I'm trying to send calls to a provider that wants the number as country code + number. When I dial in that format it matches the default rule [2-9]xxxxxxxxx and the call is not connected. I edited the default rule from 011xx. to xx. to remove the 011 prefix but if I dial 011+country code... the 1 is still prepended to the number. If I remove [2-9]xxxxxxxxx then the call is connected. I tried putting the rule as the first rule but that didn't help.
How can I remove 011 from the dialed number and keep the [2-9]xxxxxxxxx rule in place?
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Obi digit map issue
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[Anveo] Call Flow to Selectively Enable or Disable CNAM Dips
Can this be done ?
Most likely I'm looking to set criteria by area code.
It would function like a supplementary blacklist against wasting money on identifying calls from places from which I never get legitimate calls, as I am getting a lot of those this summer.
Recently, I have disabled CNAM dipping for the first time since enabling it nearly 4 years ago, but would prefer a better way to manage the setting.
Any thoughts ?
If not currently possible, then please consider this as a shoutout to Anveo for addition of a feature to make it so.
Thanks. :)
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Porting-out Fax No. from OneSuite (Local Access LLC - JSI/1)
Good day.
For over two weeks now, I've been trying to port my fax No. from OneSuite to Callcentric (Local Access LLC - JSI/1 would be the losing carrier).
I was first told by OS C.S. to use my account information for the porting-out. After a couple of days, the LNP order was rejected because of an address mismatch.
Called OS C.S. again and was told that the account address was the only address they used. In the mean time, CC had requested the CSR from Local Access, only to be told that there was no CSR.
I found OS's corporate address and had CC resubmit the LPN order with it. More time went by and it was rejected again by LA because, supposedly, the DID was associated with a DSL line. OS does not provide DSL lines.
A few more days later, I received a call from a very nice OS C.S.R. who gave me the name and address of another company and instructed me to use them. CC canceled the original LNP order and submitted a new one with the new name and address. After four days, LA rejected it, again, for address mismatch.
I can't be the only person who has attempted to port a DID out of OneSuite. Has anyone else done it successfully. If so, kindly let me know how you managed to do it (especially what LA uses for a name and address for OS).
One final thing. OS C.S. confirmed that the DID can be ported out, but I will be hit for a Jackson (don't know at which point).
Thank you.
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[Voip.ms] Small Office - Voip.ms setup help
Hi, We have a small office in Mississauga, currently setup to use voip.ms (by me) with about 6 sets. It has worked, for the most part, but we would like to get some help in fixing a few issues and generally making it more stable. Issues we are having:
- call dropping - intermittent
- sometimes phones ring, sometimes they don't (i think they are spa942s)
- i have ringgroups setup
I was able to spend more time on this in the past but lately have not had the time, and i'm sure someone from this forum would be able to sort out the issues much more quickly than me. We are okay to pay a reasonable amount for consulting.
Thanks for reading. Brian
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[Other] NetTalk question
I've been using NetTalk as my primary "landline" for the last 3 years. About 3 weeks ago a situation developed where calling out is not a problem but incoming calls don't come in. I've been in contact with NetTalk and they have sent my case to their "concerns" department but I have received no word after repeated requests. Has anybody else experienced this? The vibe I'm getting from NetTalk is they don't consider this a real priority. You would think an issue like this would get their attention. Are they in financial trouble and have bigger things to worry about? Do I need to move on VOIP-wise? Any input would be appreciated because all I get are canned responses from NetTalk.
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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, or 13 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
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 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 / fcc-blacklist-exceptions
================================================================
Add / Delete FCC Blacklist
ipt-add / ipt-del / ipt-chk
===========================
Add / Delete / Check 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-upgrade
================
Upgrade Asterisk
set-timezone
============
Set System and PHP Time Zone
regen-ssh-keys
==============
Regenerate SSH Keys
clear-cache / clear-logs
========================
Clear Cache / Logs
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[Equipment] Wireless Non-Bluetooth Headset for Cellphone?
Is there an equivalent to DECT wireless headsets, but instead of for VoIP/Deskphones, for Cellphones? Bluetooth isn't really a sustainable solution for serious business calls.
I'm thinking some sort of Wireless DECT Dock that has 3.5mm input?
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[Equipment] Obi DigitMap Question
I have an obi110 connected to my own asterisk server, and I've got asterisk configured to send calls through different services when I prefix the number with a star code (like *2 *3 *4, etc).
