Here i going explain what Iam doing and where i need help ..
Iam running Sip Express Router ,Asterisk, on same box (for testing) my Sip express router is working fine and i can accept global register requests with valid account and in front of Sip express router (SER) Iam using Mediaproxy-1.4.2 which is handler to rtp/rtcp streams between nated clients ,SER is running on port 5060
and Asterisk on 5065, here i need to forward pstn calls to asterisk and i am planning to connect asterisk to a Cisco Gateway ,
when sip client calls to pstn SER will recieve invite message and it forwards to asterisk
1)how the Asterisk will handle this call with rtp
2)and when pstn customer calls the call goes in to SER and it looks the 'location' database and it will reject call because it is not registerd user
so, we take pstn call directly to asterisk and we forward call from asterisk to SER and i want to know is how the SER handle this call
that means when SER found a sip client it invites that sip client and which mediaproxy is going to handle this call the SER's or Asterisk's ????????
Can we use only one mediaproxy for both SER and ASTERISK by loading modules in ASTERISK so that it will be easy for billing ..???
please explain me how the process will take here bcoz i am with lots of questions and confusions in this particular process
hope some body will solve my headache confusion ..Thanks in advance
Kindly regards,
Ravi.
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