Hello:
Can someone explain the purpose of Anveo's "Call Flow for Outbound Calls" and how & why one might use it?
Thanks,
Rob.
Edit:
Sorry, I'm trying to understand it with respect to using 'platform minutes' as in:
"...Anveo Platform Minutes are calculated as the total number of inbound minutes for calls received on Anveo Value phone numbers, 'Inbound SIP Trunk for Call Flow' plus 'Call Flow for Outgoing Calls' feature. Incoming calls on Anveo phone numbers are NOT counted towards Anveo Platform Minutes..."
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[Anveo] Trying to understand Anveo's "Call Flow for Outbound Calls".
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SpeedTalk Mobile
I recently learned about this T-Mobile MVNO. They offer several plans that may be interesting to this group, though I've no personal experience with them. All plans include free SIM.
For light cell phone use: $99 per year, includes 500 voice minutes, unlimited SMS and 1 GB of data monthly.
As an intermediate for porting to e.g. Google Voice: $5 for 30 days, 100 min., 100 SMS, 100 MB. There is a 14-day money back guarantee, so it could even be free.
For alarm systems or backup remote access: $60 per year, 1500 units; each voice minute, SMS or MB of data counts as one unit. Unused units roll over to next year.
For moderate cell phone use (though there are many MVNOs in this range): $249 per year, 'unlimited' talk and text, 3 GB data monthly, overage throttled to 2G speeds.
They offer nothing for heavy users.
https://www.speedtalkmobile.com/plans
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[Equipment] is Polycom Soundpoint IP 335 any good?
I have a couple of Polycom Soundpoint IP 335 phones, without power adapter, for a while. Are the phones any good to buy the adapters from eBay and use them? Thanks.
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[Other] SignalWire, FusionPBX and mod_signalwire
Has anyone tried this?
I setup a FusionPBX instance to play with and connect to SignalWire using their module, but the module does not exist in the FusionPBX installation.
According to the freeSwitch docs, the module is supposed to be included by default from version 1.8.3 on and FusionPBX is currently on 1.8.5.
Any ideas?
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2/26/19 Thru ?? - Delayed SMS To GV # Sent From Vzw Network #
See https://www.reddit.com/r/Googlevoice/comments/av6jct/delayed_receipt_of_texts_sms/#bottom-comments for more.
Apparently, since yesterday, 2/26/19, if someone sends a SMS from a # on the Vzw network, it is currently very delayed being received on a GV #, by up to a number of hours!
This has nothing to do with what service the receiver uses to get their GV # messages; it only seems to affect messages sent FROM a Vzw # to a GV #, no matter what.
It may just be me, but it looks like Google "re-did" their support forums, and I can't even figure out how to report this, nor can I find any reports of it on the Google support pages!
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[Equipment] LG Rebel 2 or LG Rebel 3 Smartphones for $9.99 from Amazon
TracFone LG Rebel 2 Or Net10 LG Rebel 3 smartphones for $9.99 from Amazon after $30 price drop
They can be used as a VoIP phone without any cellular service
TracFone LG Rebel 2 4G LTE Prepaid Smartphone: $9.99 (This was selling for $38.88 yesterday):
https://www.amazon.com/TracFone-Rebel-LTE-Prepaid-Smartphone/dp/B0727V13FG?psc=1&SubscriptionId=AKIAJWYS5JI5YLLVARAQ&tag=cl03f-20&linkCode=xm2&camp=2025&creative=165953&creativeASIN=B0727V13FG&tag=cl03f-20&smid=ATVPDKIKX0DER
Net10 LG Rebel 3 4G LTE Prepaid Smartphone: $9.99 (This was selling for $40.11 yesterday):
https://www.amazon.com/dp/B07CQ9H5QH/?tag=cl03f-20&psc=1&smid=ATVPDKIKX0DER
5″ touchscreen; 1.1 GHz quad-core processor; Android 6.0 Marshmallow
4G LTE; Wi-Fi capable; Bluetooth 4.1 wireless technology; MP3 Player
5 MP Camera/5 MP front facing Camera; Internal memory 8 GB; supports Micro SD memory card up to 32 GB (not included)
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[Voip.ms] Please add to this VOIP.MS features wish list
Apparently, VOIP.MS is going through an innovation phase.
