sipLite for Skype

sipLite for Skype is the free sip phone that could bridge Skpye and Sip.

 

Basic features

  • Accept incoming Skype call and then call out with SIP protocol.
  • Accept incoming SIP call and then call out with Skype.
  • Standard SIP softphone with the following features:
    • Support CODECS: G729, G723, aLaw, uLaw, GSM.
    • Secure for call from internet Cafe or public computers.
    • Balance display.
    • Local port and server port can be self-defined.
    • Smart keypad: 0123456789*# as DTMF. ESC to hangup. ENTER to dial or redial the last called number. BACKSPACE to backspace. + - to adjust volume.
    • Communication with other windows application through IPC.

 

Download

  • English version: 3.0.0.119 (updated 2010-02-01)
    MD5 Check Sum : fb857ea58410ff09ce83d94981ab66ea

 

  • Chinese version: 3.0.0.118 (updated 2010-02-01)
    MD5 Check Sum : 629f75f785451e93c42c7da11e3deab7

 

How to configure

To save your time the following steps you need to know when configuring the software.

 

Click [Register] tab to input sip server informations.

 

Click [Service] tab to select service type.

 

If [Sip->Skype] radio button selected, the software will convert incoming Sip call to Skype call.

If [Skype->Sip] radio button selected, the software will convert incoming Skype call to Sip call.

 

Click [Audio] tab to select audio devices.

 

Usually [audio cable] is needed to link sipLite and Skype. The easy way is to use [Virtual Audio Cable] but not physical sound cards.

 

Work with Skype

Suppose Skype is already running well on your PC. After you run SipLite for Skype, the following dialog will pop out:

 

Please select [Allow this program to use Skype] and then press [Ok] button.

 

Configure SIP server to work with [Sip to Skype] function.

Suppose you use Asterisk as Sip server. We need to configure in the following way:

 

Add the following content to /etc/asterisk/sip.conf .

 

[xlite1]
type=friend
context=sip2skype
host=dynamic
secret=12345
nat=yes
canreinvite=no
allow=all

 

[siplite]
type=friend
context=sip2skype
host=dynamic
secret=12345
nat=yes
canreinvite=no
allow=all

 

Add the following context to /etc/asterisk/extensions.conf .

 

[sip2skype]
exten => 00861082952824,1,set(CALLERID(name)=${EXTEN})
exten => 00861082952824,2,dial,sip/siplite|60
exten => 00861082952824,3,hangup

 

Reload asterisk to make the above settings effect.

Register X-Lite with xlite1 account, register sipLite for skype with siplite account.

 

Now call 00861082952824 with X-Lite you will see siplite accept the call and then Skype call 00861082952824 out.

 

Configure SIP server to work with [Skype to Sip] function.

Add the following content to /etc/asterisk/sip.conf .

 

[8804]
type=friend
host=dynamic
accountcode=8804
username=8804
secret=8804
context=skype2sip

 

Reload asterisk to make the above settings effect. Register sipLite for skype with above account.

 

Click [Service] tab.

 

 

Add 12345_ for [Access code for Skype->Sip].

Select radio button [Skype->Sip].

 

Now, call your Skype with another Skype account for example iaxtalk. You will get the following informations from Asterisk's console:

 

[Feb  4 10:54:42] NOTICE[2700]: chan_sip.c:14035 handle_request_invite: Call from ‘8804′ to extension ‘12345_iaxtalk‘ rejected because extension not found.

 

You can see you have got informations for the incoming skype call:

 

Skype caller account: iaxtalk

Service access number: 12345

 

So, the left thing for you is to write a agi for your business.

 

[skype2sip]

exten => _12345., 1, noop(${EXTEN})

exten => _12345., 2, agi, your_agi|${EXTEN:6}

exten => _12345., 3, hangup

 

Customize softphone for you

We provide sipLite softphone customize service. If you are interested in getting your own branded softphone please contact us.

 
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