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Why you need to do asterisk -vvvvc to get asterisk console? Does you asterisk running in the background? What you need to look for the console log is asterisk -vvvr not -vvvvc
I suppect your siptosis is not hooked to the asterisk ie. it is not running in the background.

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I use asterisk -vvvvc because asterisk is not started after I logon to the computer.
I just don't know how to check siptosis is hooked to asterisk.
It seems that siptosis is registered onto the Asterisk server correctly as I can see my siptosis account online when I type sip show peers in Asterisk CLI.

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What version linux you are using? Is your asterisk compiled or install via rpm or apt-get?
It seems that your siptosis peers are not configured to take the call. I don't know why from your config files as they all looked ok to me.

For your information, my linux (centos 6) and siptosis work for more than a week from the last reboot. It is running very well.

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I am using Centos 6.4.
Asterisk is download from digium in .tzr.gz format.
I think asterisk is compiled in that case.

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I see. I have not tried siptosis single channel in centos 6.4. There have been some unknown issues in preventing my trunks to start up.

Try centos 5.x, your setup should work.

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I've tried CentOS 5.9 today but it still don't work.
Anyway, thanks for your help

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回復 21# kurtor

It should work without any problem.

Follow the instruction again.

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I use asterisk -vvvvc because asterisk is not started after I logon to the computer.
I just don't kn ...
kurtor 發表於 2013-4-29 01:18


kurtor,
If you still didn't get it to work I really suggest you doing the setup from basis configuration, that means don't use the main configuration file 'siptosis.cfg' generated by stsTrunkBuilder unless you've realized every variables within the file.

Here are basis configuration for siptosis.cfg

host_port=5070
username=skypests
passwd=unimportantpassword
do_register=no

The proper trunk setting on asterisk/FreePBX

type=peer
host=<SipToSis LAN IP>
port=5070
nat=no
dtmfmode=auto
canreinvite=no
qualify=yes
incominglimit=1
outgoinglimit=1
call-limit=1
busylevel=1
context=from-trunk

Note: Be sure you've closed the SELinux & firewall on CentOS.

Good luck
OSSLab Blog :: VoIP & IT Consultant

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我的情況同樓主一樣。我是使用Elastix, CentOS5.4
SKype可以打入Elastix,但Elastix的分機不能打出Skype。
請成功的朋友提供你們的siptosis的各個配置文件sample.

Thanks!

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