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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Grandstream Wave app or Wave w/Video app?
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Iphone Voip SMS, T.38
Hey, I'm looking for a VoIP provider that offers reasonably low cost service (pay as you go per minute) that offers bi-directional SMS and calling to a VoIP app on my phone, such as Bria, for example. As well as offers T.38 fax support.
Other plus is if it's canadian based, and billing is in CAD, but that's not important at all.
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[Anveo] Anveo Direct termination rates discrepancy?
I downloaded two termination rates lists from Anveo Direct last night:
anveodirect.all.csv
anveodirect.prime.csv
I ran about 20 CONUS contacts against 'prime' and the rate_inter was the same for all: $0.00313.
This includes my Callcentric free NY DID.
I ran a handful against 'all' and none of the routes, prime or standard, had a rate that low.
Here's an example (Callcentric free NY DID) from 'all':
USA,1646547,0.00881,0.00856,prime,2,Intelepeer
USA,1646547,0.00962,0.00962,prime,23,Windstream
USA,1646547,0.00642,0.00717,standard,24,Alcazar Networks
USA,1646547,0.01188,0.01188,prime,3,Excel
USA,1646547,0.0295,0.0295,prime,4,Inteliquent
USA,1646547,0.01208,0.02186,prime,6,Global Crossing
USA,1646547,0.01004,0.009,prime,7,Earthlink Carrier
USA,1646547,0.0082,0.00862,prime,8,360 Networks
USA,1646547,0.0118,0.00795,prime,11,Pacwest
USA,1646547,0.03106,0.03144,prime,12,Verizon
USA,1646547,0.01103,0.01192,prime,13,Hypercube
USA,1646547,0.01088,0.01169,prime,14,Zone
USA,1646547,0.0115,0.0115,prime,15,Level3
USA,1646547,0.01038,0.01033,prime,16,O1 communications
USA,1646547,0.01507,0.01507,prime,17,ATT
USA,1646547,0.00962,0.00962,prime,18,ANI Networks
USA,1646547,0.01936,0.0184,prime,19,CenturyLink
USA,1646547,0.0059,0.00556,,25,Bandwidth.com
USA,1646547,0.01223,0.00844,prime,29,Onvoy
Can someone explain the apparent discrepancy?
Hope I'm not missing something.
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[Equipment] Obitalk forum: I found myself banned!
I admit I have not visited in maybe 2 months, but today I was looking for some F9 info and followed a Google link...my account has been banned?!?!?! WTH?
Have they closed the forum and this is a generic message? I can't even find a link to b!tch to a mod or admin.
Some of the information there is priceless (Digit maps, tips and tricks) - I hope it won't get lost.
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[General] Linksys SPA942 Desk phone on Samsung Officeserv 7400?
Hi There,
So I have a box of about 40 Linksys SPA942 handsets, and I was wondering if it would be possible to configure it on our Samsung Officeserv 7400 PBX? I get as far as getting the phone to register to the PBX, but when i try and make a call it says "Invalid Number". I suspect this would be the dial plan that needs setting up on the actual handset, but I have no idea how to set this up. I reset the device to factory defaults, then configured it to register to my PBX. One other thing I note is that even though the phone "registers", I don't see it consuming a 3rd party license when i log into OSDM (Option is called "3rd SIP Phone), and then as I mentioned earlier, when i dial any number the phone reports "Invalid Number"
I was wondering if there is something obvious I'm missing here? Any help would be highly appreciated!
Thanks!
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[Asterisk] Using pjsip with Simonics gateway
I was wondering if anyone has experience configuring pjsip.conf for use with Simonics gateway. I was able to get outbound calling configured without any big effort but need to know the settings for the registration section of pjsip for inbound to work. He has settings for sip.conf but so far I haven't been able to translate them.
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Beware Sangoma S300/S500 IP Phones
So, I've run into a bit of an interesting and frustrating problem here...
