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• 如何利用幻觉把星号与天线连接起来?
原标题:How to connect asterisk to skype using siptosis?

我在几个星期里一直试图这样做,但还未说明如何连接天空。 我发现的最佳进展来自这一辅导。

http://www.voipandhack.it/archives/linux/asterisk-failover-e-registrazioni-sip&ei=nqJRTNTPB8OfrAfDovGDAw&sa=X&outolate&transct=transresult&

但每当我试图发出响应123的呼吁时,我的星号就会显示。

== Using SIP RTP CoS mark 5 -- Executing [*123@phones:1] Dial("SIP/1004-00000030", "SIP/siptosisuser/echo123") in new stack == Using SIP RTP CoS mark 5 -- Called siptosisuser/echo123 -- SIP/siptosisuser-00000031 is ringing -- SIP/siptosisuser-00000031 is circuit-busy == Everyone is busy/congested at this time (1:0/1/0) -- Auto fallthrough, channel SIP/1004-00000030 status is CONGESTION

我的预测表明:

2010-07-30 10:48:21,596 Failed to select RTP format 2010-07-30 10:48:21,597 ### local descriptor=v=0 o=skypests 1280512101 0 IN IP4 192.168.. s=Session SIP/SDP c=IN IP4 192.168.. t=0 0 m=audio 63202 RTP/AVP 0 8 98 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:98 iLBC/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv a=silenceSupp:off

2010-07-30 10:48:21,597 ### remote descriptor=v=0 o=root 1729829715 1729829715 IN IP4 127.0.0.1 s=Asterisk PBX 1.6.2.8 c=IN IP4 127.0.0.1 t=0 0 m=audio 18104 RTP/AVP 3 101 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv

我先把我的IP地址改为192.168..,只是出于安全原因。 这似乎以某种方式连接起来,但我假定我一米在法典中做了一些错误的事情吗? 我也避开了推论。 单单还看了我下载的扫描包裹,但我得以在温得战网站上找到另一门辅导的复印件。

这证明是便于设置的辅导员,但似乎在排雷方面工作。 我只想呼吁响应123,似乎没有工作。 我在乌本图和大多数辅导中工作的人都在西里,因此解决我的问题并不多。 我也没有使用静态天空,这将是一个问题吗?

任何背后者/下水道都会受到非常赞赏!

感谢你的时间,并提前感谢你!

问题回答

我不熟悉Sip ToSis,但你可能想看SkypeLink(以前称为SIP的Skype)解决方案,使你能够直接使用SIP来接听电话。 与Asterisk的Skype不同,它是自由的。

如果你已经知道这一点,并且正在寻找Sip ToSis的具体特征,SkypeLink就没有提供这些特征。 Good luck.





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