Sip trunk

I have a SIP account that i can’t get to work on asterisk.
I have it working with x-lite but can’t translate it to asterisk.
Here is how I have it on x-lite:

How do I get that out bound proxy line in asterisk?

Plz send ur Sip provider peer config from sip.conf & ur outbound call context config from extension.conf
one more thing in sip.conf wher u mention user settings or extension settings. in which context sud be ur out bound calll context. plz check that also


the problem is that i can’t register on the provider, i tried several configurations but with no success.
I’m using Freepbx to configure the trunk.

This is not the freepbx forums :wink: Try asking there.

So how do i do it without freepbx

Learn asterisk:

humm… the RTFM answer.
Ok, i have sip trunks configured and running, I just can’t make this one working, i just needed to know how to configure that outbound proxy port on asterisk.

It's not the RTFM answer for nothing. I have no problem giving of my time but I hate when people just take advantage. Would you want me asking you to spoon feed you answers all day for free ?

Now that my rant is over what comes up in the CLI when you try to make a call ? What do you have in sip.conf and in extensions.conf ?

Ok i am sorry. Nobody here is obliged to give me the answer, specially for free, and i usually don’t come to the forums unless i have tried everything else.
Monday i’ll try to post the content of those files, but i dont’ even make a call because i can’t register on the sip server for that account. The provider won’t give me any help on this so i tested it with x-lite and could only make it work using that outbound proxy port you can see on the image.

You can simply set port=5070 under that specific friends section in sip.conf

Didn’t have a chance to test this yet.
Anyway just wanted to give you the information I got from the provider.

Proxy Address:
Proxy Port: 5070
Registar address:
Registar port: 5060
SIP URI: +35130201xxxx
Username: +35130201xxxx
Password: xxxxxx

As you can see there is a different port for the proxy and the registar.
Thank you for your patience.

The registration is done under the general section so that should go by the default of 5060. Keep port=5070 under the sip information section and you should not have any problems.

Sorry for the late reply.
I tested that port=5070, I found a outboundproxy= command and used it as well but I just can’t register on the provider. I get a ‘request sent’ and thats it.

Speak to the provider see if they are getting anything from you. Also run a sip trae and see what you get back.

Good luck trying to get such a response “Speak to the provider see if they are getting anything from you” from the provider. First tier support staff normally won’t have access to the packets.

Easier way is to see what is really being sent and received on the wire. Wireshark is an excellent utility for this purpose.

Normally there are 2 components with ITSP: SIP Proxy and Outbound Proxy. Without getting too technical, SIP Proxy server handles the signalling and Outbound Proxy handles the media. The SIP Proxy server usually is not exposed to the public.

Your provider is probably using as Outbound proxy and as SIP proxy.

Proxy Address:
Proxy Port: 5070
Registar address:
Registar port: 5060
SIP URI: +35130201xxxx
Username: +35130201xxxx
Password: xxxxxx

From my understanding, there is no separate configuration for outbound proxy vs. registrar within *. (my experience with * is relatively short)

So here is what I think you should do:
From your * box, configure hosts file to point (address of to

Then in the trunk configuration from freepbx,
set fromdomain and host to
Set other registration info.

You should capture the packets and see what you get if it still fails to register.

I finally got it registered with this string:
But now I can’t make any calls, I get a ‘All lines are busy now’, and the error "call failed : 603 declined.