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 [...@default:1] Answer("SIP/xx.xx.xx.xx-00000004", "") in new stack -- Executing [...@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 [...@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 [...@default:1] NoOp("SIP/xx.xx.xx.xx-00000004", "*****timeout*****") in new stack -- Executing [...@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 [...@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 0ef328f40a5fd6ca31a68dae2af75...@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 0ef328f40a5fd6ca31a68dae2af75...@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 <tles...@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 > > -- > _____________________________________________________________________ > -- 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 >
-- _____________________________________________________________________ -- 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