Asterisk OpenSource PBX
I’m testing an interesting product this week from a company called Andrews & Arnold. It’s a way of using a UK GSM mobile phone with FreePBX/Asterisk that I’ve not seen before. The service is a PAYG “SIM only” mobile phone service. You get a SIM card to go in your existing mobile phone, which must be either unlocked or on the O2 network –
One of the easiest ways to listen to new voicemails is to get Asterisk/FreePBX to e-mail them to you as an attachment. One downside of this is that if you try to listen to them on your mobile phone they will play via the speaker. This makes it tricky to listen to them discreetly! For Android there’s a little App called Earpiece that allows you to specify that media is played through the earpiece, rather than the speaker. This is quick to switch on temporarily while listening to the voicemail. It may be hardware dependent, but it works fine on my OnePlus One….
As an admin for a telephone system, possibly one of the most useful things you can do is monitoring your peers and trunks. There are 2 great reasons you should do so: 1. You can respond to and resolve issues with your system before your users know about it, and you can be in the know if someone reports “none of the phones are working” when in fact only 1 or 2 are not working 2. You can actually know when there is a problem with the system – where you otherwise might not know there is a problem until someone…
Toggl is a hosted time tracking system, used to keep track of time spent of different projects. If you use Asterisk/FreePBX and spend time on the phone to clients that should be assigned to a project you can automatically create time entries in Toggl using their API. To do this we can make a curl call from Asterisk when a call ends. To be able to do this you’ll need to ensure you have curl compiled in to Asterisk. You can check this with “core show function CURL”. I tested this on Asterisk 11 and FreePBX v2.11, but it should…
I was recently talking to someone that was looking at running Capsule CRM. In Capsule you can create a “Task” which is something that has happened or will happen. These “Tasks” can then be assigned to a customer or contact. Capsule has an API that allows automating posting of these tasks and we can use CURL in Asterisk to post the call details.This guide was written using Asterisk 11. If this doesn’t work for you ensure you have curl compiled in to Asterisk with “core show function CURL”. First we need to tell Asterisk to run a command when a…
Getting some routers working with VOIP can be a pain. The problem is caused by NAT, and the translation of a private IP address (10.0.0.0/8 or 172.16.0.0/12 or 192.168.0.0/16) in to a IP address that works on the internet. This is difficult with SIP/VOIP because IP addresses are not only included in the packet header, but also inside the packet itself. There are various methods that can be used to try and resolve this issue, and various places that ‘fixes’ can try and work. The issue is this … a local device (phone) sends a SIP packet to a device on the internet…
Setting up a trunk to a SIP call provider in Asterisk can be a pain. Even for me – and I do it a lot! When we set up a SIP call provider in Asterisk (a SIP trunk) and send calls to them our Asterisk server will send the call provider a SIP INVITE. Their system could respond in many different ways – decline the call, ask for user info, silently drop the call, process the call …. Sometimes the call provider will send back information in the SIP reply as to why the call has failed. Maybe you have…
For a while now SysAdminMan has been offering FreePBX/A2Billing hosting with OpenVPN server already installed on the server. What I really want to find is the perfect client/router that’s simple to configure and easy to deploy. We’ve been recommending OpenWRT for a while now but it can be a pain to flash the firmware and get OpenVPN configured. I’ve also used Microtik routers for a while and they are very powerful routers in such a small, reasonably priced, package. This test was done using a Microtik RB750GL I wasn’t sure how it would work though as Microtik routers only support…
I had a customer today that was struggling to get a phone to register on a server with Asterisk 11.5.1 installed, even though it would register OK on a server with Asterisk 11.3 The phone was behind a NAT firewall, with the Asterisk server on a public IP address. Looking at the SIP packets they were coming from port 50758 – — SIP read from UDP:XX.XX.XX.XX:50758 — but when Asterisk replied it was sending the reply to port 5062 – — Transmitting (no NAT) to XX.XX.XX.XX:5062 — Now with either – nat=yes or – nat=force_rport,comedia on the extension Asterisk should…
I had a customer let me know that they had improved their call quality from WiFi and 3G connections by turning on the Asterisk jitter buffers for SIP connections. If you have any extensions where connection quality is intermittent it could be worth trying. This can by done with the FreePBX SIP Settings module or by adding the following lines to – /etc/asterisk/sip_general_custom.conf jbenable=yes jbimpl=adaptive After changing you should restart Asterisk or ‘sip reload’ from the console. If you are just using Asterisk the change would go in the [general] section of sip.conf If you try this option and…