1. Run the tests built into sipxconfig and ensure they pass. 2. Ensure the phones are using the domain and not the ip address to register (registrar and proxy).
3. If not installing from ISO and letting sipx be the DNS server at setup, consider reinstalling from ISO and doing it in that way. ============================ Tony Graziano, Manager Telephone: 434.984.8430 Fax: 434.984.8431 Email: tgrazi...@myitdepartment.net LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 Fax: 434.984.8427 Helpdesk Contract Customers: http://www.myitdepartment.net/gethelp/ ----- Original Message ----- From: sipx-users-boun...@list.sipfoundry.org <sipx-users-boun...@list.sipfoundry.org> To: ronald teng <ronaldt...@gmail.com> Cc: sipx-users@list.sipfoundry.org <sipx-users@list.sipfoundry.org> Sent: Sat Apr 03 08:02:25 2010 Subject: Re: [sipx-users] calls from PSTN to operator(AA)=ok but transfer from operator to an extension=failed Hi, I will assume that you meant still not working. From the cisco logs the only thing in which was weird at first sight was that you had IPs in every SIP URI (request/from/to/contact) _but_ in the REFER Refer-To field where you had "2...@alsterph.lan". Since the cisco accepts the REFER (202 Accepted) but he doesn't generate an INVITE to "2...@alsterph.lan" in response and issue a "503 Service Unavailable", the most obvious answer is that he does support REFER but he is unable to decide where to send the INVITE to "2...@alsterph.lan". I have very basic cisco knowledge, but the answer probably lies in how the cisco internally decides where to send an INVITE to "2...@alsterph.lan". Normally if he has no pre-configured way to know it, he should issue a DNS SRV request for "alsterph.lan", from this DNS SRV request he will get the transport (tcp/udp), the host and port where he has to send his INVITE packet. So what I offered you was to first check/create a DNS SRV for "aslterph.lan", or in last resort to use an IP instead of a host/domain in the "2...@alsterph.lan" transfer target which is probably the easiest (but not nicest) way to solve your problem. Regards. ronald teng wrote: > UPDATE: DNS has been fixed so i can now ping just > 'alsterph.lan'...incoming still now working (T_T) > > would i need to use the sip call transfer commands of cisco for this? > According to this link i posted before, i'll need to have this script > named " app_h450_transfer.tcl " in my flash. Any ideas as to where i > can find a copy of this? > > _______________________________________________ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users Unsubscribe: http://list.sipfoundry.org/mailman/listinfo/sipx-users sipXecs IP PBX -- http://www.sipfoundry.org/ _______________________________________________ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users Unsubscribe: http://list.sipfoundry.org/mailman/listinfo/sipx-users sipXecs IP PBX -- http://www.sipfoundry.org/
