Anybody else on the east coast having problems with their Nettalk service? Both my DUO and TK6000 stopped working as of this morning. Reset did not bring the service back. It looks like their website is down as well (Hmmm).
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[General] Is Nettalk Down on the East Coast?
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RIP Elastix
Elastix has been trixboxed http://www.elastix.org/community/threads/what-if-3cx-acquired-elastix.132897/#post-144972
I really liked elastix it so sad to see all these open source projects being destroyed.
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Guide - Anveo Direct with Obihai's OBi202 for Faxing (or not)
I've started a how-to on setting up an OBi202 to work directly with Anveo Direct, bits and pieces can be found, but not all in one place.. Different forums, websites, and then the manual..
I have my OBi202 on it's own static IP, but would work if you were using your OBi202 as a router, for dynamic IPs, would need to log into AnveoDirect and change the IPs if/when your IP changes. Port Forwarding may come later for putting the ATA behind a router, but not a priority for me right now.
Keep in mind I am setting it up for a 2-Line fax machine, but this should only matter for the CODEC page and the Anveo Outbound Call Routing
Variables - You will want to edit these for your numbers
15556667777 - Also set as outgoing CID and CNAM
15556668888
123.125.25.156 - OBi202's static IP
AnveoDirect Trunk Prefix Code - 012345 (Alpha is ok too)
Using OBiTALK's web interface, non-default changes are mentioned
Anveo Direct 1. In normal settings Set Admin Password SP1 Settings-Gear OBiTALK Compatible Service Providers Generic Service Provider Configuration Name Anveo Direct Primary Line for Outgoing Calls - Check Both Incoming Calls Will Ring On - Line1 VoiceMail Notification - Uncheck Both Service Provider Proxy Server sbc.anveo.com Username obi202 (LOWERCASE) 2. Obi Expert 3. Voice Services Gateways & Trunk Groups Voice Gateway1 Name Anveo Direct Access Number sp1(sbc.anveo.com;ui=15556667777) Digit Map (<012345>1xxxxxxxxxx|<011:012345>xx.) AuthUserID 15556667777 4. Voice Services SP1 Service X_InboundCallRoute ph X_RegisterEnable - Uncheck X_UserAgentPort 33445 CallForwardOnBusyEnable - Check CallForwardOnBusyNumber ph2 5. Service Providers ITSP Profile A General Name Anveo Direct X_SymmetricRTPEnable - Check ITSP Profile A SIP ProxyServer SBC.ANVEO.COM OutboundProxy SBC.ANVEO.COM 6. Codecs CODEC Profile A G729 Codec Enable - Uncheck G726R32 Codec Enable - Uncheck FAX Event Enable - Check Codec Settings T38ECM - Check 7. Physical Interfaces PHONE 1 OutboundCallRoute {(Mvg1):vg1} CallWaitingEnable - Uncheck ForwardOnBusyNumber ph2 PHONE 2 OutboundCallRoute {(Mvg1):vg1} CallWaitingEnable - Uncheck
For incoming calls, I want them to ring Phone1, if Phone1 is busy, then ring Phone2, if Phone1 and Phone2 are busy, give a busy tone
Anveo Settings
Destination SIP Trunks
Title OBi202SIP URI obi202@123.125.25.156:33445
Outbound Services/Call Termination
Dialing Prefix: 012345Authorized IP Addresses: 123.125.25.156Call Routing Method: Custom Least Cost Routing ModelGet Routes/Carriers within 'Rate Cap' from: Custom Routes At the bottom check all carriers with T.38, US/Canada and WorldwideThen take top N results: 5 Routes/CarriersFinally send calls (with failover) to routes in order of: random orderSmart Route Option: - Checked
Changing the Routes/Carriers is from a suggestion by Anveo's support, Thanks!
I think this is everything I have set.. As long as the SIP Scanners don't find the UserPort, all should be good. Even if they do find it, in this case, it is my fax machine, so I won't hear it anyways.
Outstanding Tweaks:
Inbound Calls,
There is a way to enter the DIDs, so the incoming Route pays attention to the DIDs, but I could not figure it out, was initially why it wasn't working.. :|
Outbound Calls,
Have to add dialing in E164 format outside US/Canada.. Also add input checking, right now 1NPANXXXXXX works without checking for the proper NPANXX (10001005555 is accepted), and 011+ works, but dialing 44.. does not.
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voip setup question
disclaimer: total newbie here!
My current set up:
cable internet --> cable modem --> linksys router --> wired computer + 3 WiFi devices
pots line --> base station + an additional handset
modem, router and wired computer are all located in the bedroom. There's a phone jack in the bedroom but on the opposite wall. Base station is in the living room and the other handset is in the kitchen.
