The biggest red flags are Asterisk and CUCM The SIP stacks on these platforms aren't complete and REFER support is usually lacking. Trunking with these platforms requires a session border controller to interface with these platforms. How many active calls do you expect between these systems? If it's a small amount you could probably get away with the internal sipXbridge SBC. Otherwise you could look into using a Patton or Ingate SBC, or you could roll your own with FreeSWITCH: http://wiki.sipfoundry.org/display/sipXecs/FreeSWITCH+SIP+Trunking+Gateway
On Mon, Nov 5, 2012 at 10:59 AM, Chris Parker <[email protected]> wrote: > Kind of stumped with this one... > > I have Vitelity as my SIP trunk, which is configured in Asterisk to answer > with an IVR and perform some other functions. > If a call is passed from my AA in Asterisk or the DID is configured to > call an extension that belongs to my sipX box (Polycom phones), I cannot > transfer that call to any other internal extension or park that call. The > same is true if a call comes from another PBX (CUCM) - both gateways are > configured as "Unmanaged Gateway" > > If I call out from sipX to the PSTN (there is also a trunk configured with > Vitelity on sipX) I can place that outbound call in park orbit or transfer > them to another internal extension. > > I'm more familiar with troubleshooting Asterisk than sipX, so consider me > a newbie in this regard. I will gladly post configs / dump log files upon > request once I know what I'm looking for. > _______________________________________________ > sipx-users mailing list > [email protected] > List Archive: http://list.sipfoundry.org/archive/sipx-users/ > -- Josh Patten eZuce Solutions Architect O.978-296-1005 X2050 M.979-574-5699 http://www.ezuce.com
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