Despite Gigaset taking down the English version of their website (and pulling out of the North American market?), there seems to be a firmware for the Gigaset C610A IP phones (and possibly others). The new firmware version is 42.231 (422310000000 / V42.00).
I am not able to find any information about it, but I decided to install it anyway. I downloaded it through the web interface of the phone. Nothing appears to have changed visually.
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[Equipment] New firmware for Gigaset C610A IP
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[Asterisk] [How to] log RTP stats at the end of a call
For some time, I have wanted to log RTP stats at the end of every call, to assist in troubleshooting. rtpqos sometimes works, but as Trev mentioned in the other thread, its usefulness is limited. Sometimes it returns zero for everything, and I haven't been able to correct that.
Fortunately, my phone includes RTP stats in a special X-RTP-Stat header that it sends to Asterisk at the end of the call. It looks like this:
PS=211,OS=36292,PR=212,OR=36464,PL=0,JI=0,DU=4,EN=G711U,DE=G711U
Since my Asterisk PBX is configured not to proxy audio, these figures should be accurate. Many phones and ATAs do a similar thing, sometimes with a different header name. The trick is accessing it. If the far end hangs up first, the h extension runs before my phone sends the SIP packet with the X-RTP-Stat header.
A reliable way to access this SIP header is with the Dial options F (when caller hangs up, transfer called party), U (execute Gosub on called channel), g (continue with the dial plan after called party hangs up), and applying hangup handlers to the near side of the call.
[rtp-stats-test]exten => inbound,1,NoOp(Simulated inbound call) same => n,Dial(SIP/DeskPhone,,F(other-side-hung-up-first,1)U(add-hangup-for-other-side)) exten => outbound,1,NoOp(Simulated outbound call) same => n,Set(CHANNEL(hangup_handler_push)=hangup,1) same => n,Dial(SIP/some-external-destination,,g) same => n,Goto(other-side-hung-up-first,1) exten => other-side-hung-up-first,1,NoOp(Allow time for the near end to hang up.) same => n,Wait(15) same => n,Playtones(500*100/250,0/250) same => n,Wait(60) exten => hangup,1,NoOp(${SIP_HEADER(X-RTP-Stat)}) [add-hangup-for-other-side]exten => s,1,Set(CHANNEL(hangup_handler_push)=rtp-stats-test,hangup,1)
The above code works for both inbound and outbound calls, regardless of which end hangs up first.
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Getting busy signal calling any toll-free number on Vestalink
I'm getting a busy signal when I dial any toll-free number on my Vestalink trunk but they all connect OK through my Callcentric trunk. Did Vestalink suddenly stop carrying toll-free numbers? Non-toll-free numbers go through OK on my Vestalink trunk.
I first became aware of this when my mom's Philips Lifeline medical alert device kept failing to connect to an agent when my mom pressed the button on the remote unit she wears like a necklace. The Philips Lifeline device calls a toll-fee number. I config'ed my Obi110 to send the Philips Lifeline numbers over the Callcentric trunk instead of the Vestalink trunk and that resolved the issue.
Anyone else here who is still using Vestlink experience this issue, or is it just me?
Now I just gotta remember how to program my Obi110 once again to send all toll-free numbers over the Callcentric trunk instead of the Vestalink trunk.
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CallCentric Setup with FreePBX
So I have had nothing but headaches with this. So now I finally got my main incoming only number setup BUT when it comes to incoming routes what is my exact DID? I've tried everything but Im only able to have 1 incoming line. When I add a 2nd and want the 2nd line to go to an different Incoming Route thats where my problem lies as if I set the DID under incoming route to anything other than "ANY" FreePBX gives both numbers when calling a Number not in service error.
Pretty much things work fine with 1 or 2 trunks if my Incoming Route is set to "ANY" did. But then both numbers ring the same extension. Theres no way I can get 1 ring 1 ext and 2nd ring a 2nd ext due to not knowing the correct DID that CallCentric is using
I am using 1777xxxxx100 for first and 1777xxxx200 as example for my DID and its not working.
