[asterisk-dev] Overlapped dialling on SIP trunks for outgoing calls

Klaus Darilion klaus.mailinglists at pernau.at
Mon Apr 28 15:08:04 CDT 2008


Andreas Brodmann wrote:
> 2008/4/28 Tilghman Lesher <tilghman at mail.jeffandtilghman.com 
> <mailto:tilghman at mail.jeffandtilghman.com>>:
> 
>     On Monday 28 April 2008 02:15, Klaus Darilion wrote:
>      > Johansson Olle E schrieb:
>      > > 25 apr 2008 kl. 17.34 skrev Tzafrir Cohen:
>      > >> On Fri, Apr 25, 2008 at 05:25:33PM +0200, Andreas Brodmann wrote:
>      > >>> I read the docs before writing here and I used allowoverlap=yes.
>      > >>>
>      > >>> What I would like to know from you is wheter this is
>     supported for
>      > >>> phones only (as the example below shows) or wheter it is supposed
>      > >>> to work on sip trunks for 'outgoing' calls:
>      > >>>
>      > >>> 1) in sip.conf: [general] allowoverlap=yes
>      > >>> 2) asterisk sends an INVITE to a carrier
>      > >>> 3) carrier sends 484 back
>      > >>> 4) asterisk sends congestion msg to phone.
>      > >>
>      > >> Can you emulate it in the dialplan?
>      > >
>      > > Reading the source, if we get 484 we end up here:
>      > >
>      > > Case 484:       /* Address incomplete */
>      > >                          return AST_CAUSE_INVALID_NUMBER_FORMAT
>      > >
>      > > And that's an error returned to the dialplan and the dialplan will
>      > > have to try again.
>      >
>      > Does that mean that Dial() has to be extended ?
> 
>     I did some work on this a couple weeks ago, to support a slightly
>     different
>     use of incomplete matching, so I just extended it to support SIP overlap
>     dialling.  See issue #12351.
> 
> 
>  I just had a look at your issue @ bugs.digium.com <http://bugs.digium.com>
> 
> One thing I am not sure about - would your patch also cover the 
> following scenario
> for international numbers with unknown length?
> 
> exten => _000!,1,Incomplete
> exten => _000!,n,Dial(SIP/${EXTEN:1}@carrier,120)
> .
> .
> If so, is there a backport to 1.4.x for this? If not, would it be 
> difficult to create?
> Basically all I want is asterisk to wait for an interdigit timeout for 
> international numbers,
> because their length is unknown. If this combination of phones using SIP 
> overlap dialing
> and asterisk still waiting for an interdigit timeout before sending 484 
> to the phone would
> be possible that opened a completely new perspective to real 'VoIP only' 
> environments.

How do SIP phones with overlap dialing handle the following scenario:
1. User presses "1"
2. SIP phone sends INVITE sip:1 at ....
3. user presses "2"
4. what happens now? does the phone CANCEL sip:1 at ... and INVITE sip:12 at ..?

regards
klaus



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