Its always also the same direction. On Fri, 23 Feb 2007, Anton wrote: > I had yesterday the case again. > > In my case? when i do have 1way audio, it's always in the IN > direction. I mean in the scheme > > <USER>--SIP--<CHAN_SS7_BOX>----<PSTN> - the <PSTN> side > cannot hear the <USER> - but users hears PSTN. > > What's in your case? Any other behaviors? > > On 23 February 2007 13:07, asterisk@xxxxxxxxx wrote: >> I don't know exactly, but it seems to be that a working >> Call Flow looks like this: >> >> -- Executing Dial("IAX2/srv8-srv25-4", >> "SS7/W05/0043123456789") in new stack -- SS7 request >> (SS7/W05/0043123456789) format = 0x8. -- SS7 channel >> SS7/W05/0043123456789 allocated successfully. -- Called >> W05/0043123456789 >> -- SS7/W05/9 is making progress passing it to >> IAX2/srv8-srv25-4 -- SS7/W05/9 is ringing >> -- SS7/W05/9 answered IAX2/srv8-srv25-4 >> -- SS7 hangup 'SS7/W05/9' CIC=9 Cause=16 (state=7) >> >> >> and a not working call flow looks like this: >> >> -- Executing Dial("IAX2/srv8-srv25-1", >> "SS7/W05/0043123456789") in new stack -- SS7 request >> (SS7/W05/0043123456789) format = 0x8. -- SS7 channel >> SS7/W05/0043123456789 allocated successfully. -- Called >> W05/004369914014005 >> -- SS7/W05/7 is ringing >> -- SS7/W05/7 answered IAX2/srv1-srv2-1 >> -- SS7 hangup 'SS7/W05/7' CIC=7 Cause=0 (state=5) >> -- Hungup 'IAX2/srv1-srv2-1' >> >> >> This is now with chan_ss7-0.9 Asterisk 1.2.10 and zaptel >> 1.2.12 >> >> Can anyone help? >> >> Thanks >> >> Nico >> >> >> >> _______________________________________________ >> --Bandwidth and Colocation provided by Easynews.com -- >> >> asterisk-ss7 mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-ss7 > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-ss7 mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-ss7 >