Hello DSLreports folk,
I wanted to take a minute to give an update on CircleNet here first.
We have decided that we will not be taking on any new customers, we will continue to take payments and provide service to existing customers until 1/1/2020. After that date we will keep our servers up and allow customers to spend down their balances until 1/1/2021. That should give two years notice.
This isn't a decision we've come to lightly but at the moment we do not have the time to take care of ourselves and also to provide the level of service that I think our customers deserve.
A formal announcement will go out tomorrow to our customers and I will help ANY customer facilitate a move to any other provider whenever they're ready.
I doubt we'll provide voip services again in the future but I do hope to lurk around and see what becomes of this awesome forum.
Sam
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CircleNet announcement
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Which service to port/migrate off of Ooma?
We have had Ooma service now for 8+ years but as we use our home phone less and less are looking at migrating our home line to a new service.
I have been considering Google Voice but am confused about state of gateway support for Google and also don't want to spend a lot of time handling a service migration followed by lack of equipment support.
What currently does the community like for low cost residential support? In all cases will it be necessary to port out my home # to an intermediary like a pre-paid mobile SIM before migrating it to another service?
Currently with Ooma I am using the simulataneous ring feature to my mobile and the blacklist feature (which doesn't seem to do much). I also occasionally fax on Ooma.
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[CAN] CCTS success: Source Cable kept charging after PortOut
... because we still had their legacy VoIP hub/box.
Just sharing in case anyone is in the same situation and wants to copy-and-paste what I think would work.
We ported my parents' phone to voip.ms. So far, so good.
Source Cable (now a sub of Rogers) kept charging us for their digital phone service after the port. Port-outs are supposed to cancel the old service per CRTC.
Source said that "their way" is to cancel service once equipment is returned. Dunno what they're going to do with an 8 year old ARRIS that they don't even use for internet anymore.
I filed a complaint to CCTS on Saturday. CCTS emailed Source this morning. 2 hours later Source called and said they'd reverse any charges from date of port-out.
So if anyone wants to copy-and-paste a similar complaint to CCTS, here are some time-savers:
"We transferred our voice service from _________ to __________. The port completed on ____________.
When we received our _____________ bill, they continued charging us the full cost of service of $_______ + HST.
When I called ______ on _______, I stated that we thought this service would be cancelled as of port date.
Agent informed us that the charges would only be stopped upon return of the VOIP box to ________. There was no notice of
this in the bill, nor any request/process for return of equipment.
We believe that it is unfair for __________ to continue to charge full price for service that has already been cancelled. The
equipment is legacy and worth ________ (Ebay). Fair ongoing rental chg for equip would be $___/YEAR.
====What do you consider to be a reasonable resolution to your dispute?====
A full refund of all “Home Phone” service charges from the date of port completion on _______ ($___/month + HST)
If _______ believes our continued possession of this equipment must be recouped, ______ may be credited CAD$_____/month
for the box."
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How to Install naf Asterisk on Ubuntu for Obi100 and Google Voice
Here is an 'easy' install of naf Asterisk (aka GVsip).
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 may pop up with "ITU-T Telephone Code", enter 1 for United States.)sudo ./configure ;sudo make menuselect (optional)sudo 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 indicated by the phone LED (on the Obi) lighting up.
Quick Links:
Install in Mint-Cinnamon https://www.dslreports.com/forum/r32078999-How-to-Install-naf-Asterisk-on-Ubuntu-for-Obi100-and-Google-Voice~start=60
Install in old laptop and in a VM https://www.dslreports.com/forum/r32078999-How-to-Install-naf-Asterisk-on-Ubuntu-for-Obi100-and-Google-Voice~start=56
Install in Debian-Cinnamon https://www.dslreports.com/forum/r32078999-How-to-Install-naf-Asterisk-on-Ubuntu-for-Obi100-and-Google-Voice~start=127
Install in Windows 10 https://www.dslreports.com/forum/r32078999-How-to-Install-naf-Asterisk-on-Ubuntu-for-Obi100-and-Google-Voice~start=391
naf donated certificates https://www.dslreports.com/forum/r32016984-Asterisk-Google-Voice-SIP-testing-and-technical-discussion~start=1151
Certificate Extraction https://www.dslreports.com/forum/r31741105-ObiHAI-Obi100-Obi110-Firmware-Mod-Discussion~start=653
Obi200+ Now Required https://www.dslreports.com/forum/r32078999-How-to-Install-naf-Asterisk-on-Ubuntu-for-Obi100-and-Google-Voice~start=330
Adding Google Contacts https://www.dslreports.com/forum/r32078999-How-to-Install-naf-Asterisk-on-Ubuntu-for-Obi100-and-Google-Voice~start=38
Info on *67 (block your caller id) https://www.dslreports.com/forum/r32078999-How-to-Install-naf-Asterisk-on-Ubuntu-for-Obi100-and-Google-Voice~start=42
Spam blocker https://www.dslreports.com/forum/r32078999-How-to-Install-naf-Asterisk-on-Ubuntu-for-Obi100-and-Google-Voice~start=180
Asterisk/GVsip information https://github.com/naf419/asterisk/wiki
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.
