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標題: IP01 Christmas Sale [打印本頁]

作者: ckleea    時間: 2010-12-13 11:49     標題: IP01 Christmas Sale

Any customer who never bought IP01 IP PBX and AT610 IP Phone before, now have a chance to purchase sample with a special promotional price!

ONLY $99 to get an IP PBX & IP phone package for your Christmas!

Promote code: IP01isthebest
Deadline: Dec 31 2010
FXS or FXO optional
The price doesn't include delivery charge and each customer could only buy one package

Open Source Asterisk IP PBX
High performance OSLEC
Configurable IVR menu
Voice Mail,Voicemail to Email
Call forward, call waiting, call transfer
Call conference
Call queues,Ring group
SIP trunking, IAX trunking
PSTN analog trunk
Call Detail Record
Access vis: SSH/telnet/web

Interface
1 X RJ45    port
1 X RS232 port
1 X RJ11 port (FXS/FXO interchangeable)


For more information:
http://www.atcom.cn/products_ippbx.html

or send email to:
sales@atcomemail.com
作者: bubblestar    時間: 2010-12-13 13:17

$99 美金 or 人民幣?
作者: ckleea    時間: 2010-12-13 14:20

Either is a good deal
作者: lttliang    時間: 2010-12-13 14:59

你地仲咁钟意ip01?
作者: ckleea    時間: 2010-12-13 15:05

回復 4# lttliang

Remember I can make it to run Swithfin firmware. Once you switch, you are in a new arena.
作者: bubblestar    時間: 2010-12-13 15:12

本帖最後由 bubblestar 於 2010-12-13 15:14 編輯

其實我由一開始到現在,IP01都沒有問題,現在把Firmware 轉用了 Switchfin 更加穩定了(情況好像Linksys Router 一樣,有DD-WRT 固件作第三方支援),加上如果真的是 $99一部,算是不錯了。

你想像一下,SPA3102 在香港要HK$575-$625一部(以正價計算),同樣是一個FXO口,但IP01是SERVER,理論上可以加入20-30條extension,自家用夠晒了,又是圖像介面。而且兩者不能作直接比較。SPA3102我也很鍾意,好多野玩。
作者: lttliang    時間: 2010-12-13 17:08

其實我由一開始到現在,IP01都沒有問題,現在把Firmware 轉用了 Switchfin 更加穩定了(情況好像Linksys Rou ...
bubblestar 發表於 2010-12-13 15:12



    或者是个人运气吧  我是怕左这够野了    我依家用PC装左elastix用,虽然比较耗电,起码唔洗再担心他死机
作者: ckleea    時間: 2010-12-13 17:33

我都相信,可能是你唔好運,買嘅IP01 改唔到。
作者: 角色    時間: 2010-12-15 07:34

看来ATCOM的食水非常深,现在99元就有两件。估计是不包O口或者S口的Analogue卡吧!

角色
作者: ckleea    時間: 2010-12-15 15:48

同意,你看 Nortel SIP phone 係 ebay 價錢就知,不過最重要係backend 嘅firmware and software!
作者: Qnewbie    時間: 2010-12-17 23:09

Ckleea, where you got the info?
作者: ckleea    時間: 2010-12-18 05:45

回復 11# Qnewbie


    Can't tell. But worth a try. It is very good deal.
作者: 亞星    時間: 2010-12-20 09:44

本帖最後由 亞星 於 2010-12-20 14:46 編輯

之前未買過先得?
作者: ckleea    時間: 2010-12-20 17:55

I can say the price is real.
作者: Qnewbie    時間: 2010-12-20 18:38

Just send email to them for query regarding the payment & shipping cost(to Mainland China)...

If it is US$99, it would be great.
作者: 亞星    時間: 2010-12-21 12:01

Price: IP-01(IP PBX) with 1 FXO or FXS ,AT-610(IP Phone) at $99   Shipment charge at $5    Total is $104 (寄到香港)

1, delivery: within 1 day after your payment received

2, payment: 100% payment in advance

可以用 PayPal 俾
作者: ckleea    時間: 2010-12-21 20:21

價錢是理想!
作者: ddwrt_voip    時間: 2010-12-22 07:57

Shipping charge at $5usd . 真的嗎? 好像不太可靠.
作者: lttliang    時間: 2010-12-22 14:08

唔知会不会似我买个一部  是够次品
作者: Qnewbie    時間: 2010-12-22 23:39

699CNY including shipping cost to mainland, China.

Looking forward to the BUGGY IP01
作者: 亞星    時間: 2010-12-23 12:27

今日收到貨了, 有時間就要開始學習
作者: bubblestar    時間: 2010-12-23 14:01

回復 19# lttliang


   
想證明是否次貨,可以看看有購買的師兄檢查 一下表面及試一試基本設定,然後打出打入作測試便知道。

聽說AT610也是好東西,支援IAX的。如果是AT620就更好了。
作者: bubblestar    時間: 2010-12-23 14:07

回復 20# Qnewbie


   
Good price for 2 pieces of VoIP devices.  Last Year, IP01 and AT610 were sold at ¥950.00 and ¥320.00 respectively.  You can see how much you have saved in this purchase.

