SIPTAPI FAQs

Q: Is SIPTAPI compatible with Windows 10

A: Yes. In our tests we do not have experienced any problems. But we often receive reports from users that it does not work. Most of the times the problems are some personal firewalls, anti virus software and false positives in Windows Defender.

Note: Since Win 8 the "Debug" versions are not supported anymore. Thus you have to install the "Release" versions of SIPTAPI.

Q: Why does SIPTAPI not remember my settings in Windows 10

A: Generally, since Windows 7 Windows is very strict when configuring TAPI. You should login as Administrator to install or configure SIPTAPI.

SIPTAPI stores it configuration in the Registry. For unknown reasons Windows deletes the respective Registry keys on upgrades, e.g. from Win 8 to Win 8.1, or when installing the Windows 10 Anniversary Update. When this happens, SIPTAPI can not save its configuration anymore. The workaround is very simple: just uninstall and reinstall SIPTAPI from within the "Phone and Modem" control panel.

Q: How does SIPTAPI work?

A: SIPTAPI allows you to trigger outgoing calls from TAPI applications (click-2-dial). When you trigger the call in your TAPI application (e.g. the Windows Dialer, also known as dialer.exe), SIPTAPI calls your SIP phone. When you answer the phone call, SIPTAPI will perform a call transfer (blind transfer) and instructs your SIP phone to transfer the call to the actual called target. This call transfer is done using the SIP REFER method. Therefore the SIP server or your SIP phone (see next question) must support the REFER method.

For inbound indication (commercial version only), SIPTAPI registers to the SIP server to receive incoming call indication. The INVITE sent to SIPTAPI is never answered - it purely serves to indicate the incoming call and CLI to SIPTAPI.

Q: How do I configure SIPTAPI?

A: SIPTAPI allows two configurations: the "normal" configuration and the "alternative" configuration. In the normal configuration SIPTAPI "talks" to the SIP server, whereas in the alternative configuration SIPTAPI talks directly to the SIP phone. In the normal configuration, when the SIP server is a PBX (e.g. like Asterisk), usually the call transfer (REFER) is intercepted and handled by the SIP server. If the SIP server is just a proxy (e.g. like Kamailio), the REFER will be forwarded to the user's SIP phone and the call transfer will be handled by the SIP phone.

In both cases, the "SIP Domain" is usually the domain/IP of the SIP server. The Outbound Proxy is either the IP of the SIP server, or the IP of the SIP phone.

If you use the commercial version for inbound call indication, you

See the README for details.

Q: Does SIPTAPI allow call control?

A: No. Once SIPTAPI initiates the call transfer, SIPTAPI does not have any control over the call.

Q: My phone starts ringing but when I answer the call there is no outgoing call.

A: There can be many issues. Often the dialed number does not match the dialing plan on the SIP server. Please check the Windows dialing rules to match the dialing plan on the SIP server. You may also inspect the "Refer-to" header in the REFER request.

Q: Does the commercial version work with Asterisk?

A: Yes. But Asterisk does not support multiple registrations under the same SIP account (unless you use Asterisk >=12 with chan_pjsip). Therefore, you must configure dedicated SIP accounts for every SIPTAPI line. On incoming calls Asterisk must be configured to ring both SIP accounts, on outgoing calls SIPTAPI must be configured with the extension of the associated SIP accounts.

For example you want to use SIPTAPI for the existing SIP account 123. Then, in sip.conf create a dedicated SIP account for the SIPTAPI and configure SIPTAPI this account, eg:

;existing phone
[123]
type=peer
secret=...

;new account for SIPTAPI
[123siptapi]
type=peer
secret=...

Further, in the extensions.conf configure a "call forking" to forward the incoming call to both accounts:

[to_sip_phones]
exten => 123,1,Dial(SIP/123&SIP/123siptapi)

In the SIPTAPI configuration configure the "phone user" with the phone number of the associated SIP account, e.g. in above case:

username=123siptapi
phone-user=123