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:new: 2009-05-17

Skype ⇔ SIP Interlink

To connect Skype to a SIP-PBX (e.g. Asterisk), there’s only one free application available:

There are also some (Windows-)apps for money, like:

Since you can’t integrate these apps into Asterisk, you need to add an extension for each Skype account (1 per PC).

If somebody calls via Skype, you can let SipToSis dial a specific number (internal/external) via the Asterisk PBX. Most probably you want it to dial your own SIP extension.

If you call the Skype extension, you can let SipToSis dial a specific Skype contact. Maybe you could add another virtual PBX so that you could dial <SkypeContact>@<SipToSis-IP> on your VoIP-phone to call a specific Skype contact - but your VoIP-phone then needs to be registered with this virtual PBX.

SipToSis (SippySkype)

Since this program is free and written in Java, it’s the perfect choice for now. Configuration might be a bit tricky though.

Configuration

In this example, there are following values:

Value Description
192.168.1.162 IP of the PC running Skype and SipToSis
192.168.1.245 IP of the Asterisk PBX
88 SipToSis-extension on the Asterisk PBX
44@192.168.1.212 Extension and IP of the VoIP-phone to use for incoming Skype calls

siptosis.cfg (former sippyskype.cfg)

via_addr=192.168.1.162
host_port=5060
contact_url=sip:88@192.168.1.245:5060
from_url="Skype Gateway" <sip:88@192.168.1.245:5060>
username=88
realm=asterisk
passwd=skype
do_register=yes

So this general description should work:

Setting Note
via_addr IP of SipToSis/SippySkype
host_port Desired Port of SipToSis
contact_url sip:Asterisk-Skype-Extension@192.168.1.245:5060
from_url some name + the contact_url
username Asterisk-Skype-Extension
realm might be not used
passwd Asterisk-Skype-Ext-Passwort
do_register Should SipToSis register itself in the PBX? Yes!

The remaining options can be left at default values.

SkypeToSipAuth.props

This file defines the receivers of the calls. You can route incoming calls of different Skype contacts to different SIP accounts. But in most cases you want to receive all calls on one specific VoIP phone.

*,sip:44@192.168.1.212:5060

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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