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, [EMAIL PROTECTED] 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
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