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advice for new voip system
hope this is the right forum for what im looking for...if not if anyone has a rec on some other forums that might be able to help me. thx
kind of a newbie to the world of voip and doing my best to learn as much as possible. i have a current voip system with my own pbx and at&t circuit. not sure what i have but i know they are ripping me off and want to leave them. my business model is an answering service for clients that forward their calls to me when they leave for the day. i have 100 phone numbers with one assigned to each client so that when they forward to their assigned number i know how to answer the calls that come in. most of the time its just one to two calls at a time coming in with lots of spare time in between calls.
im looking for advice on which way i should go with a new system. i currently have 2 phones setup to receive calls and of course im limited to being at the one particular office with the pbx.
what i think i want is a system that would allow me to setup a phone at each of my 3 locations which are several hours apart. when calls come in i want all 3 to ring as if they are in the same office. that way i dont have to keep someone at my main location 24/7 to catch calls. i can utilize employees at the other 2 locations can catch the calls as well
i was quoted a virtual pbx from 8x8 with transferring my numbers from at&t. they said they set up the account and i just plug each of the phones into my routers at each location for internet access and boom they all work. is 8x8 considered a good voip company?
is this my best method to pull off what im trying to do? im looking for good call quality and reliability while being cost conscience on my monthly charges. btw all 3 locations have good low ping high speed cable internet connections.
if its really cost effective i would consider putting in a voip in a box system at one of the locations if it could feed the other 2 but not sure how that works and what kind of cost and recurring monthly cost im looking at. (i am a network/computer engineer by trade just never used or learned voip systems till now)
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Numerous Calls With Caller ID 1004 / 1002 / 11100
Hi,
Starting Saturday night we have received tens of calls with a caller ID of 1002 / 1004 / 11100. When we answer there is no audio. Any idea of what it is and what we can do about it.
Thanks
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Google Voice OAuth2 tokens expire
On Google Voice Gateway I am seeing users' OAuth2 refresh tokens occasionally expiring. This most often happens when the user changes his Google password. The philosophy behind this is debatable -- I believe tokens should remain valid regardless of password change -- but it is what Google has chosen to implement.
There seem to be other cases not explained by password change. Searching developer forums, I see a lot of virtual shrugs and inquiries to Google about it.
Expiring tokens will affect Google Voice Gateway accounts, Asterisk integrations using the patched OAuth2 XMPP library, and Obi devices. The solution is to reauthorize the application. On Google Voice Gateway, log in at https://simonics.com/gw and use the Reauthorize link to do so.
I'm working on an e-mail notification that will let users know when their GVGW tokens quit working.
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H323 helper One way audio over OpenVPN
I have an Avaya 4602 IP phone which uses H323 protocol, which is connected to Tomato 1.28 Shibby firmware. Router has an OpenVPN connection to VoIP server.
The phone appears to register fine with the VoIP server with H323 helper enabled (RTSP + SIP helpers disabled on the Tomato router) .
Problem I'm having is that callers can hear me, but I cannot hear them. Nor can I hear a dial tone when picking up the handset , but I can establish a call (albeit without audio).
I installed tcpdump on the Tomato router and monitored the Br0 interface , and noticed that RTP audio stream was being transmitted from the VoIP phone (192.168.1.199) to the VoIP server (192.168.0.4), however I couldn't see a RTP stream from the VoIP server back to the VoIP phone (i.e. 192.168.0.4 > 192.168.1.199)
I have had this behaviour before with our Linux routers (EdgeMax), and believe the issue was resolved by modifying INDIRECT-MEDIA and INDIRECT-SIGNALLING which is linked to the SIP Conntrack module.
I have tried various different combinations of switching off/on the helpers, none of the combinations appear to fix this issue.
My Linux knowledge is very limited, and wanted to know the following;
1. Does anyone know if this Tomato firmware has INDIRECT-MEDIA and INDIRECT-SIGNALLING parameters that can be enabled / disabled? if so how?
2. Is there a way to inspect/modify the conntrack module configuration? I have sniffed around using SSH and couldn't find config files.
3. Does anyone have any other ideas how to fix this?
I have spend several hours trying to troubleshoot without any success.
thanks
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[Asterisk] from V1.6 to 11 to 14
Not so important but out of curiosity, mostly. I have been using an extension registered to my home run asterisk from outside. Under the same router conditions, it works perfectly well under the older version (Ubuntu 10.04), half-works under V 11 (ubuntu 14.04), it does not work at all under V14 (ubuntu 16.04). I do not recall making any changes to linux settings, such as IP tables, for example. Extensions within my LAN work as expected routing/accepting calls through/from SIP providers as well as Google through Motif (for the newest versions). Any ideas why the different behaviour?
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[Unlock] New Grandstream ATA used by Vonage
Looks like Vonage uses a new Grandstream ATA for their service, model HT802.
