FreePbx Configuration Guide
©© If you are looking to install an Asterisk PBX server, follow our tutorial here, and then come back here to configure it with SipCel.
To configure SipCel with your existing FreePbx installation, we have to do the following:
- login into your admin panel in the web gui, and go to Setup/ trunks/ add SIP Trunk, then we need to add the following details:
- Trunk Name: SipCel
- Outbound Caller ID: –Leave it empty, cause you may setup the caller ID in extensions configuration, it’s more better, and personalised, unless you want to turn into absolute caller ID for all over your extentions, so in that case you can setup this here.
- Maximum channels: leave empty
Then we need to go to Outgoing settings, and:
- Trunk Name: SipCel
- Peer details:
Option 1: if your PBX server have statical IP (Fix IP), you may only setup the following:
Option 2: If your PBX server have dynamic IP (changeble, and NO fix IP), then you may login into asterisk server by using the following details:
username=Your Sip username, whcih you can find it under the tab Sip/Iax in your web pannel, it’s NOT your web login for the customer portal
secret=Your Sip secret, whcih you can find it under the tab Sip/Iax in your web pannel, it’s NOT your web password for the customer portal
host=sip.sipcel.com ==> Or if you’re in Europe, you can use sip.sipcel.eu
fromuser=Set same as your username, above
context=sipcel ; or change for proper context
trustrpid = yes
sendrpid = yes
canreinvite = no ==> If your Asterisk Server is v. 1.8 or later, and you have turn on directmedia in your sip.conf, you may change this entry to directmedia=yes, instead.
;;inf you face incoming calls failing, or interconnection errors caused by your firewall, you may configure your firewall to read your asterisk configuration, or just add the directive insecure, as: insecure=port,invite
Then if you have a DID number, or your expect to receive calls from SipCel, you may setup the incoming service also, so, in Incoming Settings we may do the following:
User context: your username, which you use above
Then, in the text field, we have to add the following:
secret=your sip password which you use above
Then the string configuration, if you’re using dynamic IP, you need to setup the string for your incoming calls, if you’re using statical (fix IP), you don’t need it, so just skyp this.
To setup the string we have to do as: username:secret@host; so:
Then submit changes.
Reload the server from the red bar in the top, and then we need to setup the incoming calls, if we’re expecting to recieve calls from SipCel. So we go to Setup/Inbound Routes in the left panel, and then select Add incoming Route.
If the DID which you’re expecting to receive calls are only one, or you don’t have any other DID more in your box, so you just need to allow whatever incoming calls by doing the following:
Description: whatever you want to use, for your own fun
DID Number: _. ==> Which this manner, all incoming calls will be accepted and routed. If you have several numbers, so we may need to introduce the number as it’s coming from SipCel. To know how it’s going to be sent from SipCel go to your customer portal, then under the DID tab look into the Destination row, you will see, if DID is destinated to your SIP/Account_ID, then you’ll recieve what you can read in the DID field. If you see for example Sip/your_did_number_or_somethingelse@your_host; then you’ll receive what you have before the at @. You can edit that yourself on your own fun, but it will be waiting for SipCel Network Operations Center approval, as it may not be correct, so your DID when it’s edited may not going to handle calls more, until approval, if you need it faster, you can rase a ticket for fastest review.
For example, to send the caller identification number to the sip server in the SIP destination box use the format SIP/nnnnnnnn@sipserver, where nnnnnn is the DID number which you have declared in your Pbx and sipserver is the IP address or FQDN of the Pbx server.
Setting your outgoing calls:
Go to Setup/ outbound routes, and then add route.
If you want to send all your calls to SipCel, just put the route name as you like for your fun, and in Dial Patterns that will use this Route, in the match pattern route put X., then in trunk sequence put SIpCel in the 0 option, as default, or whatever you like, save and reload.
That’s all, insure that you’re connected and registered properly by calling to 123 from your extension, so you may hear SipCel echo testing message, if so, all it’s working properly.
If you need any further assistance, you’re welcome to raise a ticket from your customer panel.