Hi Steve, Thanks for reply. Is this method will follow on DID incoming calls too?
I mean when we call on DID then call will come to my server and then I want to move this call to any SIP extension. But call will not come to extension just got message *"device not in use". *But device already registered into asterisk server. But thanks you clear my concept into Voip Call routing too. On Thu, Jun 9, 2011 at 12:15 AM, Steve Edwards <[email protected]>wrote: > On Wed, 8 Jun 2011, virendra bhati wrote: > > I have working experience of asterisk with PRI lines. Recently I have took >> VoIP routes from my provider. My basic issue is that now how asterisk will >> behave in such case. I mean in PRI call will come as below process >> >> PRI - -> Digium Card - -> Dadhi/Zap - -> Extensions.conf >> >> What will be the VoIP calling call flow in Incoming and outgoing calls? >> > > Eth[x] -> sip.conf -> extensions.conf > > -- > Thanks in advance, > ------------------------------------------------------------------------- > Steve Edwards [email protected] Voice: +1-760-468-3867 PST > Newline Fax: +1-760-731-3000 > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > -- ----- Thanks and regards Virendra Bhati +91-9172341457 Asterisk Engineer
-- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