So I propose that we create a wish list of features or
enhancements that we'd like to see implemented on
their website or in their telephone functions. As this thread
progresses, I'll occasionally comb through the suggestions,
number them, and put them into an updated list.
My starter list is below. Please add suggestions of your own.
As always with brainstorming, 90 percent of what's
proposed will be undoable for some reason. But please
suggest with abandon, to give VOIP.MS a large list from
which to choose some ideas for consideration.
======================================
1. Create a second level of login (perhaps labelled a
"user" login) that can't make account changes but can
enter the fax portal to send faxes. (The original type
of login could be labelled "admin" and would retain
full control of the account.) Reason: sometimes a
small business assigns faxing to an employee who needs
only to fax and thus needn't be accorded full account
control. This is in keeping with fundamental account
security. (Related request: allow for multiple logins
-- perhaps 10 -- each of which could be designated
as either type "user" or type "admin".)
2. On the web interface, allow a user to make a one-
time choice about how many subaccounts or DIDs to
display at once. As it is, each time you go in to look at
a long list, the quantity defaults to the lowest quantity,
which is 10. So if I choose ALL, it would stay on ALL
for my login name.
3. Allow subaccount names to have more than 12
characters after the underscore, to allow for more
descriptive names. 25 would be nice.
4. Implement MMS (finally), with full image embedding.
This might need to be a fairly expensive feature, but
some of us would pay for it. I'd pay $15 per month per
DID for fully functional MMS. The bottleneck here may
be the limited SMS-only texting available from Vitelity,
VOIP.MS's main upstream provider.
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[Equipment] Grandstream: Some New DECT Cordless IP Phones
Full specs here with some suggested prices.
http://www.grandstream.com/company/news/grandstream-debuts-new-generation-dect-cordless-ip-phones
Prices look decent....
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[Equipment] Multi line phone model options
I have been looking around for a two line phone. I was hoping that there would be some typical DECT6 handset/base in the common form factor and in a mid kind of price range. But it seems that 2-line phones are more marketed at office use and people who prefer a giant form factor to cover a lot of desk space. Unfortunately I don't use a desk, so even if I did want a big phone, I would have to put it on some kind of arm or stand, and for a large piece of equipment, that's not really ideal. Basically, I've got no more than 4"x4" to dedicate to a base station.
So, are there really no 2 line DECT6 phones that meet my criterion?
I would consider an IP phone for this but, unless I've missed it, they don't appear to take a landline as one of the inputs. Anyway, this is not a huge priority, but I'd like comments and suggestions to consider.
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[General] FCC chairman warns of regulatory intervention on carriers
https://www.theverge.com/2019/2/13/18223396/fcc-spoofing-ajit-pai-robocalls-warning
https://www.fcc.gov/call-authentication
Let's hope we see changes by years end.
Have managed to get home phone under control with GV and CC but cell phones are out of control.
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[General] Anybody Tried Signalwire.com?
These are the FreeSwitch guys with a cloud offering.
Very similar to Twillio, but much lower cost. Lots of programability features.
US DIDs are $0.08 per month. Free in-porting.
US Termination is $0.00255
US Origination is $0.009
No minimums.
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How to Install naf Asterisk on Ubuntu for Obi100 and Google Voice
Here is a 'easy' install of naf Asterisk.
There is no GUI, I prefer it this way.
I have pre-configured it for up to 10 GV accounts (except for personal info). You can easily add more.
You will need oAuth2 credentials for each GV account.
For hardware I used:
An old Core 2 Duo (2008) with 2GB ram and a 20GB hard drive. It uses 50 Watts continuous power.
