[asterisk-users] DTMF issues
Diego Iastrubni
diego.iastrubni at xorcom.com
Thu Apr 19 02:24:11 MST 2007
Hi all,
I am trying to indentify a problem: I have 2 machines, one with Asterisk
1.0.11, the second with Asterisk 1.2.17. Both running with the same zaptel
(1.2.16). Asterisk 1.0.11 running on Sarge with AMP's dialplan and the 1.2.17
running on Etch with FreePBX's dial plan.
Now on both machines, I have some FXS connected (yes, I am talking about
Astrinbanks...). The problem is that when I call one FXS to another on the
1.0.11 machine I can press one of the DTMF buttons and have a solid sound. On
the 1.2.17 machine, the sound is cut off and only the first 0.1 seconds at
the beginnin and end are passed to the other side.
I can also confirm that with another dial plan the machine with cut off DTMF
sound does pass the full DTMF sound (we usually leave the keys pressed for
several seconds to test the sound quality).
Now the fun stuff:
The zapata.conf files are identical on the AMP and FreePBX setup. The other
configuration I tested, is too different to help me. The default values on
both machines in the end of this mail.
I am probably missing something very stupid, and I would like to hear what do
you think it is.
[channels]
language=en
threewaycalling=yes
transfer=yes
usecallerid=yes
echocancel=yes
echocancelwhenbridged=yes
;echotraining=800
faxdetect=incoming
;;;;;;[401]
signalling=fxo_ls
record_out=Adhoc
record_in=Adhoc
mailbox=401
immediate=no
group=5
echotraining=yes
echocancelwhenbridged=no
echocancel=yes
context=from-internal
callprogress=no
callerid=ZAP channel 1 <401>
busydetect=no
busycount=7
channel=>1
;;;;;;[402]
signalling=fxo_ls
record_out=Adhoc
record_in=Adhoc
mailbox=402
immediate=no
group=5
echotraining=yes
echocancelwhenbridged=no
echocancel=yes
context=from-internal
callprogress=no
callerid=ZAP channel 2 <402>
busydetect=no
busycount=7
channel=>2
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