I'm using sipgate.de as my sip provider. When I'm using xlite as client
on sipgate.de, everything works fine: I call number, hear ringing (real
progress tone form called party, not one generated in xlite) and then
talking with called person.
But, when I'm using Asterisk as sip client on sipgate.de, I don't hear
progress tones: I hear only one (locally generated) ring tone, and then
quiet till somebody (called party), answer. After that I can normally
talk with called person. That's happening both with Xlite as sip client
on Asterisk, and with Cisco 12sp/30vip as skinny/sccp client on Asterisk
(so, my conclusion is that problem is in Asterisk).
The biggest problem there is that some numbers have free voice messages
(for example: "number is changed, call xxx-xxxx") and I do NOT hear
that
messages.
*CLI> -- Executing Dial("SIP/xlite1-f896",
"SIP/xxxxxxxxxx@sipgate|50|tr") in new stack
-- Called xxxxxxxxxx@sipgate
-- SIP/sipgate-ce45 is making progress passing it to SIP/xlite1-f896