If you have a newer computer it will probably use less power. A newer Celeron would be great (under 15 Watts).
Connect the ethernet port of the soon-to-be Ubuntu/Asterisk server to your router.
Connect the Obi100 to the same router.
I used a DVD with Ubuntu Server 17.1 to do a fresh install and then:
sudo do-release-upgrade (installs latest Ubuntu version 18.04.1)
sudo apt install gcc (installs C compiler)
On your regular computer:
Download the ZIP file I posted and extract the four configuration files.
Load pjsip.conf into Notepad (or similar).
Choose which SIP account (1001, 1002... through 1010) you'll be using. You'll only be editing in that account.
Where it shows 'agoodpassword' change this to a made-up password you'll use later in the Obi.
Enter your Google Voice 10 digit phone number in place of 1112223333 in TWO places.
Add your oAuth credentials. Use copy and paste to avoid typing errors. Don't leave any spaces before or after.
In place of 444555666 enter your Obi number (written on the bottom of the Obi). You can make up a number if you wish of around 9 digits.
Save the file (without any formatting).
Copy the newly edited file and the other three .conf files to a USB stick.
Then back to the Ubuntu computer:
cd /usr/srcsudo git clone https://github.com/asterisk/asterisk.gitcd asterisksudo contrib/scripts/install_prereq install (runs the installation script) (It will pop up with "ITU-T Telephone Code", enter 1 for United States.)sudo ./configure --with-pjproject-bundledsudo make menuselect.makeoptssudo menuselect/menuselect --enable chan_mobile --enable cdr_mysql --enable app_macro menuselect.makeoptssudo make (compiles C into objects)sudo make install (links objects, downloads stuff)sudo make config (Configure as service at bootup)sudo cp configs/samples/*.* /etc/asteriskcd /etc/asterisksudo find . -name "*.sample" -exec sh -c 'mv "$1" "${1%.sample}"' _ {} \; ;Insert your USB stick with the four .conf files.lsblk (this will list drives. The USB will probably be sdb1)sudo mkdir /usb (make a USB directory (on HDD) to mount the USB drive)sudo mount /dev/sdb1 /usb (USB contents are now available in /usb)sudo cp /usb/*.conf /etc/asterisk (copy our .conf files)sudo umount /dev/sdb1 (log out of USB stick) ;remove your USB stick reboot
At this point Asterisk will be running in the background and your Obi will connect if you have already configured it.
Tips on using a running Asterisk:
sudo asterisk -r (continues running but also gives you the Asterisk CLI (command line).
core stop now (stops Asterisks if you need to make configuration changes.)
sudo nano /etc/asterisk/pjsip.conf (edit the pjsip.conf file)
sudo Asterisk -cvvvv (re-starts Asterisk with verbose level 4)
Configuring an Obi100 (other ATA's, IP phones, and softphones may be similar):
Open a browser on a computer on the same LAN and log into your router to find the IP address of the
Ubuntu/Asterisk server and the Obi100.
Log into the Obi and go to Service Providers, ITSP Profile A, SIP
Uncheck ProxyServer and type in the address of the Asterisk server. Something like 192.168.1.5
Uncheck ProxyServerPort and type in 5085
Scroll down to the bottom and click on Submit (and OK).
Go to RTP (on the menu).
Enable RTCP and X_RTCPMux
Click on Submit and OK.
Go to Voice Services, SP1 Service, SIP Credentials, AuthUserName, 1001 or whatever account # you configured earlier.
AuthPassword, Enter what you used in place of 'agoodpassword'.
Scroll down to bottom and click on Submit and OK.
Click on Reboot in the upper right.
It should register with the Asterisk server and the phone LED (on the Obi) will light up.
Edit1: Github file location for Asterisk
New configs.zip file. Has new versions of pjsip.conf and extensions.conf.
Edit2: New configs.zip file. Has new versions of pjsip.conf and extensions.conf.
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So a question for some of the more SIP savvy folks...
