Hello Masters
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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Woodoo People .pGa!
2006-Jun-01 02:33 UTC
[Asterisk-Users] connecting asterisk to pstn help
look for SER and Asterisk on voip-info. I think, you plan to got to UA->SER-(mediaproxy)->Asterisk->PSTN if yes, ser will communicate UA (user agent) on one leg, and asterisk on other. you can use your asterisk to billing and pstn connection. on incoming call dial $phone/ip.address.of.ser> 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.> _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > Asterisk-Users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users-- WoodOO-[P]an[G]alaktikan[A]gent-People <][> http://shadow.pganet.com wpeople@shadow.pganet.com]iCQ#33118021[wpeople.on.iRCNet]wpeople@RedHat.users