So I've got this in my Obi digitmap:
*[2-6]1[2-9]xx[2-9]XXXXXXS0
It allows the call, but there's always a long delay before it will send the call out (unlike 1[2-9]XX[2-9]XXXXXXS0 which sends the call immediately).
I thought there might be some interference from the Star Code Profile, so I deleted all of the default codes. But it made no difference.
I also tried putting {t=di2} after the *[2-6] to have it play a second dial tone after the star code, but it did nothing.
Any ideas?
I did this with my Polycom easily with a digitmap: *[2-6],1[2-9]xx[2-9]xxxxxx (with timeout=0)
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Needing a SIP line
I need a SIP line for use with a small PBX system which calls that go to my mobile are instantly fowarded to the SIP number which calls my mobile so callers are placed on hold. But it seems like with CallCentric I can only have a plan for just outbound and only a plan for inbound? But I need both!
Any suggestions for this?
Thanks!
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[Unlock] WRTP54G-NA Unlock Procedure
Howdy,
Picked up a used WRTP54G-NA adapter. Yes it is the NA.. The device doesn't have the -NA stamped on it but the box does with the matching MAC and serial number. Firmware Version: 3.1.24
Trying to use CYT 4.6 to unlock it, this version (after trying a few of the 3. versions) sees the device..
Silly question, does CYT only work for Vonage units? I thought it worked for all of the providers.
It is trying to register to ip.wrtp54g.sipura.rgns.net Galaxy Telecom, which after a couple buy-outs and name changes is currently TNW tnwcorp.com.
Tried contacting them just in case they knew or would release the Admin password, was told it was used on Xplornet's telephone service. But they told me they didn't know what the password was. I didn't know that Xplornet had VoIP service until this week, but contacting Xplornet about anything is a waste of time to say the least. :(
Trying to use CYT to unlock this, I get this far
CYT-Unlocker - Local network version v4.6_ Device IP Address: 192.168.15.1 CYT Model: WRTP54G Your device has several login accounts. The security-level for each is different. You should assign your device unique passwords to keep it secure. Login Name Default Password Description admin admin ROUTER-level access (lowest) user tivonpw USER-level access (medium acc) Admin (unique value) ADMIN-level access (full acc) Note: Some devices change the Login Name.. These passwords listed below will be set on your device. ADMIN-Level login: Admin ADMIN-Level password: Admin USER-Level login : user USER-Level password : user --------------------------------------------------------------------------------Make changes to the values above & press [ENTER] to begin updating your device STATUS: Please Verify Settings Your Device is: WRTP54G
Pressing Enter..
Simple XML provision server is waiting for your device..This should only take up to one minute.If it takes more than one minute, you may stop the XMLprovision server by pressing the [ESC] key.
But no further, and yes I completely disabled the firewall (after the pop-up asked me if I wanted to allow it), also tried running as Administrator.
Ping hack, if I am doing it right is not enabling the bootloader console. The pings do show on the console though.
DSLR wouldn't let me post in the thread about the CYT tool, too old.. haha So creating this one.
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CircleNet is backing away from retail VOIP.
Good Morning DSL reports,
I wanted to bring some news about CircleNet LLC. Weve noticed over the last year an increasing number of local agencies starting to apply taxes/fees to VOIP services. Weve been trying to work with each as they do this to make sure that we stay compliant but the varied nature and complexity of this is really mind boggling. This process has been very difficult for us because of the prepaid nature of our service.
We came to a realization about 2 weeks ago that we were spending much more time and money completing paperwork to stay compliant than we were actually working with technology. We dont see this trend of increased regulation at the local level or federal changing anytime soon.
This combined with some personal issues has led us to make a decision.
We will not be taking new customers for the VOIP business also we will take no further payments after 12/24/2016. Well continue operating the infrastructure until 12/30/2017 if there are customers still using it.
This only affects retail customers we will continue to provide wholesale VOIP services as is to companies with FCC499s in place, virtual servers, ISP services in our area and other tech services. In the future and Ill keep lurking here and please feel free to PM me with questions.
A notice similar to this will go up on our website soon.
Guys, thank you for what was a pretty cool time.