If you were a SIP provider and I had an issue with making out going calls even though my PBX was showing as registered and I provided you a tcpdump showing the full SIP conversation, would you not be able to look at the invite and response to determine if there was a value being sent not compatible with your system?
The issue I am seeing is that my system registers with the provider and a tcpdump showing the SIP conversation shows that the register comes back with a 200 OK and the PBX shows it is registered. Place an inbound call to the associated DiD number works perfect and the extension tied to the DiD rings and one can have a normal conversation.
Where I am having the issue is with outbound calling. Trying to make the call, the invite is sent then the provider returns a "Proxy Authorization Required" message then a "Backend Unavailable" message.
I ask this because I put in a ticket and included a dump of the full register and of a failing outbound call and they came back asking for server logs. Not sure how anything in my server log will give them any more info than the SIP dump already does.
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[General] Looking for the cheapest World Wide DID's
I’m doing affiliate marketing to companies by providing them leads.
They are calling those leads and try to turn them into customers.
For every customer I bring them, I am getting paid.
Usually it takes anywhere between 1 hour to 3-day Max for the leads to become a customer.
I have a strong feeling those companies I provide leads to, are shaving me.
Since I have the phone numbers of the leads, I was thinking about setting up a forwarding number for each lead and then have a recording for each call made by the companies to those leads.
The problem is I’m working on many different countries (Non USA) and on big volumes it could add up to a lot of money spent on buying daily DID’s.
Do you think there’s a company out there that would “rent” a DID for cheaper based on Days/Hours used?
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[General] Sponsoring the first 100 registrations in ENUMER in March, 2019
ENUMER has been announced here ~1 year ago ( http://www.dslreports.com/forum/r31739482-General-ENUM-is-live-again-with-ENUMER ).
During this year, there are registered 76 individual numbers, established gateway to US toll-free numbers, and our public ENUM-resolver receives ~2000 ENUM-requests per day. I have no idea, how many requests are handled within local resolvers. Established partnerships and cooperation with companies and groups, develops of SIP switches: sipXcom, PortaOne, OpenSIPS.
Softphone "Voip by Antisip" also added the option "search in the ENUMER".
The independent company Postmet created and runs automatic phone validator ( https://enumer.bitname.ru/index.php?lang=en ) , which generates correct ENUM-records and sends them to customer's wallets. This is a convinience fee 1EMC (~$0.28), just to cover expenses for VPS and verification phone call.
As a 1 year milestone, we are making a promotion: we are sponsoring 100 registrations to forum members! Promo ends Apr, 1, 2019, or after 100 requests, whatever is first.
Just send an e-mail request to enumer[X]emercoin.com (substitute [X] to @), contains your EMC address from your Emercoin wallet. The 1st 100 addesses will receive 1.05EMCs, enough for cover your phone validation expenses and ~100 record updates thereafter. ENUMER validation site: ( https://enumer.bitname.ru/index.php?lang=en ).
More details and useful materials about ENUMER see on the web-site: http://enumer.org/
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VOIP fun tonight
Some might be out a' bar crawling...
Some might be out at the casino (like I was last night :D)...
Some might be snugly in bed to get up early tomorrow for work...
And some might be... dealing with a failing(?) Obi110 that works OK... until it's gotta be power cycled, then it shows an alternating red/green power light :(. This has happened to my Obi110 the last few times I've had to reset it. Before tonight, it powered up OK if I left it unplugged long enough (a few minutes), or tried another 12V power brick (like I did the last time), but not tonight. I even tried a 12V 1A power brick... no dice. I'm gonna try a factory reset and see if that gets it going again.
Until then... I swapped it out for my hacked HT701 and am gonna run with that for my Callcentric service. The HT701 rings all four of the phones in my house OK- an old (2006) Panasonic cordless in my living room, an old wall-mount AT&T trimline in my kitchen (buttons backlit with line power, electronic ringer, I'm guessing early to mid 90's era), a basic corded line-powered phone at the bottom of my basement stairs (electronic ringer), and an AT&T 1855 desk phone with speakerphone (AC power for the speakerphone and built-in caller ID display and answering machine, but can make and receive calls without AC power).
