Category Archive: VoIP

Sep 17 2016

OAuth 2.0 Support for Asterisk 13, so XMPP connections to Google Voice can be made using the more secure oAuth authentication

If you are running Asterisk 13 (or are ready to upgrade to Asterisk 13) and are using it to connect to one or more Google Voice accounts, you can now use oAuth authentication instead of the problematic username/password, without resorting to the use of a pre-built distribution that may contain features you don’t need and …

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Sep 07 2016

How to hack the FreePBX blacklist for better call blocking capability, take 2 – adding TrueCNAM scoring

This article is mostly a repeat of the article How to hack the FreePBX blacklist for better call blocking capability, the only difference being that this article adds the ability to use TrueCNAM to help determine if an incoming call is from a telemarketer or robo-caller. It’s not foolproof, of course, and some spam calls …

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May 26 2016

Read this before you pay $10 to Obihai for support

As you may be aware, I removed all of the Obihai-related articles on this blog in protest of Obihai’s decision to attempt to charge users an additional $10 support fee in order to get firmware updates, and I no longer recommend Obihai devices. However, if you previously purchased an Obihai and find that you need …

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Aug 01 2015

Install Elastix Unified Communication Server – Unixmen

Elastix is an unified communications server software that brings together IP PBX, email, IM, faxing and collaboration functionality. Source: Install Elastix Unified Communication Server – Unixmen

Jun 02 2015

Link to forum thread: Obihai sucks – buyers paid for Google Voice, now they want more!

Obihai now wants more money to give you the functionality you originally purchased! This was kind of buried in a previous thread but I thought it deserved its own thread. I have been a supporter of Obihai devices in the past, but no longer! I have an Obihai OBi100 and when I first got it …

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May 26 2015

How to use a custom SIP header to pass the DID (or other data) between two Asterisk servers that jointly handle extensions

If you have multiple Asterisk or FreePBX servers at different locations that pass Intra-Company traffic between each other using SIP trunks, you may have wished for a way to pass the Calling DID number (or some other bit of data stored in an Asterisk variable) from one server to another. For example, let’s suppose that …

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May 24 2015

Link: One Way Audio SIP Fix

The Problem When making audio calls using SIP the phone rings but when it is answered there is only one way audio or no way audio. What Cause One Way Audio The cause of one way audio is a combination of NAT and STUN (which we’ll come onto later). Let’s talk about NAT first. Full …

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May 12 2015

Forum thread: User-specific Caller ID/whitelist/blacklist in Asterisk/FreePBX

Occasionally I see a forum thread that contains enough interesting material that it could be the start of an article, but because it’s stretched through a number of posts, I can only link to the thread. And also, because I’m not the author of any of the original posts, I can’t just copy and paste …

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Apr 16 2015

Stop SOME SipVicious attacks from reaching your Asterisk, FreeSwitch, YATE, etc. PBX server

This tip was posted by user “infotek” on the FreePBX site but applies to all software PBX systems that use the iptables firewall. “infotek” wrote: By default the SipVicious scanner uses the ua : “friendly-scanner”. To block this ua, you can have iptables search the packet for that text. add the following line to /etc/sysconfig/iptables …

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Apr 15 2015

Link: Quick trick for Missing Forwards calls to Google Chat Option in Google Voice Account

Forwards calls to Google Chat is an important option for people who are using Google Voice as free VOIP solution. In order to able to receive calling, forwards calls to Google Chat option must be activated so the VOIP adapter can acts as a Google Chat to receive calling, eg: OBi Devices. Without that, the …

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