AT610 comes with 1 WAN and 1 LAN.  Good deal.
作者: ddwrt_voip    時間: 2010-12-23 23:15

恭喜,恭喜
作者: Qnewbie    時間: 2010-12-24 05:57

Accurately, this is not my first choice(two ATAs are broken within a few months. Currently there is not ATA for me).

I just want one simple ATA as backup of SPA3102(still on its journal to my home). It is IAD100T from Wuchun(?) with price tag of 265CNY. But they did not have it in stock!

My asterisk don't need to handle many channels simultaneously(might be max 6 channels). Maybe IP01 just fit for this purpose!
作者: bubblestar    時間: 2010-12-24 09:43

Sorry to hear about that.  Why the ATAs are broken so easily??  Were they bought from TB?  

That is why I insist, if possible and within own budget, buy ATA from Hong Kong or the US.  At least, we can have reliable maintenance.  You know, the costs of buying two defective ATAs + a replacement ATA will definitely more expensive than getting a brand new one in the first place.

Except the SPA1001, my two SPA3102 and 1 PAP2T were all bought from Hong Kong.  They are still loyal to work for me without any problem.  There is a Chinese saying - 貴買平用 - that is always true indeed.

Absolutely, IP01 is quite enough for home and private use if we don't need to take care of big crowds of heavy users.
作者: Qnewbie    時間: 2010-12-24 17:43

One is from TB. The other one is from online store(it was a genie one).

Two functions I wish for IP01 are callback and DISA. Does anyone have experience on those functions?
作者: ckleea    時間: 2010-12-24 20:18

回復 27# Qnewbie


    DISA should not be difficult. You can just follow the manual and add the right pin for secruity. I have no experience to use call back. It would be great to hear.
作者: 亞星    時間: 2010-12-25 13:56

我應該用 IP01 本身的介面學習還是轉用 Switchfin firmware 開始學習好呢?
作者: ckleea    時間: 2010-12-25 14:11

Basically they are the same. Using the better interface in switchfin, it is more user friendly and responsive.

May be you need to learn APL.
作者: 亞星    時間: 2010-12-25 14:48

APL 係咪 A Programming Language 呀? 我唔識0架, 我真係 LuLu 一名黎0架
如果要轉 firmware 去 Switchfin 可能都要請教各位師兄師組
作者: ckleea    時間: 2010-12-25 15:26

APL = Asterisk Programming Language
APL is the same for ATCOM or Switchfin. Just GUI looks much better in switchfin

For switching switching firmware, please see my previous posting.

http://www.telecom-cafe.com/foru ... page%3D1&page=2

It is not difficult but unfortunately some tricks in between. You need to get ready a number of tools
作者: 亞星    時間: 2010-12-25 15:41

要用少少時間爬下文準備一下先, 如果有問題搞唔掂再向師兄請教
作者: ckleea    時間: 2010-12-25 15:43

No problem. Only now I do not have IP01 on hand. It is being shipped to UK.
作者: ckleea    時間: 2010-12-25 15:45

本帖最後由 ckleea 於 2010-12-25 15:46 編輯

Please also follow on the posts in that thread. There are a lot of follow up discussions. 3 IP01 have been tried to convert. 2 succeeded.
作者: 亞星    時間: 2010-12-25 17:02

師兄有無下面 firmware 既 download link 呀?

atcom 舊firmware e.g. 0.3.7,要 ext2版,不要 md5 版
switchfin 舊D firmware e.g. svn 37X,要md5 版
switchfin 最新firmware e.g. svn 438,要md5 版
作者: 角色    時間: 2010-12-25 17:43

听说switchfin稳定很多,所以晚一些我也把我的IP01转成switchfin。

角色
作者: ckleea    時間: 2010-12-25 18:05

回復 36# 亞星

Please PM your email, I will send you.
作者: ckleea    時間: 2010-12-25 18:06

回復 37# 角色
My latest switchfin firmware is SVN 432. So far it is stable. The last uptime was > 35 days.
作者: 亞星    時間: 2010-12-26 13:55

想問一下用 GUI 轉 switchfin 要用 HTTP URL 定係 TFTP Sever 呢?
作者: 亞星    時間: 2010-12-26 14:48

試咗幾次都唔成功, 要過幾日買到 console cable 先有得再試

圖片附件: GUI with TFTP.jpg (2010-12-26 14:48, 159.45 KB) / 下載次數 940
http://telecom-cafe.com/forum/attachment.php?aid=337&k=5d5359f3591f8676be1dc0e07f342939&t=1732551315&sid=8RexyE



圖片附件: Upgraded FAILED.jpg (2010-12-26 14:48, 149.52 KB) / 下載次數 867
http://telecom-cafe.com/forum/attachment.php?aid=338&k=687115c503cc99c0942822eb02ceb65b&t=1732551315&sid=8RexyE


作者: 亞星    時間: 2010-12-26 16:51

本帖最後由 亞星 於 2010-12-26 16:55 編輯

成功了
http://www.telecom-cafe.com/foru ... =2963&pid=10582
作者: ckleea    時間: 2010-12-26 17:25

回復 42# 亞星


    Congratulated to your success.
作者: 亞星    時間: 2010-12-26 20:01

本帖最後由 亞星 於 2010-12-26 20:20 編輯

下一步係 setup 隻 IP01(Switchfin), 當我乜都唔識(我用緊 SPA3000)由零開始我應該要 setup d 咩先呢?
作者: ckleea    時間: 2010-12-26 22:38

