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(Asterisk / Trixbox) connect OpenSIPS

Posted on 2009-05-17
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Last Modified: 2013-12-21
I need settings my SIP server(OpenSIPS) can be make the outbound call to PSTN network via Asterisk Gateway
How to configure the Asterisk and OpenSIPS?

Many Thanks
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Question by:ycTIN
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Expert Comment

by:michofreiha
ID: 24407318
Dear Sir,

You are asking for a BIG question here and cannot help you like that..You should begin working on the installation and the configuration and let us know about issues that you are facing

Regards
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Author Comment

by:ycTIN
ID: 24408573
Hi michofreiha,

Thanks for your reply, my case is i has build up two server OpenSIPS and Asterisk(Trixbox), and they are working fine.

I known need settings some  route between OpenSIPS and Asterisk , but in google i only found the out dated information about OpenSER.

Best Regards,
ycTIN
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Expert Comment

by:michofreiha
ID: 24409816
Dear Sir,

Are you making any kind of load balancing between OpenSIPS servers and asterisk?Mean are you using carrierroute module for example on OpenSIPS?

regards
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3 Use Cases for Connected Systems

Our Dev teams are like yours. They’re continually cranking out code for new features/bugs fixes, testing, deploying, testing some more, responding to production monitoring events and more. It’s complex. So, we thought you’d like to see what’s working for us.

 
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Author Comment

by:ycTIN
ID: 24409975
No, i only need forward the call to PSTN network.

I don't care the performance and steps, because this just for study


SIP Phone <--> OpenSIPS <--> Asterisk <--> PSTN

SIP Phone connected in OpenSIPS,
make outbound call to PSTN via Asterisk
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Accepted Solution

by:
feptias earned 250 total points
ID: 24413436
Perhaps the answer is something like this (this is for guidance only and is well short of being a complete script):

if (method=="INVITE") {
    rewritehost("<ip_of_asterisk>");
    if (!t_relay()) {
        sl_reply_error();
    };
};

Before accepting the INVITE, you may want to authenticate the calling device using functions such as proxy_authorize and proxy_challenge.

You just need to configure Asterisk like this:
In sip.conf set a default context or specify the context for all calls from the OpenSIPS host address by defining a SIP peer with the correct values for context= and host=.
In extensions.conf, define a section for that context and put steps as follows (this is just an example):
[mycontext]
exten => _0.,1,Dial(DAHDI/g1/${EXTEN},30)

To explain:
_0. will accept any number starting with zero; You can use whatever pattern is needed for your PSTN target numbers. Another example would be _X. which would accept any dialled number. Dial(DAHDI/g1/${EXTEN} will place a call on a free channel in the first DAHDI group. Use Dial(ZAP/g1/${EXTEN} if you are using Zap drivers.
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Expert Comment

by:michofreiha
ID: 24414436
Do you need to match any prefix before sending call to asterisk server or you need to route all numbers to asterisk without checking the prefix?
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Author Comment

by:ycTIN
ID: 24437650
@feptias: many thanks for your reply and clear answer, i will try this late.

@michofreiha: my first goal is can forward the call, if can filter the prefix like PBX is perfect
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Assisted Solution

by:michofreiha
michofreiha earned 250 total points
ID: 24439719
You can use something like the following:
First you should scan the prefix if you have one like all numbers that begin with 00 as follow:

if($rU =~ "^00.*")
{
route(8);
exit;

}

route[8]
{
sethostport("Asterisk_IP");
        record_route();
            append_hf("P-hint: outbound\r\n");
             
}
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Featured Post

3 Use Cases for Connected Systems

Our Dev teams are like yours. They’re continually cranking out code for new features/bugs fixes, testing, deploying, testing some more, responding to production monitoring events and more. It’s complex. So, we thought you’d like to see what’s working for us.

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