My mom's Philips Lifeline medical alert device works OK with the HT701 and Callcentric (I tested it).
I might just stop using voip.ms for my outgoing calls and use Callcentric for outgoing also, even if it costs me another $3-4 a month. I'm currently using the Personal Unlimited for incoming + North America 120 for outgoing + voip.ms pay-by-the-minute plan for outgoing (wasn't using the Callcentric minutes, but adding them was only $0.45 more). It was a pain to program my Obi110 to use Callcentric for 411, 911, my mom's medical alert device (because it didn't work reliably with voip.ms), voicemail, and voip.ms for all other calls. I'm a programmer, but setting up dialplans are kind of a pain (I'm not that good with regular expressions). And I'm going to lose that programming when I factory reset the Obi110 (I backed up the config, but not recently, and really don't feel like reprogramming it again).
So that was my fun for tonight.
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[Equipment] Configuration Utility for OBi100/110/200/202/212/300/302
OBiCfg.exe is a Windows program intended to provide configuration assistance for OBi100, OBi110, OBi200, OBi202, OBi212, OBi300, and OBi302 devices. Dial plans are generated using North American Number Plan (NANP) rules.
The main window consists of a Phone panel (upper) and a Service Provider panel (lower).
Phone panel options include:
1. Available trunk(s)
2. Primary line trunk
3. Second dial tone
4. 911 trunk and optional redirect number
5. 411 trunk and optional redirect number
6. Auto Attendant (**0)
7. Configuration (***)
8. Ad Hoc Gateway (n*ddddddddddd)
9. Speed Dial
10. Redirection of international calls
11. Redirection of 11-digit calls
12. Redirection of 10-digit calls
13. Redirection of 7-digit calls
14. Redirection of toll free calls
15. Redirection of iNum calls
16. Redirection of SIP URI calls
Service Provider panel options include:
1. Service codes (x11)
2. User dialing rules
3. 11-digit dialing
4. 10-digit dialing (with option to add '1')
5. 7-digit dialing (with option to add area code and '1')
6. International (00/011) dialing
7. iNum dialing (8835100xxxxxxxx)
8. SIP URI dialing (with option for IP dialing)
9. Direct access to the LINE port (#)
10. Trusted callers
11. Incoming call routing and forking
Notes:
Changing the 'Device' setting resets all options to default values. Default values for the currently selected device may be reset by selecting Edit -> Defaults.
The Service Provider selected as the 'Primary Line' may not be disabled.
The 911/411 'Redirect To:' number must be a valid phone number.
The seven 'Redirect ... Calls To:' options pertain to numbers dialed without a service route access code (**n). The target trunk and Primary Line trunk must both support the applicable dialing format. Any transformations (7-Digit Dialing Add Area Code, 7-Digit Dialing Add '1', and Ten Digit Dialing Add '1') specified by the target trunk will be performed.
User rules must be separated by a '|'. No validation is performed.
The 7-digit dialing 'Add Area Code' number must be a valid area code.
Trusted callers must be separated by a '|'. Numbers must be entered as they appear in the OBi Call History. Digit Map notation may also be used. No validation is performed. Trusted callers use the Auto Attendant.
Forked call destinations must be entered in TK format and separated by a "|". For example:
sp1(18005551212)|sp2(userid@192.168.1.175:5064)
A maximum of four destinations may be specified, including any selection of AA, PH1, and PH2.
Single Stage Dialing may use either the OBiTALK network or direct SIP-to-SIP communications. If a trunk other than OBiTALK is selected, that trunk must be configured for SIP. An OBi can be a client, a gateway, or both. An OBiON app can only be a client. On an OBi202 or OBi302, the gateway Primary Line is that of Phone Port 1.