I have a couple of the new Sangoma phones (S300 and S500) that I've been putting through the paces, as they have some really slick integrations with FreePBX, and I'd love to be able to use them in use-cases where that would be beneficial. There have been some bumps along the way, one of which was resolved by me sticking to an older firmware version (where dialplan functionality works properly), and the other addressed by spending five minutes writing a simple Grasemonkey script. (There's an issue with the commercial EndPoint Manager module not rewriting certain characters in the dialplan, which then breaks the XML. I've put a few requests in, and they keep saying it's fixed, but it's not. I'd fix it myself in a matter of minutes, but the module source is encrypted, leaving me with zero control of it. Finally grew tired of fighting over it and just wrote a Greasemonkey script to fix the issue on the browser side.)
Now, I've run into a major issue that I've not yet been able to find a workaround for: These phones claim to support "HD Voice" (or wideband audio, whatever you want to call it.) There's a nice "HD" logo on the handset. The specs show the phone supports G.722, and it even negotiates G.722 audio with the PBX if configured to do so. However, even on a G.722 call, these phones flat out refuse to output a wideband audio signal.
I can tell a MAJOR difference in the audio on a G.722 call between my Sangoma S500 and old Cisco SPA508G. Using G.722, calls on the Cisco have plenty of response on the low/high ends of the spectrum, whereas the Sangoma phone just sounds "flat" (almost like G.711.)
Here's what I've tried and found: Switching the Sangoma between G.711 and G.722 (verified on the PBX that it was, in fact, using the codecs it should be), there's no discernible difference in the audio between the two. Doing a side-by-side comparison (call to the same MoH source, which *is* HD, one phone to each ear) with both phones using G.722 renders that the Cisco sounds MUCH cleaner than the Sangoma. I also tried a side-by-side comparison with the Sangoma running G.722 and the Cisco running G.711, and they both sound about the same at that point.
I've verified it isn't the handset itself, as I can connect the Sangoma handset to my Cisco phone, and the audio is amazing (it even has more bass response than the Cisco handset, almost too much really.)
I decided to lodge a ticket with Sangoma on this, because it seems like a fairly critical issue to have a phone that advertises "HD", supports G.722, yet won't actually produce audio any better than what you'd get from G.711. Before lodging a ticket, I connected the audio output of the handset (not headset) port to my computer (I had to make a special cable to do this), and ran some tests using spectrum analyzer software on my machine to show what frequencies were present in the audio (I've attached the results of that here.) Using the same HD source, with G.722 in both tests, I found the Sangoma drops off sharply once you hit 4000Hz, whereas the Cisco starts dropping off around 7000Hz (as expected for G.722), and passes a little bit all the way up to 8000Hz.
I submitted a ticket to Sangoma, including the same spectrum analyzer output that I've attached here. I figured it'd be a simple matter for them to take that information and fix the problem. Instead, the responses I've received (after having included the spectrum analyzer output) are along the lines of "we feel their [sic] is no voice quality issues and on par with our competitors in our price range we are on", "everyone has their own feeling on what they feel is good audio quality and in the end these phones are not some high def stereo system", "nobody else is complaining and it has passed all of our test [sic]", "G722/HD voice is such a crock as once its [sic] transcoded by asterisk to ulaw it sounds worse than standard ulaw would be all the way through", etc.
Uhh... your competitors (Cisco, Obihai, and Polycom I have all personally used) are supporting G.722 properly, and it works well with their phones. (Just last night I carried on a conversation with a broadcast buddy of mine, he was using a Polycom 650 and I was on a Cisco SPA508G, and the call was definitely proper HD.) I wouldn't consider the output of a spectrum analyzer coupled with the official specs on G.722 (which are not being met) to be considered my own feeling of good audio quality. I consider those to be hard facts. I'm not opening a ticket to debate the (de)merits of transcoding G.711G.722 on Asterisk (I know, there's a whole school of argument on that, but we've made the decision to use G.722 network-wide for any device that supports it, so it is what it is.)
