Failover trunk - Asterisk - VoIP 研究室 - 電訊茶室

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16^# 跳轉到 »

發表於 2011-2-14 07:25 | 顯示全部帖子

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.

JABBER: asterisk INCOMING:
-- Accepting AUTHENTICATED call from 192.168.xxx.x:
> requested format = alaw,
> requested prefs = disabled,
> actual format = alaw,
> host prefs = disabled,
> priority = reqonly
-- Executing [9xxxxxxxx@DLPN_DP1:1] Macro("IAX2/6101-4113", "superdial,SIP/133xxxxxxxx@hkbn2b") in new stack
-- Executing [s@macro-superdial:1] Set("IAX2/6101-4113", "GROUP()=") in new stack
-- Executing [s@macro-superdial:2] Set("IAX2/6101-4113", "GROUPCOUNT=0") in new stack
[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:
0 >
^
[Feb 14 07:48:32] WARNING[30286]: ast_expr2.fl:472 ast_yyerror: If you have questions, please refer to doc/tex/channelvariables.tex.
-- Executing [s@macro-superdial:3] GotoIf("IAX2/6101-4113", "0?104") in new stack
-- Executing [s@macro-superdial:4] GotoIf("IAX2/6101-4113", "1?macro-superdial,s,6") in new stack
-- Goto (macro-superdial,s,6)
-- Executing [s@macro-superdial:6] GotoIf("IAX2/6101-4113", "1?macro-superdial,s,8") in new stack
-- Goto (macro-superdial,s,8)
-- Executing [s@macro-superdial:8] GotoIf("IAX2/6101-4113", "1?macro-superdial,s,10") in new stack
-- Goto (macro-superdial,s,10)
-- Executing [s@macro-superdial:10] Dial("IAX2/6101-4113", "SIP/133xxxxxxxx@hkbn2b,,,") in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Called 133xxxxxxxx@hkbn2b
-- No one is available to answer at this time (1:0/0/0)
-- Executing [s@macro-superdial:11] Goto("IAX2/6101-4113", "s-NOANSWER,1") in new stack
-- Goto (macro-superdial,s-NOANSWER,1)
-- Executing [s-NOANSWER@macro-superdial:1] GotoIf("IAX2/6101-4113", "1?macro-superdial,s-NOANSWER,3") in new stack
-- Goto (macro-superdial,s-NOANSWER,3)
-- Executing [s-NOANSWER@macro-superdial:3] NoOp("IAX2/6101-4113", "") in new stack
-- Executing [9xxxxxxxx@DLPN_DP1:2] Macro("IAX2/6101-4113", "superdial,SIP/pstn-spa3k-d1/xxxxxxxx") in new stack
-- Executing [s@macro-superdial:1] Set("IAX2/6101-4113", "GROUP()=") in new stack
-- Executing [s@macro-superdial:2] Set("IAX2/6101-4113", "GROUPCOUNT=0") in new stack
[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:
0 >
^
[Feb 14 07:48:33] WARNING[30286]: ast_expr2.fl:472 ast_yyerror: If you have questions, please refer to doc/tex/channelvariables.tex.
-- Executing [s@macro-superdial:3] GotoIf("IAX2/6101-4113", "0?104") in new stack
-- Executing [s@macro-superdial:4] GotoIf("IAX2/6101-4113", "1?macro-superdial,s,6") in new stack
-- Goto (macro-superdial,s,6)
-- Executing [s@macro-superdial:6] GotoIf("IAX2/6101-4113", "1?macro-superdial,s,8") in new stack
-- Goto (macro-superdial,s,8)
-- Executing [s@macro-superdial:8] GotoIf("IAX2/6101-4113", "1?macro-superdial,s,10") in new stack
-- Goto (macro-superdial,s,10)
-- Executing [s@macro-superdial:10] Dial("IAX2/6101-4113", "SIP/pstn-spa3k-d1/xxxxxxxx,,,") in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
-- Called pstn-spa3k-d1/xxxxxxxxx\
-- SIP/pstn-spa3k-d1-00000007 is ringing
-- SIP/pstn-spa3k-d1-00000007 answered IAX2/6101-4113
== Spawn extension (macro-superdial, s, 10) exited non-zero on 'IAX2/6101-4113' in macro 'superdial'
== Spawn extension (DLPN_DP1, 9xxxxxxxx, 2) exited non-zero on 'IAX2/6101-4113'
-- Hungup 'IAX2/6101-4113'

複製代碼

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17^#

發表於 2011-2-14 09:00 | 顯示全部帖子

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.

