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標題: Connecting 2 asterisk machines [打印本頁]

作者: ckleea 時間: 2011-1-28 07:20 標題: Connecting 2 asterisk machines

本帖最後由 ckleea 於 2011-1-28 11:01 編輯

Would like to discuss on the best methods to connect 2 asterisk machines

Two methods exist
1. SIP
A machine is acted as sip provider and allows B machine to connect as sip client, vice versa.
It is similar to what one login to their own VOIP provider

E.g. in sip.conf
register = 3250:3250T03250@192.168.xxx.xxx/ast18
[sipast18]
disallow = all
allow = ulaw
allow = alaw
host = 192.168.xxx.xxx
fromdomain = 192.168.xxx.xxx
insecure = no
secret = 3250T03250
type = peer
username = 3250
context = from-ast18

in extensions.conf, I have the following for outbound

[CallingRule_ast18]
exten = _3XXX,1,Dial(SIP/sipast18/${EXTEN},,r)
exten = _3XXX,n,Hangup()

for inbound,
[from-ast18]
exten = ast18, Dial(SIP/6100,,r)

2. IAX - inter
The protocol provided by asterisk
my example is similar to what I set in the sip.conf. need also inbound and outbound setting in extensions.conf

further examples can be found here
http://www.voip-info.org/wiki/view/Asterisk+Connect+2+servers

Sometimes, IAX connection not working well.

How about your experience?

作者: bubblestar 時間: 2011-1-28 09:57

Are you talking the environment in same LAN ?

For me, IAX in same lan works fine but I cannot see the exact CALLER ID and the extension number of the real caller except the registered one for connection.

In IP01, when typing iax2 show registry and iax2 show peers, I can see all registered IAX peers details. However, when in Asterisk 1.8 CLI, iax2 show registry shows nothing but it can still operate normally.

My impression is that IP01 GUI under 1.4 and Asterisk 1.8 APL connection is not perfect.
Most probably, Asterisk 1.8 still have some BUGS here and there. Sometimes, we may encounter strange and surprise.

作者: ckleea 時間: 2011-1-28 10:39

Both within the same LAN and over internet

For same LAN, has been no problem at all.
Over internet, it is a different issue. For me, I have test on SIP e.g. YH has been providing trunks for other users and I have done so as a provider before.
For IAX, it is an interesting option as it needs to have only one port open.

GUI is always a problem for us.

作者: 角色 時間: 2011-1-29 20:35

IAX does not use the "friend" and instead use the peer and the user.

YH

作者: bubblestar 時間: 2011-1-29 22:13

回復 4# 角色

But your previous tutorial did suggest using "friend" for IAX connection and said it can let us see the caller ID at both sides.

複製代碼

Is the above settings still valid? Would you post your present IAX settings for our reference, please.

Many thanks

作者: 角色 時間: 2011-1-29 22:41

Oh! I almost forgot what I have said.
I recalled from my memory that Qnewbie is using the method that I said the last-two post.

YH

作者: bubblestar 時間: 2011-1-29 22:48

I cannot see extensions at both sides when using "friend" method.

作者: 角色 時間: 2011-1-30 08:21

I cannot see extensions at both sides when using "friend" method.
bubblestar 發表於 2011-1-29 22:48

I am able to recall my memory that I use iax for one way and use sip for the another way.

SZ ---> HK using IAX
HK ---> SZ using SIP

YH

作者: bubblestar 時間: 2011-1-30 10:55

That means your Dial out method is using IAX but Dial in is using SIP。

Haha, this blending method can now be called YH's IAX-SIP trunk.

作者: 角色 時間: 2011-1-30 12:11

No!
If the members in SZ who want to make calls to HK, the dial plan is using IAX.
If the members in HK who want to make calls to SZ, the dial plan is using SIP.

I believe that both ways can be used in either SIP or IAX to have telephone numbers shown on the callee's telephone LCD.

YH

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