El 19/08/10 15:07, Kathryn Jones escribió:
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
<mailto: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 <http://www.digium.com> &
www.asterisk.org <http://www.asterisk.org>
I suggest you to debug DTMF and core, enabling them in logger.conf:
console => notice,warning,error,debug,dtmf
And issuing a "logger reload" command in asterisk CLI.
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.
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.
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.
Good luck in your debugging,
--
Ing. Miguel Molina
Grupo de Tecnología
Millenium Phone Center
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