Some development on this topic... Watching the SIP debug in Asterisk, I see that when I try to transfer the call it actually asks Asterisk to dial the target extension and Asterisk has no clue how to deal with that since it owns the 1xxx group while sipX owns 2xxx. If I put in a line that says to send _20xx back to sipX it works! Although due to how hacky this is, calls sent to Park are lost forever and all transfers are blind. At least this is progress in a way. I tried to create a SIP trunk between Asterisk and sipX but it was sort of wonky; it couldn't call back into Asterisk to reach the 1xxx and the calls sent over from Asterisk still exhibited the same broken transfer - long story short I tried and failed somehow at SIP trunk, so it's back to an unmanaged gateway.
On Nov 6, 2012, at 2:40 PM, Chris Parker <cparke...@me.com> wrote: > The call volume is going to be very low. If I understand this correctly, I > would create a trunk under Gateways in sipX for my Asterisk system and create > the other end in Asterisk accordingly, rather than calling it an Unmanaged > Gateway. > And to answer another question, yes the sipX and Asterisk system are on the > same subnet whereas CUCM is in a different subnet but has unrestricted access > to that subnet. > > On Nov 6, 2012, at 12:00 PM, sipx-users-requ...@list.sipfoundry.org wrote: > >> An unmanaged gateway is just that. Can I assume that the address for both >> systems are on the same subnet? Unmanaged gateways would assume that the >> other and knows how to handle the SIP REFER method. >> >> Asterisk as a sip trunking system is not exactly compliant. >> >> If REFER is not supported, then the media needs to be anchored by sipx once >> it accepts the call and hold the REFER internally, at which point you would >> setup a manual sip TRUNK not a gateway. >> > _______________________________________________ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/
