(Asterisk / Trixbox) connect OpenSIPS

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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feptiasConnect With a Mentor Commented:
Perhaps the answer is something like this (this is for guidance only and is well short of being a complete script):

if (method=="INVITE") {
    if (!t_relay()) {

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):
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.
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

ycTINAuthor Commented:
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,
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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?

ycTINAuthor Commented:
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
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?
ycTINAuthor Commented:
@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
michofreihaConnect With a Mentor Commented:
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.*")


            append_hf("P-hint: outbound\r\n");
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