2.9 KiB
| title | layout | created | updated | toc | tags | |||||
|---|---|---|---|---|---|---|---|---|---|---|
| Skype ⇔ SIP Interlink | default | 2009-05-17 13:06:06 +0200 | 2009-05-17 13:06:06 +0200 | false |
|
To connect Skype to a SIP-PBX (e.g. Asterisk), there's only one free application available:
- SipToSis (former SippySkype)
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)
{% highlight ini %} 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 {% endhighlight %}
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