akhilesh chand
2013-Oct-20 05:35 UTC
[asterisk-users] Problem with call transfer from one server to another server
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/4001 at 192.168.14.110,120,tT) exten => 8561,n,hangup() exten => 8562,1,Dial(SIP/4001 at 192.168.14.110,120,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/4001 at 192.168.14.110,120,tT) exten => 5001,n,hangup() Regards Akhilesh -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20131020/f9bd0a36/attachment.html>
Mitul Limbani
2013-Oct-20 05:44 UTC
[asterisk-users] Problem with call transfer from one server to another server
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" <omakhileshchand at gmail.com> 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/4001 at 192.168.14.110,120,tT) > exten => 8561,n,hangup() > > exten => 8562,1,Dial(SIP/4001 at 192.168.14.110,120,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/4001 at 192.168.14.110,120,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 >-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20131020/a6823826/attachment.html>