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@xxxxxxxxxxxxxxxx
Subject: RE: 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@xxxxxxxxx]
Sent: Tuesday, March 20, 2007 10:38 PM
To: asterisk-ss7@xxxxxxxxxxxxxxxx
Subject: Re: 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 Y?cebas 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@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@xxxxxxxxxxxxxxxx
> [mailto:asterisk-ss7-bounces@xxxxxxxxxxxxxxxx] On Behalf
> Of Anton Sent: Sonntag, 18. M?rz 2007 21:57
> To: asterisk-ss7@xxxxxxxxxxxxxxxx
> 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 Y?cebas <ercan@xxxxxxxxxxxxxx>:
> > > 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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