ckleea

The next trial will be to get support from other users in HK. so that we can have failover from PCCW, HGC, NWT, HKBN, etc

bubblestar

What a GREAT idea!

ckleea

This is our VOIP world!!

ckleea

Not only the cheapest but reliable.

角色

The next trial will be to get support from other users in HK. so that we can have failover from PCCW ...
ckleea 發表於 2011-2-13 22:32

CK's asterisk server can be acted as a super hub of our general asterisk servers.

YH

ckleea

回復 20# 角色


    I am just at the beginning. Still many not sure and not understand. However, with mutual co-operation, we can do more in a reliable and economic way.

ckleea

Give you a feel of the log message on failover.

You can see 2b has no answer then it goes to dial via pstn line.

1. JABBER: asterisk INCOMING:
   
2.     -- Accepting AUTHENTICATED call from 192.168.xxx.x:
   
3.        > requested format = alaw,
   
4.        > requested prefs = disabled,
   
5.        > actual format = alaw,
   
6.        > host prefs = disabled,
   
7.        > priority = reqonly
   
8.     -- Executing [9xxxxxxxx@DLPN_DP1:1] Macro("IAX2/6101-4113", "superdial,SIP/133xxxxxxxx@hkbn2b") in new stack
   
9.     -- Executing [s@macro-superdial:1] Set("IAX2/6101-4113", "GROUP()=") in new stack
   
10.     -- Executing [s@macro-superdial:2] Set("IAX2/6101-4113", "GROUPCOUNT=0") in new stack
    
11. [Feb 14 07:48:32] WARNING[30286]: ast_expr2.fl:468 ast_yyerror: ast_yyerror():  syntax error: syntax error, unexpected $end, expecting '-' or '!' or '(' or '<token>'; Input:
    
12. 0 >
    
13.     ^
    
14. [Feb 14 07:48:32] WARNING[30286]: ast_expr2.fl:472 ast_yyerror: If you have questions, please refer to doc/tex/channelvariables.tex.
    
15.     -- Executing [s@macro-superdial:3] GotoIf("IAX2/6101-4113", "0?104") in new stack
    
16.     -- Executing [s@macro-superdial:4] GotoIf("IAX2/6101-4113", "1?macro-superdial,s,6") in new stack
    
17.     -- Goto (macro-superdial,s,6)
    
18.     -- Executing [s@macro-superdial:6] GotoIf("IAX2/6101-4113", "1?macro-superdial,s,8") in new stack
    
19.     -- Goto (macro-superdial,s,8)
    
20.     -- Executing [s@macro-superdial:8] GotoIf("IAX2/6101-4113", "1?macro-superdial,s,10") in new stack
    
21.     -- Goto (macro-superdial,s,10)
    
22.     -- Executing [s@macro-superdial:10] Dial("IAX2/6101-4113", "SIP/133xxxxxxxx@hkbn2b,,,") in new stack
    
23.   == Using SIP RTP TOS bits 184
    
24.   == Using SIP RTP CoS mark 5
    
25.     -- Called 133xxxxxxxx@hkbn2b
    
26.     -- No one is available to answer at this time (1:0/0/0)
    
27.     -- Executing [s@macro-superdial:11] Goto("IAX2/6101-4113", "s-NOANSWER,1") in new stack
    
28.     -- Goto (macro-superdial,s-NOANSWER,1)
    
29.     -- Executing [s-NOANSWER@macro-superdial:1] GotoIf("IAX2/6101-4113", "1?macro-superdial,s-NOANSWER,3") in new stack
    
30.     -- Goto (macro-superdial,s-NOANSWER,3)
    
31.     -- Executing [s-NOANSWER@macro-superdial:3] NoOp("IAX2/6101-4113", "") in new stack
    
32.     -- Executing [9xxxxxxxx@DLPN_DP1:2] Macro("IAX2/6101-4113", "superdial,SIP/pstn-spa3k-d1/xxxxxxxx") in new stack
    
33.     -- Executing [s@macro-superdial:1] Set("IAX2/6101-4113", "GROUP()=") in new stack
    
34.     -- Executing [s@macro-superdial:2] Set("IAX2/6101-4113", "GROUPCOUNT=0") in new stack
    
35. [Feb 14 07:48:33] WARNING[30286]: ast_expr2.fl:468 ast_yyerror: ast_yyerror():  syntax error: syntax error, unexpected $end, expecting '-' or '!' or '(' or '<token>'; Input:
    
36. 0 >
    
37.     ^
    
38. [Feb 14 07:48:33] WARNING[30286]: ast_expr2.fl:472 ast_yyerror: If you have questions, please refer to doc/tex/channelvariables.tex.
    
39.     -- Executing [s@macro-superdial:3] GotoIf("IAX2/6101-4113", "0?104") in new stack
    
40.     -- Executing [s@macro-superdial:4] GotoIf("IAX2/6101-4113", "1?macro-superdial,s,6") in new stack
    
41.     -- Goto (macro-superdial,s,6)
    
42.     -- Executing [s@macro-superdial:6] GotoIf("IAX2/6101-4113", "1?macro-superdial,s,8") in new stack
    
43.     -- Goto (macro-superdial,s,8)
    
44.     -- Executing [s@macro-superdial:8] GotoIf("IAX2/6101-4113", "1?macro-superdial,s,10") in new stack
    
45.     -- Goto (macro-superdial,s,10)
    
46.     -- Executing [s@macro-superdial:10] Dial("IAX2/6101-4113", "SIP/pstn-spa3k-d1/xxxxxxxx,,,") in new stack
    
47.   == Using SIP RTP TOS bits 184
    
48.   == Using SIP RTP CoS mark 5
    
49.     -- Called pstn-spa3k-d1/xxxxxxxxx\
    
50.     -- SIP/pstn-spa3k-d1-00000007 is ringing
    
51.     -- SIP/pstn-spa3k-d1-00000007 answered IAX2/6101-4113
    
52.   == Spawn extension (macro-superdial, s, 10) exited non-zero on 'IAX2/6101-4113' in macro 'superdial'
    
53.   == Spawn extension (DLPN_DP1, 9xxxxxxxx, 2) exited non-zero on 'IAX2/6101-4113'
    
54.     -- Hungup 'IAX2/6101-4113'

複製代碼

ckleea

The other possibilities for our world is skype trunks. I have 4 at the moment and can add more if my ATOM D525 can be made working. I have migrated the D330 server images into D525 but not working well. Somethings outside asterisk not working as expected. Not enough time to deal with.

Besides, another thing is that we need a better internet line. To may understanding and experience, I feel HGC is still better for my use. I do encounter problem with my 2b account while I am in HKBN network.

Qnewbie

WOW, 4 skype trunks+PCCW, HGC, NWT, HKBN etc. You are running a small voip company!

ckleea

No, only one PCCW line and a 2b line. Skype trunk is from the siptosis software.

Qnewbie

Is it true that the Skype limits the number of skype clients per ip?

There are 4 skype clients might used at the same time at home. If I add one more to run siptosis using monthly subscription plan, there are 5 skype users per ip! So the limitation rumours rises my concern.

ckleea

Now I have 5 skype trunks added to my asterisk server.

All are working.

Qnewbie

That's great!

ckleea

回復 28# Qnewbie

It will be much better for me to get my D525 running to host more skype trunks.

How is your skype - voip setup? Can you tell us how you make it work? Do you have a linux server or use windows for skype connection?

Qnewbie

The skype trunk is set up with siptosis in plain windows, no other fancy stuff.

It should be installed to ATOM PC(former asterisk server with slow CF-card) latter if the number of skype is not limited.