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【角色茶桌】—— Skype to Asterisk 的重要性

如果大家经常在外,特别在一些国家的VoIP禁止的地方,5060经常会给当地的电信局封杀,所以拨打不同,但是可以上网和上Skype,很少听说Skype会被block,主要Skype本身的穿透力是非常强。Skype在全球有很多个Servers,如果常用的port,如5060被block,它可以用port 80,443,或者其他ports,IP address也能走位,所以很少听说过某些软件能封杀Skype。

如果你在一些VoIP被禁止的地方,可以使用Skype拨打别的Skypename,作用是通过Skype而离开被禁止VoIP的地方,然后Skype to Asterisk Gateway,再拨打其他电话。还值得注意的地方时Skype to Skype 的bangwidth要求很低都能有不错的语音通讯效果。

角色

One benefit for Skype is that most 3G-ISPs treat Skype as data-traffic not VOIP traffic. VOIP traffic is counted as voice and hence much expensive than data-traffic.

TOP

I have asked this question to my colleague that he also agreed that Skype is not treated as VoIP but data flow only.

YH

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回復 3# 角色


    How is your skype trunk reliably? If you play along long, sometimes it may disappear.

TOP

Yeah exactly as you experienced. Once I call the Skype trunk, it turns on again.

YH

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大家不要看少Skype-to-Asterisk,起码你可以某些地方走出来,而且效果也非常不错!

TOP

角色兄!!不知道你有無留意呢隻野

http://www.soundwin.com/zh-tw/1_ ... 9B%92_id119828.html

我台灣出品的!!

TOP

好似唔係好work

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回復 7# dreamy2k

這個我有一對, 最初是不太穩定, 不過漸趨穩定, 適合像我這些用IP01或NAS做Asterisk的人用!
Welcome to my TaoBao shop: http://mandymak520.taobao.com/

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回復 9# 雯雯

我想問下呢隻野有無解決SKYPE -> SP110 DMTF的問題??

因為我自己系家做了一個環境將FREETALK FXS 連了去 SIP ATA FXO上,如果用SIP PHONE 打去ATA FXO上之后跳去FREETALK再撥出去SKYPE系無問題,但問題系SKYPE打去FREETALK之后想轉去SIP SERVER上的分機就吾得!!之后找來找去先知道系DMTF出了問題

我想問下SP110有無解決呢個問題呀

TOP

We use siptosis bridge skype <--> asterisk. Working quite well

Sound quality in general better than VOIP.

DTMF in general no problem in our setup

TOP

TOP

I have used the software siptossis for a number of times on both Windows and Linux platforms. The voice quality is pretty good.

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回復 13# 角色


Unfortunately, the development and support for siptosis has stopped.

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回復 10# dreamy2k

之前試過DTMF是不太好, 最近比較忙沒有再試.
Welcome to my TaoBao shop: http://mandymak520.taobao.com/

TOP

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