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Emulate a T1 environment using two asterisk servers and cross over cable

Posted on 2007-12-06
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Hello Gurus!  I have a question about Asterisk.

We are planning on doing some development work using Asterisk server.  When completted the server  will use T1 lines coming into digium cards in the asterisk server.  My question is how can I simulate a T1 environment in my test environment for my devlelopers without having to actually bring in a T1?  T1's are expensive and I'd like to simulate the environment if possible.

What has been proposed is we set up two asterisk servers, each with a single port T1 card, and connect them via cross over cable.  Server A will be the development server from which all calls will be initiated.   Server A to initate a call out from it's T1 card and using the cross over cable Server B will recive the call and emulate the T1.  So server B is acting like the phone company.   We are only testing outbound calls from Server A to ServerB.

This seems doable in theory but has anybody done it practice?  Can anybody offer me a dial plan or some steps that will allow me accomplish this? I'm an asterisk newbie but follow directions well.   My hope is this will allow the developers an environment that will simulate production so when we do roll the server into production and hook it up to an actual T1 there wont be any surprises.

Not interested in VOIP suggestions just how to get the cross over to work and emulate a T1 environment.  
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Question by:codefaze
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by:grant300
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All you need to do is create an RJ45 T1 crossover cable.  Then you can connect the two T1 cards back-to-back.

WARNING:  The RJ45 T1 Crossover cable is NOT the same as an Ethernet crossover cable.  The Digium card user manual has the pinouts which will allow you to create the cable.

Regards,
Bill
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feptias earned 500 total points
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I have done this for E1 with the Digium card. (Probably the same model of Digium card, just with the jumpers in a different position for T1 or E1). My E1 cross-over leads have the following pin connections, but you should check in case T1 uses different pins:
Pin 1 to pin 4;   Pin 2 to pin 5;   Pin 4 to pin 1;   Pin 5 to pin 2

You must set the signalling parameter correctly in zapata.conf. Server B emulates the Network/CO so you should set signalling=pri_net. The development server, Server A, will need signalling=pri_cpe. There are other settings in zapata.conf that are important too, such as switchtype, but I don't know what you should use in your location. Check this link:
http://www.voip-info.org/wiki/index.php?page=Asterisk+config+zapata.conf

Although you say you don't want VoIP suggestions, you might consider configuring an IP phone/softphone to register with Server B and directing all your test calls to it in the dial plan. This will allow you to answer the calls, check that they are playing the expected announcement and also allow you to check what caller ID is being sent from server A.
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by:codefaze
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Thank you, this helps.  Do you have an example of an actual dial plan?
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by:feptias
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The dial plans - i.e. contexts in the file /etc/asterisk/extensions.conf - will depend on what you are using to initiate outbound calls from Server A and answer inbound calls on Server B. Will you be making calls from Server A using IP or analogue handsets (i.e. someone dialling the number) or are you using some kind of automated dialler? If the latter, what method do you plan to use? What do you plan to use to answer the calls on Server B - a softphone maybe?

Basically, Server A can call server B using something like this:
exten => _X.,1,Dial(ZAP/G1/${EXTEN},60)
 This will take whatever number you dial and forward the call to Zap Group 1 (your T1 circuit) with a 60 second timeout.

Server B would answer it using something like this:
exten => _X.,1,Dial(SIP/101,40)
 or
exten => _X.,1,Dial(ZAP/25,40)
 (The first to call a SIP softphone configured as extension 101, the second to call an analogue phone that appears as channel 25 based on the assumption that channels 1-24 are your T1 circuit).

While setting up, always use the command "set verbose 3" from the Asterisk CLI (or start Asterisk with asterisk -vvvc from the Linux command prompt) because this will give you step by step output as the lines in your dial plan are executed. It makes it much easier to see if your plan is working.

It is hard to give you all the details required to get started with Asterisk. There is too much to know. You should be able to find some online tutorials or you could buy a book. If you have a specific question feel free to post another comment here or, better, ask a new question in the EE Asterisk topic area.
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by:codefaze
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Thank you, this is helpful.  I can use this to get myselft started.
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