Hello,
El Wednesday 18 July 2007 10:02:26 Marc LEURENT
escribió:
Does anyone succeed in redirecting SIP calls like
[0-9]
*(a)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.
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Jesus Rodriguez
VozTelecom Sistemas, S.L.
jesusr(a)voztele.com
http://www.voztele.com
Tel. 902360305
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