An OBi serving as a gateway must have one or more clients listed in the 'From:' field, separated by a '|'. If the OBiTALK network is being used, clients are identified by their OBiTALK number. If SIP is being used, clients are identified by their SIP userid. For example:
OBiTALK:
From: 200173590|200486255|290842739
SIP:
From: 17771234567|'x'lite
M, m, S, s, X, and x are reserved Digit Map characters and must be enclosed in single-quotes if they are part of a SIP userid.
An OBi participating as a client must have one or more gateway entries listed in the 'To:' field, separated by a '|'. Gateway entries consist of a gateway number (2-99), a '=', and a gateway. If the OBiTALK network is being used, gateways are identified by their OBiTALK number. A Speed Dial will be created for each OBiTALK gateway entry. If SIP is being used, gateways are identified by their hostname or IP address. Speed Dials are not created for SIP gateway entries. For example:
OBiTALK:
To: 2=200173590|3=200486255
SIP:
To: 2=mydomain.dyndns.org|3=192.168.1.175:5064
When dialing a number from a client, the call will be placed via the local OBi if dialed in the normal manner. For example:
18005551212 -> Local PrimaryLine
**1 18005551212 -> Local SP1 Service
**2 18005551212 -> Local SP2 Service
**8 18005551212 -> Local LINE Port
**9 200123456 -> Local OBiTALK Service
If a dialed number is prefixed with a gateway number followed by a '*', the call will be placed via the specified gateway. For example:
2*18005551212 -> OBi #2 PrimaryLine
2**1 18005551212 -> OBi #2 SP1 Service
2**2 18005551212 -> OBi #2 SP2 Service
2**8 18005551212 -> OBi #2 LINE Port
2**9 200123456 -> OBi #2 OBiTALK Service
3*18005551212 -> OBi #3 PrimaryLine
3**1 18005551212 -> OBi #3 SP1 Service
3**2 18005551212 -> OBi #3 SP2 Service
3**8 18005551212 -> OBi #3 LINE Port
3**9 200123456 -> OBi #3 OBiTALK Service
File -> Output XML File creates a file suitable for loading directly into an OBi using System Management -> Device Update -> Restore Configuration. This file can also be imported into the OBiTALK Web Portal. Only parameters generated by OBiCfg are altered.
File -> Output Text File creates a file that lists all parameters generated by OBiCfg. A 'Warnings' list is output at that bottom of the OBiCfg.txt file for any option requests that could not be performed.
OBiCfg configurations may be saved with File -> Export Configuration and restored with File -> Import Configuration.
If OBiCfg is executed with a command line option of a configuration filename (for example: OBiCfg.exe OBiCfg.cfg), the specified configuration file will be loaded at startup.
Nothing is installed and no modifications are made to Windows by running OBiCfg.exe.
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[Anveo] How to disable Anveo's Outgoing message?
I did set up my outgoing message, "Hi, I cannot answer, please leave a message".
When I call my number, I hear that, AND then I hear Anveo's message:
"The called party is not available. Please leave a message after the beep...".
How can I stop Anveo's message?
Unfortunately, Arne Bolen hasn't had time for his Anveo Knowledge Base. :(
Mark484
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[Voip.ms] Voip.ms API Updates
Just received the below email.
All the team at VoIP.ms is excited to announce that we just released a set of new methods that can be used through our API: Local Number Portability!
These methods allow you to start a portability process, add an invoice, retrieve the current status and much more.
You can now integrate these functionalities on your website, intranet, extranet or customer portal… whatever fits your needs!
For more information about our API, you may retrieve all the related documentation by clicking here.
Thanks for being a VoIP.ms customer and we hope you will enjoy these new methods.
Best regards,
VoIP.ms Team
https://www.voip.ms
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[Equipment] is Polycom Soundpoint IP 335 any good?
I have a couple of Polycom Soundpoint IP 335 phones, without power adapter, for a while. Are the phones any good to buy the adapters from eBay and use them? Thanks.
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