【角色茶桌】—— QNAP TS-119 Asterisk 1.8 Installation - Asterisk - VoIP 研究室 - 電訊茶室 - Powered by Discuz!

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16^# 跳轉到 »

發表於 2011-10-14 23:41 | 只看該作者

本帖最後由角色於 2011-10-15 11:36 編輯

打电话时没有问题，但是cli出很多的warnings：

[Oct 14 23:26:19] WARNING[24044]: chan_sip.c:24433 handle_incoming: Misrouted SIP response '401 Unauthorized' with Call-ID '49a5c7b46b5f20152a31083a2dacca59@xxx.homeftp.org', too many vias
[Oct 14 23:26:19] WARNING[24044]: chan_sip.c:24433 handle_incoming: Misrouted SIP response '401 Unauthorized' with Call-ID '4aa9469639d296680abc03032c1c40f6@xxx.homeftp.org', too many vias
TWTS119*CLI>
Disconnected from Asterisk server
Executing last minute cleanups

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可能是把之前Asterisk 1.4的东西全搬过去，所以出了这样的问题。现在从新清空sip.conf，extensions.conf，然后慢慢加入基本的东西，而没有问题。

知道问题在那里，就是ET263 register => 出问题。

角色

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17^#

發表於 2011-10-14 23:48 | 只看該作者

I believe that it might be due to the incorrect package installed during the "make install" process.

YH

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18^#

發表於 2011-10-14 23:53 | 只看該作者

Just talk with YH using his new Asterisk 1.8 in QNAP NAS. The voice and sound quality is very good. Hope YH can get a fruitful success.

Rank: 2

19^#

發表於 2011-10-15 06:27 | 只看該作者

After installation, if you exit the xterm / ssh login, when you start the asterisk, you may experience the following error:

/opt/asterisk-1.8.7.0/sbin/asterisk
/opt/asterisk-1.8.7.0/sbin/asterisk: error while loading shared libraries: libssl.so.0.9.8: cannot open shared object file: No such file or directory

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This can be resolved by

export PATH=/opt/bin:$PATH
export LDFLAGS="-L/opt/lib"
export CFLAGS="-I/opt/include"
export LD_LIBRARY_PATH=/opt/lib

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20^#

發表於 2011-10-15 06:35 | 只看該作者

本帖最後由角色於 2011-10-15 06:56 編輯

Asterisk changed from 1.4 to 1.8, the iax gives the following error:

[Oct 15 06:26:50] ERROR[13845]: chan_iax2.c:4911 handle_call_token: Call rejected, CallToken Support required. If unexpected, resolve by placing address 90.130.18.241 in the calltokenoptional list or setting user 2201 requirecalltoken=no

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This can be resolved by placing

[2201]
requirecalltoken=no

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in the user 2201.

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21^#

發表於 2011-10-15 07:11 | 只看該作者

回復 20# 角色

This is usually required. Perhaps you have not used IAX clients as frequent as I have.

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22^#

發表於 2011-10-15 07:35 | 只看該作者

There is some problem in recognising the incoming digit in the IVR. I do not know whether it comes with default voice in gsm form. What I used in the NAS Asterisk 1.4.22 is u-law.

YH

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23^#

發表於 2011-10-15 07:43 | 只看該作者

There is some problem in recognising the incoming digit in the IVR. I do not know whether it comes w ...
角色發表於 2011-10-15 07:35

I am sorry that I forward my ivr to an incorrect one. Now the ivr using HKBN 2b is fixed.

YH

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24^#

發表於 2011-10-15 09:53 | 只看該作者

The date/time in cdr is incorrect, it use GMT in the cdr.conf. We have to modify accordingly.

[csv]
;usegmtime=yes ; log date/time in GMT. Default is "no"
usegmtime=no ; YH 15-10-11

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YH

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25^#

發表於 2011-10-15 10:13 | 只看該作者

回復 23# 角色

Couldyou post your sip.conf [general] and hkbn2b setting again?

