I got it working: On users.conf: ... [span_1] group = 1 hasexten = no switchtype = national signalling = ss7 trunkname = Span 1 trunkstyle = digital ; GUI metadata hassip = no hasiax = no context = DID_span_1 dahdichan = 2-15,17-31 ... What hw? (what cards, how many, bus, servers) Digium Wildcard TE110P T1/E1 Card SuperMicro server Debian lenny 2.6.26-2-amd64 asterisk 1.6.2 dahdi 2.2.0 libss7 1.0 What do you do with calls? SS7 to SIP Thank for your prompt reply /Pedersen On Wed, Jul 29, 2009 at 1:07 PM, Krzysztof Drewicz < krzysztofdrewicz at gmail.com> wrote: > 2009/7/28 Domjan Attila <adomjan at tvnet.hu>: > > Yes it's a bug, I pointed it for a half years ago... > > In the ss7_start_call() function the diaplan execution started 1st, > > after come the variables setup... > > > > If you want try my version from my svn, this bug is fixed there, I have > > ~20k calls/day. > > What hw? (what cards, how many, bus, servers) > What do you do with calls (SIP/IAX, codec transcoding, isdn routing, > IVR etc...) > > _______________________________________________ > --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 > -- Peers -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-ss7/attachments/20090729/6c152eda/attachment.htm
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