如何使用 siptosis 将 asterisk 连接到 Skype？
我已经尝试这样做几个星期了，但仍然没有弄清楚如何连接到 Skype。我发现的最佳进展来自本教程
但是每当我尝试进行 echo123 呼叫时，我的星号就会显示
== 使用 SIP RTP CoS 标记 5 -- 在新堆栈中执行 [*123@phones:1] Dial("SIP/1004-00000030", "SIP/siptosisuser/echo123") == 使用 SIP RTP CoS 标记 5 -- 称为 siptosisuser/echo123 -- SIP/siptosisuser-00000031 正在响铃 -- SIP/siptosisuser-00000031 线路繁忙 ==此时大家都很忙/拥挤(1:0/1/0) -- 自动失败，通道 'SIP/1004-00000030' 状态为 'CONGESTION'
，我的吸血现象将显示：
2010-07-30 10:48:21,596 无法选择 RTP 格式 2010-07-30 10:48:21,597 ### 本地描述符=v=0 o=skypests 1280512101 0 IN IP4 192.168.. s=会话 SIP/SDP c=IN IP4 192.168.. t=0 0 m=音频 63202 RTP/AVP 0 8 98 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8PCMA/8000 a=rtpmap:98 iLBC/8000 a=rtpmap:101 电话事件/8000 a=fmtp:101 0-15 a=发送接收 a=silenceSupp:off
2010-07-30 10:48:21,597 ### 远程描述符=v=0 o=根 1729829715 1729829715 在 IP4 127.0.0.1 s=星号 PBX 1.6.2.8 c=IP4 127.0.0.1 t=0 0 m=音频 18104 RTP/AVP 3 101 a=rtpmap:3 GSM/8000 a=rtpmap:101 电话事件/8000 a=fmtp:101 0-16 a=silenceSupp:关闭 - - - - a=时间:20 a=sendrecv
出于安全原因，我已将 IP 地址更改为 192.168。.。它似乎以一种或另一种方式连接，但我假设我在代码中的某个地方做错了什么？我也没有接触 siptosis.cfg，因为我指定的教程没有做到这一点。我下载的 siptosis 包中也没有 cfg 文件，但我可以从 nerdvittle 网站上的另一个教程中找到副本
这应该是一个可以轻松设置的教程，但它似乎不起作用在我的身上。我只想打电话给 echo123，但似乎不起作用。我正在 ubuntu 上工作，大多数教程都是在 CentOS 上，所以没有太多可以解决我的问题。我也没有使用静态 Skype，这会有问题吗？
任何提示/提示/答案将非常感激！
感谢您的宝贵时间，并提前致谢！
I've been trying to do this for weeks and still have not yet figured out how to connect to skype. The best progress I've found out was from this tutorial
But whenever I tried to make an echo123 call, my asterisk would show
== 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'
and my siptosis will show:
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
I've changed my IP address to 192.168.. just for security reasons. It seems to be connecting one way or another but I'm assuming I'm doing something wrong somewhere in the code? I also haven't touched the siptosis.cfg because the tutorial I specified didn't do it. The cfg file also didn't come with the siptosis package I downloaded but I was able to find a copy from another tutorial on the nerdvittle website
This is suppose to be the tutorial that makes it easy to setup but it doesn't seem to work on mine. I just want to have some call to echo123 and it seemed to be not working. I'm working on ubuntu and most tutorials are on CentOS so there's not much to solve my problem. I also am not using static skype, will that be a problem?
Any hints/tip/answers would be very much appreciated!
Thank you for your time and thank you in advance!
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我不熟悉 SipToSis，但您可能想了解一下 Skype Connect（以前称为 Skype for SIP）解决方案，该解决方案允许您使用 SIP 直接向 sip.skype.com 拨打和接听电话。与 Skype for Asterisk 不同，它是免费的。
如果您已经了解这一点并且正在寻找 Skype Connect 未提供的 SipToSis 特定功能，我们深表歉意。祝你好运。
I'm not familiar with SipToSis but you may want to look at the Skype Connect (previously known as Skype for SIP) solution which allows you to make and receive calls direct to/from sip.skype.com using SIP. Unlike Skype for Asterisk, it's free.
Apologies if you already know this and are looking for specific features of SipToSis which are not provided by Skype Connect. Good luck.