https://www.sangoma.com/press-releases/sangoma-announces-strategic-acquisition-voip-innovations/
↧
VoIP innovations a Sangoma company
↧
[Equipment] Ringdown with an ATA
I did not see anything on here with the search, so here goes:
Does anybody know how to set up a ringdown line with an ATA? Ideally, just an HT-802 or equivalent 2 port FXS ATA stand-alone.
I have seen several folks just use the Viking boxes, but I was sitting here thinking you should be able to do it FXS port to FXS port on an ATA. But I did not see any that had done so.
Any suggestions? Hints?
Thanks.
KA0OUV [Tim]
↧
↧
[PBX] Replace Norstar systems with VOIP for under $10K
I'm soliciting suggestions for a slide out/slide in business grade voip system, to replace existing norstar 8/16 line phone systems.
I would like Avaya, but I'm only familiar with big iron.
Can anyone offer suggestions as to what they would use if you were in my situation? I have a number of clients that are running Norstar CICS/ Callpilot 100's and they are all getting upgraded next year to voip.
Fire away, and thanks for your recommendations!
↧
[Anveo] How can I receive SMS on Anveo Direct via HTTP request?
Between the asterisk rows below are Anveo Direct's instructions for receiving SMS
messages. (The regular Anveo service (not Direct) offers various SMS delivery options,
but Anveo Direct seems to offer only the API option.)
I'd like to enable this on my CPanel webhosting account, but I assume that some
backend preparation is necessary. Can any of you explain how the URL format
example in the instructions below could be made to work on one of my domains?
************************************************************************
************************************************************************
Anveo.com - SMS API to receive incoming SMS using HTTP
• Anveo can forward incoming SMS Text messages received on Anveo phone
number to another server using HTTP request.
• A number of special placeholders/variables can be used in URL which Anveo will
use to forward incoming SMS Text Messages. Placeholders will we replaced with the
actual values from incoming SMS message.
$[from]$ - contains the phone number from which SMS message was received.
$[to]$ - contains Anveo Phone Number which received SMS message.
$[message]$ - contains the actual text of the message.
Example [I severed the https from the start to make sure that the URL would
display in full on this forum, which typically truncates fully formed URLs]:
www.mydomain.com/smscallback?from=$[from]$&phonenumber=$[to]$&message=$[message]$
************************************************************************
************************************************************************
↧
[General] I may be interested in residential VOIP
Greetings,
I think I'm interested in VOIP for my residence and I'm not too knowledgeable about it other than it's a phone service operated via the Internet and it is potentially cheaper. I currently have cable Internet and phone service via Suddenlink. And the Caller ID service does not work for me and seems that SL can't fix it and I need for the Caller ID service to work for phone line monitoring for scam filtering.
I think I want to change so I can save money and have more calling features and for better scam filtering. In the meantime I will educate myself more about VOIP.
What are the better/best services available for residential use and what is involved in changing over to VOIP? What is the general amount of time needed to port a phone number? Do most voip services include e911 services? I'm interested in what you have tried and did it work best.
I'm looking for some general information that I can use that will help to guide me in the right choice to make. Any good websites for a newbie to use such as myself?
Thanks for your time and suggestions.
↧
↧
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.
↧
[Voip.ms] Caller ID filtering for invalid / malformed
Hello,
I don't get any international calls on my voip.ms DIDs so I've enabled caller ID filtering based on invalid / malformed numbers that do not respect the North American dialing plan, in order to stop phone scammers that often have such numbers. My question is, does this filtering rule also match blank / unknown / private CID? I would like to let those calls through - for example, medical calls often don't send CID for privacy reasons.
Thanks,
cinergi
↧
Callcentric will not register to my OBi2182
I have been having problems setting up Callcentric on SP2 of my OBi2182. Previously I had one of my GoogleVoice numbers on SP2, but I have been having no audio problems for a while. I have several extensions on my Callcentric account and want to route a Voxbeam DID to Callcentric via SIP URI. This works ok on my OBi200.