回復 44# 亞星

It is the same as usual asterisk. Either you do it via GUI or via APL. The latter is cleaner and more adapted to your own requirement.
作者: 亞星    時間: 2010-12-27 10:11

我第一步是否先建立用戶? 有那些參數要設定呢?
作者: lttliang    時間: 2010-12-27 10:45

我第一步是否先建立用戶? 有那些參數要設定呢?
亞星 發表於 2010-12-27 10:11



    先创建拨号方案  不过无所谓  你创建分机时 如果没有创建拨号方案  或提示你的
作者: 亞星    時間: 2010-12-27 13:04

先创建拨号方案  不过无所谓  你创建分机时 如果没有创建拨号方案  或提示你的 ...
lttliang 發表於 2010-12-27 10:45

是這兩個嗎? 有無例子可以參考和說明一下?

圖片附件: DP01.jpg (2010-12-27 13:04, 83.89 KB) / 下載次數 955
http://telecom-cafe.com/forum/attachment.php?aid=347&k=09eaeee7931d5c0b4cc996ca7bbbcfa3&t=1732551315&sid=8RexyE



圖片附件: CR01.jpg (2010-12-27 13:04, 96.11 KB) / 下載次數 890
http://telecom-cafe.com/forum/attachment.php?aid=348&k=80f853c96923f4e879874b6a080134db&t=1732551315&sid=8RexyE


作者: 亞星    時間: 2010-12-27 13:24

找到這個網頁, 跟住佢咁設定可以嗎?
http://wiki.osslab.org.tw/VoIP/I ... Addons/Asterisk_GUI
作者: lttliang    時間: 2010-12-27 13:47

一般是这样
1.创建分机有用的DialPlan,也可以创建只可内部打的DiualPlan
2.创建分机,每一个分机都可以选择你创建的DialPlan
3.创建trunk
4.创建out route
5.进入DialPlan添加你在第4创建的out route
6.创建incoming route
至于voicemenus  你可在incoming route指定
其他要你慢慢摸    我已经冇再用ip01了  所以冇法比你图示
作者: ckleea    時間: 2010-12-27 13:52

和lttliang 一樣,冇IP01試俾你知。
作者: 亞星    時間: 2010-12-27 14:01

多謝兩位師兄指點, 我初步用 iptel 已經成功 register, 其他的要一步一步再試

圖片附件: Registered.jpg (2010-12-27 14:00, 99.84 KB) / 下載次數 828
http://telecom-cafe.com/forum/attachment.php?aid=349&k=ee8f9c990a28a62cc0809a79cf303d88&t=1732551315&sid=8RexyE


作者: lttliang    時間: 2010-12-27 15:23

如果有人可以在ip01中加入打出或打入时  都可自动录音就好了   
Elastix有这个功能,之前我都有问过,但是冇人能解决
作者: ckleea    時間: 2010-12-27 15:57

回復 53# lttliang


Switchfin firmware has this function.
作者: lttliang    時間: 2010-12-27 16:22

本帖最後由 lttliang 於 2010-12-27 16:23 編輯
回復  lttliang


Switchfin firmware has this function.
ckleea 發表於 2010-12-27 15:57



    咁就不错
刚刚升级左freepbx  多左个custom Context
终于可以似asterisk-gui一样  指定不同分机使用不同DialPlan了,
我发觉freepbx是最有心做好gui的,经常会加新功能上去,以完善功能的不足
作者: bubblestar    時間: 2010-12-27 16:38

應該說的是,freepbx 仍有不足,所以Elastix的出現是補其缺漏。大家互補長短。

你可以看看,細分之下,Elastix 像是有兩個GUI 提供給用家使用一樣似的,既可用GUI,亦有custom context 的地方給大家自行寫入自己的DP或使用方式。
作者: lttliang    時間: 2010-12-27 16:58

本帖最後由 lttliang 於 2010-12-27 16:59 編輯
應該說的是,freepbx 仍有不足,所以Elastix的出現是補其缺漏。大家互補長短。

你可以看看,細分之下,Ela ...
bubblestar 發表於 2010-12-27 16:38



    Elastix主要是加左一D新功能  比如call center  IM等等(仲有好多)  所以我同你的看法都一样  只是補其缺漏,但elastix中PBX里100%的操作都是与联接到freepbx中的,就好像手机一样,freepbx就是手机上的按键,而Elastix就是在手机外面又套左一个自己牌子外观的壳,所以只要freepbx升级左之后,elastix pbx这一部份的操作与显示就都会同样发生变化(只是同步elastix PBX 中显示的内容,冇显示的部份仲需要进入freepbx)

所以我觉得elastix pbx 完全可以直接联到freepbx  而冇必要加多一层PBX GUI,不过可能这样做  可以方便AGI的对接吧  我也不太懂  只是发表一下个人的睇法
下面是我的图示
k.jpg

g.jpg

不过这样可能会侵权了  呵  毕竟是两大不同的gui公司

圖片附件: k.jpg (2010-12-27 16:58, 85.32 KB) / 下載次數 1047
http://telecom-cafe.com/forum/attachment.php?aid=350&k=9a0f0313dc160aa8f2bd5b1577d0a996&t=1732551315&sid=8RexyE