Sam
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[General] Unlimited Lifetime SIP Trunk $85 with free Sprint cell service
We're a Johnny-Come-Lately to the RingPlus world, but this is a deal worth considering. If you join their new Member+ program ($69.99 for a lifetime* membership), you can then sign up for the new (one-time $15 fee) Eternity Free Plan which gives you free unlimited calling in the U.S., free unlimited text messaging in the U.S., and 1.8GB of 4G LTE free data each month using any Sprint-compatible smartphone. The silver lining is that RingPlus also provides free SIP calling on your same account, and it works great with any Asterisk PBX. Translation: $85 gets you an unlimited Asterisk SIP trunk for your lifetime or the lifetime of RingPlus... which ever comes first. :-)
Offer expires this Sunday, March 13. If you have Sprint service in your neighborhood, this is a No-Brainer. If you have an Asterisk PBX and no Sprint service, it's still a No-Brainer just for the free U.S. calling via SIP trunking.
Additional details and Asterisk trunk configuration here: http://nerd.bz/1p8lOKo
RingPlus summary and fine print: https://social.ringplus.net/discussion/460/eternity-free-plan-member/p1
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[General] Small call center info.
Hello,
we would like to setup a call center for 6 restaurants. All restaurants have 2 landlines. Client wants all calls to main number forwarded to call center from 9am to 6pm for reservation purposes.
Client needs music on hold and call audio recording. If somebody wants to talk with manager call can be transferred from call center to second landline at restaurant.
We were thinking to install a traditional pbx with 6 phones but we saw potential in voip providers and cloudpbx. We were also thinking to install local server with Freepbx or Elastix.
Any suggestions?
Thanks
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[Other] FYI: Level 3 Toll Free Outage
For those of you with inbound toll free where Level3 is the carrier:
08/29/2016 12:49:55 GMT - The Voice NOC reports toll free numbers are not completing in multiple markets impacting voice services. 911 services are not impacted. Multiple teams have been engaged and troubleshooting is in progress to identify and resolve the issue. Please be advised that updates for this event will be relayed at a minimum of hourly unless otherwise noted. The information conveyed hereafter is associated to live troubleshooting effort and as the discovery process evolves through to service resolution, ticket closure, or post incident review, details may evolve.
https://www.reddit.com/r/networking/comments/504xbo/level_3_voice_outage_global_ticket_being_worked/
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Grandstream Wave app or Wave w/Video app?
I want to try some non-CSipSimple softphones on Android 6.x over WiFi and 4G LTE...
It seems that Grandstream's Wave softphone on Android has a separate app that supports video calling. Has anyone determined why two Wave apps... why not just the one development that supports video, assuming video calling is optional?
OE
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[Asterisk] Dial plan injection attack
If you administer Asterisk or FreePBX or similar and you have never heard of a dial plan injection attack I strongly suggest that you read the following:
https://github.com/asterisk/asterisk/blob/master/README-SERIOUSLY.bestpractices.txt
This is NOT something new, but I have been running Asterisk for over a year and I just stumbled across it.
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[General] A few e911 questions
What do I add to my dialstring to be able to dial 911?
Does voip.ms charge a fee every time 911 is dialed?
Is there a way to make a test call to ensure that the correct address is being relayed?
Thanks in advance.
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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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Washington DC 911 center down for 90 minutes due to kill switch
The District's (Washington DC) 911 emergency line went down for 90 minutes beginning late Saturday because a contractor trying to shut off an alarm accidentally hit a master turnoff switch, officials have concluded.
The contractor, working through a company hired by the Department of General Services, is no longer working at the Office of Unified Communications, officials said. Meanwhile, a review is underway to determine whether changes should be made in the way the shut-off switch is set up to avoid the issue in the future.
The outage lasted from about 11:35 p.m. to 1:15 a.m. During that time, residents trying to reach police, fire and medical crews had to dial a 10-digit number to seek emergency help. A backup center should have immediately begun accepting all 911 calls, but that system also failed, officials said. They said they are working to determine the cause and whether it is linked to the switch....
https://www.washingtonpost.com/local/public-safety/contractor-hit-wrong-switch-knocking-out-911-for-90-minutes-in-district/2016/08/29/649d2140-6e1e-11e6-8533-6b0b0ded0253_story.html
"We have met the enemy and he is us".
---Pogo
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[CallCentric] NEW: Free Conferencing Service Backed by CallCentric
Anonymous Calling by All Conferees
Unlimited Conferees
Unlimited Conference Duration
5-second Setup
No Sign Ups; No Credit Cards
Free Service except for the call to New York number
Backed by CallCentric
If this sounds like your kind of conferencing service, then give Anonochat a try.
You can set up your conference with a web browser. If you do it from an iPhone, you're automatically connected and logged in to the conference room.
If you're really paranoid, we've provided a BASH script on the PIAF Forum.
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