I have set-up my OBi2182 with the same settings, and Callcentric will not register. I am getting...
Connect Failed: 407 Proxy Authentication Required (server=204.11.192.160
In my credentials I have
Username: 1777*******102 that's Extension 102
Password: ***********
I manage all my configurations using the Local UI.
There must be some little thing that I am missing.
Any ideas?
↧
Channelized SIP trunks
Best per-channel rate you have seen or been quoted? Just curious. That pricing method isn’t usually discussed here.
↧
↧
[CircleNet] SIP server appears to be down
Good day.
For about an hour now, my OBi302 has been unable to register on sip.circlenet.biz.
The status web page shows the following (emphasis added):
SP1 Service Status
Parameter Name Value
Status Register Failed: No Response From Server
(server=23.25.121.90:5060; retry in 99s)
I was monkeying around with the OBi302, but didn't change any server settings for the CN SP.
↧
What are optimal voice settings for SPA-112?
I experience what I can only describe as garbled or broken audio sometimes on some calls (ie audio that I hear, while my voice as heard by the other party apparently sounds fine). I haven't ruled out house-wiring problems (from where the ATA is in the basement to where my phone is wired to up stairs). I have just updated the ATA to 1.4.1 SR4 and was poking around and wanted to try using G722 but it's not one of the codec options (kinda sucks). Anyways, looking for optimal voice settings for this thing, will primarily be talking to others that also have SPA112 if that means anything, all using voip.ms if that matters.
↧
[Equipment] Replace My ATA or Purchase Sip Phones
After a long and productive life my Obi 200 has died. I run a very simple 3 DID residential phone system wherein 3 GV numbers are forwarded to the DIDs at voip.ms. Nothing fancy. I have been looking at the Yealink 60 as a possible replacement and perhaps adding a Yealink 56 extension. OTOH I could purchase a new Obi or other ATA. Curious about anyone's experiences and or recommendations. Thanks
↧
[Voip.ms] troubleshooting phone/device to voip.ms sub-account connectivity
Perhaps I'm missing something already in the UI, but I don't think so.
Aside from getting someone on chat or opening a service ticket, is there anyway to "look at the log files" when I create a new sub-account and program a new phone and discover that it doesn't work?
It seems the only feedback I can get is "connected" or "not connected" and there is considerable (but understandable) lag with that.
Perhaps my best action is to dig out the hub and fire up wireshark to look at the traffic?
I was thinking of asking VoIP.ms for a sub-account connectivity "Wizard" You press the button on the website and it sends you an email after a bit (or does a redirect to a web page) The wizard would give such feedback as "wrong password", "wrong port" "working, but traffic not encrypted", "TCP, was expecting UDP", "unsupported codec" and of course if there was no attempt to use that sub-account it would tell you to check the phone settings of sub-accouint name, server, etc)
↧
↧
Alternatives to VOIP.ms with low monthly and low per-minute rates
Can anyone please suggest alternatives to VOIP.ms with low monthly rates (VOIP.ms is $0.85) and low per minute rates? The ability to choose your own number from a large pool of numbers is a major plus. Thanks.
↧
[PBX] FreePBX for the Raspberry Pi
https://www.dslreports.com/forum/r30661088-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 16-GVSIP on a Raspberry Pi.
iptables, dnsmasq, and exim4 are also installed.
Installation takes a 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, 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
gvsip-ver will display the currently installed version.
gvsip-upd may be run if updates to GVSIP become available (GVSIP will be installed if not already present).
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-boot
========
Set Boot PARTUUID (or /dev/mmcblk0)
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)
↧
Cost for incoming 1-800 calls
I'm looking at doing VOIP with incoming 1-800 calls (a few thousand minutes per month). The providers I've checked so far around around 1.9c/min. Am I missing someone (reliable) who is less? It seems like 1-800 calls cost a lot more than other incoming calls but perhaps that's just the way it is. Thanks.
