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93 lines
2.9 KiB
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93 lines
2.9 KiB
Markdown
---
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title: Skype ⇔ SIP Interlink
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layout: default
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created: 2009-05-17 13:06:06 +0200
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updated: 2009-05-17 13:06:06 +0200
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toc: false
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tags:
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- know-how
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- software
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- sip
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- skype
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- conference
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---
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To connect Skype to a SIP-PBX (e.g. Asterisk), there's only one free application available:
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* [SipToSis](http://www.mhspot.com/sts/siptosis.html) (former *SippySkype*)
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There are also some (Windows-)apps for money, like:
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* [Uplink Skype to SIP](http://www.nch.com.au/skypetosip/)
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* [PSGw](http://www.rsdevs.com/psgw.shtml)
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Since you can't integrate these apps into Asterisk, you need to add an extension for each Skype account (1 per PC).
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If somebody calls via Skype, you can let SipToSis dial a specific number (internal/external) via the Asterisk PBX. Most
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probably you want it to dial your own SIP extension.
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If you call the Skype extension, you can let SipToSis dial a specific Skype contact. Maybe you could add another
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virtual PBX so that you could dial `<SkypeContact>@<SipToSis-IP>` on your VoIP-phone to call a specific Skype contact -
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but your VoIP-phone then needs to be registered with this virtual PBX.
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SipToSis (SippySkype)
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=====================
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Since this program is free and written in Java, it's the perfect choice for now. Configuration might be a bit tricky
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though.
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Configuration
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-------------
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In this example, there are following values:
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| Value | Description |
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|:--------------:|:---------------|
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|192.168.1.162 |IP of the PC running Skype and SipToSis|
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|192.168.1.245 |IP of the Asterisk PBX |
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|88 |SipToSis-extension on the Asterisk PBX |
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|44@192.168.1.212|Extension and IP of the VoIP-phone to use for incoming Skype calls|
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### siptosis.cfg (former sippyskype.cfg)
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{% highlight ini %}
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via_addr=192.168.1.162
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host_port=5060
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contact_url=sip:88@192.168.1.245:5060
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from_url="Skype Gateway" <sip:88@192.168.1.245:5060>
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username=88
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realm=asterisk
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passwd=skype
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do_register=yes
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{% endhighlight %}
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So this general description should work:
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|Setting |Note |
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|:------------|:-------------|
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|`via_addr` |IP of SipToSis/SippySkype|
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|`host_port` |Desired Port of SipToSis |
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|`contact_url`|`sip:`*Asterisk-Skype-Extension*`@192.168.1.245:5060`|
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|`from_url` |some name + the `contact_url`|
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|`username` |Asterisk-Skype-Extension |
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|`realm` |might be not used |
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|`passwd` |Asterisk-Skype-Ext-Passwort |
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|`do_register`|Should SipToSis register itself in the PBX? Yes!|
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The remaining options can be left at default values.
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### SkypeToSipAuth.props
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This file defines the receivers of the calls. You can route incoming calls of different Skype contacts to different SIP
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accounts. But in most cases you want to receive all calls on one specific VoIP phone.
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*,sip:44@192.168.1.212:5060
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<p><div class="noteclassic" markdown="1">
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You have to specify the IP of the **VoIP-phone** which should receive the calls. **NOT** the IP of the Asterisk PBX.
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</div></p>
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