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El 19/08/10 15:07, Kathryn Jones escribió:
<blockquote
cite="mid:AANLkTi=eaNY-wDHNKrXsi_FSrCPyh_msvdKExjj9stTa@mail.gmail.com"
type="cite">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 <a class="moz-txt-link-abbreviated" href="mailto:0ef328f40a5fd6ca31a68dae2af75219@xx.xx.xx.xx">0ef328f40a5fd6ca31a68dae2af75219@xx.xx.xx.xx</a> for seqno
102 (Critical Response) -- See doc/sip-retransmit.txt.<br>
[] WARNING[1235]: chan_sip.c:3780 retrans_pkt: Maximum retries exceeded
on transmission <a class="moz-txt-link-abbreviated" href="mailto:0ef328f40a5fd6ca31a68dae2af75219@xx.xx.xx.xx">0ef328f40a5fd6ca31a68dae2af75219@xx.xx.xx.xx</a> 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 moz-do-not-send="true"
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>
Check us out at: <a moz-do-not-send="true" href="http://www.digium.com"
target="_blank">www.digium.com</a> & <a moz-do-not-send="true"
href="http://www.asterisk.org" target="_blank">www.asterisk.org</a><br>
<font color="#888888"></font><br>
</blockquote>
</div>
</blockquote>
I suggest you to debug DTMF and core, enabling them in logger.conf:<br>
<br>
console => notice,warning,error,debug,dtmf<br>
<br>
And issuing a "logger reload" command in asterisk CLI.<br>
<br>
A rxgain of 20 is too much for me, leave them in rxgain = 0.0 and
txgain= 0.0. Maybe 20dB gain is high enough to distort the audio signal
and make DTMF detection more difficult.<br>
<br>
Look at the DTMF events in your CLI, that way you can debug better. You
can enable core debug if you want issuing the CLI command "core set
debug X", with X on 1 or 2, and setting it off when you want.<br>
<br>
If your call is received from the PSTN, asterisk will detect the inband
DTMF tones in the audio signal. The rfc2833 configurations are only for
VoIP endpoints.<br>
<br>
Good luck in your debugging,<br>
<pre class="moz-signature" cols="72">--
Ing. Miguel Molina
Grupo de Tecnología
Millenium Phone Center
</pre>
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