Paste ur chan_dahdi.conf n sip.conf on both servers to pastebin and fwd link here.
Mitul On Oct 20, 2013 11:07 AM, "akhilesh chand" <[email protected]> wrote: > Dear All, > > I have pri with E1 facility that have 30 line and 100 pri number which is > provided by service provider.Number started like 23568561,23568562,23568563 > and so on. Service provider provide last four digit number for did mapping > like 4561,4562,4563. > > > exten => 8561,1,Dial(SIP/[email protected],120,tT) > exten => 8561,n,hangup() > > exten => 8562,1,Dial(SIP/[email protected],120,tT) > exten => 8562,n,hangup() > > Call comes into first server successful.But problem with second server > when call came into second server i got following error: > > * chan_sip.c:20063 handle_request_invite: Call from '' to extension > '4001' rejected because extension not found.* > > In one more scenario: > > when i create one extension and call forwarding with this extension that > time I'm able to transfer call successful the code is given below: > > exten => 5001,1,Dial(SIP/[email protected],120,tT) > exten => 5001,n,hangup() > > > Regards > Akhilesh > > -- > _____________________________________________________________________ > -- 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 >
-- _____________________________________________________________________ -- 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
