Are you saying this won't pass the required sip info to asterisk for it to know how to route a call that came in through Sipx? -----Original Message----- From: Josh Patten <[email protected]> Date: Tue, 25 May 2010 16:19:05 To: Gerald Harper<[email protected]> Cc: <[email protected]> Subject: Re: [sipx-users] Asterisk Dial through SIPX
Yes, that's exactly what it would be. Unfortunately I don't think there is a way to pass DID information down through to Asterisk from sipX this way. Josh Patten Assistant Network Administrator Brazos County IT Dept. (979) 361-4676 On 5/25/2010 4:17 PM, Gerald Harper wrote: > Can you expand a little on the "Create an extension and have asterisk > register to the sipX system" option. Would this be a trunk off the > Asterisk system that registers as a phone on the sipX system? > > -------------------------------------------------- > From: "Josh Patten" <[email protected]> > Sent: Tuesday, May 25, 2010 2:09 PM > To: <[email protected]> > Subject: Re: [sipx-users] Asterisk Dial through SIPX > >> There are two ways to do this: >> Remove permission requirements from your trunks which is not generally a >> good idea >> or >> Create an extension and have asterisk register to the sipX system. >> >> The reason why sipX won't allow a passthrough is because trunks don't >> have any permissions so if an outbound trunk requires the "Local >> Dialing" permission then the call will be rejected. >> >> Josh Patten >> Assistant Network Administrator >> Brazos County IT Dept. >> (979) 361-4676 >> >> >> On 5/25/2010 11:36 AM, Gerald Harper wrote: >>> Content-Type: text/plain; >>> charset="utf-8" >>> Content-Transfer-Encoding: 8bit >>> Organization: SipXecs Forum >>> X-FUDforum: 08063afcdd00a6e76393c5b9527381e8<47162> >>> Message-ID:<[email protected]> >>> >>> >>> >>> So I have started playing around with an Asterisk box, I >>> downloaded and installed Asterisk and was presented with a >>> FreePBX gui, after a bit of messing around I have been able >>> to make the two machines talk to each other. >>> >>> Now I am trying to figure out if I can make a call that >>> originates on the Asterisk dial out via a SIP trunk on the >>> sipx box. I can see the digits coming across from the >>> Asterisk server but I am guessing that the call can not be >>> authenticated by the sipx. >>> >>> Has anybody done this before? Any hints or tips? >>> _______________________________________________ >>> sipx-users mailing list [email protected] >>> 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 [email protected] >> 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 [email protected] 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 [email protected] 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/