Edit3: Added 'Quick Links' above to jump to pertinent info.
New configs.zip file. Has new versions of pjsip.conf, extensions.conf, and modules.conf.
Edit4: New "method" param added to pjsip.conf under transport_tls to match pjproject change made in Asterisk past 6-27-2019.
Updated extensions.conf file.
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Google voice / voip box in 2019
Need to connect my Google Voice number to a good old-fashioned handset. Wondering what the state affairs is in December 2019.
Just ordered one of these “ OBi202 2-Port VoIP Phone Adapter with Google Voice “ on Amazon for 70 bucks. Just need a handset for outgoing calls for emergency use only / maybe use once a month. Not looking for anything but local calls, not willing to pay a monthly fee. Google voice number doesn’t get a lot of spam or so it seems. Had it for past 10+ years.
Am I on the right track here? Are their other better solutions, and what kind of headset / cheap I can wall mount with work with this little white box. Of course it has to be somewhat reliable :)
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Getting a DID (phone #) from Localphone might be slow
I've obtained phone numbers (DID) from various providers and the process is generally pretty quick.
Usually it's right away, or if it's going to take a day or two days for a particular DID, the provider will let you know up front.
Trying to get a DID from Localphone, it said numbers were available for the US city I wanted, so I added funds to my balance and then I clicked to pay for the number.
AFTER clicking, a notice came up which said something like "It may take up to 14 days to get your number. It will probably be a lot less but be advised...."
This was not a special number or anything like that, and as said, it was in a city they said was available.
So my gripes with Localphone:
1) How on earth can it take 14 days (even "up to" 14 days) in an American city they said was available? How exactly are they obtaining these numbers if that be the case? Does it involve waiting for someone to pass away?
2) The "up to 14 day warning" did not appear until AFTER clicking to purchase the number.
Now, they DO allow you to "cancel" afterwards which I did after an hour or so.
Yes, I admit I am an impatient person, but I didn't want to have to wait "X" number of days which could have been 2 or 6 or 14.
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Localphone has good service and good prices.
It's the lack of upfront notice that I object to.
Again, this was for a new number, NOT a porting.
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[Voip.ms] VoIP.ms e911 address validation...
FYI...
I revisited e911 address validation on VoIP.ms to confirm/tweak some details. A previously validated (2015) street name and city in mixed case would no longer validated in mixed case. The suggested changes specified all caps. So, I changed full name, street name and city to all caps (manually... the green checkmarks to auto-update the fields did not work as advertised) and proceeded through validation.
I suppose if validation wants it that way, it should be that way... all caps... probably for readability on various downstream devices/displays.
OE
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VOIPo Server Location
When I do a trace route on sip.voipwelcome.com it goes 209.212.145.160 which a couple of internet sites indicate is in Arlington Heights near Chicago. Previously tracert took a circuitous scenic tour to Dallas via Chicago. Have server locations been changed or am I interpreting the data incorrectly?
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Polycom speakerphone quality
A friend needs to replace his desk phone. He uses speakerphone almost exclusively. Doubletalk performance is important; he often has trouble interrupting the other party, being soft spoken and sitting about two feet from the mic.
I've been looking at the Polycom VVX series. The simplest/cheapest model (VVX 300) has all the functionality he needs, but he would consider a more expensive one if it sounded clearer to him and/or the remote party.
Can someone with experience with this series please comment on whether the higher end phones sound any better? Also, if you have both Polycom and another brand that sounds better, please post details.
Thanks.