All I want is for the device I've purchased on the premise that it will support "HD" audio (it has the logo on the handset, and the phone actually utilizes G.722 to the PBX), to actually support the "HD" audio I've paid for. Had it not advertised "HD" support, frankly I would probably have never bought the thing. If I want G.711 audio, I'll use an old phone and ATA, or I'll get a cheap Linksys/older Polycom IP phone. Would definitely be much cheaper than the S500 I have.
I won't even talk about how the retaining clips/tabs for the handset cord have already broken off the S500, and I've only had it a month. (If you ever have to replace the handset cord for any reason, you stand a good chance of breaking those trying to get it out -- seems they are very thin plastic.)
If I hear anything back from them/make any progress on this, I'll definitely post an update. I'd love to be able to use these phones, but if they aren't supporting what is advertised (and support refuses to fix it), I have a very hard time recommending this product to anyone. Mind you, if they'll fix it, I'll probably feel different, but that's just how I feel about it right now.
TL;DR - Sangoma S300/S500 phones claim to support "HD" audio, do not actually do so, when provided with factual proof that it isn't doing what it should, support feels like the lackluster audio quality is good enough for the price, and refuses to fix it.
EDIT: The first image is the output of the Sangoma S500 running in G.722, second image is output of the Cisco SPA508G running in G.722.
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Porting problems
Hey guys,
Hopefully I can get some advice on what to do concerning some porting issues I'm having with two voip companies. I started this port in July and it's now September and it doesn't seem like it's gonna get resolved anytime soon.
I'm trying to port multiple numbers from one carrier to another mainly to save some money as the winning carrier is much cheaper. After a month maybe I finally got a few of the numbers ported but I had to jump through many hoops to accomplish this. Lots of emails back and forth between the two companies. I realize porting multiple numbers may require dealing with different CLEC's and this was probably part of the problem. This was the first time that I can remember when porting numbers that I had to play go between with the two carriers. Usually all ports I ever initiated completed without any interaction from me.
Even so, the different CLEC's should all have the same personal info about me?
The last few numbers which still haven't gotten an FOC date is another matter.
After the winning carrier resubmitted the remaining numbers, the port was rejected again because the information did not match what was on file at Bandwidth.
That information that didn't match was apparently my name.
The losing carrier insists that the winning carrier is not putting in my correct name on the port request and that is why it is being rejected. The winning carrier insists that they are putting in my correct name.
I think that someone is really jerking me around and that this problem might not ever get resolved. I've already paid an extra month at the losing carrier for these numbers and it looks like I may have to pay for another month.
Any advice on what I can do at this point or should I just hunker down and wait for the two companies to figure out what they need to do to complete this port.
The names of the two companies are not really important. Let's just say they are two reputable companies that are discussed on here from time to time.
Thanks guys.................................
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[Anveo] procedure to set up voice mail
Hi. I'm a new Anveo user with a Personal Unlimited plan running on a zoiper softphone on my Samsung Note cellphone. I want to set up the voice mail for my new number and after poking around different web resources, I have come to the conclusion that I need a procedure with steps to get me going on this. I also did a search on this site but did not see anything pop up. (I confess I only scanned the filtered post titles and did not go through every post.) Does anyone know if there is a previous post with this info or if a step-by-step procedure exists to set up the voice mail? I also run a zoiper app on my laptop and need to set it up here too. Thanks in advance for any info or wayfinding.
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[Anveo] Anveo Direct termination rates lookup script
Attached is a simple Bourne shell script to look up Anveo Direct termination rates for a phone # prefix starting with country code.
Requires UNIX, Linux or Cygwin environment.
This is for people who are comfortable with a command line interface.
After copying the script file to your system, set execute permission on the file.
Invoke the script with no arguments to get a usage summary.
