Just obtained an HT812 to replace a PAP2 that failed yesterday. I've got it connected to my router via the WAN port, have tried with both the default DHCP and static IP addresses; have enabled the web server via the phone interface.
I see the device via an arp -a both from my router and from my PC, but it does not respond to either a ping or the web interface.
Any suggestions?
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[Equipment] Issues with Grandstream HT812
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[Equipment] RPi 0 and Rpi 0 W on sale US $3.14 @MicroCenter
I just posted here about the subject. You can grab them from your local MicroCenter store before the offer expires and/or out of stocks. I am sure you can use the Rpi 0 as a JTAG tool and probably Rpi 0 W as a PBX, etc.
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Business SIP TLS provider?
Interested in business level SIP for small office with SIP TLS/SSL support, mobile app and desktop softphone support.
I've been with Appia for a while, but engineering is unable to resolve outstanding issues. I'm not sure if this could be related to their new owners, but service had been great. Disappointed to even be looking.
Any recommendations?
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[Voip.ms] What happens when POP setting in ATA not same as voip.ms portal?
I had the pop settings in both my ATA and voip.ms DID portal set to toronto6, and it's been working fine for weeks if not months. Last week, maybe friday, I was noticing that every 5 or 6 minutes I was losing registration for about 2 minutes, and this was a repeating cycle. I thought it might have something to do with using a new router (Edgerouter ER3) and maybe the NAT-keepalive timing needed some tweaking. I think voip.ms says to use 300 seconds for keep-alive, but I was set for 180. And the ER3 has been in use for more than a week prior to this.
Anyways, in my ATA's (2 of them, SPA112) I changed the POP's to toronto5, and that seemed to help quite a bit. I wasn't seeing registration loss anymore. Outbound calling was working, but I didn't test inbound until today. I try calling in (from another voip.ms did on a separate account) and I get half a ring and then fast busy. I look at inbound call logs and they all show failed. At this point the DID pop's are still toronto6 in the portal. I change them to toronto5 (to match what the ATA's are using) and bingo - incoming calls are getting through.
So I'm wondering if it should have mattered that my ATA's were using toronto5 while the DID account setting was set to toronto6? And by the way, even in that state, voip.ms portal was showing the ATA's were registered and using toronto5.
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[Voip.ms] Get dialtone, ATA registered, incoming calls ok, can't make calls
I started the recent thread "What happens when POP setting in ATA not same as voip.ms portal?" and I thought my problems were over when I changed everything from toronto6 to toronto5.
Turns out a new problem developed today, don't know exactly when, but I was dealing with a live-chat but had to cut it short - and I don't think they can help!
New problem is that I get dialtone, I dial any number (even a did belonging to another voip.ms account) I think I get 1 or 2 rings, then I get fast busy. This is with two ATA's (SPA112) that were working perfectly for a month or two (I had the problem with them where I think someone had logged into the user account and did a call forward to a UK number). These ATA's are no longer login-accessible from the web. The receive calls fine. Calls that are attempted don't show up in voip.ms call log.
Voip.ms thinks that the ATA's outgoing traffic is somehow "trapped" in our lan, but I don't buy that. Another theory is that there's too much sip-port traffic hitting our wan-side - we don't have a good handle as to how to see that yet. We were messing around with an ER3 lite but went back to our old router (never turned it off) and still this problem remains. Power-cycled everything and still can't dial out. But dialing in is fine.
Any ideas what's going on here?
Maybe next thing I'll try is hard-code toronto5 server IP into the ATA.
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[Equipment] Amplified Cordless Phones
Looking for a setup with amplified handsets for my mom. Can be either VOIP base or standard for rj11.
I have an ata in place on her standard inside jacks but could also just as easily replace that with a setup where the base does the VoIP.
Any recommendations?
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[General] Vonage / Motorola VT2142
I have one of these sitting around so I thought I would unlock it and then use Simonics Google Voice Gateway for my brother. I started reading old forums and could not find any info on whether that is possible and really questioned it when I read this:
said by Toro years ago:
One important note: the VT2142 accepts numeric SIP usernames. The only other characters that are accepted in the SIP username are the lowercase characters a,b,c and d. On the other hand, the Betamax services use alphanumeric usernames to login to the web site, and the same username is also used for the SIP settings. So when you create your Betamax account, make sure you use only the characters mentioned above and numbers.That seems like a serious restriction however, my brother does not yet have an account with Simonics (nor do I) so maybe follow the advice given for Betamax users? Any info is appreciated.
Thanks.