圖片附件: g.jpg (2010-12-27 16:58, 202 KB) / 下載次數 1045
http://telecom-cafe.com/forum/attachment.php?aid=351&k=585b2728fc94911af59bea97a6f478d7&t=1732551315&sid=8RexyE


作者: 亞星    時間: 2010-12-28 09:04

新手問題, IP01 作為一個獨立既 server 其實我係咪無需要向其他 SIP provider(例如 iptel) register?
我在 IP01 新建的 extenstions 是否要在各 SPA3000/3102 重新設定?
作者: bubblestar    時間: 2010-12-28 09:20

基本上是對的。

但IP01只能作私人SERVER,你雖然可以任意設定extension號碼給親友註冊使用,但若想要跟IP01以外的朋友溝通而又不方便提供自己的extension 給他,最好也註冊一些常用的VoIP Provider,例如Google Voice,iptel,ET293 之類,而呢D第三方Provider註冊資料,不再需要在SPA3000 等的ATA內註冊,全部可放在ip01 Server 便可以,一經註冊,所有由你授權的內線都可以照用,是無需逐部電話機SET。
作者: 亞星    時間: 2010-12-28 09:38

即是一個 iptel account 在 IP01 可以分配給 所有/指定 extensions 打出和打入使用(打入可以嗎)?
另外我建立了兩個家中的 extensions 和一個海外用的 extension, SPA3000/3102 如何設定才能把它們連上測試, 我是否要有一個給 IP01 的 Hostname?
作者: bubblestar    時間: 2010-12-28 10:01

一個iptel 帳戶也可以了。

接入的來電可以指定某一內線響及接聽,也可以兩條家中一齊響及接聽,甚至加埋海外個條extension 接聽也可以。

如果SERVER 在家中,那麼用內部IP Address 註冊本身Server 便足夠了。若是海外最好有Fixed IP 或 DDNS 給海外extension 連入。  實際上,只要用三個Parameters參數,基本上便已做好必要的設定了。
其他一些細微的調較,一般是話音質素的,之後再慢慢探討吧。

SPA3102.png

圖片附件: SPA3102.png (2010-12-28 10:01, 14.92 KB) / 下載次數 905
http://telecom-cafe.com/forum/attachment.php?aid=364&k=b132a76cef2887ce9947e5ac6a4f9f38&t=1732551315&sid=8RexyE


作者: bubblestar    時間: 2010-12-28 10:08

雖然已轉了Switchfin,但ATCOM本身的Manual 仍然好有用的,裡面也有一些基本使用設定,可供參考。

http://atcom.cn/download.html
作者: 亞星    時間: 2010-12-28 10:21

等我今晚收工返屋企試試先
另外想知 DP 係咪可以由 IP01 全權負責 SPA3000/3102 可以唔洗 set?
作者: bubblestar    時間: 2010-12-28 10:35

無錯。

但遇到有些個別情況時,例如你想使用http digest 功能遙距使用遠方的SPA3000/3102打當地PSTN電話時,或許需要在ATA上作一些設定配合。
作者: 亞星    時間: 2010-12-28 10:41

如果我 ATA 將 iptel 和 IP01 都 registered, 我要選擇用  iptel or IP01 打出電話是否要由 DP 決定?
作者: bubblestar    時間: 2010-12-28 11:41

如果在SPA3102兩條線之中,Line 1 註冊給 IPTEL,它會變成獨立線,所以由Line 1 DP 決定。  若Line 2 入面註冊的是IP 01 入面設定好的IPTEL帳戶,它會由IP01 DP決定。

建議你最好申請兩條iptel 線,一條作獨立後備用,一條給IP01 用。儘量不要一線由多個DP控制,一亂起上來,找源頭較煩。
作者: 亞星    時間: 2010-12-28 13:59

剛才試了公司的 SPA3000 連不上家中的 IP01, 不知道甚麼地方出錯

圖片附件: Failed.jpg (2010-12-28 13:59, 19 KB) / 下載次數 921
http://telecom-cafe.com/forum/attachment.php?aid=365&k=44b5bc30ee5cf51156329474205a7c7b&t=1732551315&sid=8RexyE



圖片附件: SPA3000 setting.jpg (2010-12-28 13:59, 61.64 KB) / 下載次數 866
http://telecom-cafe.com/forum/attachment.php?aid=366&k=9a87ffea15841d8b444ed0ac51c32013&t=1732551315&sid=8RexyE



圖片附件: Extensions.jpg (2010-12-28 13:59, 72.15 KB) / 下載次數 846
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圖片附件: Extensions setting.jpg (2010-12-28 13:59, 95.97 KB) / 下載次數 879
http://telecom-cafe.com/forum/attachment.php?aid=368&k=1a1f394656581220c68d42e5ee8a23c3&t=1732551315&sid=8RexyE


作者: ckleea    時間: 2010-12-28 14:03

Check your router if port 5060 is open or not. You may need to enable port redirection at the router.
Normally, your SPA3012 should connect.

Please also check if you have reload the users setting you just create in IP01
作者: 亞星    時間: 2010-12-28 14:15

本帖最後由 亞星 於 2010-12-28 14:58 編輯

Router 已經 set port redirection 5060 去 IP01, users setting 都 reload 過, Proxy set 左用屋企 IP 都係唔得...