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[Asterisk] PJSIP in Asterisk is hot stuff if you configure it right
I have a very small "family and friends" system running Asterisk and FreePBX that I've been using as a gateway to Google Voice, among other things. With the ongoing demise of XMPP connectivity and a couple of other fortuitous events, I decided it was time to build a new system and of course I wanted to include naf's GVSIP packages, although that turned out to be a short-lived experiment. But I figured that if I had to use PJSIP anyway, I might as well see if I could get it to work for my extensions and trunks. I had a very low opinion of PJSIP going in and my initial tests were not encouraging; I had lots of issues. But to make a long story short, the thing that helped the most was adding this line as the very first line in /etc/asterisk/pjsip_custom_post.conf:
endpoint_identifier_order=auth_username,username,ip
BUT you do not need to add that line manually any more, because now you can configure the Endpoint Identifier Order in the Chan PJSIP Settings in FreePBX as shown in the second screenshot above. Also in that screenshot, note that enabling TLS is optional and if you don't use it you can turn it off; it was only originally enabled for GVSIP.
But here is the thing that impressed me. I set all the extensions to use PJSIP, but I figured I was going to have a world of problems because many of them were connecting on ports other than 5060. In some cases it was a two line device connecting on ports 5060 and 5061; in other cases it was to avoid problems with routers or ISP's that seemed to think that traffic on port 5060 should be messed with. So my approach was going to be, bring up the new system and see what failed to connect, and either switch that extension back to Chan_SIP or if possible, try changing the device back to using port 5060. But you can imagine my shock and wonder when EVERY extension apparently connected without issues. And if I run pjsip show endpoints in the Asterisk CLI, the Contact: field shows the port each device is using. More wondrous is that the connections on port 5061 don't seem to interfere with the TLS stuff.
Maybe I have a fundamental misunderstanding of how SIP is supposed to work, and therefore I should not have been surprised, but it was never that easy in Chan_SIP. It seems to me that the biggest problem with PJSIP is the default configurations. For example, the default of IP matching first caused me a LOT of problems.
I only wanted to post this to say that maybe PJSIP is not something we should all avoid as long as possible. Sure, FreePBX needs to support it a bit better (how do I add the new keep_alive_interval=90 in the [global] section, per naf's instructions?) but I suspect they are learning too. I'm just happy as a clam that I converted all the extensions and trunks to PJSIP, and if you'd asked me a month ago I would not have had a good word to say about it!
Shown above are my current General SIP settings (with IP address redacted, of course) and Chan PJSIP Settings pages. I'm not saying those are the optimum settings (comments are welcome) but they are part of what got it working for me.
(The following is GVSIP related information that is no longer relevant, but I moved it to the bottom of the post for historical reasons):
Also, if you set tls - 0.0.0.0 - All to YES in the Chan_PJSIP settings as shown in my Chan PJSIP Settings screenshot, be sure that if you have GVSIP trunks configured in /etc/asterisk/pjsip_custom_post.conf you have the transport line in the registration section for each trunk set to
transport=0.0.0.0-tls
and NOT to the original
transport=transport_tls
from naf's original instructions. If they are set to the original value you can change them using
sed -i -e 's/transport=transport_tls/transport=0.0.0.0-tls/g' /etc/asterisk/pjsip_custom_post.conf
Thanks to uid://1917447 for the above information about changing the transport= line.
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VOIPo down again
I know, I sound like a broken vinyl record. Both 618-632 #### and 618-624 #### lines are down and I guess they were down yesterday too but I did not follow-up on wife's complaint yesterday. Anyway today incoming rings after calling party's call already went to voicemail. Outbound calls do not connect but sorta ring the calling part with one ring but no connections. Courtesy ringback generally does not work.
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[Anveo] Anveo Direct's "Geo POP" setting is not what it seems for US?
According to Anveo Direct support, since they do not proxy RTP media, incoming calls go directly from carrier's (Voxbone or Bandwidth) server to our PBX.
In my Anveo Direct DID configuration, there are only two US locations to choose from: NY, NY and LA, CA. Since my PBX is located in Texas, neither is the most ideal and I'd really want a central location like Dallas.
So my initial assumption was that if I were to choose say NY, my calls will be routed to NY first, then to my PBX in TX.
But upon analyzing some CDRs, it seems all the incoming calls actually originate from Dallas IPs (using IP geolocation lookups).
It kind of makes sense since my DID's carrier is Bandwidth.com CLEC per online databases. They are so massive so I actually can't imagine them having only two POPs in the US and having to route calls from/to central via east/west coasts.
Does this mean the "GEO POP" setting in Anveo Direct's portal is kind of for "reference" only, and the actual carrier (Bandwidth.com) will always attempt to connect calls using their own routing logic to choose the best POPs/servers?
And finally, I assume this setting definitely has nothing to do with outbound calls on Anveo Direct, which lets us choose from a wide variety of carriers, right?