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[General] Ring.To/RingTo ends on 31 March 2018
I didn't get this in an e-mail, just happened to notice when I went to their web site tonight. Service ends at the end of March. The first part of their post is below. They're recommending SideLine but I don't know if you can port in a number to SideLine much less bringing in multiple numbers.quote:RingTo is saying goodbye
It's been a pleasure working with customers like you these last few years on RingTo. While we've learned a lot over the course of our journey with RingTo, we're focusing our ongoing development elsewhere. So with that said, we'll cut to the chase here: we are sunsetting RingTo on 3/31/2018.Edited to add: Yes, SideLine does allow porting in at least one number; unsure about multiple numbers.
Port-in info: https://sideline.zendesk.com/hc/en-us/articles/214936063-iOS-Porting-FAQs
Port-in from RingTo specifically: https://sideline.zendesk.com/hc/en-us/articles/360000127846-Porting-your-Number-from-RingTo
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[Asterisk] No ring back tone from Google Voice outbound calls
I have been experimenting with Google Voice on freePBX on and off since last year. Unfortunately my PC's hard drive died a couple of weeks ago and I had to re-install everything on a new one. I installed freePBX 14 (Asterisk 13/Motif/OAuth2) on a Debian Jessie. Everything seems to be working except that I don't have the ring back tone from outbound calls. The phone, hooked up to Asterisk through an HT802 ATA, is dead silence until the other end answers. I have found a solution to get the ring back tone by changing Trunk Dial Options (Connectivity->Trunks) from "T" (Asterisk default) to "Tr". But I remember I did not have to do anything like this last time. So I am wondering if this is the correct solution and if I have missed some necessary configurations when I set up the system. Thanks in advance for any help I can get.
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[CallCentric] Service Affecting Issue - Inbound Calls to Canada Numbers
Trying to call my DID today resulted in Busy Tone. An email to Customer Service resulted in them acknowledging that there is indeed a problem, but haven't heard anything since (noon EST). Meanwhile service seems to have been restored but ticket is still open, and they have this on their dashboard:
Service Affecting Issue - Inbound Calls to Canada Numbers
We are aware of Transport issues affecting inbound service (specifically call completion) to a limited amount of local numbers in Canada. We are currently working with our Underlying Carrier for a resolution.
Can't believe I'm the only one, anyone else affected?
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[PBX] FreePBX for the Raspberry Pi
The included script (install) and archive (install.tar.gz) will build
FreePBX 2.11, 12, 13, or 14 plus Asterisk 11, 12, 13, 14, or 15 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, 13, or 14 Debian Stretch 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 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 (FreePBX 13 or 14 only)
Answer IPv6 option ('No' recommended)
Review selections
Set Hostname (Item 2 / 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 a root.
Installation will complete.
Utility scripts included in /root:
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-set-ptuuid / image-shrink / image-mount
==========================================================================
Backup / Check / Set PTUUID / Shrink / Mount an Image of the System SD Card
upgrade
=======
Upgrade / Update Linux
asterisk-13to14
===============
Upgrade Asterisk 13 to Asterisk 14
asterisk-upgrade
================
Upgrade Asterisk
set-boot
========
Set Boot PARTUUID (or /dev/mmcblk0)
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
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 simultanous 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)
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[General] obi202-FPL-ph2 reserved for alarm system, How to configure
Hi!
I am currently using only one provider (FreePhoneLine).
ph is connected to my home phone
ph2 is connected exclusively to the alarm panel (CID format)
I added the following to Star code profile A
5149041234,alarm1 mode,set($Noji1,200),set($Noec1,1),set($Cdm1,3),set($Lbdt,1),call(5149041234)
If setting ITSP Profile A/General/DTMF Method to InBand
then alarm testing is ok
also voice test (1-416-342-9562) is ok
but using star code (*98 on FPL) get me onto voicemail correctly but does not recognize any touchtone commands menu (3 or *)
So I am very near the perfect solution. Any ideas ?
Note that DTMF Method=Auto works perfectly for voice test and voicemail inquiries but not with alarm testing.
One solution maybe to have 2 profiles (one for ph with Auto) (one for ph2 with InBand)
but I really don't know how to configure this solution.
other solution is correcting the problem with recognition of touch tone commands in FPL's voicemail control.
Every hints will be appreciated.
Thanks and have a nice day
ItiBi
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Seeking opinions on VoIP Monitor (voipmonitor.org)
Can anyone give a review on this software?