Download Anveo Direct's "All Routes" .all.csv termination rates file here:
http://www.anveodirect.com/prices/outbound
Examples:
To look up only prime routes:
% ./anvrates.sh 1646547 prime anveodirect.all.csv
To look up routes in all tiers, including unspecified tier:
% ./anvrates.sh 1646547 '.*' anveodirect.all.csv
Please don't ask me questions about UNIX regular expressions -- read the egrep manual page.
NOTE: Knowledge of UNIX regular expressions is not required for basic usage.
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[Equipment] ATA with FXO / obi110 or ?
I'm currently using an obi110 ATA that has an FXO port I use with my landline and configured to work with Asterisk.
My problem is that it seems to be a lemon. If it gets unplugged or loses power, pretty soon after powering back up it goes nuts on the network and you can't even access it. The only way I've found to fix it after that is to unplug/plug and then right away do a factory reset and restore my settings. Then it's usually fine until the next incident. Sometimes even a network outage will cause the problem.
I hate to get another obi110 after my experience with this one, especially since they've increased the prices and they're now going for $70 (I think I got my old one for $40).
Is there any other ATA with an FXO port that's as good as or better than the obi110 performance wise? I've got an old spa3000, but it gets a lot of echo.
I recently read about a new "HTek Unicorn 3112" ATA with an FXO port, but it's $89, and I haven't heard of anybody using it.
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Getting Cisco signed SSL certificates
Does anyone here provision Cisco/Linksys ATAs through HTTPS ? How did you get get your web server cetificates signed by them ?
I manage about 20-30 Linksys/Cisco ATAs (like SPA2102, PAP2T, WRP400, SPA1x2) and their configurations are all stored on a web server (encrypted). I would like to switch that to use HTTPS, but I need some certificates generated by Cisco.
I found this online app which I assume it might help me: https://webapps.cisco.com/software/edos/home but when I try "Signup for SPA" I get a message "Please contact edos-support@cisco.com to request access to tool".
I sent an email to that address, but nobody has replied in several days.
Does anyone know if that is the right place to obtain the SSL certificate ? I tried to ask in the Cisco forums but nobody there answered yet.
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[General] microsip registration issue
I am trying to run microsip http://www.microsip.org/ on Windows 10 as a un-privileged (non-admin) user. I do not see the page to register and even if I register as admin user, the software does not work for non-admin user. Is this normal? Can you suggest any other Windows open source softphones that can do command line dialing? I need something like "application.exe 408-555-1212" and the SIP client should dial the number.
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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] Buyers BEWARE of Grandstream "Enterprise Phones"
List of my issues and a story of my struggle with grandstream.
I have filed numerous bugs in the past 6 month for GXP 2170, none of which have been fixed yet even after 4 FW releases! Software support is very bad!
Helpdesk is useless, they say that we submitted the bug and there is nothing else they can do, not even provide an estimated resolution time.
Phone is still unstable and unpredictable in a real Enterprise environment (LLDP etc)
If you think Enterprise phones from grandstream are suitable for REAL Enterprise you are insane!
GXP2170 Phone looks appealing but once you get a chance to work with it you realize that all these features are pointless if they dont work properly. Lack of support and the amount of persistent bugs is staggering! Beware!!!
I am not going to go into too many details since a lot of issues have been covered in their forum already, so here is my quick list:
#1. LLDP is still not working. Apparently it was fixed in v.65 but I still see no change from before!
LLDP tables are not populated properly with the cisco switch, implementation of LLDP is still broken which renders the phone a useless door stopper. (Real Enterprise jewel)
#2. Cannot place an outgoing call or use the phone features if WAN connection to Internet is down FW v.1.0.7.65! Another Enterprise Feature
If phone is local to the PBX and the WAN is down, nobody can use the phone because it has a stupid INTERNET DOWN popup which locks the phone completely until connectivity is restored!
No testing on the software team part at all it seems!
