Нашел интересный сервис, кто знаком с ним? http://www.iaxtalk.com Downloads sipLite for Skype (Version: 3.0.0.119 updated 2010-02-01) sipLite softphone (Version: 3.0.0.138 updated 2010-01-29) sipLite softphone with SMS sending / callback triger / advertisement banner supported (Version: 3.0.0.146 updated 2009-08-28) iaxLite softphone normal installation version (Version: 2.4.0.10 updated 2008-05-22) iaxLite softphone portable version (Version: 2.3.0 updated 2008-01-02) iaxLite softphone old version with AU-100 USB phone supported iaxLite softphone with SMS sending / callback triger / advertisement banner supported UniDialer GSM modem ANI callback triger windows GUI version (Active hangup mode) GSM modem ANI callback triger console version with passive hangup mode (Available for our customers only) GSM modem SMS callback triger windows GUI version
для скачивания предлагается sipLite for Skype (Version: 3.0.0.119 updated 2010-02-01) не пойму, как его увязать со скайпом? может кто сообразит?
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