暫時 remote 改左屋企隻 SPA3012 connect 到 IP01

圖片附件: Home SPA3102.jpg (2010-12-28 14:58, 71.35 KB) / 下載次數 835
http://telecom-cafe.com/forum/attachment.php?aid=369&k=77739e2f6f48493d69f9a07b48499b7c&t=1732551315&sid=8RexyE


作者: bubblestar    時間: 2010-12-28 14:59

IP01 裡面的 PBX features >> Outgoing Calling Rules / Incoming Calling Rules / Dial Plans  設定好味?
作者: ckleea    時間: 2010-12-28 15:22

回復 69# 亞星

You need the real IP not the internal lan IP. Will you use dynamic DNS host for simpler management?
作者: 亞星    時間: 2010-12-28 15:24

我設定正確嗎?

圖片附件: Outgoing call Rule.jpg (2010-12-28 15:24, 66.96 KB) / 下載次數 867
http://telecom-cafe.com/forum/attachment.php?aid=370&k=a3f2500a4e66328bbc4665f9e57cce8e&t=1732551315&sid=8RexyE



圖片附件: Incoming Call rule.jpg (2010-12-28 15:24, 56.78 KB) / 下載次數 838
http://telecom-cafe.com/forum/attachment.php?aid=371&k=7a7c59a8aa484f86c350d06a39fcf195&t=1732551315&sid=8RexyE



圖片附件: DialPlan.jpg (2010-12-28 15:24, 60.42 KB) / 下載次數 835
http://telecom-cafe.com/forum/attachment.php?aid=372&k=65b6f737e2ee6348c6c0d6a30ab97079&t=1732551315&sid=8RexyE


作者: 亞星    時間: 2010-12-28 15:28

回復  亞星

You need the real IP not the internal lan IP. Will you use dynamic DNS host for simpler ...
ckleea 發表於 2010-12-28 15:22

真 IP 和 DDNS 都試過, 真 IP: Port 和 DDNS: Port 都試過全部都唔得...
作者: ckleea    時間: 2010-12-28 15:31

Can you try a software client to remote connect your IP01? Please get the log message from connection.
作者: ckleea    時間: 2010-12-28 15:33

回復 72# 亞星


    These setting governs your incoming and outgoing calls. Not related to user extension connection.
作者: 亞星    時間: 2010-12-28 15:50

Can you try a software client to remote connect your IP01? Please get the log message from connectio ...
ckleea 發表於 2010-12-28 15:31

試左都係唔得

圖片附件: X-lite.jpg (2010-12-28 15:49, 58.18 KB) / 下載次數 869
http://telecom-cafe.com/forum/attachment.php?aid=373&k=f7b77d44b92426e2c8b5d269c923a3b3&t=1732551315&sid=8RexyE



圖片附件: X-lite Setting.jpg (2010-12-28 15:49, 57.93 KB) / 下載次數 855
http://telecom-cafe.com/forum/attachment.php?aid=374&k=a15d2e5e9a36075fbe6d26c48ed45583&t=1732551315&sid=8RexyE