It's Grandstream newest generation of ATAs, with 2 ports.
Currently available at Walmart and BestBuy for $10.
If unlockable, this could be a nice, inexpensive 2 port ATA. Interesting that they are using a MicroUSB port for power.
I don't have a way of getting any of these soon to test as I am located in Canada. However, I am sure there are similarities between unlocking those and the BasicTalk HT701 (though probably they will not accept Mackey's firmware without some modifications).
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[Equipment] Linksys SPA942
Hi,
as many people before me, I purchased used SPA942 only to find out it's password protected. It's linked to telio.no by provisioning config.
In order to configure lines, admin password must be known, so for now the 942 is more of a paperweight.
Here's what I've been able to track:
Default config (non-telio) has custom admin password, however upon performing factory reset, within 30 seconds, telio's config is downloaded and the phone restarts. After that the admin password is changed to "1111" which lets me to configure profile rules BUT admin account is disabled in webconfig (returns 403 - Forbidden). In non-telio configuration, webconfig is accessible for admin, but the password is different, and likely not numeric, as I've tried running hydra's brute-force against 942's /admin http-get form with 1000000 combinations to no avail.
The phone tries to access several DNS names after factory reset:
ns1.telio.no
ns2.telio.no
ns3.telio.no
ntp.telio.no
dns1.as2116.no
prov.telio.no
port-prov-01.telio.no
However closer monitoring with wireshark didn't reveal any http packets going outside to prov.telio.no. I've tried redirecting all these DNS names to my server, running tftpd32, but nothing showed in the log.
Could someone please advice me how to perform reset of admin's password so I could (sic) finally access webadmin?
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[Voip.ms] VoIP.ms - 911 has been dialed
My wife hit a buck head-on today at 74mph so I called for a tow and called 911 (first time ever) to alert the highway patrol. She is fine and OBi/VoIP.ms 911 worked (I asked them to confirm my address). The deer is dead and the Volvo looks totaled. Here is the e-mail notification VoIP.ms sends (if configured):
****
Dear Customer,
This is a courtesy message from VoIP.ms
911 has been dialed from your account "xxxxxx_xxx-H2"
Date and time: 2016-11-16 at 18:42:23
Caller ID: xxxxxxxxxx
Do not reply to this email, we will not receive your message. Email our support directly if you have questions regarding this email.
Regards,
VoIP.ms
****
Good to know.
OE
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[General] Cloud at cost CS offline?
About a week ago, I noticed that I cannot access one of my server instance from the control panel. I submitted a ticket but never got any reply. Yesterday, I noticed that I cannot access the server by SSH anymore, I updated my ticket and sent an email to the CS department. No action has been take so far. Anybody know if there's anyway to contact them directly or their CS does not work anymore?
I know I get what I paid for but still ....
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[Voip.ms] OBIHAI and Alarm Monitoring Center
Hello, looking for some help on figuring this out. I'm looking to replace my landline with VOIP for the Alarm (Vista 15p).
The OBI200 is setup with VOIP.ms and Voip.ms is setup with g711 codec enabled only & I tried several DTMF options but no luck. From a hardware perspective, I got an ethernet coupler that connects the Alarm and OBI200 device.
I ran several tests and found that calls are being made but not being picked up I guess or there is no acknowledgement.
We are planning to dedicate the obi200 only for the alarm.
Any thoughts would be greatly appreciated.
Disclaimer: Well aware of using voip for Alarm.
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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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[Equipment] Build your own Digital PBX under US $6 using a POGO-V4-A1-01
I just posted this post and decide if you will take advantage of the offer to build your choice of a digital PBX under US $6.
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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
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
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[General] CRTC signs agreement with U.S. regulator to combat robocalls and
A good start:
http://news.gc.ca/web/article-en.do?nid=1155379
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PSTN: choosing an area code for optimal routing/reliability
I fully realize this *probably* doesn't really matter in the grand scheme of things, but what the heck.
I will be obtaining a new DID in the near future that truly doesn't matter where it is.
At this point, I'm looking for whatever is the most likely to remain up and stable, no matter what happens, whether it's massive flooding, a cavalcade of tornadoes, a rash of earthquakes, or nuclear attacks.
I have a passing familiarity with the PSTN and how call switching is handled, but I don't know a lot of the intricacies of how the network is set up. I do gather that it's a bit less "self-healing" than, say, a full IP set-up with BGP: a number is (more or less) tied to a specific wire center or rate center (unless it is explicitly moved by LNP).