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26^#

發表於 2011-10-15 11:01 | 只看該作者

回復 25# ckleea

1. Put the following line to /etc/hosts

203.80.89.135 s2hkbntel.net s21.hkbntel.net

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2. sip.conf

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3. extensions.conf

;HK PSTN Gateway, with CID
[trunk-hk-pstn-yh9]
exten => _9XX.,1,Dial(SIP/${EXTEN:1}@hkbn2b,,r)
;HKBN 2b dial-in
[from-hkbn]
exten => 36112233,1,GotoIf($[${CALLERID(num)} = 96xxxyyy]?YH-call)
exten => 36112233,n,GotoIf($[${CALLERID(num)} = 64xxxyyy]?Paul-call)
exten => 36112233,n,Dial(SIP/2004,,)
exten => 36112233,n,Hangup()
exten => 36112233,n(YH-call),GoTo(internal,1130,1)
exten => 36112233,n(Paul-call),Goto(internal-paul-ivr,1131,1);
[internal]
;TW IVR
exten => 1130,1,Answer
exten => 1130,n,Wait(3)
exten => 1130,n,Background(vm-enter-num-to-call)
exten => 1130,n,WaitExten
exten => 1130,n,Playback(vm-goodbye)
exten => 1130,n,Hangup
exten => i,1,Playback(pbx-invalid)
exten => i,2,Goto(1130,1)
exten => t,1,Playback(vm-goodbye)
exten => t,2,n,Hangup
[inter-paul-ivr]
exten => _1,1,Dial(SIP/21xx,,)
exten => _2,1,Dial(SIP/21yy,,)
exten => 1131,1,Answer
exten => 1131,n,Wait(3)
exten => 1131,n,Background(vm-enter-num-to-call)
exten => 1131,n,WaitExten
exten => 1131,n,Playback(vm-goodbye)
exten => 1131,n,Hangup
exten => i,1,Playback(pbx-invalid)
exten => i,2,Goto(1131,1)
exten => t,1,Playback(vm-goodbye)
exten => t,2,n,Hangup

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27^#

發表於 2011-10-15 11:18 | 只看該作者

My only difference is the hosts file, I can only register at S22, not s21

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28^#

發表於 2011-10-15 11:47 | 只看該作者

回復 24# 角色

How about the /var/log/Asterisk/messages? Is the time also correct?

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29^#

發表於 2011-10-15 11:50 | 只看該作者

回復 28# bubblestar

The time is correct

[Oct 15 11:16:29] NOTICE[20130] chan_sip.c: Peer '2023' is now Reachable. (272ms / 2000ms)
[Oct 15 11:26:11] NOTICE[20130] chan_sip.c: Peer '2023' is now Lagged. (3303ms / 2000ms)
[Oct 15 11:26:22] NOTICE[20130] chan_sip.c: Peer '2023' is now Reachable. (273ms / 2000ms)
[Oct 15 11:27:22] NOTICE[20130] chan_sip.c: Peer '2027' is now UNREACHABLE! Last qualify: 1541
[Oct 15 11:27:33] NOTICE[20130] chan_sip.c: Peer '2027' is now Reachable. (963ms / 2000ms)
[Oct 15 11:39:07] NOTICE[20130] chan_sip.c: Peer '2023' is now Lagged. (2444ms / 2000ms)
[Oct 15 11:39:18] NOTICE[20130] chan_sip.c: Peer '2023' is now Reachable. (624ms / 2000ms)

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30^#

發表於 2011-10-15 12:40 | 只看該作者

本帖最後由 bubblestar 於 2011-10-15 12:44 編輯

YH, Congratulations for running both Asterisk 1.8 and Asterisk 1.4 parallel on the same machine of QNAP NAS. It might or might not be a breakthrough but sure it is a very good try to manipulate and maximize the use of our limited resources. The test just made between you and me proved very successful. Please keep this on and share with us what you have done.

Many thanks