That part seemed to work, but I kept getting sipx not being found as a peer, even though my context in sip.conf was [sipx]
On Nov 7, 2012, at 0:10, Josh Patten <[email protected]> wrote: > When using a SIP trunk you will need to have Asterisk point to port 5080. > > > On Tue, Nov 6, 2012 at 10:25 PM, Chris Parker <[email protected]> wrote: >> 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 <[email protected]> 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, [email protected] 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 >> [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 > > _______________________________________________ > sipx-users mailing list > [email protected] > List Archive: http://list.sipfoundry.org/archive/sipx-users/
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