DIDX – Sample Configurations for Asterisk
( These are some of the configurations we have found to work best. )
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:
67.15.128.14 - sip1.didx.net
67.15.128.18 - sip2.didx.net
209.62.66.242 - sip3.didx.net
198.101.50.4 - sip4.didx.net
198.101.50.2 - sip5.didx.net
74.55.75.30 - sip6.didx.net
174.133.195.194 – sip7.didx.org
74.52.4.234 - sip9.didx.net
88.208.247.34 – eu1.didx.org
88.208.208.34 – eu2.didx.org
67.19.199.170 – sip1.virtualphoneline.com
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 |
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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.
[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.
IAX.conf /etc/asterisk/iax.conf
[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.
Extensions.conf /etc/asterisk/extensions.conf
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
[didx]
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,1,Answer
exten => radio,2,MP3Player(http://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 though fine to your asterisk.
Support
We recommend you to visit
About Us
http://www.voip-info .org
If you are still having problems, We suggest you contact an asterisk consultant, or best would be to contact www.Digium .com
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Michael Robertson, CEO of Sipphone (the first IP communications company to join DIDX) and also Mp3, shares thoughts on the success of DIDX
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Humberto Correa, formerly CEO of FonicaPBX, and now with BBCOM CLEC as well as Alan Pesatty, business consultant, discuss in Spanish the benefits of telecoms using DIDX to buy and sell phone numbers.
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John Lodden, CEO of Michigan Networks, explains the advantages of selling DID on DIDX during a Cluecon conference.
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Jay Xia and Rehan Allahwala discuss how DIDX providesDID phone numbers from around the world to telephone companies around the world, signup for a free trial account now.
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Praise shared by Hunter Newby of telx and Suzanne Bowen of Super Technologies Inc. and DIDX in a video interview
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ITEXPO East 2007 – Ft. Lauderdale, FL – Jan. 23 – 26, 2007 and other World-Famous Telephony Conference
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Douglas Kimani, DIDX member shares the need for DIDX among global service providers in English and Swahili.
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Worldworkz at Spring VON 2007
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Connectbynet at Spring VON 2007
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Trixbox at Spring VON 2007
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Rich Tehrani of TMC Internet Telephony
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DS2 Jose Lucini at NXTCOMM 2007
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Richard Koch, RNK Communications at Comptel 2007
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Efrain Betancourt and Danny Argudo of DEG Telecom at GTM 2007
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CallThePlanet’s CTO Steven Harrison at EXPOcomm 2008 in Mexico City
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Serge Kruppa of Peerant and Jessica Ruiz of DIDX at EXPOcomm 2008 in Mexico City
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Luis Alberto Moreno (DIDX Partner) of Bas Computers at EXPOcomm 2008 in Mexico City
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Alexandro Apan NeoCenter on Call Centers, DIDX, EXPOcomm
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