My idea was to get Obi202 and hook it up like this:
cable internet --> cable modem --> obi in bridge mode --> linksys router
Ideally, I would like to keep both handsets where they are now. What do you guys suggest?
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RESOLVED - Configuring CSipSimple for Anveo (Android 5.1)
I'm really not as ignorant as I look...really!
Should be simple but won't configure no matter how/what.
Maybe I should have gone with Voip.ms, no? (Sipdroid and Linphone are about the only app that will config, but I want CSipSimple.)
Thanks for any help you can provide.
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[General] Most reliable DID provider to port an *Alaska* number to?
I'm looking to port an Anchorage-based landline number over to a DID provider and route it to my Asterisk box.
The catch about this particular question is that not all of the "major"/recommended providers will work with Alaska numbers. A few (now! finally!) offer some Alaska DIDs, and several more (including ones that don't actually sell Alaska DIDs) appear to support porting my specific number, at least based on the testers on their websites.
In addition to some specific recommendations, I do have a general question: is there any benefit to porting the number to a provider that also resells Alaska DIDs? I'm left wondering if the providers who will take my number by porting but don't sell any numbers in Alaska may not be as reliable. Nothing to base that on, since I assume the traffic flows from the DID owner's TDM switch to me via IP regardless, but it's just a gut-check I have.
Anyway, what I'm looking for is primarily reliability; for origination, cost is a secondary concern (within reason). Also, I'll need E911, as this is a primary residential line.
So, I've uncovered the following:
VoIP.ms: sells Alaska DIDs and will port Alaska numbers; offers E911; proxies media
Flowroute: will port Alaska numbers; offers E911; direct media
Callcentric: sells Alaska DIDs and will port Alaska numbers; offers E911; direct media?
Vitelity: may sell Alaska DIDs and will port Alaska numbers; offers (free?) E911
CallWithUs: will port Alaska numbers; no E911
FutureNine: unknown; offers E911
Anveo Direct: no Alaska DIDs/porting; no E911; direct media
Alcazar: no Alaska DIDs/porting
VoIP Innovations: unknown--company based a few miles from me in Pittsburgh, though!
DID Logic: sells Alaska DIDs, does not port (AFAIK), supports G.722 (for whatever good that does here)
As for pricing, the only one that's really somewhat uncompetitive is FutureNine's $20 fee for number porting. The others that port Alaska numbers are within the same ballpark, so reliability/call quality is my main concern. That said, all other things equal, VoIP.ms and Flowroute come in at the low end of the inbound calling spectrum, at least among providers that work with Alaska number porting, though Vitelity has a slight edge if it's true that they have free E911. (VoIP.ms currently also has a promo with free porting.)
I have personally had good experience with both origination and termination with VoIP.ms and Flowroute, though Flowroute doesn't sell Alaska DIDs, so it's back to my first question about reliability with that arrangement. OTOH, VoIP.ms has a potential single-point-of-failure (since I must register to a single server that also proxies the RTP stream), whereas Flowroute has a few different servers with DNS SRV and does not proxy media, which I appreciate.
So, my question is, given all of the above, what would your recommendation be? I'm kind of leaning towards either VoIP.ms or Flowroute, but if CallCentric is going to be much more reliable, they may be worth the (small) premium; Vitelity remains an unknown, and the rest probably won't work for me. Am I missing anything?
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I also have a related question about termination: Anveo Direct has (by a good margin) the lowest prices for calling Alaska (907 area code) numbers. How do they compare, especially for Alaska calling?
Back when I worked for and set up a PBX with a local company in Anchorage, I had experience with VoIP.ms for outbound calling to Alaska numbers. VoIP.ms's value route of 3 cents per minute was OK, but I had a lot of complaints from my sales staff about poor call quality especially to rural Alaska, so I ended up just moving intrastate calls to our PRI at 16 cents per minute for long distance with the local ILEC (ACS) versus 18 cents per minute for VoIP.ms's premium route. After moving state, I later discovered Flowroute's 5 cent per minute rate for Alaska, and my own limited personal testing seems to indicate that it is fairly reliable--at least it's much better than VoIP.ms's value route for Alaska.
But--Anveo Direct has rates as low as 0.2 cent per minute for many numbers in urban Alaska and peaking at 3.2 cents per minute--much lower than almost every other provider--for even rural Alaska calls, and even when looking at their intrastate rate table (that's another ball of wax). Anyone have any comments on the reliability of Anveo Direct's rates and routes, especially as it applies to Alaska? Any other things to look at for outbound termination as it relates to Alaska?
