On Wed, 2009-07-29 at 12:07 +0200, Krzysztof Drewicz 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) 2 servers, one of sangoma a108d, and a digium TE420P + EC sun 4100/2200 servers I have 2 interconnection, the main connected to sangoma, I have 4 E1s ~100 cllas on peak, the system load/cpu usage are very low... > What do you do with calls (SIP/IAX, codec transcoding, isdn routing, > IVR etc...) > MGCP/NCS packet cable users with my heavy patched chan_mgcp + packetcable cops support :) I have ~6k MTAs https://issues.asterisk.org/view.php?id=12950 no codec transcoding, just g711a The dialplan is very complex an difficult, the standard asterisk CDR variables not enough, I need many more. The system emulate a real telco sitch. I'm using realtime only in the dialplan (call forwarding and many more call controlling parameters) > _______________________________________________ > --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... Name: not available Type: application/pgp-signature Size: 197 bytes Desc: This is a digitally signed message part Url : http://lists.digium.com/pipermail/asterisk-ss7/attachments/20090729/247ece4e/attachment.pgp
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