作者: ckleea    時間: 2010-12-28 17:48

I have not yet found a solution in your case but please try to reflash again the firmware and allow a full configuration reset.
  1. [Dec 28 16:50:14] WARNING[17267] res_smdi.c: No SMDI interfaces were specified to listen on, not starting SDMI listener.
  2. [Dec 28 16:50:24] NOTICE[375] chan_sip.c: Peer '6002' is now UNREACHABLE!  Last qualify: 20
  3. [Dec 28 16:50:24] WARNING[17267] pbx_config.c: The use of '_.' for an extension is strongly discouraged and can have unexpected behavior.  Please use '_X.' instead at line 244
  4. [Dec 28 16:50:24] WARNING[17267] pbx_config.c: The use of '_.' for an extension is strongly discouraged and can have unexpected behavior.  Please use '_X.' instead at line 245
  5. [Dec 28 16:50:24] WARNING[17267] pbx_config.c: The use of '_.' for an extension is strongly discouraged and can have unexpected behavior.  Please use '_X.' instead at line 246
  6. [Dec 28 16:50:24] WARNING[17267] pbx_config.c: The use of '_.' for an extension is strongly discouraged and can have unexpected behavior.  Please use '_X.' instead at line 247
  7. [Dec 28 16:50:24] WARNING[17267] pbx_config.c: The use of '_.' for an extension is strongly discouraged and can have unexpected behavior.  Please use '_X.' instead at line 248
  8. [Dec 28 16:50:24] WARNING[17267] res_musiconhold.c: Cannot open dir /var/lib/asterisk/moh or dir does not exist
  9. [Dec 28 16:50:25] NOTICE[17267] indications.c: Removed default indication country 'us'
  10. [Dec 28 16:51:05] NOTICE[375] chan_sip.c: Peer '6002' is now Reachable. (114ms / 2000ms)
  11. [Dec 28 16:51:05] NOTICE[375] chan_sip.c: Peer '6001' is now Reachable. (58ms / 2000ms)
  12. [Dec 28 16:51:24] WARNING[17458] app_dial.c: Unable to create channel of type 'IAX2' (cause 20 - Unknown)
  13. [Dec 28 16:51:52] WARNING[375] chan_sip.c: Maximum retries exceeded on transmission MzFkOTIwMzA3NDk4MTdkMzFkYmMxMTFiMWIwYTc0Y2U. for seqno 2 (Critical Response) -- See doc/sip-retransmit.txt.
  14. [Dec 28 16:51:52] WARNING[375] chan_sip.c: Hanging up call MzFkOTIwMzA3NDk4MTdkMzFkYmMxMTFiMWIwYTc0Y2U. - no reply to our critical packet (see doc/sip-retransmit.txt).
  15. [Dec 28 16:51:58] WARNING[17519] app_dial.c: Unable to create channel of type 'IAX2' (cause 20 - Unknown)
  16. [Dec 28 16:52:09] WARNING[17546] app_dial.c: Unable to create channel of type 'IAX2' (cause 20 - Unknown)
  17. [Dec 28 16:52:17] WARNING[17571] app_dial.c: Unable to create channel of type 'IAX2' (cause 20 - Unknown)
  18. [Dec 28 16:53:51] WARNING[17668] app_dial.c: Unable to create channel of type 'IAX2' (cause 20 - Unknown)
  19. [Dec 28 16:54:36] NOTICE[375] chan_sip.c: Peer '6001' is now Reachable. (57ms / 2000ms)
  20. [Dec 28 16:54:41] NOTICE[375] chan_sip.c: Peer '6002' is now Reachable. (55ms / 2000ms)
  21. [Dec 28 16:54:46] WARNING[17832] app_dial.c: Unable to create channel of type 'IAX2' (cause 20 - Unknown)
  22. [Dec 28 16:54:54] NOTICE[375] chan_sip.c: Auto-congesting SIP/6002-00000014
  23. [Dec 28 16:55:03] WARNING[17867] app_dial.c: Unable to create channel of type 'IAX2' (cause 20 - Unknown)
  24. [Dec 28 16:55:11] NOTICE[375] chan_sip.c: Auto-congesting SIP/6001-00000016
  25. [Dec 28 16:55:40] NOTICE[375] chan_sip.c: Peer '6001' is now UNREACHABLE!  Last qualify: 57
  26. [Dec 28 16:55:45] NOTICE[375] chan_sip.c: Peer '6002' is now UNREACHABLE!  Last qualify: 55
  27. [Dec 28 16:55:46] NOTICE[17986] cdr.c: CDR simple logging enabled.
  28. [Dec 28 16:55:46] WARNING[17986] res_smdi.c: No SMDI interfaces were specified to listen on, not starting SDMI listener.
  29. [Dec 28 16:55:47] WARNING[17986] pbx_config.c: The use of '_.' for an extension is strongly discouraged and can have unexpected behavior.  Please use '_X.' instead at line 244
  30. [Dec 28 16:55:47] WARNING[17986] pbx_config.c: The use of '_.' for an extension is strongly discouraged and can have unexpected behavior.  Please use '_X.' instead at line 245
  31. [Dec 28 16:55:47] WARNING[17986] pbx_config.c: The use of '_.' for an extension is strongly discouraged and can have unexpected behavior.  Please use '_X.' instead at line 246
  32. [Dec 28 16:55:47] WARNING[17986] pbx_config.c: The use of '_.' for an extension is strongly discouraged and can have unexpected behavior.  Please use '_X.' instead at line 247
  33. [Dec 28 16:55:47] WARNING[17986] pbx_config.c: The use of '_.' for an extension is strongly discouraged and can have unexpected behavior.  Please use '_X.' instead at line 248
  34. [Dec 28 16:55:47] WARNING[17986] res_musiconhold.c: Cannot open dir /var/lib/asterisk/moh or dir does not exist
  35. [Dec 28 16:55:47] NOTICE[17986] indications.c: Removed default indication country 'us'
