Switchfin firmware on ATCOM IP01 - VoIP 研究室 - 電訊茶室

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發表於 2011-1-15 16:16 | 顯示全部帖子

受够了ATCOM的FIRMWare, 有没有Switchfin的Firmware可以下载？
http://anyhub.net/file/1gOI-v373-ast14.uimage-md5 好像没有了

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2^#

發表於 2011-1-15 16:28 | 顯示全部帖子

太过分了，已upgrade 经到 switchfin 373. 一点问题都没有。
到437有没有需要？

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3^#

發表於 2011-1-15 16:38 | 顯示全部帖子

更新到437，现在看看如何用Switchfin,哪里有教程？

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4^#

發表於 2011-1-15 21:04 | 顯示全部帖子

i just find a issue with 437, the login panel is not compability with IE, only chome show the login prompt?

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5^#

發表於 2011-1-16 00:28 | 顯示全部帖子

IAX ATCOM AG188N
use AG188N instead of SPA2102/3000 to connect ATCOM IP01, when swith to Switchfin, the user cannot be athenticate with the following error in log:
Call rejected, CallToken Support required. If unexpected, resolve by placing address 192.168.xx.xx in the calltokenoptional list or setting user 6xxx requirecalltoken=no

after searching a while, add the following to the user.conf under 6xxx
requirecalltoken=no,
then the 6xxx can autheticate to IP01

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6^#

發表於 2011-1-18 17:37 | 顯示全部帖子

my active channel for IP01 seems dead, when dial, the following message appear in log.
"chan_sip.c: Call from '6002' to extension '9XXXXXXXX' rejected because extension not found"
Outgoing dail rule is set to trunk 1 dial pattern : "9XXXXXXXX"
any command line to debug the issue?

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7^#

發表於 2011-1-18 17:45 | 顯示全部帖子

sorry just fix it. typo error.
working for VOIP trunk now.

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8^#

發表於 2011-1-18 18:43 | 顯示全部帖子

to add ET263 to IP01 SWITCHFIN, does it need to follow yhfung's method, edit
users.con ET263 registersip=no
sip.conf [general] context，
register=588034567:et263_password@sip.etelephone.cn:10002/588034567
? or any other easier way?
yes, i forget to add an _ in the outgoing plan

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9^#

發表於 2011-1-18 21:36 | 顯示全部帖子

[attach]414[/attach]

unlucky

[Jan 18 21:14:18] NOTICE[363] chan_sip.c: Peer '588060123' is now Reachable. (330ms / 2000ms)
[Jan 18 21:14:18] WARNING[363] chan_sip.c: Forbidden - maybe wrong password on authentication for NOTIFY
588060123 has no problem to log in by 3CX.
Any suggestion?

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10^#

發表於 2011-1-18 21:38 | 顯示全部帖子

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11^#

發表於 2011-1-19 00:34 | 顯示全部帖子

finally got it:
add user.conf registersip=yes改为registersip=no

in Configure Trunk Voip add 10002

in Admin -> SIP -> Misc -> register field add
registersip=yes改为registersip=no
588060123:XXXXX@sip.etelephone.cn:10002/588060123

finally need reboot. for change to user.conf
then the status show registered.

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12^#

發表於 2011-1-20 02:14 | 顯示全部帖子

Now setting up the SPA3102. Previously I setup the SPA3102 PSTN Line as an extension on the IP01 and eone would call the extension and then get a dial tone to dial out.
According to http://forum.voxilla.com/cisco-l ... asterisk-28807.html
A custom trunk with custom dial string: SIP/$OUTNUM$@300 can work with http authentication. is it possible?
want to kave a better way to handle the SPA3102.

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13^#

發表於 2011-1-23 19:35 | 顯示全部帖子

SWITCHFIN log file error.
If i view today's log, then i will be lock up in ip01 login screen. i.e. repeat display of login screen no matter what is my input. (except today's log, there is no issue for other log file)
I i disable logger function, and hardward reboot the IP01, then I can work flawlessly on my IP01.
The issue is that now i am tuning the IP01. is there any command to reset the log file?

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14^#

發表於 2011-1-24 00:54 | 顯示全部帖子

there is not log error, but i check the /var/log/messages size, it is very big.
Until i enter asterisk -rx "logger rotate", the system do not start normally.
I believe the switchfin will clean up the log file every night, and the log message size is reduced to 10K.
So i want to reset the message size every half day until i finish my ip01 setup.
Is there any automatic way to handle it?

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15^#

發表於 2011-1-28 01:53 | 顯示全部帖子

可以upload你的compile Version吗到Web。我直接用你的firmware update.