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Nik Pakar nikpakar at gmail.comHi Andreas, i think i mis-interpret my initial question. I should have asked whether LCR can have a chan_lcr for yate like it has for asterisk. If its not there can you please give me some idea on how to build one. Thanks for your help. Rgds Nik On Thu, Mar 1, 2012 at 11:05 AM, Nik Pakar <nikpakar at gmail.com> wrote: > Hi Andreas, > > Wouldnt it be better if we can integrate OpenSBC to some thing like yate > which already have extensive ss7, sigtran, camel and map parts implemented. > > It will be a perfect marriage if we can do it as both openbsc and yate > have its own strength which can make a full fledge mobile switching > environment. > > How difficult would it be either, > > 1. Use OpenBSC Nitb as its on LCR and implement a gsm channel driver for > yate. > or > 2. Use extensive SCCP, MAP implementation already on yate and implement > full A interface integration towards yate. > > Lets throw some ideas on this way, as im sure this will be a major > implementation if we can do it. > > Rgds > Nik > > On Thu, Mar 1, 2012 at 7:01 AM, Andreas Eversberg <andreas at eversberg.eu>wrote: > >> >> >>>> channel.c -o bchannel.po >> >>>> bchannel.c:26:27: error: mISDN/mISDNif.h: No such file or directory >> >>>> bchannel.c:28:31: error: mISDN/mISDNcompat.h: No such file or >> directory >> >>>> bchannel.c:29: error: 'MISDN_AF_ISDN' undeclared here (not in a >> function) >> >>>> bchannel.c:30:24: error: mISDN/q931.h: No such file or directory >> >>>> >> >> hi, >> >> currently the chan_lcr only works with mISDN. the chan_lcr directly >> opens the bchannels in order to exchange audio with asterisk. this only >> works if the call is connected to isdn phone or isdn line, since only >> these interfaces support mISDN. gsm and sip interface cannot be used >> with chan_lcr (anymore). >> >> i plan to forward audio between lcr and chan_lcr using the unix-socket >> and remove the export/import code from lcr. >> >> currently you can only link gsm insterfaces to asterisk using sip. with >> jolly/rtpmux branch of openbsc, you can even forward rtp traffic >> directly between openbsc and the remote sip endpoint (asterisk or other >> sip gateway), without routing traffic through lcr. >> >> if you have any suggestions about further development, let me know. >> >> regards, >> >> andreas >> >> p.s. i committed the appbridge.cpp fix of alexander. thanx! >> >> >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.osmocom.org/pipermail/openbsc/attachments/20120301/7c2944a9/attachment.htm>