Connecting 2 asterisk machines

ckleea

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?

bubblestar

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.

ckleea

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.

角色

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

YH

bubblestar

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.

1. Interconnection Between TW Asterisk Server and SZ Asterisk Server using IAX
   
2. 下面用IAX，雙方都可以顯示對方電話號碼。
   
3. 
   
4. TW Asterisk Server
   
5. 
   
6. iax.conf
   
7. 
   
8. [general]
   
9. bindport=4569
   
10. bandwidth=high
    
11. jitterbuffer=yes
    
12. forcejitterbuffer=yes
    
13. maxjitterbuffer=1000
    
14. maxjitterinterps=10
    
15. resyncthreshold=1000
    
16. tos=ef
    
17. autokill=yes
    
18. 
    
19. [trunk-tw-sz]
    
20. type=friend
    
21. host=sz.3322.org
    
22. trunk=yes
    
23. disallow=all
    
24. allow=alaw
    
25. qualify=yes
    
26. context=from-sz
    
27. 
    
28. 
    
29. extensions.conf
    
30. 
    
31. exten => _20XX,1,Dial(IAX2/trunk-tw-sz/${EXTEN},,r)   ;Using IAX2
    
32. 
    
33. [from-sz]
    
34. exten => _12XX,1,Dial(SIP/${EXTEN},,r)
    
35. 
    
36. SZ Asterisk Server
    
37. 
    
38. iax.conf
    
39. 
    
40. [general]
    
41. bindport=4569
    
42. bandwidth=high
    
43. jitterbuffer=yes
    
44. forcejitterbuffer=yes
    
45. maxjitterbuffer=1000
    
46. maxjitterinterps=10
    
47. resyncthreshold=1000
    
48. tos=ef
    
49. autokill=yes
    
50. 
    
51. [trunk-sz-tw]
    
52. type=friend
    
53. trunk=yes
    
54. host=tw.3322.org
    
55. disallow=all
    
56. allow=alaw
    
57. qualify=yes
    
58. context=from-tw
    
59. 
    
60. 
    
61. extensions.conf
    
62. 
    
63. exten => _12XX,1,Dial(IAX2/trunk-sz-tw/${EXTEN},,r)
    
64. 
    
65. [from-tw]
    
66. exten => _20XX,1,Dial(SIP/${EXTEN},,r)

複製代碼

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

Many thanks

角色

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

bubblestar

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

角色

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

bubblestar

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.

角色

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