Does anyone still have an unlocked HT701 in use (either originally from BasicTalk or factory unlocked).
Could you please check the Status page and see if it lists a firmware version under CPE ? Also, does the power LED stay on after boot up is completed ?
I have a bunch of HT701s from ACN and I noticed the power LED comes on during boot up then it goes off. Seems to be regardless of the firmware installed. They appear to work fine otherwise, although they don't display a firmware version under CPE.
I don't remember seeing something like this on the BasicTalk HT701.
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Anyone has a HT701 in use ?
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[VOIPo.COM] Have VOIPo's client emails been hacked
Earlier this evening I received the email (blatant spam) shown above. The entire email content is shown.
It was obviously hacked from my VOIPo account listing.
I always use the names of the entities with whom I register as the alias attached to my domain name ... so that I know where emails come from. In this case, my email address registered with VOIPo is voipo @ mydomain.com.
Fortunately, I always check the provenance of suspicious incoming emails.
Anyone else receive a similar email? :(
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[VOIPo.COM] No incoming or outgoing service on VOIPo
I have no incoming or outgoing service. I can't connect to Voipo using VPanel. Help!!!
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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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Ooma xmit audio volume issues
It seems my not so tech savvy brother helped my mother to switch to Ooma.They live far away from me. She seems happy with it however when she calls me or I call her, her xmit volume is excessive. I have to turn my Cisco phone down 2 clicks on the volume control before she stops blowing my ear out. She is the loudest caller I get. Then there is still the distortion this excessive volume causes. I was also wondering if this may be caused by CODEC or CODEC translation somewhere along the line,.
As I am not the subscriber so I can not post in ooma forums.
I was wondering if anyone is aware of this issue with Ooma. I did find similar complaints on Ooma forums but did not find a fix posted. It would seem logical that there is a transmit volume adjustment, but just do not know if it is user accessible, nor where to tell someone to look for it.
Thanks,
Mark
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[Equipment] Problems Modifying GrandStream Dial Plan
Hello.
Last year, I purchased a GrandStream IP Dect phone combination consisting of a DP750 base and a DP720 hand set. I'm very happy with them and believe that I got more than my money's worth (paid around $70.00 for the combo on Amazon).
As I became more familiar with them, I modified the Dial Plan for one of the Profiles by adding a rule to allow placing toll-free call from CC via SIP Broker. That change worked fine.
Last week, in order to dial CC and VoIP.ms Speed Dial codes with only two digits, I added a new rule at the beginning of the DP, as follows:
{<=*75>[0-9][0-9]|x+|*x+|*xx*x+|**275*xxxxxxxxxxx}It prepended "*75" as intended and SD calls connected. However, I later found out that the new rule would prepend "*75" to the first two digits of any dialed numbers greater than one digit, and call the corresponding SD code.
I then moved the new rule from the beginning to the end of the DP, as follows:
{x+|*x+|*xx*x+|**275*xxxxxxxxxxx|<=*75>[0-9][0-9]}After this change, presumably because the "x+" rule at the beginning of the DP "captured" the two digits dialed, "*75" would not be prepended and calls would hang.
For the next iteration of the DP, I deleted the "x+" rule from the beginning of the DP, as follows:
{*x+|*xx*x+|**275*xxxxxxxxxxx|<=*75>[0-9][0-9]}The result was the same as with the first DP shown above.
Finally, I added a new rule at the beginning of the DP to allow for calls to eleven digits North American numbers, as follows:
{1[2-9]xx[2-9]xxxxxx|*x+|*xx*x+|**275*xxxxxxxxxxx|<=*75>[0-9][0-9]}The result was the same as with the first and last DPs shown above.
Not sure why the last DP did not work as intended.
By the way, I could find no GS "timer" equivalent to OBi's "Sx" (did I miss it?).
Any ideas?
Thank you.
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Rob Thomas may release open source fork of FreePBX
I have never cared much for Sangoma as a company but now it looks like they have gone and pissed off former FreePBX developer Rob Thomas, who unleashed this Twitter stream:
https://twitter.com/xrobau/status/1266204816134422529
The short version is that Sangoma went and removed a bunch of old blog posts from Rob and other developers from the FreePBX blog, and he notes that "The only common factor is that the blog posts that were erased were all by people (such as myself) that are very pro Open-Source software." So if you read to the end it looks like he intends to release his own open source fork, which I for one would welcome. (Edited to clarify they were blog posts, not forum posts.)
