Please test the following. Calling from PSTN to AA and try to transfer to an extension that is not forwarded.
You have to ensure your ITSP is sending the INVITE to port 5080 in order for any transfers to succeed. If not, what Gerald said... how to get around that is to create a dialplan (56+10 digits sends 10 digits to a specific "other" gateway), which can accomplish calls coming from one provider but the forward dialplan sends it out through another. That should fix you. On Fri, Apr 20, 2012 at 1:08 PM, Tommy Laino <[email protected]> wrote: > > Content-Type: text/plain; > charset="utf-8" > Content-Transfer-Encoding: 8bit > Organization: SipXecs Forum > X-FUDforum: 08063afcdd00a6e76393c5b9527381e8 <67729> > Message-ID: <[email protected]> > > > > I have 3 IP trunks on my test system. I am trying to have an > option from the auto attendant transfer to a cell phone. If > i do it from a local or remote polycom phone it works fine. > Once an external call comes into a trunk and it chooses the > option the caller gets disconnected. Anything that I might > be missing. > -- > Tommy Laino > Dome Technologies > _______________________________________________ > sipx-users mailing list > [email protected] > List Archive: http://list.sipfoundry.org/archive/sipx-users/ > -- ~~~~~~~~~~~~~~~~~~ Tony Graziano, Manager Telephone: 434.984.8430 sip: [email protected] Fax: 434.465.6833 ~~~~~~~~~~~~~~~~~~ Linked-In Profile: http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4 Ask about our Internet Fax services! ~~~~~~~~~~~~~~~~~~ -- LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: [email protected] Helpdesk Customers: http://myhelp.myitdepartment.net Blog: http://blog.myitdepartment.net
_______________________________________________ sipx-users mailing list [email protected] List Archive: http://list.sipfoundry.org/archive/sipx-users/