  36. [Dec 28 16:56:20] NOTICE[375] chan_sip.c: Peer '6002' is now Reachable. (47ms / 2000ms)
  37. [Dec 28 16:56:22] NOTICE[375] chan_sip.c: Peer '6001' is now Reachable. (51ms / 2000ms)
  38. [Dec 28 16:56:36] WARNING[18135] app_dial.c: Unable to create channel of type 'IAX2' (cause 20 - Unknown)
  39. [Dec 28 16:56:45] WARNING[18162] app_dial.c: Unable to create channel of type 'IAX2' (cause 20 - Unknown)
  40. [Dec 28 16:57:24] NOTICE[375] chan_sip.c: Peer '6002' is now UNREACHABLE!  Last qualify: 47
  41. [Dec 28 16:57:28] NOTICE[375] chan_sip.c: Peer '6001' is now Reachable. (50ms / 2000ms)
  42. [Dec 28 16:57:45] WARNING[18259] app_dial.c: Unable to create channel of type 'SIP' (cause 20 - Unknown)
  43. [Dec 28 16:57:45] WARNING[18259] app_dial.c: Unable to create channel of type 'IAX2' (cause 20 - Unknown)
  44. [Dec 28 16:57:50] WARNING[18272] app_dial.c: Unable to create channel of type 'SIP' (cause 20 - Unknown)
  45. [Dec 28 16:57:50] WARNING[18272] app_dial.c: Unable to create channel of type 'IAX2' (cause 20 - Unknown)
  46. [Dec 28 16:58:32] NOTICE[375] chan_sip.c: Peer '6001' is now UNREACHABLE!  Last qualify: 50
  47. [Dec 28 16:58:52] NOTICE[375] chan_sip.c: Peer '6001' is now Reachable. (55ms / 2000ms)
  48. [Dec 28 16:59:16] WARNING[18481] app_dial.c: Unable to create channel of type 'IAX2' (cause 20 - Unknown)
  49. [Dec 28 16:59:24] NOTICE[375] chan_sip.c: Auto-congesting SIP/6001-0000001e
  50. [Dec 28 16:59:56] NOTICE[375] chan_sip.c: Peer '6001' is now UNREACHABLE!  Last qualify: 55
  51. [Dec 28 17:00:00] NOTICE[375] chan_sip.c: No compatible codecs, not accepting this offer!
  52. [Dec 28 17:03:16] NOTICE[18762] cdr.c: CDR simple logging enabled.
  53. [Dec 28 17:03:16] WARNING[18762] res_smdi.c: No SMDI interfaces were specified to listen on, not starting SDMI listener.
  54. [Dec 28 17:03:17] WARNING[18762] pbx_config.c: The use of '_.' for an extension is strongly discouraged and can have unexpected behavior.  Please use '_X.' instead at line 247
  55. [Dec 28 17:03:17] WARNING[18762] pbx_config.c: The use of '_.' for an extension is strongly discouraged and can have unexpected behavior.  Please use '_X.' instead at line 248
  56. [Dec 28 17:03:17] WARNING[18762] pbx_config.c: The use of '_.' for an extension is strongly discouraged and can have unexpected behavior.  Please use '_X.' instead at line 249
  57. [Dec 28 17:03:17] WARNING[18762] pbx_config.c: The use of '_.' for an extension is strongly discouraged and can have unexpected behavior.  Please use '_X.' instead at line 250
  58. [Dec 28 17:03:17] WARNING[18762] pbx_config.c: The use of '_.' for an extension is strongly discouraged and can have unexpected behavior.  Please use '_X.' instead at line 251
  59. [Dec 28 17:03:17] WARNING[18762] res_musiconhold.c: Cannot open dir /var/lib/asterisk/moh or dir does not exist
  60. [Dec 28 17:03:17] NOTICE[18762] indications.c: Removed default indication country 'us'
  61. [Dec 28 17:04:28] NOTICE[375] chan_sip.c: Peer '6001' is now Reachable. (53ms / 2000ms)
  62. [Dec 28 17:04:29] NOTICE[375] chan_sip.c: Peer '6002' is now UNREACHABLE!  Last qualify: 0
  63. [Dec 28 17:04:56] NOTICE[375] chan_sip.c: Peer '6002' is now Reachable. (10ms / 2000ms)
  64. [Dec 28 17:04:57] NOTICE[375] chan_sip.c: Peer '6002' is now Reachable. (10ms / 2000ms)
  65. [Dec 28 17:05:07] WARNING[375] chan_sip.c: Maximum retries exceeded on transmission NDUxODY3MTE0YmVjOTE3ZTNlMWY5ZjY2MGRiZWViMDc. for seqno 2 (Critical Response) -- See doc/sip-retransmit.txt.
  66. [Dec 28 17:05:07] WARNING[375] chan_sip.c: Hanging up call NDUxODY3MTE0YmVjOTE3ZTNlMWY5ZjY2MGRiZWViMDc. - no reply to our critical packet (see doc/sip-retransmit.txt).
  67. [Dec 28 17:05:32] NOTICE[375] chan_sip.c: Peer '6001' is now UNREACHABLE!  Last qualify: 53
  68. [Dec 28 17:06:21] NOTICE[375] chan_sip.c: No compatible codecs, not accepting this offer!
  69. [Dec 28 17:06:35] NOTICE[375] chan_sip.c: Peer '6001' is now Reachable. (56ms / 2000ms)
  70. [Dec 28 17:06:51] WARNING[19568] app_dial.c: Unable to create channel of type 'IAX2' (cause 20 - Unknown)
  71. [Dec 28 17:07:01] NOTICE[375] chan_sip.c: No compatible codecs, not accepting this offer!
  72. [Dec 28 17:07:06] NOTICE[375] chan_sip.c: No compatible codecs, not accepting this offer!
  73. [Dec 28 17:07:39] NOTICE[375] chan_sip.c: Peer '6001' is now UNREACHABLE!  Last qualify: 56
  74. [Dec 28 17:08:13] NOTICE[375] chan_sip.c: No compatible codecs, not accepting this offer!
  75. [Dec 28 17:14:53] NOTICE[351] chan_sip.c: Peer 'vr2ilhome' is now Reachable. (334ms / 2000ms)
  76. [Dec 28 17:15:28] NOTICE[351] chan_sip.c: Peer '6001' is now Reachable. (52ms / 2000ms)
  77. [Dec 28 17:15:56] NOTICE[351] chan_sip.c: Peer '6002' is now Reachable. (76ms / 2000ms)
複製代碼