I put a lot of work into setting up HOMER SIP Capture in my company's environment only to find it disappointing. It doesn't handle video sessions well, the UI is annoying, and the query engine is super slow no matter how many resources I throw at it. I'm ready to move on to something else.
VoIP Monitor seems to have more polish on the UI, a better capture agent, and a feature we really want, which is RTP analysis.
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Grandstream HT802 Can't Login Into Web UI
First try to configure a brand new unlocked Grandstream HT802, but I can't seem to get into the Web UI for some strange reason. Never had problems with their other models like the HT502 and HT702. The lack of a second LAN port doesn't help either as I have to connect and disconnect the Ethernet cable between the modem and the PC.
So far this is what I did:
1. Connected Ethernet cable from my router's LAN port to the WAN port on the HT802.
2. With a phone connected to Port 1 dial *** and then 02 to get the IP address.
3. Voice says the IP address is 192.168.1.112
4. Reconnect Ethernet cable from the WAN port HT802 to the PC's Ethernet port
5. Open browser and enter 192.168.1.112, but no login window pops up
What am I missing here ? I already did a reset with the paperclip in the back to no avail, anymore ideals on what I should try ?
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[Future9] Am I crazy? 1-213-458-7977 - Different on cell VS F9?
Am I crazy?
If I dial 1-213-458-7977 using F9 VOIP it is some kind of scam line.
BUT
if I dial 1-213-458-7977 using my ATT cell it works and takes it to the SKYPE line for a meeting I was attending?
I have no idea what is going on here?
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[General] nettalk
Hello do you have nettalk issues today, i cannot dial or received phone call
my ligne is disable...i cannot received or make call
the message when i try to call my number is the number your are dial is disconnected or not in service please check...
thank you
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[Equipment] VVX 400, 802.3af and Cisco 3750G POE?
Any reason that combo shouldn't work?
I suspect some ding-dong did some bad cabling, but anyone have that combo of switch and phone running?
An older Polycom Soundpoint conference unit is working fine, the VVXs not powering up, and the cisco not seeing and CDP or LLDP traffic on the ports.
3750G POE version does standards-based 802.3af, and as best I can tell the phones do too...
Remote cable tests off the switch show me two things of note, but I can't find good info on the standard that says whether POE should still work:
- The two main data pairs are intact, but crossed (I suspect someone bought a bunch of x-over patch cables)
- The two extra pairs show shorted, but not sure if I'm seeing the POE transformers on the phones as a short
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[Asterisk] New Kid on the Block: Introducing VitalPBX 2.0
If you haven't tried VitalPBX 2,0, you're missing quite a treat. It's a formidable contender for the best VoIP aggregation out there, and it's free.
Nerd Vittles review: http://nerdvittles.com/?p=25166
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[Equipment] Sick and tired of Grandstream HT-702 issues
Had this for slightly over 2 years and it has been plagued by problems connecting and keeping connections up.
It is always line#1 which is having issues. Tried switching port##, using UDP and TCP, mucking around with NAT to no avail.
Never mind that it takes forever to boot up and connect, often after a reboot it just won't register until I go into the line# and hit Apply, then it is magically registered on line#2. This trick does not work for line#1 but shortly after line#2 is registered, line#1 would register too.
This was bought to replace a retiring PAP2T which started rebooting every 10 seconds. Never had any issues registering with PAP2T!
Grandstream support blames the router w/o even looking into the issue. They never asked what sort of router, just told me to replace it. FU.
It seems to work much better with TCP and NAT keep alive turned on for both line#1 and #2. At least it is not dropping registration that often. Line#2 keeps registration 100% of the time once it registers, while line#1 keeps losing registration several times a day.
Ports 5060 and 5061 worked fine with PAP2T, they never seem to work with HT702, which wants 20000 and 20001 for some reason. And they only work if there is a NAT entry to forward to HT702, where PAP2T did not need any NAT set up at all on the router. What am I doing so wrong with Grandstream which Linksys just figured out all by itself?
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[Other] Softphone that can autodial dtmf tones
Hello,
I have a small business and need to call companies with interactive voice response systems. I'm often dialing the same strings of numbers and I struggle with dialing as the 10 key on the keyboard is opposite of the key numbers on a phone. I'm looking for a soft phone that you can set up buttons to auto dial numbers while in a call or that I could paste numbers into a field to auto dial.
The other issue I have is with my voice and the interactive voice response systems. I don't think I have a deep voice, and while I speak slowly and clearly, the systems often can't understand what I say and I have to go thru the whole process again.
Any suggestions would be helpful!
Thanks!
Eoin
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