#3. BLFs are broken too if you use Reserved (Red), Reserved (Green) and Inverse profiles.
They will not reset back to the original state once triggered - Bug#64206
Users are forced to stare all day at +10 Super Bright blinding LEDs because GS cannot fix this little bug in over 6 month? BLF states are messed up as well and not updating properly.
Another example of Enterprise stewardship! ;(
#4. Overall usability is unrefined, BLFs are hidden during the call or when dialing
BLF LEDs are not working when phone is in use as they are turned off and are not functioning. Really suitable feature for Enterprise environments!
#5. Build quality is hit and miss too, some phones have rattling handset noise inside, other have a loud speaker making funny noises (vibrational) when set to maximum volume.
#6. Noise Suppression on Bluetooth Hands Free makes it sound as if the call has dropped and the other party would constantly say "hello, are you there"
There should be a way to configure different levels for noise cancellation, otherwise this neat feature is useless as so many other things on this phone! Another useless Enterprise feature!
#7, 8, 9, 10
I can continue for hours but I have to go
Just read the forum...
At the end I am left with the Buyer's remorse, big headache and lots of lost time and productivity!
I wonder how GS is still in business and selling phones!
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Callcentric IP addresses
I need to use SIP/2.0 300 Multiple Choices to redirect SIP endpoints (each with valid Callcentric credentials entered into them) over to the Callcentric servers from my own server.
Unfortunatley, Cisco SPA112's used by the majority of my Customers seem unable to do a DNS lookup when they receive a 300 Multiple Choices response, so I need to pass them an IP address. My server does not support DNS SRV, so I have to hard code an IP address into the 300 Multiple Choices packet.
The problem is - which IP address? I don't understand the security at Callcentric since they beefed it up. Devices register with my SIP server and not with Callcentric, but I believe you are allowed to make calls without being registered as long as the credentials are correct.
My question: If I randomly pick a Callcentric IP address and insert it into my 300 Multiple Choices packets and redirect calls, should they in theory be accepted by Callcentric?
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[Equipment] New Panasonic dect SIP phone KX-TGP600
I just noticed Panasonic has a new dect SIP phone:
http://panasonic.net/pcc/products/sipphone/products/kx_tgp600/index.html
In addition to the usual dect handsets, it looks like they have a dect desk phone for it, also.
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How to deal with sip censorship in UAE?
In UAE SIP was already blocked along with every other VOIP service since many years. People have been using VPN to get access for quite sometime. However recently there has been a law passed threatening to heavily fine and jail those using VPN services which is really sad. People now feel scared to use VPN services. I wanted to ask this group what should be the best practice for those wanting to avoid the censorship for VOIP/SIP specifically in order to avoid getting caught.
If there are any tools available which one can use to setup a proxy server to obfuscate the traffic and to avoid getting detected by the DPI firewalls? Or are there any ReadyMade SIP services which offer level of obfuscation for SIP to avoid detection?
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[PBX] Can you help me transfer my asterisk PBX to a larger server?
Here is what I want to do:
1. Backup an existing asterisk PBX on a CaC VPS (OldVPS) to, say, my PCs hard disk.
2. Create a new, larger CaC VPS (NewVPS).
3. Install my backup on the NewVPS so that it is exactly like OldVPS, only larger.
I am not sure of which backup type to use to accomplish this: file backup or image backup.
Note: CaC = Cloud at Cost
Thanks,
Rob.
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[Anveo] line takes too long to pick up and start the audio prompt
Hi,
We use anveo for all our incoming calls.
When I try to call one of my anveo phone numbers, there is a 4-5 second delay from the moment anveo picks up to when the welcoming audio prompt begins.
Can we get rid of the delay?
anveo support was not helpful
see attached photo for the full call flow.
the audio prompt is part of the "extension" widget.
Also note that my office number is not ported into anveo, instead it forwards to the anveo number.
thanks
p.s. how do i get notifications on this thread to my email?
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