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
>
>
>
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