DIDX provides simple call forwarding Service, does not offer SIP or IAX2 accounts (PEERS) to register on our network.
Which means that you must allow DIDX to send you calls on your asterisk server from our IP Addresses.
Our IP Addresses:
| *IP Addresses | *DNS |
|---|---|
| 162.243.32.115 | sip4.didx.net |
| 142.93.9.107 | sip10.didx.net |
| 159.203.27.198 | ca.didx.net |
| 188.166.168.173 | eu2.didx.net |
| 46.101.80.13 | eu3.didx.net |
| 198.199.87.53 | us1.didx.net |
| 162.243.107.173 | us2.didx.net |
You should be able to receive calls from DIDX over sip or iax2
Asterisk Sample Configurations
Sample extensions.conf
Sample iax.conf
SIP.Conf Sample File Location: /etc/asterisk/sip.conf
Since the call is going to you over GENERAL Context, you will need to add the following lines to make your asterisk work with DIDX properly. Otherwise you will face errors and will think that DID is not working.
We will explain below why you need to add each particular line.
context=default ; This is very important, as this is where the call from DIDX will land to. If the context does not exist in your extensions.conf the call will not come to your asterisk, and will return “404 not found” to DIDX, very common error at our end.
port=5060 ; The port where DIDX sends the call to. For sending calls on a different port.
bindaddr=X.X.X.X ; Please bind to your main IP address that you are using.
srvlookup=yes ; Enable DNS SRV lookups on outbound calls
dtmfmode=rfc2833 ; If you need DTMF and you do not have this line, there may be errors in getting DTMF from DIDX.
relaxdtmf=no
disallow=all ; Disallow all codecs
allow=ulaw ; Required for DTMF
allow=alaw ; Required for DTMF
allow=g729
allow=gsm
maxexpirey=30
defaultexpirey=180
canreinvite=yes
nat=0
UserAgent= Asterisk
echocancel=yes
echocancelwhenbridge=yes
[1000] ;A Sip User – Nothing to do with DIDX
type=friend
username=12126559343
Notice that there is no registration information of ours on your server, because we do not require you to register any users / peers on our network. We use standard SIP and IAX2 forwarding.
[general]
bindport=4569
bindaddr=0.0.0.0 ; Your server ip address
jitterbuffer = yes
disallow=all
allow=alaw
allow=ulaw
dtmfmode = rfc2833 ; To get DTMF Properly from DIDX
context=Default ; This is where your call will land to if you do not
send it to a user IE
[asterisk@yourdomain.com/1111111111](mailto:asterisk@yourdomain.com/1111111111)
allow=all ; Codec which you want to use for DIDX
[guest]
Context=default ;Where you want the calls to go into from DIDX if u send it to [guest:guest@domain.com/12126555763] (mailto:guest:guest@domain.com/12126555763)
Disallow= all
Allow= ulaw;Codec on which calls will come to your asterisk server
dtmfmode= rfc2833 ; To get DTMF Properly from DIDX
Notice that there is no registration information of ours on your server, because we do not require you to register any users / peers on our network, we use standard SIP and IAX2 forwarding and the calls are going to land on your guest user, or you can land them to any other user of your own.
Extensions.conf has to know where the call you are getting from has to go to.
We are going to assume that you are using context didx where you want to send your calls to
exten = _X.,1,Dial(SIP/123456@fwd.pulver.com)
exten = _X.,2,Hangup
This will send all the calls to the freeworlddialup account number 123456
This is just a SAMPLE for you to go ahead and configure it properly.
Trouble Shooting your problems of call not coming to your asterisk
Most of the time we get request that the call is not going though, or voice is not coming on the did, this is why we give 2 FREE did’s so that before you attempt to buy anything, you can check the setup at your and our end, this helps us trouble shoot the problem.
Whenever you have a problem with any did number, you should first use the free did to test the same problem, because the problem can be at providers end also, but the free did’s are toughly tested before we give them to you.
Playing a MP3 file from your server:
Playing a MP3 file from your server will help you easily detect some of the errors, simply enter this code in your extensions.conf default contact defined in your general sip.conf section.
exten = radio,2,MP3Player(https://www.didx.net/jesus.mp3)
then send the call from didx to your server to radio@yourdomain.com or radio@yourip
This will play a song on the phone, and will show that the call is going through fine to your asterisk.
Support
We recommend you to visit About us, VOIP information
If you are still having problems, We suggest you contact an asterisk consultant, or best would be to contact www.Digium.com