原帖由 Mr.Tom 於 2009-2-3 15:26 發表

不會是派發給員工的尾糧掛

香港雷曼好似都係用call manager

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The existing config in my house:
cisco 7961G(custom XML config w/sip fw)<--LAN-->SIP PROXY(ASTERISK)<--SIP trunk cross WAN-->IpTel(registered with user1@iptel.org)
Incoming Call Test :
user2@iptel.org dial to user1@iptel.org, 7961 will ring

Outgoing Call Test :
7961 dial 31001,asterisk remove fist digit and forword the call to 1001@iptel.org(iptel's voice menu)

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原帖由 yhfung 於 2009-2-3 18:03 發表
刚才有一幅帖子说在香港有很多VoIP公司倒闭!没有办法,很多PSTN如PCCW拿了很多资源,很多接入香港的PSTN都会经过它(如果我错,请大家提出),但是我始终认为将来是SIP Phone的天下。PSTN电话将会成为博物馆的事务。

大家想想,以 ...

you are misunderstinding the meaning of pstn, pls check what "pstn" stand for first. Then u will know why nothing can replace it.

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原帖由 yhfung 於 2009-2-3 21:08 發表


Thank you for providing the use of iptel and SIP phones, both SIP phones can call each other with the Asterisk IP PBX. However, without using Asterisk IP PBX, your example can be realised by just us ...

Here is just an example to show you how the call be made from SIP proxy to outside line, and vice versa. You can change or add more outside line to asterisk, and let the proxy to choose the best line to place your call. Pls think about it, you have a direct SIP trunk(cheaper cost)  to US via internet and a traditional line to PSTN via  carrier(eg pccw), when the internet connection is broken, you can still place the call via traditional without any dialing method change at user side. This is just one of all advantages for SIP proxy, for more feature about IP PBX, e.g. voicemail, voice menu, conference bridge and group pickup please google it.

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已開支付寶戶口,但好似無信用卡增值,大家有咩方法?

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原帖由 yhfung 於 2009-2-8 23:33 發表
PSTN的未来是。。。

1、请看图一,我们日常打电话,一般在同一国家,打地区号,然后继续拨打你所需要的电话号码。如果不同国家,前面加上国家号码。

图一
799012

2、请看图二,现在增多,出现很多的VoIP公司,它们的客户内网是免 ...

不可能,當愈來愈多人使用網絡傳送影音,ISP公司出海的頻寬要求愈高,成本增加會直接反影於上網的月費上,同時,VOIP對網絡延遲十分敏敢,ISP要同時提供低延遲和高數據量傳送,對自身網絡的LOADING要求很高, 以現今ISP的網絡架構,是不足以提供保證低延遲的出海網絡服務

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pls pm, thx

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