Tom Bolton 11.May.12 07:48 AM a Web browser Audio/visual services 8.5.2 IFR1 All Platforms
I have some questions about connecting SUT Lite with PSTN behind our VoIP provider.
I've deployed Asterisk PBX and connected it to our VoIP provider with SIP Trunk. User extensions connected to Asterisk can call externall PSTN users well. But when I'm trying to connect Sametime Lite client via Proxy / Registrar component and Asterisk to reach the PSTN end user it makes me some problems.
The call is reaching the destined PSTN user but when I am answering the phone the line is closing.
Is it possible to connect SUT Lite with VoIP provider fe. Skype, Ipfon (polish provider), etc. through Software PBX Gateway (Asterisk or other)?
I preffer Asterisk because it is free for our testing purposes.
I've read the SUT Lite interoperability testing program page, but maybe something has changed and haven't updated on that page.
p.s. there is a strange thing when I try to make a call from SUT Lite.
Choosing the phone number fe. 0-22 4444-333-222 it first creates
the SIP connections with callerid sip:firstname.lastname@example.org and then with