↧
Broke FreePBX on my Pi
Tonight after doing Buster kernel and FreePBX updates, I'm getting this:
Error: Class 'Locale' not found in file /var/www/html/admin/libraries/BMO/View.class.php on line 1050Stack trace: 1. Error->() /var/www/html/admin/libraries/BMO/View.class.php:1050 2. FreePBX\View->getBaseLocale() /var/www/html/admin/libraries/BMO/View.class.php:459 3. FreePBX\View->setLanguage() /var/www/html/admin/bootstrap.php:223 4. require_once() /etc/freepbx.conf:9 5. include_once() /var/lib/asterisk/bin/fwconsole:12
Tried modifying freepbx.conf and fwconsole files to get past the "include_once" or "call_once" references, but it just breaks it further.
Anyone know what I did before I revert this to previous, working backup?
TIA
↧
↧
[Unlock] Unlocking the BasicTalk ATA
Important NOTE: a better unlocking method has been posted later in this thread. Please see this post for a permanent unlock developed by uid://1479488. My soft unlock may help in some cases if the ATA has "called home".
I have some good news for those of you looking for an inexpensive ATA.
I've just got my hands yesterday on a couple of BasicTalk ATAs (I've had my eyes on them for a few months but I live in Canada and don't go to US that often) and I put together a small tutorial for unlocking them.
The ATA is a Grandstream HT701 with a customized firmware.
I posted it on my website at http://voipfan.net/unlock/ht701bt.php
I will leave the access open to everyone for a couple months then make it available to registered users (like my other unlocking tutorials).
Enjoy and if you run into any trouble please post here.
--
Providers (through asterisk): voip.ms, freephoneline, smartcall.ro, ipcomms, callcentric. Hardware: Vonage VDV21, Moto VT2x42, Linksys SPA series, Grandstream HT series, Panasonic KX-TGP5x0
http://www.voipfan.net
↧
How to use IAD2431 as PRI-SIP device?
So many of you may or may not have known that I have a Nortel Meridian Option11c. It is a powerful machine and right now, it has two TMDI cards (basically, a card that allows the Meridian to accept ISDN PRI connections.
As of this moment, I have it connected to a Windstream service and I also have another connection to a FreePBX box that is basically a gateway to a vanilla Asterisk server using another internet connection.
In other words,
Meridian ---> FreePBX ---> vanilla Asterisk box (on another IP address and no, I don't have static ip with Comcast) and vice-versa.
I was planning on re-purposing the FreePBX box into something more practical and in hopes of making it work, I bought myself an IAD2431-1T1E1 running Version 12.4(15)T12 as of right now. I got the basic configuration to work and whenever I call up an outside line to the IAD, it gives me a dialtone (which I think is not supposed to happen as the call is supposed to be without dialtone and all. Also, I can't dial anything out obviously.
What I was thinking was,
Would a ITSP such as Voip.Ms work in this case?
How would I register a SIP proxy in this device?
I know I might ask too much but looking at Cisco's website, most of the links give me a 404 so that is not much help.
↧
[Voip.ms] Hangup vs Busy on Spam Calls
Looking through my CDR I just saw two batches of four calls within 2 seconds to the same DID from "2457883364". (245 is an unassigned area code)
Currently I use caller-id filtering to "hangup" calls from this area code.
Presumably the spammer's kit retries calls 3 times before giving up (nice to see such dedication to service :-) ), then tries the same number again about five minutes later.
Ideally I would like a call treatment that:
Is unlikely to cause the spammer to questions their caller id (so they don't change their behaviour),
Costs the spammer money and/or time,
Doesn't cost me anything.
Unfortunately the only call treatments available on voip.ms which don't cost me money are "Hang Up" and "Busy". I'm thinking of switching to "Busy" as this may make a spammer less likely to be realize that they are being blacklisted, and might stop the immediate retries as the spammer's equipment is more likely to understand a busy signal.
Has anybody noticed any downsides to doing this?
My guess is that spammers will retry more often at longer intervals (not so obvious on the CDR), but some hard data would be useful.
↧