Asterisk as a SIP UA

Asterisk Asterisk General
1

Hy Guys, I’m looking for solution for my big problem.
I’ve one asterisk A with realtime, and I need to connect 3 another asterisk server (B,C,D) to my asterisk A via SIP account. I create 3 SIP account on my asterisk A server, and add extensions to extensions.conf. if I use “register => user:password@asterisk_A_IP”, then asterisk A cannot see this servers and I can’t create calls. is there any way how I can connect my 3 others asterisk to this asterisk A server without “registry” on both sides ??

THX
Pls help me.

2

Use static host addresses.

3

ok THX for your reply.
I created 3 SIP users on Server_A with “host=IP_Address_server_B, C and D”, and on Server B,C,D use “register =>user:pass@IP_Address_Server_A” but, this not working… I think Server B,C and D don’t send any register invite to Server_A, and I don’t know why… When I set SIP debug on Server_A I can’t see nothing about Servers B, C and D, nothing invite and nothing else.

4

You don’t need to register when the addresses are static. If everything has static addresses, there should be no registers at all.

5

ok, you mean something like this?:
table users on Server A

type=peer secret=mypass name=2000 host= 22.33.44.55 (IP server B,C,D) nat=no

and how I can register from server B,C or D like user 2000??? Like I said, I was try register but it’s not working for me…when and when I try on server B,C,D this:

type=peer secret=mypass name=2000 host= 10.20.30.40 (IP server A) nat=no
It’s not send register invide to server A, it’s just new user on this server (B,C,D)…

pls help, it’s very important for me…

6

Why do you want to register? I have a feeling that you don’t understand the purpose of register. It is for use when you do not know the client address in advance.

7

I don’t want register, but I think this is only possibility… I’ve one asterisk A (with realtime) and I want route some calls to remote servers like B,C and D… for example I created user 1500 on server A, and I want all calls for this user route to Server B… It sound like VOIP provider, but I don’t. I want just 3 servers…Is there any way, how to do it?? Look I just want create on server A 3 SIP users, and send login to my co-workers with username, pass and IP server A…When there is way without registration, pls show me how…

8

On Server A:

exten=> 1500,1,Dial(SIP/${EXTEN}@ServerB

Note that multiple registrations from the same address cause confusion for calls in the reverse direction, but if you did so, you would have to have a host=dynamic entry, in sip.conf, for every extension in the system.

There may be better ways with things like IAX switches and dundi, but I’m not familiar with them.

I believe that the last posting was the first one where you indicated you wanted to register all the remote extensions, ratehr than just the servers.

9

THX, I’ll try it… It sound like I don’t have to create SIP user on server A and use register on others servers… I just create extensions on both side, and that’s all???

THX

10

OK, I tried it. I created extension

and on B server

In console on server A, I seen:

-- SIP/37.9.170.146-00000060 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0)
and on server B is nothing…

11

IP address will bypass any passwords and codec selections at the near end. You would normally use the sip.conf section name for the other server.

You also need to provide the logs from the other end. My guess is that you have a message about extension 1500 not being in whatever the context associated with the calling server is.

Each extension number should exist on the all the servers that use it.

There should be a sip.conf peer for each other server that can be the destination or source of calls for the current server.

(As described, A should have entries for B, C, and D, and each of those should have entries for A.)

The context associated with those sip.conf entries should contain all the extension which are local to the server.

The context for locally originated calls should contain all the extensions.

Again, IAX trunks may be a better solution. and there are other mechanisms for supporting unified numbering schemes.
Local extensions should Dial the final device. Remote extensions should dial the distant server, with dialed digits included.

12

hi dave, it’s not working… :frowning:
I sand PM to you.

13

I deleted the PM because you should not send PMs to forum contributors requesting private help.

14

ok, that’s my config
Server_A (realtime)
[ul]sip_table
name-2000
type-friend
context-internal
host-dynamic
secret-secret
[/ul]
sip.conf

[general] register => 5000:secret@IP_Server_B
extension_table

[internal] exten => 2000,1,Dial (SIP/${EXTEN}@IP_Server_B)

Server_B
sip.conf

[5000]
name=5000
secret=secret
context=internal
host=dynamic
type=friend

register => 2000:secret@IP_Server_A[/code]

[i]extensions.conf[/i]
[code]exten => 2000,1,Dial(SIP/200)
exten => 2000,2,Hangup[/code]

call from Server_A is routed to Server_B, it's good, but I've problem 
[code]Call from '' (IP_Server_A:5060) to extension '2000' rejected because extension not found in context 'default'.

I’ve everything in context internal on both servers, and i use include => general but it’s not working.

real thx