作者: bubblestar    時間: 2010-12-28 20:20

TO: 阿星

請檢查一下
1.  DP internal 有沒有剔選/包括iptel ??
2.  Incoming Call Rule 不要用 s ,用返自己IPTEL 號碼 _101234
3.  Extension insecure 一欄,選用 insecure = very
作者: 亞星    時間: 2010-12-28 20:24

已經重新 reflash 過, 等 ckleea 兄先幫我 remote 檢查一下
作者: 亞星    時間: 2010-12-30 21:22

始終都係要交隻 IP01 俾 ckleea 兄搞
作者: ckleea    時間: 2010-12-30 21:53

你的network setting 有D奇怪!
作者: ckleea    時間: 2010-12-30 21:57

Just reset the configuration, 已進入GUI
作者: ckleea    時間: 2010-12-30 22:13

Completed configuration. The problems are
1. network is set as default for 192.168.1.100, hence on a different lan, one can't identify the IPO1 box
2. somehow the /persistent/etc/network.conf is messed up. A reset of configuration works.

The audio problem has resolved.
作者: 亞星    時間: 2010-12-30 22:17

咁快搞掂, 唔該晒
作者: ckleea    時間: 2010-12-30 22:25

本帖最後由 ckleea 於 2010-12-30 22:28 編輯

Please see this page for the update.

http://www.telecom-cafe.com/foru ... age%3D1&page=14
作者: ckleea    時間: 2010-12-30 22:29

回復 84# 亞星

你個IP01 box quality 好似唔錯。
作者: 亞星    時間: 2010-12-30 22:49

國貨黎講外売手工算唔錯
作者: bubblestar    時間: 2010-12-31 12:00

回復 86# ckleea


   
可能每一個生產批次的材料都有些少分別。最緊要個內含夠壯健,最好是一個 BAD BLOCK 都無就至佳。
作者: ckleea    時間: 2011-1-3 11:15

回復 88# bubblestar


    我覺得呢批IP01唔錯,起碼少D Bad NAND RAM Block,另外ATCOM GUI 可以直上switchfin
作者: 亞星    時間: 2011-1-3 11:43

都過左年應該回復正價
作者: ckleea    時間: 2011-1-3 17:48

回復 84# 亞星


Your device works for 2 days.

圖片附件: Status.png (2011-1-3 17:48, 64.84 KB) / 下載次數 1020
http://telecom-cafe.com/forum/attachment.php?aid=397&k=c794b5571d7cb584fac1f4a9a9713bc2&t=1732551315&sid=8RexyE


作者: Qnewbie    時間: 2011-1-3 17:56

Wait for IP01, which should be arrived next week.
作者: ckleea    時間: 2011-1-3 21:39

回復 91# ckleea


已完成跟進,在運作中。
發覺新的firmware有D 唔同,係GUI,networking setting 要DHCP = yes, 唔可以 set auto;否則就變回憶192.168.1.100
如果入唔到GUI,就要改 呢兩個 files
/persistent/etc/network.conf
/persistent/etc/asteisk/rc_org.conf
作者: 亞星    時間: 2011-1-4 17:37

本帖最後由 亞星 於 2011-1-4 17:41 編輯

啱啱收到 ATCOM Sale 個 email, 一個關心客戶使用 atcom 產品情況的 email
令人覺得 ATCOM 售後服務真的不錯.
作者: bubblestar    時間: 2011-1-4 17:59

ckleea 兄的無私奉獻及幫忙,也不能不提。沒有他幫手,我地大家沒有咁多野玩,也沒有咁多野學。
作者: 亞星    時間: 2011-1-4 21:05

Support
作者: ckleea    時間: 2011-1-4 21:08

回復 94# 亞星

No need to tell them you have made the changes.

The hardware itself is good. Your price is really excellent. But ATCOM GUI is   
   
Another good GUI is from Spotel.

http://www.nicherons.com/ippbx08.html

I have been seriously looked into the product. Very well. Only too expensive and too many ports for me.
作者: ckleea    時間: 2011-1-4 21:10

回復 96# 亞星

Your another box is ready to get. How is your box at home?

I don't know why the DHCP client at latest switchfin firmware is less stable as before.
作者: 亞星    時間: 2011-1-7 10:27

本帖最後由 亞星 於 2011-1-7 10:31 編輯

用 SPA3000 可以 register 同埋見到轉綠燈, 但用隻 AT610 IP Phone 見到 registered 但無轉綠燈

圖片附件: Extension.jpg (2011-1-7 10:27, 78.27 KB) / 下載次數 996
http://telecom-cafe.com/forum/attachment.php?aid=401&k=4ae7c8528981552fa155b2556058b26a&t=1732551315&sid=8RexyE



圖片附件: AT610.jpg (2011-1-7 10:31, 67.08 KB) / 下載次數 1003
http://telecom-cafe.com/forum/attachment.php?aid=402&k=381673f760eae00066e8218aab803f5f&t=1732551315&sid=8RexyE


作者: bubblestar    時間: 2011-1-7 11:46

回復 99# 亞星


   
印象中,ATCOM本身不建議用第一條extension 6000 做 registration 的,因為它預設給了auto-attendant 使用,查找之下發覺它的MANUAL真的是這麼說的。
雖然你現在改用了Switchfin,但也試試用6002再看看怎麼樣。

6000.png

圖片附件: 6000.png (2011-1-7 11:46, 79.11 KB) / 下載次數 988
http://telecom-cafe.com/forum/attachment.php?aid=403&k=f87f59bb6f27a93a519cb683458cafd9&t=1732551315&sid=8RexyE






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