Skype ⇔ SIP Interlink

By | 17 May 2009

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:

ValueDescription
192.168.1.162IP of the PC running Skype and SipToSis
192.168.1.245IP of the Asterisk PBX
88SipToSis-extension on the Asterisk PBX
44@192.168.1.212Extension 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:

SettingNote
via_addrIP of SipToSis/SippySkype
host_portDesired Port of SipToSis
contact_urlsip:Asterisk-Skype-Extension@192.168.1.245:5060
from_urlsome name + the contact_url
usernameAsterisk-Skype-Extension
realmmight be not used
passwdAsterisk-Skype-Ext-Passwort
do_registerShould 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

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