As well, while PSTN latency is virtually negligible, photons and electrons still do have a speed limit, so why not minimize that effect as well? I assume that a number in a rural location has to go through more switches and layers, whereas a number in a major city will be topologically "close" to major "class 1" switching centers--much the same as a server in a datacenter with a direct connection to 111 8th Ave in NYC or 350 E Cermack in Chicago or 1 Wilshire in Los Angeles is going to have a better chance of lower latency and avoiding congestion than a server located in Pierre, SD. That shorter distance also reduces the risk of a physical issue (such as a line cut) causing a problem; Fairbanks, AK was actually cut off from the outside world for a few days when I lived there due to a flood washing out the railroad tracks along which GCI's primary fiber-optic line was laid.
Anyway, so I'm just wondering--even if it really in practice doesn't really matter--where I should choose a number to be based to optimize its reliability and its geographical centrality. Here are my thoughts:
New York City: clearly a major nexus of telecommunications, both IP and PSTN. Some risk of infrastructure attacks and weather but a generally stable power grid and a major priority for the restoration of services if something does happen due to its key standing.
Northern Virginia (Ashburn, etc.): probably one of the best-connected centers for IP transit (it is where MAE-East was!) with relatively low risk of natural disasters, but that doesn't mean it's a PSTN nexus
Los Angeles: weather basically a non-issue and another major access point on the national data and telecommunications networks. Some risk of geological instability.
Chicago: Probably the single biggest crossroads of connectivity in the country and reasonably safe from natural disasters and infrastructure-disabling weather.
Dallas: Another major crossroads of connectivity in the country but some potential for disabling weather events (tornados, ice storms, flooding).
On the other hand, Wikipedia's article on the PSTN's topology (https://en.wikipedia.org/wiki/PSTN_network_topology) says that the nine Regional Centers (Class 1 switches) were in:
White Plains, NY
Wayne, PA
Pittsburgh, PA
Norway, IL (nuclear-hardened)
Conyers, GA
St. Louis, MO
Dallas, TX
Denver, CO
Sacramento, CA
That said, I don't know if all of those still function as Class 1 switches or if Class 1 switches are even a thing anymore--but it does make sense that at least some of those are still serving as sort of "central routing switches" for the PSTN even today.
I think at this point I would lean towards either Chicago or Dallas. Both are absolutely massive centers of data transit and both are listed as having a heritage of high access on the PSTN (well, Chicago's was a bit out west in Norway) with probably many, many different redundant routes (and likely good for a PSTN-IP cross-connect, if need be). Both are also relatively isolated from potential disabling disasters, with Dallas being perhaps a bit better connected to the west coast but Chicago being better connected to the east coast (http://www.webhostingtalk.com/showthread.php?t=969940).
On the other hand, my current hometown of Pittsburgh actually ranks as one of the safest major metros from natural disasters (http://www.bestplaces.net/docs/studies/safest_places_from_natural_disasters.aspx) and is (or was) home to one of the nine regional Class 1 switches, but its days of being a center of connectivity seem to be over; early Internet hosting pioneer Pair Networks hasn't even been successful at getting an IXP going there, and the bulk of my 100mbps FiOS data goes to Ashburn before going elsewhere.
A minor consideration is termination rates; I haven't fully analyzed all the rate tables, but some VoIP providers like Anveo Direct do discriminate not only by area code but even prefix for calling rates, varying costs from dirt-cheap (some rural wire centers, where costs may be barely a cent a minute) all the way down to basically free (some urban mobile numbers, where it's sometimes less than a hundredth of a penny per minute). My assumption is that highly-connected urban wire centers may be cheaper to call than far-flung rural mountain wire centers connected only by 200 miles of microwave repeaters or something.
Anyway, I thought this might be an interesting subject to a few people here and look forward to any insights others can share.
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CallCentric for small biz
I've already sent an inquiry to them online about this, but figured I'd ask here as well. Does CallCentric offer unlimited outbound calling for small businesses?
I have first-hand experience with CallCentric and would like this small business to use them. But their site only offers an unlimited outbound plan for residential customers.
Does CallCentric allow CID spoofing? I can always configure all their outbound calls to show their main number, but if they'll allow passing individual DID's so the remote caller sees it as a direct number instead of the general main number, it would be ideal.
TIA
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