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 Address are: 67.15.180.14 67.15.128.14 67.15.128.18 208.44.220.237 208.44.220.234 209.62.66.242 64.246.22.119 70.84.58.18 You should be able to receive calls from DIDX over sip or iax2. Asterisk Sample Configurations ** Sample sip.conf ** Sample extensions.conf ** Sample iax.conf To learn how to change Forwarding Settings Click Here To Learn more about Other DIDX Features visit www.didx.net/school

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. [general] 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=64.246.22.119 ; 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 allow=ulaw   <<< Required for DTMF allow=alaw   <<<< Required for DTMF 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 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 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.