[asterisk-users] WaitExten() always times out

[Kathryn Jones]

Thanks for your reply :)

I added Answer to my dialplan:

exten => xxx,1,Answer()
exten => xxx,n,Background(welcome)
exten => xxx,n,WaitExten(7)

exten => _X,1,AGI(agi.php)
exten => _X,n,PlayBack(vm-tocallnumber)
exten => _X,n,Dial(SIP/voiptrunk/${NUM})

exten => t,1,Noop(*****timeout*****)
exten => t,n,Playback(pbx-invalid)
exten => t,n,Hangup()

cli output:

-- Executing [xxx at default:1] Answer("SIP/xx.xx.xx.xx-00000004", "") in new
stack
    -- Executing [xxx at default:2] BackGround("SIP/xx.xx.xx.xx-00000004",
"welcome") in new stack
    -- <SIP/xx.xx.xx.xx-00000004> Playing 'welcome.slin' (language 'en')
    -- Executing [xxx at default:3] WaitExten("SIP/xx.xx.xx.xx-00000004", "7")
in new stack
    -- Timeout on SIP/xx.xx.xx.xx-00000004, going to 't'
    -- Executing [t at default:1] NoOp("SIP/xx.xx.xx.xx-00000004",
"*****timeout*****") in new stack
    -- Executing [t at default:2] Playback("SIP/xx.xx.xx.xx-00000004",
"pbx-invalid") in new stack
    -- <SIP/xx.xx.xx.xx-00000004> Playing 'pbx-invalid.gsm' (language 'en')
    -- Executing [t at default:3] Hangup("SIP/xx.xx.xx.xx-00000004", "") in new
stack
  == Spawn extension (default, t, 3) exited non-zero on
'SIP/xx.xx.xx.xx-00000004'
[] WARNING[1235]: chan_sip.c:3780 retrans_pkt: Maximum retries exceeded on
transmission 0ef328f40a5fd6ca31a68dae2af75219 at xx.xx.xx.xx for seqno 102
(Critical Response) -- See doc/sip-retransmit.txt.
[] WARNING[1235]: chan_sip.c:3780 retrans_pkt: Maximum retries exceeded on
transmission 0ef328f40a5fd6ca31a68dae2af75219 at xx.xx.xx.xx for seqno 102
(Critical Response) -- See doc/sip-retransmit.txt.

I still can't read the DTMF input :(

I also tried adding:

dtmfmode = rfc2833
rfc2833compensate = yes
relaxdmtf = no ; should be no because setting it to yes cause talkoff

to sip.conf and chan_dahdi.conf
and increasing rxgain=20 (I wasn't sure how much was appropriate)

Nothing seems to help.

ANY tips or ideas will be apreciated.


On Thu, Aug 19, 2010 at 1:19 PM, Tilghman Lesher <tlesher at digium.com> wrote:

> On Wednesday 18 August 2010 16:52:38 Kathryn Jones wrote:
> > I must not be receiving them properly, since I can't make it work. I just
> > can't see why :P.
> >
> > My asterisk version is 1.6.2.6. Like I said before, for outgoing .call
> > files WaitExten works fine, it's on incoming calls that I cannot receive
> > the number I need.
>
> There's your answer.  On outgoing calls, the other end signals the line
> into
> answered state, whereas on incoming calls, you must explicitly answer the
> channel before listening for DTMF.
>
> --
> Tilghman Lesher
> Digium, Inc. | Senior Software Developer
> twitter: Corydon76 | IRC: Corydon76-dig (Freenode)
> Check us out at: www.digium.com & www.asterisk.org
>
> --
> _____________________________________________________________________
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