Now before a certain individual pops in and tries to say there is already such a thing, I will just point out that in my opinion he could have owned this space if he wasn't so in love with all his old crap addons that were never that useful back in the day but that now only make his fork bloated with stuff that hardly anyone wants or uses. For a while he offered a "lean and mean" version but for whatever reason that fell by the wayside. If he would just offer a version sans all his not-so-wonderful addons, in other words JUST his fork of FreePBX with Asterisk and NOTHING ELSE, he'd probably get more users, or maybe might have if we weren't afraid he's soon drop it and start trying to force all his added crap on people again. Rob has never played those kind of games with people, so if he really goes through with it I'd love to see what he's offering. Besides that, Rob was something of a rarity among developers in that (at least in the early non-corporate days of FreePBX) he'd actually listen to users and even on occasion come up with bug fixes in nearly real time. So this whole paragraph is just my opinion (and there's one guy I block that I'm sure will feel the need to try to troll me on this, so if everyone will will kindly refrain from quoting him I'll never see his posts and life will happily go on) but honestly I do not think FreePBX is going to get any better now that all their good developers are no longer with the company (and I wonder if Asterisk will suffer a similar fate).
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[General] VoIP providers not blocked for "account validation"?
So far I'm finding that even with SMS enabled, all of my voip.ms and callcentric DIDs fail for online account verification for google services, twitter, facebook, etc. Are there any providers known to work for this or have the tech bros decided that being in possession of a physical cell phone is a requirement (for marketing/tracking purposes, I'd assume)?
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[General] Problem with Google Voice Sign Up
Yesterday, a friend asked if it is still possible to sign up for a free and new Google Voice number and my answer is yes. During the sign up process, she had to provide her mobile number to receive SMS # for verification. The process went smooth up to the end when clicked the FINISHED button where she was brought back to the sign up process for a new Google Voice number. Before she clicked the FINISHED button, it clearly showed the following message indicating her mobile number has been successfully linked her Google Voice number. Has anyone encountered such a problem?
Phone number added
(xxx) xxx-xxx has been successfully linked to your account and will ring when someone calls your Google Voice number.
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[Voip.ms] Access accounts from PSTN (without DID)
Recently Voip.ms announced that iNum support would be discontinued by the upstream provider. I have an iNum that depending on who wants to call me (country/area) I will provide them with an local (for them) iNum PSTN access number, then they can dial my iNum which links to my softphone via my Voip.ms sub-account.
With the termination of the service I don't have a similar solution anymore, and I certinally don't want to buy multiple DID's so my family can phone me once in a blue moon. Just FYI before iNum I used SIP Broker, but that service seems to be rather broken gone, and Voip.ms doesn't use sip.voip.ms anymore, you should point to the server where the sub-account is registered, and SIP Broker doesn't do that. So, no luck with moving back to that solution.
I did get a confirmation from Voip.ms that their wiki is up-to-date in the instructions what tell you by assigning a internal extension to a sub-account you can receive calls via SIP URI. However, unless I am missing a very basic piece of the puzzle that is of no help to me, as the URI requires x@y:port addressing, and I don't see how someone dialing into a service such as the below iNum, SIP Broker, or any other providers PSTN access numbers could dial that. Voip.ms also offers a Virtual SIP number, but it also requires the x@y:port addressing, so I don't see a difference between it, and using a sub-account, which is free (Virtual SIP is not, but it is cheap).
I think I'm stuck!
Example iNum PSTN access number: 1-416-800-4303
Example SIP Broker PSTN access number: 1-613-686-1602
thanks for any ideas,
RDP
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[CallCentric] Porting from Callcentric to Spectrum Voice
Hi all... I'm about to port my home phone number from Callcentric to Spectrum Voice because my Obi110 is on its last leg (*) and I don't feel like coughing up another $50-$80 for another Obi or Ooma device. What account #, account PIN # and address would I give to Spectrum for the port? I need to keep a home phone line because my 81 year old mom who lives with me is homebound (never leaves the house except for doctor appointments) and uses a medical alert device.
Thanks for any help/advice you can provide!
* My Obi110 displays an alternating red/green power light when it's plugged back in after being unplugged. I can eventually getting going again by fooling around with plugging in and unplugging the ethernet cable and a couple of other 12V power bricks I have, but there's going to come a day when it's going to go totally belly up. I turned off automatic firmware updates and disconnected my Obi110 from the Obitalk panel years ago.
