[Asterisk-Users] terminate sip calls from a 3rd party sip proxy into asterisk. and then to gnugk

[Voip Business]

Hello list,

your help will be apreciated in this regard.

i have an asterisk server (working just as SIP-H323 translator) I have
a customer that has a SNOM 4S sip proxy and with 60 sip users in
there, I want to terminate calls for them.

I wish to see Asterisk as a sip gateway from the snom part.


question 1: what I need to put in sip.conf in order to accept all
users invited from that sip proxy?

I dont want to create all thse 60 users in my sip.conf.

Question 2:
I terminate calls to a h323 gnugk where is my billing software, Does
the sip account and password are passed to the gnugk? in order to do
the billing there?

I hope I explain correctly or as much I can.

regards


Humberto