On Mon, Apr 8, 2013 at 4:26 PM, A J Stiles <asterisk_l...@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(${EXTEN}) > > exten => 17036361355,3,Goto(testdtmf|s|1 > > ;Ring on Elle mobile phone. > > ;exten => s,1,Answer() > > ;exten => s,n,Dial(SIP/17037171234,150,r,t,) > > > > > > [general] > > register =>1112530146:albany!@#1...@sip3.voipvoip.com/1112530146 > > registertimeout=20 > > context=incoming > > allowoverlap=no > > bindport=5060 > > bindaddr=192.168.1.10 > > srvlookup=no > > ;context=incoming > > > > ; The SIP provider > > [voipvoip.com] > > canreinvite=no > > username=1112530146 > > fromuser=1112530146 > > secret=albany!@#123 > > context=incoming > > type=friend > > fromdomain=s...@voipvoip.com > > host=69.90.209.57 > > dtmfmode=rfc2833 > > disallow=all > > allow=alaw > > allow=ulaw > > nat=force_rport > > insecure=port,invite > > > > Thoughts please? I think something to do w/ "incoming" is incorrect. > > You only have one extension, "17036361355" in the [incoming] context in > your > dialplan. Are you sure that "17036361355" is exactly what the SIP provider > are actually sending to your end ? > > I'd put an "s" extension with a NoOp(${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,)
-- _____________________________________________________________________ -- 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