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18^#

發表於 2011-2-14 21:07 | 顯示全部帖子

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

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19^#

發表於 2011-2-14 22:14 | 顯示全部帖子

本帖最後由 ckleea 於 2011-2-14 22:38 編輯

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

All are working.

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20^#

發表於 2011-2-15 07:36 | 顯示全部帖子

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?

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21^#

發表於 2011-2-15 19:45 | 顯示全部帖子

I try to increase to 8 now. But not yet able to work.

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22^#

發表於 2011-2-15 21:29 | 顯示全部帖子

Just complete. 9 trunks all working

stsTrunk_01/stsTrunk_01 192.168.1xx.xx                                     5072    OK (3 ms)
stsTrunk_02/stsTrunk_02 192.168.1xx.xx                                     5073    OK (3 ms)
stsTrunk_03/stsTrunk_03 192.168.1xx.xx                                     5074    OK (4 ms)
stsTrunk_04/stsTrunk_04 192.168.1xx.xx                                     5075    OK (4 ms)
stsTrunk_05/stsTrunk_05 192.168.1xx.xx                                     5076    OK (3 ms)
stsTrunk_06/stsTrunk_06 192.168.1xx.xx                                     5077    OK (3 ms)
stsTrunk_07/stsTrunk_07 192.168.1xx.xx                                     5078    OK (3 ms)
stsTrunk_08/stsTrunk_08 192.168.1xx.xx                                     5079    OK (3 ms)
stsTrunk_09/stsTrunk_09 192.168.1xx.xx                                     5080    OK (1 ms)

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23^#

發表於 2011-2-15 21:30 | 顯示全部帖子

回復 32# bubblestar

Of course, it will takes some time. As noted, it is for failover, some time lapse is expected.

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24^#

發表於 2011-2-15 21:57 | 顯示全部帖子

You can set up your skype trunk by siptosis in a linux machine. Then, use IP01 within LAN to connect as sip trunk

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25^#

發表於 2011-2-15 22:47 | 顯示全部帖子

One interesting thing I observe, even you have all skype accounts set up and run. In windows skype, some may actually not appeared online. Don't know why.

Anyway when dial in, it works.

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26^#

發表於 2011-2-15 22:58 | 顯示全部帖子

How is the performance with multiple active calls?
Linux CPU usage under varying conditions.

Virtualbox (not an ideal setup) in 32 bit mode using a Vista 64 Host Dual Core AMD Athlon X2 4850e.
Skype sound device setup is a 2 card snd-dummy configuration. SIP codec for all channels is PCMU.
6 calls were from the same Vista host using softphones. The other 4 were made from another PC using softphones.
The Ubuntu distro is a 32 bit 9.0.4 (desktop) with 1.5GB total ram allocation. Approx 800MB was used for the entire setup.
10 Idle channels - There really are 10 they just are not "top" processes.
10 Active channels to echo test - Despite the apparent high load, there was no issue.
6 Active channels to echo test

Dedicated Ubuntu 64 bit 10.04 (desktop) on Dual Core AMD Athlon X2 4850e and 4GB ram.
Skype 2.1.0.81 using a single card snd-dummy configuration. SIP codec for all channels is PCMU.
10 calls were from another PC using softphones.
Top on 10.04 shows weird cpu % sometimes for idle processes.
10 Idle channels - There really are 10 they just are not "top" processes.
10 Active channels to echo test
6 Active channels to echo test
10 Active channels to echo test - This is using sndShare for sound device

Windows Vista CPU usage under varying conditions.
These shots are on a Vista 64 Host Dual Core AMD Athlon X2 4850e.
Skype sound device setup using on-board sound device. SIP codec for all channels is PCMU.
6 calls were from the same Vista host using softphones. The other 4 were made from another PC using softphones.
10 Idle channels
10 Active channels to echo test - This system maxed out at 8. It struggled to get 10 going.
6 Active channels to echo test