[asterisk-ss7] Answer() is needed but it costs to thecallingparty while the extension is still ringing!

[Ercan Yücebas]

Dear All

Maybe, i find the reason

It's because the standards, chan_ss7 doesn't send address completed
after getting iam but then after if the extension starts ringing,
chan_ss7 is sending alerting and here our provider cancels the call,
because security reasons, just because this behaviour is not followed by
ITU standards.

We have to work on this issue, 

Does any body know how to debug the fully but only ISUP signalling on
chan_ss7?

BR

Ercan









-----Original Message-----
From: asterisk-ss7-bounces at lists.digium.com
[mailto:asterisk-ss7-bounces at lists.digium.com] On Behalf Of Tim Danner
Sent: Mittwoch, 21. März 2007 07:25
To: asterisk-ss7 at lists.digium.com
Subject: RE: [asterisk-ss7] Answer() is needed but it costs to
thecallingparty while the extension is still ringing!

Meaning immediate ANM or some SSP just responding without full trunk
processing in siganlling. Not trying to mix bearer and signalling here,
but here's the point.  Programmers.

-----Original Message-----
From: Tim Danner 
Sent: Tuesday, March 20, 2007 11:18 PM
To: asterisk-ss7 at lists.digium.com
Subject: RE: [asterisk-ss7] Answer() is needed but it costs to
thecallingparty while the extension is still ringing!


Almost sounds like a ISUP timeout issue, or just some timeout issue.
Have we tryied anthing with ISUP CPG???
What's you're post dial delay, PDD.  It's usually between IAM and ACM,
buit could be satified with CPG.  I'm thinking this because if ANM does
the job....

-----Original Message-----
From: Anton [mailto:anton.vazir at gmail.com]
Sent: Tuesday, March 20, 2007 10:38 PM
To: asterisk-ss7 at lists.digium.com
Subject: Re: [asterisk-ss7] Answer() is needed but it costs to
thecallingparty while the extension is still ringing!


Works of coarse. Try putting Ringing() before Dial() or 
check your SIP route. Strange that REL is received on your 
SS7 - not sure, but looks like your telco side is releasing 
call. Put here more messages and your dialplan.

On 20 March 2007 23:46, Ercan Yücebas wrote:
> Any new ideas, working ideas?
>
>
> Just putting Dial(SIP...) didn't work, did you tried and
> it's working on your system?
>
> Extension rings only one time and then asterisk hangs up
> the call, sip cancel is coming from asterisk to
> extension, then the provider switch sends the call a
> second time, because the first one as too quick, then it
> happens the same and no more try from switch.
>
> I compared the debug with and w/o answer() (debug level
> 10), I'm getting this difference
>
> Without Answer()
>
> Mar 20 18:58:03 DEBUG[5693] mtp.c: Got MSU on link 'l1'
> sio=5 slc=9 m.sls=0 bsn=1/97, fsn=1/38, sio=c5, len=13:
> a0 0f 4b 90 09 00 0c 02 00 02 83 a9
> Mar 20 18:58:03 DEBUG[5693] l4isup.c: processing ISUP
> message, typ=REL, CIC=9
> Mar 20 18:58:03 DEBUG[5693] channel.c: Soft-Hanging up
> channel 'SS7/siuc/9'
>
> With Answer()
>
> Mar 20 19:02:23 DEBUG[5694] chan_sip.c: Allocating new
> SIP dialog for (No Call-ID) - NOTIFY (No RTP)
> Mar 20 19:02:30 DEBUG[5694] chan_sip.c: Acked pending
> invite 102 Mar 20 19:02:30 DEBUG[5694] chan_sip.c:
> Stopping retransmission on
> '3069a97536c9b3b56e105d1001324e66 at 212.23.245.87' of
> Request 102: Match Found
> Mar 20 19:02:30 DEBUG[5694] chan_sip.c: SIP response 200
> to standard invite
>
>
>
> BR
> Ercan
>
>
>
>
> -----Original Message-----
> From: asterisk-ss7-bounces at lists.digium.com
> [mailto:asterisk-ss7-bounces at lists.digium.com] On Behalf
> Of Anton Sent: Sonntag, 18. März 2007 21:57
> To: asterisk-ss7 at lists.digium.com
> Subject: Re: [asterisk-ss7] Answer() is needed but it
> costs to the callingparty while the extension is still
> ringing!
>
> Just put Dial(SIP...) or Ringing than Dial there... Don't
> put Answer in dialplan if you do not mean it.
>
> On 19 March 2007 03:51, Mitul Limbani wrote:
> > Hello Ercan,
> >
> > Quoting Ercan Yücebas <ercan at goldenphone.ch>:
> > > Dear All
> > >
> > > Is there other ways to not answer the channel in the
> > > dialpla for an inbound pstn call and just pass the
> > > signalling through and lets the sip extension
> > > ringing? After 200 ok, sure we have to answer the ss7
> > > channel. With Answer() in first position of a
> > > dialplan, the calling party starts to pay, without
> > > having been really connected!
> > >
> > > BR
> > > Ercan
> >
> > Did you try to ring extensions directly without putting
> > Answer() in your dial plan ?
> >
> > i.e. exten => s,1,Dial(SIP/${EXTEN})
> >
> > ??
> >
> > Thanks & Regards,
> > Mitul Limbani,
> > Founder & CEO,
> > Enterux Solutions,
> > The Enterprise Linux Company (TM),
> > www.enterux.com
> > _______________________________________________
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