[asterisk-users] extensions.conf / test DID

[Satish Barot]

On Mon, Apr 8, 2013 at 4:26 PM, A J Stiles <asterisk_list at earthshod.co.uk>wrote:

> On Monday 08 April 2013, Thomas Perron wrote:
> > I am trying to make sure my DID and SIP account details are working
> > properly and engaging the extensions.conf and dial plan.
> >
> > I have a successful SIP session registered:
> >
> > Connected to Asterisk 11.3.0 currently running on Asterisk (pid = 922)
> > Asterisk*CLI> sip show registry
> > Host                                    dnsmgr Username       Refresh
> > State                Reg.Time
> > sip3.voipvoip.com:5060                  N      1112530146         105
> > Registered           Mon, 08 Apr 2013 06:02:09
> > 1 SIP registrations.
> > Asterisk*CLI>
> >
> > Here is the dial plan:
> > [incoming]
> > exten => 17036361355,1,Playback(beep)
> > exten => 17036361355,2,SayDigits(${EXTEN})
> > exten => 17036361355,3,Goto(testdtmf|s|1
> > ;Ring on Elle  mobile phone.
> > ;exten => s,1,Answer()
> > ;exten => s,n,Dial(SIP/17037171234,150,r,t,)
> >
> >
> > [general]
> > register =>1112530146:albany!@#123 at sip3.voipvoip.com/1112530146
> > registertimeout=20
> > context=incoming
> > allowoverlap=no
> > bindport=5060
> > bindaddr=192.168.1.10
> > srvlookup=no
> > ;context=incoming
> >
> > ; The SIP provider
> > [voipvoip.com]
> > canreinvite=no
> > username=1112530146
> > fromuser=1112530146
> > secret=albany!@#123
> > context=incoming
> > type=friend
> > fromdomain=sip3 at voipvoip.com
> > host=69.90.209.57
> > dtmfmode=rfc2833
> > disallow=all
> > allow=alaw
> > allow=ulaw
> > nat=force_rport
> > insecure=port,invite
> >
> > Thoughts please?    I think something to do w/ "incoming" is incorrect.
>
> You only have one extension, "17036361355" in the [incoming] context in
> your
> dialplan.  Are you sure that "17036361355" is exactly what the SIP provider
> are actually sending to your end ?
>
> I'd put an "s" extension with a  NoOp(${EXTEN}) in there, just to catch the
> actual extension number they were sending.
>
> --
> AJS
>
> Answers come *after* questions.
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>                http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users


I don't think s extension will work on SIP channel. s extension is a
catch-all extension for Analog calls and Macros (reference:
https://wiki.asterisk.org/wiki/display/AST/Handling+Special+Extensions)

Just for the sake of testing I would have something like,
 [incoming]
 exten => _X.,1,NoOp(EXTENSION=${EXTEN})
 exten => _X.,2,Playback(beep)
 exten => _X.,3,SayDigits(${EXTEN})
 exten => _X.,3,Goto(testdtmf|s|1)
;Ring on Elle  mobile phone.
;exten => s,1,Answer()
 ;exten => s,n,Dial(SIP/17037171234,150,r,t,)
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20130408/d6006d9c/attachment.htm>