2008/12/17 Artifex Maximus <[email protected]> > Hi all! > > Is anyone using the $subject setup? > > What I would like to do the following setup: > 1. OXE is setup for receiving calls, handling Agents > 2. Asterisk as external IVR on extension 9xxx connected with ISDN (Q.931) > PRI > > The incoming calling route: > 1. OXE handles incoming calls, answer > 2. Transfer to extension 9xxx > 3. Asterisk answer (using one channel) > 4. IVR is handling calls > 5. If needed IVR transfer back to specified Pilot in OXE with Dial > (using two channels) > 6. Asterisk hangup (free both channels) > 7. OXE connect the PSTN incoming line with Pilot as extension transfer does > > I've talked with support person at Alcatel and he said that Q.931 > cannot handle this situation because after calls "leave" OXE it does > not know anything so I cannot hangup in Asterisk and call will use two > channel. Is it right? He said that ABCF2 or Q.SIG is able handling > this situation because Q.SIG is an extension to Q.931. I take some > search on topic and find out that Asterisk's Q.SIG not fully > implemented. Is Asterisk implementation enough for this kind of setup? > > I am using Asterisk 1.6.0.3-rc1 with dahdi-*-2.1.0 on Ubuntu Server 8.10. > > Thanks, > Zsolt
Hi, What is needed is that the Asterisk box should either : - forward incoming call to the right endpoint, using a single channel, - open a second channel and remain in media path till it ends. I'm not an authority on this topic, but I would say that, as OXE and asterisk are connected through an E1/T1 link, - you must upgrade OXE and Asterisk to Q.SIG to get forwarding option (and check asterisk's QSIG supports Call Deflection), - casual PRI is enough if you stick with 2 channels option. If you don't expect to get more than 15 (or 12) calls at a time, I don't see any real downside to use option 2. > > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
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