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[Voip.ms] How to circumvent the new Caller ID regulations ?
Ahead of January 1st, 2020, I found out that both Voip.ms and Anveo no longer pass on the CID in local format (9 digits with 0 being the first one).
Now I am stuck with two choices.
Either I am using Groundwire and its push to get my incoming calls (it seems to work reliably) or have those calls forwarded to my cell through Freephoneline (which sadly does not pass the Caller ID on).
Would anyone suggest a smarter workaround ? I am afraid the provider involved won't accept to pass those calls in an international format.
Thanks,
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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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ClearlyIP Acquires Montreal Based Modulis.ca Inc
https://www.clearlyip.com/pressrelease/clearly-ip-inc-acquires-montreal-based-modulis-ca-inc/
Modulis has joined the Clearly IP family. This combines two teams long known for a passion for Open Source real-time communications, bringing together the original people that brought you FreePBX the world’s most popular open-source PBX platform and the largest certified Asterisk team in Canada! Read more at the link above...
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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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Aquire a specific DID
First post here.
I am looking to get a specific DID but I see they are allocated to a specific provider. That particular provider won't let me pick the DID I would like preventing me from getting it an transferring later. From all appearances they are not in use. I checked with both my providers to see if I can get them and it appears not.
Is there any way to get the DID(s)?
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VoIP Holiday Greetings 2019
(Apologies to Clement Clarke Moore and/or Henry Livingston Jr.)
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Twas the night before Christmas, when all through the house
Not a creature was stirring, not even Dell mouse.
The Obi's were hung by the chimney with care
In hopes that Kris Kringle soon would be there.
The children were nestled all snug in their beds,
While flow charts of Anveo danced in their heads;
And mother with her Ooma, and I with a PAP,
Had just settled down for a long winter's nap;
When out on the lawn there arose such a clatter,
I sprang from the bed to see what was the matter.
When, what to my wondering eyes should appear,
But a miniature sleigh, and Future Nine tiny reindeer,
With a little old driver, full of vigor and vim,
I knew in a moment it must surely be him.
He was full of Magic, his eyes they did gleam,
No Local was this, he came on Vox Beam.
More rapid than eagles his coursers they came,
And he whistled, and shouted, and called them by name;
"Now, Dasher! now, Dancer! Now, Voipo and CircleNet!
On, CallCentric! On Flowroute! On, Voip.MS and some I forget!"
"And you reindeer make sure that no reins are crossed
Or else "I Be Mad" if some toys are lost!"
As I drew in my hand to look for a ground
Down the chimney he came with a bound.
Ten Yealinks he had flung on his back,
And he looked like a peddler just opening his pack.
His eyes -- how they twinkled! His face full of verve!
He said, "Come, drink a toast to good power reserve!"
Then he sprang to his sleigh, to his team gave a shout,
"Remember to switch if a server goes out!"
"To the top of the porch! to the top of the wall!
Now dash away! dash away! dash away all!"
Then I heard him exclaim, ere he drove out of sight,
"Merry Christmas to all, and to all a good-night!"
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[Voip.ms] Grandstream Wave - Video softphone...
I tried GS Wave (no video) when first released, had connection issues, and returned to Zoiper. At that time, GS had two versions, with and without video support. Now they just have GS Wave - Video.
I'm trying a Google Fi data only nano SIM in a Samsung Galaxy S6 and decided to give Wave another trial. I uninstalled Zoiper and installed Wave and it seems to be working great... not one glitch yet... seems very polished. VoIP.ms does not support video, but SMS works.
Although Wave offers a SIP account template for VoIP.ms, I decide not to use it. Here are the Wave settings I'm using with a VoIP.ms sub account that has TLS SRTP call encryption enabled:
Settings\Account Settings\+ Add New Account
SIP Account
Account Name = VoIP.ms
SIP Server = washington1.voip.ms:5061
SIP User ID = xxxxxx_OE-M1
SIP Authentication ID = xxxxxx_OE-M1
Password = *****
Voicemail UserID = *97
Display Name = OE-M1
select checkmark to save
Account Settings\VoIP.ms
CALL SETTINGS
Ringtone = select it
SIP SETTINGS
Transmission Protocol = TLS
Register Expiration (m) = 5
CODEC SETTINGS
SRTP Mode = Enabled And Forced
Custom Settings
Theme = select it/dark
Comments welcome. Has anyone used Wave video successfully with a SIP provider that supports video?
OE
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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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