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

[Anton]

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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