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[Equipment] X_InboundCallRoute Only Rings the First Rule
Hello.
I'm trying to configure the SP2 Service on my OBi1022 IP Phone to ring the IP Phone and a VoIP.ms SIP URI that the GS Wave softphone on my Android would register with when the GS Wave App is started.
I chose the GS App because it does not require Google Play and can be turned on and off at will. It would only be turned-on when I'm on the road and expecting a call on the number that rings on the SP2 Service. For example, when parked outside my barbershop waiting for a call to go-in for a haircut.
My OBi1022 IP Phone is Hardware Version 1.1 and runs on Firmware Version 5.1.11 (Build: 4858EX.1311-olisom5b).
If I set "ph1" as the first rule and the SIP URI as the second rule, the phone rings, but the SIP URI doesn't:
X_InboundCallRoute: {ph1},{SP2(XXXXXX5@newyork7.voip.ms;ui=$1)}According to what uid://1910647 said here:
http://www.obitalk.com/forum/index.php?topic=8857.0
"Processing move[s] left to right and stops when the call is routed."
On the other hand, if I set the SIP URI first and "ph1" second, the SIP URI rings as long as the GS Wave is registered to it. If the GS Wave isn't registered, the incoming call fails.
X_InboundCallRoute: {SP2(XXXXXX5@newyork7.voip.ms;ui=$1)},{ph1}The corresponding OBI1022 Call Log entries are as follows;
Call 6 06/22/2020 11:45:17 11:45:17 From '[REDACTED]' SP2(REDACTED]) To SP2(XXXXXX5@newyork7.voip.ms)11:45:17 Call Ended (503 Service Unavailable)Is a 503 SIP error code considered a "routing" of the call that prevents ringing the next rule?
Thank you.
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[CallCentric] Can't receive incoming calls on CallCentric
I have the DID Dollar Unlimited and NA Basic. All of a sudden , not receiving calls. It was working fine. I do have a GV number forwarded to it and set as my CID but even when I call the DID number directly I get silence. It does not connect? I am usinga Sipura 1001
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is there anyway to call between phone systems?
Is there anyway to dial extensions from panasonic tda 50 to freepbx system? I had somebody text me asking if they could do that and I said there is no way of doing that so is there anyway of doing this? Because I already had convert from customers isp equipment that haves fxo port to obi110 to make it sip connection.
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[Equipment] Poly 212
So I was taken back by the pricing from amazon.ca on the Obihai Technology OBI212 Universal Voice Adapter with FXS Phone and FXO Gateways Ports Support for Google Voice and SIP ... this is the model that replaces the Obi202
on Amazon CDN they want $235.28 includes FREE SHIPPING
The last time I purchased the Obi202 I pad $70 + taxes but did including shipping
I guess Poly is recovering their investment in Obihai :-)
I guess I will pay that price cause I do really like Obi Tech ... never had any problems with obi insofar as ATA's are concerned and I have a ton of those out in the field with clients.
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ObiHAI Obi100/Obi110 Firmware Mod Discussion
So many of us have the Obi1xx series devices that recently stopped connecting to google servers due to a certificate update. This thread is intended to discuss the possibility of modifying the firmware to update the certificate and let these devices work with Google Voice again.
These devices are based on a MIPS-X processor similar to the Sipura ATAs and there is not a lot of tools/docs out there about them except for a Yahoo Group mostly related to DVD player chipsets. The venerable DogFace05 who was an expert with these types of devices once posted that he was able to extract this firmware sucessfully. Not sure if he is still around. Anyone else familiar with this architecture?
It seems that the place to start looking is the end of the firmware update file which contains some kind of table. Then there seems to be a loader section which presumably decompresses one or more other sections and loads them to RAM before executing the firmware.
So the questions are:
Can we extract, modify, and repack the firmware and create proper checksums/signatures?
Where is the certificate stored and in what format?
Can we drop in a new certificate without messing up other things (e.g. if the length of the certificate has changed) or do we need to move the certificate and patch the code pointing to it?
Is updating the certificate enough or is the codebase missing support that is necessary (e.g. if key length has changed)?
Anyone who wants to participate please post your thoughts.
Thanks
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[Asterisk] New Cellphone Integration for Asterisk-Based PBXs
For those that support phone systems for medical practices, or service-oriented companies such as HVAC, home repair/construction, plumbing, real estate, or any other organization with a receptionist and a mobile workforce, our new cellular integration experiment for Asterisk-based servers may be of interest.
