Yes it is needed my libss7 version. This chan_dahdi.c tested and compile with asterisk-1.6.0.9. svn co https://observer.router.hu/repos_pub/libss7/trunk libss7 svn co https://observer.router.hu/repos_pub/chan_dahdi/trunk chan_dahdi but it needed many additional configuration options in chan_dahdi.conf for properly operation (timers, etc). If you decide to try it I can help you with a sample configuration. A On Fri, 2009-06-05 at 12:55 +0100, Zoltan Markella wrote: > Attila, > > You're a genius! Thanks for the quick reply - it was indeed the problem. > I have tried to grab chan_dahdi from your svn (and replace the stock > one), but I had quite a few compilation errors, so I abandoned that > route. (Do I need to get you libss7 as well to make it work btw?) > > I've seen this p->dialing = 0 problem in the list archives, but I wasn't > sure where it was missing from. > > Anyway, thanks again for the help! > > Cheers, > Zoltan > > Attila Domjan wrote: > > I think it is the bug in the chan_dahdi, which is introduced by the > > p->dialing not implemented proberly in the ss7 part of the chan_dahdi. > > > > Check wheter exists p->dialing = 0; after the p->progress = 1; > > in static void *ss7_linkset(void *data) function at the > > > > case CPG_EVENT_INBANDINFO: > > case ISUP_EVENT_ACM: > > > > > > On Fri, 2009-06-05 at 11:24 +0100, Zoltan Markella wrote: > > > >> Hi, > >> > >> I've been trying to get SS7 working with asterisk the last couple of > >> days, but had on luck. > >> > >> My configuration: > >> - server with a Digium TE420 card, another server with a Digium TE120 > >> card crossover cable between > >> - libss7-1.0.2 > >> - dahdi-2.1.0.4 > >> - asterisk-1.6.1.1 > >> > >> The connection between the two servers is working fine. I have set up a > >> test pri_cpe/pri_net signalling and was able to do a SIP->DAHDI->SIP call. > >> > >> /etc/dahdi/system.conf (both machines): > >> span=1,1,0,ccs,hdb3,crc4 > >> bchan=1-15,17-31 > >> mtp2=16 > >> chocanceller=mg2,1-15,17-31 > >> > >> /etc/asterisk/chan_dahdi.conf (server1) > >> context=from-ss7 > >> signalling = ss7 > >> ss7type = itu > >> linkset = 1 > >> pointcode = 20 > >> adjpointcode = 25 > >> defaultdpc = 25 > >> ss7_called_nai=dynamic > >> ss7_calling_nai=dynamic > >> networkindicator=international > >> > >> cicbeginswith = 1 > >> channel = 1-15 > >> cicbeginswith = 17 > >> channel = 17-31 > >> sigchan = 16 > >> > >> /etc/asterisk/chan_dahdi.conf (server2) > >> pointcode = 25 > >> adjpointcode = 20 > >> defaultdpc = 20 > >> [otherwise same as server1's config] > >> > >> After starting up both server, SS7 comes up successfully: > >> MTP2 link up (SLC 0) > >> --- SS7 Up --- > >> Resetting CICs 1 to 15 > >> Resetting CICs 17 to 31 > >> Got reset acknowledgement from CIC 1 to 15. > >> Got reset acknowledgement from CIC 17 to 31. > >> > >> Here's my call scenario: > >> SIP/600 (Grandstream phone) -> server 1 -> SS7 -> server 2 -> SIP/555 > >> (Nokia E71) > >> > >> Output from server 1: > >> == Using SIP RTP CoS mark 5 > >> -- Executing [1000 at default:1] Dial("SIP/600-08887770", > >> "DAHDI/g1/1000,55,tTo") in new stack > >> -- Called g1/1000 > >> -- DAHDI/1-1 is proceeding passing it to SIP/600-08887770 > >> -- DAHDI/1-1 is ringing > >> -- DAHDI/1-1 answered SIP/600-08887770 > >> > >> Output from server 2: > >> -- Executing [1000 at from-ss7:1] Ringing("DAHDI/1-1", "") in new stack > >> -- Accepting call to '1000' on CIC 1 > >> -- Executing [1000 at from-ss7:2] Dial("DAHDI/1-1", "SIP/555,50,tTo") > >> in new stack > >> == Using SIP RTP CoS mark 5 > >> -- Called 555 > >> -- SIP/555-0890bab0 is ringing > >> -- SIP/555-0890bab0 answered DAHDI/1-1 > >> > >> So the call is set up properly. BUT there's no audio on either end!!! > >> With dahdi_monitor I can see activity on both card's first channel (and > >> no other channels, so there's no CIC mismatch), but on server1 I only > >> have RX and on server2 I only have TX. > >> > >> Could anybody give me a hint where my problem could lie? > >> > >> Cheers, > >> Zoltan > >> > >> _______________________________________________ > >> --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 -------------- next part -------------- A non-text attachment was scrubbed... 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