Thanks!
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[CircleNet] Anyone still using CircleNet for outgoing calls?
I know they are shutting down their retail business but I like the prices and call quality. :)
Problem is, I haven't been able to make any calls since at least last Monday.
Anyone able to successfully make calls? I logged a ticket but haven't received a reply - not that I expected one.
I was hoping to use up the last of my money with them in the hopes of avoiding getting a refund... I may have to rethink this.
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[Other] RingPlus just ate their words
I know they cannot keep doing this but it comes too soon.
Is there any decency left in US businesses? Too bad!
This is what I received today from the email to me:
"RingPlus has to switch to a new system. Sadly, because of this switch, we will have to discontinue your current plan.
For Important Information, please visit http://bit.ly/Important-Plan-Info"
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MaxEmail Fax Service taken over by j2 Global®
We've talked here before about Voip customers using MaxEmail for sending and receiving faxes.
MaxEmail was a good service, reasonably priced.
All of a sudden the company was bought by j2 Global®, who already own eFax® and various other brands. :(
Some people choose not to do business with j2 Global® due to various policies.
https://www.dslreports.com/forum/r26580959-MyFax-eFax-500-if-you-port-out-AND-we-ll-take-it-back
Existing customers should be receiving an e-mail about the change, which includes new rate plans and new TOS.
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[CallCentric] Rate plans and per minute charging ?
This is for those who are familiar with CallCentrics rate plans, I have a Free NY number on a pay per call plan, I had $9.91 for a balance, I placed a call to another number, I did not answer and hung up the phone. My balance went from $9.91 to $9.89. If the calls are $0.0198 How is it I was charged $0.03 for a simple dial and hangup call. I want to get feedback before I make a fool of myself turning in a trouble ticket, for something I did not see.
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[General] SIP Broker Toll Free termination via CC peering
Hi everyone,
I've been using the SIP Broker Toll Free termination service via CC peering from for a while now and it was always reliable. Coincidentally as I made some changes to my OBi110 star codes I noticed TF calls were now rejected by the service provider (was this CC or SIP Broker?) with a Reason 500. I contacted CC and they insinuated that being a free service I should try and reach it via another VoIP service. Has anyone tried a TF call recently via Callcentric peering?
This thread from a while ago led me to configure and use it:
https://www.dslreports.com/forum/r29122754-General-How-to-make-no-charge-8xx-calls
Also SIP Broker lists all TF numbers as working:
http://www.sipbroker.com/sipbroker/action/providerWhitePages
Thanks in advance,
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[Equipment] Verizon Prepaid CDMA - LG Optimus Zone 3 4G LTE
I just posted this in the HotDeals forum. I believe this smartphone can be used as an inexpensive WiFi SIP phone, Google HangOuts phone, Skype, WhatsApp, etc. So, grab it before it is gone.
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[General] G.729a patent expiry
Anyone heard anything about when the patents on this codec are going to expire? The ITU-T recommendation was published 20 years (plus 6 days) ago, so unless there were any submarine patents, is it now free?
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Good (cheap) Call routing/voicemail for beginning real estate use
My wife and I are starting out in real estate. The nature of the business is that we will likely be "poor" for the first six months. We are looking for a reliable call routing system (virtual/online) [e.g. press "1" for me, "2" for my wife] which can forward calls to our cells and take voicemails (and do other things that good call routing systems do. Since we will have little to no income the first six months, we are looking to do this as cost effectively as possible. We were going to try Google Voice (since it's free), but it seems that reliability and feature limitations don't make it a good choice for this type of business (correct?). Any suggestions greatly appreciated.
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[Equipment] OBis on sale
From my inbox...
OE
Amazon 12/12 to 12/16, 12/14 to 12/15 & 12/18 to 12/19*
- From 12/12 to 12/16, get $20 off all OBi IP Phones (OBi1062, OBi1032 and OBi1022).
- From 12/14 to 12/15, Amazon will feature the OBi200 for $37.99.
- From 12/18 to 12/19, the OBi202 will be on sale for $49.99.
Newegg 12/16 to 12/17 & 12/20 to 12/21*
- From 12/16 to 12/17, Newegg will feature the OBi202 for $49.99.
- From 12/20 to 12/21, the OBi200 will be on sale for $37.99.
Newegg.ca 12/23 to 12/27, 12/28 to 1/1, 12/15 & 1/3*
- For our Canadian customers, on 12/15 and from 12/28 to 1/1, Newegg.ca will feature the OBi202 on sale at C$69.99 + free shipping.
