18 Jul
2007
18 Jul
'07
9:27 p.m.
Hello,
El Wednesday 18 July 2007 10:02:26 Marc LEURENT escribió:
Does anyone succeed in redirecting SIP calls like [0-9] *@sip.test.com to a SIP/PSTN gateway provider without using asterisk?
I suposse you need to use de UAC module. http://www.openser.org/docs/modules/stable/uac.html
You only need UAC module if digest authentication is mandatory (and UAC module has some limitations with authentication).
The easiest way is that your provider authenticates your traffic by source ip address (and maybe a prefix).
Regards.
Saludos JesusR.
------------------------------------ Jesus Rodriguez VozTelecom Sistemas, S.L. jesusr@voztele.com http://www.voztele.com Tel. 902360305 -------------------------------------