Asterisk and SIP Lines


I have an Asterisk PBX which is running ok with the internal calls (betwen internal extensions).

Now, I also have 2 SIP lines (not SIP trunks) each one with itself userid, password, domain and proxy register server. And I don’t know if it is possible to make the outbound calls through this SIP lines (like if they were RDSI or Analog lines).

The data of SIP Lines are:

Line 1:

User ID: 1001
Password: XXXXXX
Proxy Register Server:

Line 2:

User ID: 1002
Password: XXXXXX
Proxy Register Server:

If somebody has try this, please tell me how can I configure my Asterisk to make external calls through this SIP Lines.

Thank you far all.

“Making calls through SIP lines” is not a very meaningful concept for SIP. What do you actually want to do?

Whether you can do it may depend on your contract with whoever routes your inbound calls.

The exactly situation is:

My provider doesn’t offer me a SIP trunk, only offers me a pair of SIP Lines (SIP Extensions) each one with the data I have show before (UserID, Password, Domain and Proxy Register Server).

I tested with two IP Phones (I registered each one with the data my provider gave me) and they worked totally ok.

What I want to do is to make all the external calls from my Asterisk (outgoing and incoming) through this SIP Lines. Some days ago I had two RDSI Lines and I could course all the external calls from my Asterisk through this RDSI Lines.

My question is if I can use me new SIP Lines (SIP Extensions from my provider) to course all the external calls like if they were a RDSI or Analog Line.

To place outbound SIP calls to PSTN telephones, you need to set up an account with a SIP termination provider. To receive calls from PSTN telephones, you need a SIP DID provider who will provide a telephone number and route it to your SIP server. There are plenty of DID and termination providers (some offer both services) at different prices. Google is your friend!

What does your provider mean by SIP “Line”?

What outgoing call services does your provider supply.

The nature of SIP is such that one has to go out of ones way to limit the system to one call at a time. Your provider may well do that for commercial reasons.