Anton wrote: > As I understand this would require some patching, since > there is no native functionality yet? Actually call > divert/forward is a very required feature... Possibly any > extra dialplan function required, like Forward() ? > I would prefer that Alan answer this, but no patching should be required (all the required code is merged into libss7 and chan_dahdi), although Alan will have to be the one to tell how it is implemented, since he wrote it. Matthew Fredrickson > > On Thursday 10 July 2008 02:02, Alan McMillan wrote: > >> I have successfully implemented a limited NORTEL version >> of RLT on asterisk. I am using the DMS-250 flavor... I >> can try and answer questions. >> >> Alan McMillan >> >> -----Original Message----- >> From: asterisk-ss7-bounces at lists.digium.com >> [mailto:asterisk-ss7-bounces at lists.digium.com] On Behalf >> Of Matthew Fredrickson >> Sent: Wednesday, July 09, 2008 1:08 PM >> To: asterisk-ss7 at lists.digium.com >> Subject: Re: [asterisk-ss7] Asterisk/libSS7 functionality >> question >> >> Mark Wilkinson wrote: >> >>> Hello, >>> I have a question about the functionality of SS7 in >>> Asterisk. From what I can work out, at present, I can >>> not do the following :- >>> >>> I want a call to be placed from a PBX to Asterisk. >>> Have Asterisk playback a message or do other clever >>> stuff, then release the call back to the original PBX, >>> without the call remaining looped (tromboned) through >>> Asterisk. >>> >>> From what I understand of SS7, it should be possible >>> to get a SS7 PBX switch to take back the call and >>> transfer it elsewhere - Somewhat similar in >>> functionality to a SIP refer. >>> >>> If I've understood, and please correct me where I'm >>> wrong, the chan_dahdi driver >>> in the asterisk trunk needs the 'transfer' >>> functionality adding to allow it to signal >>> to the PBX a request for call take back possibly by >>> implementing the FAR FAC FAA >>> messages ? >>> >> Yes, this is definitely possible in libss7, although I >> believe right now it takes some dialplan changes to make >> it work. Alan McMillan (I believe he's on this list) >> implemented it, so he would be the one to tell how to >> configure it. >> >> Matthew Fredrickson >> >> _______________________________________________ >> --Bandwidth and Colocation Provided by >> http://www.api-digital.com-- >> >> asterisk-ss7 mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-ss7 >> >> >> _______________________________________________ >> --Bandwidth and Colocation Provided by >> http://www.api-digital.com-- >> >> asterisk-ss7 mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-ss7 >> > > _______________________________________________ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-ss7 mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-ss7 >
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