18 Jul
2007
18 Jul
'07
7:47 p.m.
Hi,
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
Ciao