We're just beginning trials of a new cellular phone interface to Asterisk which will let you deploy cellphones in the U.S. with access to all four of the major cell networks and integrate those cellphones into your existing Asterisk infrastructure using PJsip extensions. In this way, incoming calls to every cellphone number would also ring on office SIP phones or perhaps at a receptionist desk. Calls answered on the cellphone or initiated from the cellphone could take advantage of the complete Asterisk feature set on your PBX including call transfers, conferencing, etc.
We expect pricing to be about $25/month for each cellphone with unlimited calling, messaging, and a 2GB data plan. If this would be of interest, please visit the VoIP-info Forum and participate in our new thread. SIM cards are compatible with all Android and iPhone devices and automatically select the nearest tower using AT&T, T-Mobile, Sprint, and Verizon networks.
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Faxing Experiences Wanted
Good afternoon,
I used the search function but I'm not getting any results past 2011 and I figure the situation has likely changed since then, so...
The business I work at requires the use of physical fax machines. We've moved as much as possible to scan and e-mail but there are still daily situations where we have to be able to send or receive an actual fax.
I am trying to come up with a solution to use IP based transport so that we can cancel the $60+ POTS lines we currently have. I've been testing T38fax with a Cisco SPA112 ATA. It works ok, but I get inexplicable failures from time to time that doesn't inspire a lot of confidence in me. I also hate that every T.38 offering I have found is still using version 0, so we're limited to 14.4Kbps transmission speeds.
I've seen people say to use an HTTPS based fax service, I guess the ATA uses HTTPS transport instead of T.38, but I have no idea if that works at v.34/33.6Kbps or if its any more or less reliable than T.38.
If anybody has had any real world experiences they could share, I'd appreciate it.
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[Asterisk] Preventing transcoding
I use Snom phones and no matter what Codec I choose, the phone sends audio in slin16 format to the asterisk which then has to transcode it to whatever codec I've chosen to use:
Name: SIP/209-00001951
Type: SIP
UniqueID: 1592899852.11183
LinkedID: 1592899852.11183
Caller ID: XXXX
Caller ID Name: XXXX
Connected Line ID: XXXX
Connected Line ID Name: CID:XXXX
Eff. Connected Line ID: XXXX
Eff. Connected Line ID Name: CID:XXXX
DNID Digits: XXXX
Language: en_GB
State: Up (6)
NativeFormats: (g722)
WriteFormat: slin16
ReadFormat: slin16
WriteTranscode: Yes (slin@16000)->(g722@16000)
ReadTranscode: Yes (g722@16000)->(slin@16000)
-- Streams --
Name: audio
Type: audio
State: sendrecv
Group: -1
Formats: (g722|alaw)
When calling internally I get g722 along the entire length of the call:
Name: SIP/209-00001955
Type: SIP
UniqueID: 1592900175.11188
LinkedID: 1592900175.11188
Caller ID: 209
Caller ID Name: Work
Connected Line ID: 206
Connected Line ID Name: Bedroom
Eff. Connected Line ID: 206
Eff. Connected Line ID Name: Bedroom
DNID Digits: 206
Language: en_GB
State: Up (6)
NativeFormats: (g722)
WriteFormat: g722
ReadFormat: g722
WriteTranscode: No
ReadTranscode: No
-- Streams --
Name: audio
Type: audio
State: sendrecv
Group: -1
Formats: (g722)
My phone is currently configured with the following codecs: g722,pcmu,pcma,gsm,g723,g726-32,aal2-g726-32,g729,telephone-event
Can anyone shed any light on the matter?
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[Voip.ms] VOIP.MS beta for MMS is now live
[This topic deserves its own thread.]
Thanks to VOIP.MS for opening a public beta for MMS today.
The MMS feature is a little funky, as you'd expect for a beta. Within a few weeks, they'll polish it.
To use it, you'll need Beta Access Program enabled in your account. To enable it, click the relevant link to the right of your name on your account's main page.
The MMS portal is under DID Numbers >> SMS/MMS Message Center
The wiki page at https://wiki.voip.ms doesn't yet contain this key info:
• Permitted attachment types (outbound): jpg jpeg gif png mp3 wav midi
• Maximum attachment size (outbound): 1300 KB
• Maximum characters (outbound): 2048
I haven't yet tested for any inbound restrictions.
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