Thanks for your reply :)<br><br>I added Answer to my dialplan:<br><br>exten => xxx,1,Answer()<br>exten => xxx,n,Background(welcome)<br>exten => xxx,n,WaitExten(7)<br><br>exten => _X,1,AGI(agi.php)<br>exten => _X,n,PlayBack(vm-tocallnumber)<br>
exten => _X,n,Dial(SIP/voiptrunk/${NUM})<br><br>exten => t,1,Noop(*****timeout*****)<br>exten => t,n,Playback(pbx-invalid)<br>exten => t,n,Hangup()<br><br>cli output:<br><br>-- Executing [xxx@default:1] Answer("SIP/xx.xx.xx.xx-00000004", "") in new stack<br>
-- Executing [xxx@default:2] BackGround("SIP/xx.xx.xx.xx-00000004", "welcome") in new stack<br> -- <SIP/xx.xx.xx.xx-00000004> Playing 'welcome.slin' (language 'en')<br> -- Executing [xxx@default:3] WaitExten("SIP/xx.xx.xx.xx-00000004", "7") in new stack<br>
-- Timeout on SIP/xx.xx.xx.xx-00000004, going to 't'<br> -- Executing [t@default:1] NoOp("SIP/xx.xx.xx.xx-00000004", "*****timeout*****") in new stack<br> -- Executing [t@default:2] Playback("SIP/xx.xx.xx.xx-00000004", "pbx-invalid") in new stack<br>
-- <SIP/xx.xx.xx.xx-00000004> Playing 'pbx-invalid.gsm' (language 'en')<br> -- Executing [t@default:3] Hangup("SIP/xx.xx.xx.xx-00000004", "") in new stack<br> == Spawn extension (default, t, 3) exited non-zero on 'SIP/xx.xx.xx.xx-00000004'<br>
[] WARNING[1235]: chan_sip.c:3780 retrans_pkt: Maximum retries exceeded on transmission 0ef328f40a5fd6ca31a68dae2af75219@xx.xx.xx.xx for seqno 102 (Critical Response) -- See doc/sip-retransmit.txt.<br>[] WARNING[1235]: chan_sip.c:3780 retrans_pkt: Maximum retries exceeded on transmission 0ef328f40a5fd6ca31a68dae2af75219@xx.xx.xx.xx for seqno 102 (Critical Response) -- See doc/sip-retransmit.txt.<br>
<br>I still can't read the DTMF input :(<br><br>I also tried adding:<br><br>dtmfmode = rfc2833<br>rfc2833compensate = yes<br>relaxdmtf = no ; should be no because setting it to yes cause talkoff<br><br>to sip.conf and chan_dahdi.conf<br>
and increasing rxgain=20 (I wasn't sure how much was appropriate)<br><br>Nothing seems to help.<br><br>ANY tips or ideas will be apreciated.<br><br><br><div class="gmail_quote">On Thu, Aug 19, 2010 at 1:19 PM, Tilghman Lesher <span dir="ltr"><<a href="mailto:tlesher@digium.com">tlesher@digium.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;"><div class="im">On Wednesday 18 August 2010 16:52:38 Kathryn Jones wrote:<br>
> I must not be receiving them properly, since I can't make it work. I just<br>
> can't see why :P.<br>
><br>
> My asterisk version is 1.6.2.6. Like I said before, for outgoing .call<br>
> files WaitExten works fine, it's on incoming calls that I cannot receive<br>
> the number I need.<br>
<br>
</div>There's your answer. On outgoing calls, the other end signals the line into<br>
answered state, whereas on incoming calls, you must explicitly answer the<br>
channel before listening for DTMF.<br>
<br>
--<br>
Tilghman Lesher<br>
Digium, Inc. | Senior Software Developer<br>
twitter: Corydon76 | IRC: Corydon76-dig (Freenode)<br>
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