Note: This guide was written for Asterisk 1.6. While most of the content still applies, newer versions of Asterisk and FreePBX may work differently than described here.

Learn how to set up your MegaPath SIP Trunking service with IP-PBX vendor FreePBX. Conduct And Practices Handbook Course Cph Pdf To Word. Warning: Invalid argument supplied for foreach() in /srv/users/serverpilot/apps/jujaitaly/public/index.php on line 447. Use a pc on your network that has a web browser and connect to your Trixbox box using HTTP://PutYourTrixboxIpaddressHere. Click on the Asterisk menu. Click on Config Edit; Click on sip.conf; Make the following changes: [general].

Update Feb 10, 2015: I realized Asterisk 1.6 doesn't support insecure=very, article has been changed to reflect this. Contents • • • • • 1. Prerequisites Ok, for this guide we'll need to cover some basic requirements: • A functioning Asterisk server with FreePBX. We'll be using trixbox 2.8 running Asterisk 1.6. • SIP trunk info from a SIP provider. We'll be using Broadvoice. • An extension assigned to an IP Phone.

We'll be using extension 2000. To keep you hydrated and thinking clearly. Set up the SIP trunk Now, with the above covered, let's get started! Open up a web browser and go to your Asterisk server web interface. In our case we'll go to Now, on the left, under Basic click Trunks. You should see a selection of trunk types, like this: Click Add SIP Trunk.

We are now greeted with a page that we must fill in with our trunk info. In the Outbound Caller ID field, you can enter a caller ID, but it may not do anything. So, we'll skip this field.

We'll also leave the Never Override CallerID unchecked. For the Maximum Channels field, we'll put in 1. This is because the plan we are using in this guide only allows 1 incoming call at a time.

Leave the Disable Trunk and Monitor Trunk Failures at their defaults and go down to Dial Rules under Outgoing Dial Rules This is where the phone number gets 'conditioned' before it gets sent to the SIP servers. For this guide, we'll use dialing rules to condition numbers for US 10-digit dialing.

Example are these dial rules. 1+NXXNXXXXXX 1912+NXXXXXX Let's examine what these mean: We'll start with the first one.

( 1+NXXNXXXXXX) 1+ means prepends '1' to the number. N means match any number between 2 and 9. X means match any number between 0 and 9. This would match a number like, say 912-555-1234 and turn it into 1-912-555-1234 before sending it to the SIP servers.

So, the next one ( 1912+NXXXXXX) goes like this: 1912+ prepends '1912' to the number. N matches any number between 2 and 9. X matches any number between 0 and 9. So then this one would match a number like, 555-1234 and turn it into 1-912-555-1234 before sending it to the SIP servers. See, so far it isn't so bad.

But, then again, we're not finished yet either.:-) The Outbound Dial Prefix field prefixes a number to all numbers dialed through this trunk. For what we're doing, we will leave it blank. Although, like the help says, if this is a trunk to another Asterisk server or a Centrex line, you many need to put '9' in this box to access an outside line. Under Outgoing Settings, we see the field Trunk Name.

This is for, well, the trunk name. We'll put 'Broadvoice' in this box. Cmos Pld Programming Hardware And Software Support more.

Now, here comes one of the trickiest parts of setting up a SIP trunk, the peer details(settings). These are to tell Asterisk how to connect to the SIP provider. Here is an example set of settings with descriptions of each one: disallow=all This should always come before any allow directives allow=ulaw This allows use of the G.711 u-law codec. Most SIP providers support this codec. Context=from-trunk This is the context that Asterisk will dump calls coming from the trunk into this dialplan context. Without this set to a proper context, incoming calls will not work. Dtmfmode=auto This tells Asterisk how to interpert DTMF tones.

It can be auto(auto-negotiates what mode to use - recommended), inband(sends DTMF as sounds in the audio stream), rfc2833(recommended if your SIP provider supports it), or info(DTMF info is sent inside the SIP header) fromdomain=sip.broadvoice.com This tells the SIP provider what domain the call comes from. Some SIP providers require this for authentication.

Fromuser= This is the username to authenticate to the SIP provider with. Host=sip.broadvoice.com This is the host to connect with to send calls insecure=port,invite This determines if Asterisk should authenticate calls coming in.