Time
callcentric.com:5060 Y 17778379998 49 Registered Wed, 04 Dec 2019 14:58:10
1 SIP registrations.
CHANSIP
Name/username Host Dyn Forcerport Comedia ACL Port Status Description
callcentric/17778379998 204.11.192.159 Yes Yes 5060 Unmonitored
callcentric1/17778379998 204.11.192.22 Yes Yes 5060 Unmonitored
callcentric10/17778379998 204.11.192.135 Yes Yes 5060 Unmonitored
callcentric11/17778379998 204.11.192.159 Yes Yes 5060 Unmonitored
callcentric12/17778379998 204.11.192.160 Yes Yes 5060 Unmonitored
callcentric13/17778379998 204.11.192.161 Yes Yes 5060 Unmonitored
callcentric14/17778379998 204.11.192.162 Yes Yes 5060 Unmonitored
callcentric15/17778379998 204.11.192.163 Yes Yes 5060 Unmonitored
callcentric16/17778379998 204.11.192.164 Yes Yes 5060 Unmonitored
callcentric17/17778379998 204.11.192.169 Yes Yes 5060 Unmonitored
callcentric18/17778379998 204.11.192.170 Yes Yes 5060 Unmonitored
callcentric19/17778379998 204.11.192.171 Yes Yes 5060 Unmonitored
call
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How should I perform the sale of a valuable phone number?
I have a phone number for sale in a major US city. It consists of 7
identical digits. The ultimate sale price might be roughly between
$100,000 and $250,000. The number transfer would likely occur
as a change of ownership of the telco account on which the
number is active.
If the ultimate sale price were likely to be $2000, then the buyer
and I could probably do the transaction over PayPal. (I've used
PayPal for transactions of that size for the sale of domain names.)
The risk for each party is small because the price is fairly low.
For a large transaction, even in person, there's a pitfall: one
party would end up performing their part of the transaction (either
payment or account transfer) before the other, and, in the absence
of a referee, the party that acts first would naturally feel anxious
about whether the other party will follow through.
In the sale of other types of assets, the solution is escrow, in
which a trusted third party holds the funds from the buyer,
oversees the transfer of the asset, then releases the funds to
the seller.
In real estate, for example, there's a clear and verifiable pathway
for the transfer documents, which consists of the recording of
deeds with the county assessor, among other acts. State laws
lay out the process, and the professionals involved (escrow
person, real estate agent) are state-licensed.
But for the sale of a phone number, there's only a thin legal
framework for ownership and transfer.
And technically, phone numbers are owned by telephone
companies, not subscribers. (However, no reputable phone
company would summarily yank a phone number from
a subscriber, as long as the account is in good standing.)
One possible solution would be to attach the phone number
to a business entity (such as a corporation or LLC or even
a sole proprietorship), then sell the business entity such
that the phone number is contractually stipulated as part
of the package. The business entity in question would be
essentially fictitious, with no significant assets apart from
the phone number. This method might allow a phone
number sale to be performed as the sale of a business.
When a business sells, the phone number is often
formally transferred with it, because the number is a
valuable part of what is known legally as goodwill.
This a type of transaction for which there is a standard
legal framework, and thus might allow for the use of an
escrow person.
I may ask a small-business lawyer about all of this. But
first, I'd like to get some feedback here. I don't know much
about law or commerce, and my assumptions above are
likely incomplete or wrong.
My questions:
- If you were selling a phone number for a large sum,
how would you do it?
- Is there a way to impose a solid legal framework
on the sale of a phone number?
↧
Telnyx "beyond free" porting Black Friday
From the promo:
"Paying to port your numbers? Get paid instead. Get $1 free credit for every number you port to Telnyx during Black Friday week only. Hurry, offer ends soon!"
https://telnyx.com/products/number-porting
US numbers are $1/mo. each. + $0.0075/min. geographic or $0.015/min. toll-free.
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