- From 12/23 to 12/27 and on 1/3, the OBi200 will be on sale for C$49.99 + free shipping.
* WARNING: The Amazon and/or Newegg promotions happening or not is dependent on several factors outside of Obihais direct control. At the time of writing and distribution of this newsletter, to the best of our knowledge, the aforementioned promotions are scheduled to occur on the dates indicated. We apologize if there are any inconsistencies with actual promotion dates, including the occurrence of the promotion, in the first place.
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[General] Softphone Suggestions?
Hi everyone! :) I've read a few threads from searching around, but can't find anything conclusive. I want to use a softphone on my laptop and desktop for my VoIP.MS service. I was using the free 3CX version, but it turns out 3CX has deprecated the software and it isn't supported. Also, my laptop has a 4K screen and the softphone is one for ants at that resolution since 3CX doesn't scale.
I know there is bria.. but that's not free. And I tried Linphone, never ever again. Anyone have any suggestions? (Yes, I've read the softphone setup guide on VoIP.MS' website to get some ideas... many of them seem discontinued or not updated for a long time)
Cheers for the help!
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FreePBX for the Raspberry Pi
The included script (install) and archive (install.tar.gz) will build
FreePBX 2.11, 12, or 13 plus Asterisk 11, 12, 13, or 14 on a Raspberry Pi.
iptables, dnsmasq, exim4, and pygooglevoice are also installed.
Installation takes a little over an hour to complete on a Raspberry Pi 3.
Download the latest Raspbian image.
For FreePBX 12 or 13, Debian Jessie Lite is recommended:
https://downloads.raspberrypi.org/raspbian_lite_latest
For FreePBX 2.11, Debian Wheezy is required:
https://downloads.raspberrypi.org/raspbian/images/raspbian-2015-05-07/2015-05-05-raspbian-wheezy.zip
Write the image to an 8 GB or larger SD card. To accomplish this, I recommend imageUSB:
http://osforensics.com/downloads/imageusb.zip
Prior to ejecting the SD card, 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 root password
Select time zone (in US, use America, not US)
Select FreePBX version
Select Asterisk version
Answer Edge option (FreePBX 13 only)
Answer IPv6 option (No recommeded)
Review selections
Expand Filesystem (Item 1)
Boot Option (Item 3 / B1)
Advanced Options (Item 9 / A2 / Hostname: FreePBX)
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 a root.
Installation is complete.
Utility scripts included in /root:
abn / dbn / qbn
===============
Add / Delete / Query Blacklist Number
add-fcc-blacklist / del-fcc-blacklist
=====================================
Add / Delete 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-shrink
===========================
Backup / Shrink an Image of the System SD Card
upgrade
=======
Upgrade / Update Linux
asterisk-13to14
===============
Upgrade Asterisk 13 to Asterisk 14
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-fax
===========
Install Hylafax Server
add-fax-extension
=================
Add Hylafax Extension
del-fax-extension
=================
Delete Hylafax Extension
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[GSM - SIP] Looking forward cell phone calls and SMS to SIP
I have a cell phone in another country and would like to make and receive calls and SMS from that line to my US number (Google Voice but I don't mind having a separate SIP number for this).
I have a raspberry pi 1, a SIM dongle supported by chan_dongle, and an ObHai 110 (although I don't think I need it for this)
How would I get something like this set up?
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Grandstream HT812?
Does anyone here have one of these registered with an ITSP that allows dialing by SIP URI? I'd like to try some test calls to see exactly how it handles Opus.
I have mine registered with OnSIP, so it can pass any codec to a SIP URI without being proxied.
In test calls to local end-points (Polycom VVX, Grandstream GXV, Bria, Jitsi) I've not been able to get it to use Opus. Each Opus implementation is different, so the codec simply isn't accepted in these cases.
I haven't resorted to Wireshark yet. I thought it worth trying calls between two HT812s first.
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[PBX] PRI vs SIP w/no caps, no termination fees, 23 DID
We have a PRI, hardware PBX onsite, with 30 extensions, each with voice mail, caller ID, call forwarding etc.
The call center will swing from 6-20 CSR operators depending on the season. 6 for about 3 months out of the year gradually scale up to 20, then scale back down to 6.
I really need to know how much the PRI is and how many calls we average before being able to give him my feedback about moving to VoIP. So far this is what I've found:
Buy 30 new IP phones, pay $18 per SIP, $2 per DID; for 5,000 minutes per DID.
OR
Buy a new voip gateway, replace the PRI with a SIP trunk; for calls by the minute.
OR
Keep the PBX, keep the PRI, keep the phones, because POTS comes out cheaper for the call volume we do yearly.
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