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asterisk at iancoetzee...
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 Posted: Wed Jan 30, 2008 2:21 am Post subject: [asterisk-users] Problem with DTMF dialing |
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Hi all
I have a small problem here. I asked this question on another asterisk
mailing list, but nobody seemed to be able to help me there.
We are running
* Asterisk 1.4.17
* Libpri 1.4.3
* Zaptel 1.4.8
on a 1.6 dual core, 2GB ram and a digium TDM800P wildcard, hardware echo
cancelation and a quad FXO card.
We have 4 analog lines, one of which is a Cellphone line for least cost
routing.
The problem I am having is dialing out using DTMF signalling. At the
moment I am making do with Pulse dialing through the 3 analog lines. I
can recieve calls on the Cellphone line without any problems, but cant
dial out through it, as a cellphone cant do pulse dialing. I have run
"ztmonitor 1 -f gains", where 1 is the zap channel where the cellphone
is located, while dialing the number 072 031 1294. I then went to
audacity, on my own pc, and converted the raw file into mp3 format,
which is available for download at
http://www.iancoetzee.za.net/tone_dial.mp3. After listening to the
playback I concluded that the DTMF signals being sent is totally wrong.
The relevant pieces of my configs are below
Your help in this matter will be greatly apreciated.
Regards
Ian
--
www.vddi.co.za <http://www.vddi.co.za/>
I Coetzee
IT Technician
Telephone : 012 664 2300
Cellphone : 079 522 6519
Fax : 012 644 2902
E-mail : ian at vddi.co.za
Skype : vddb_igcoetzee
*/etc/asterisk/zapata.conf*
; Span 1: WCTDM/0 "Wildcard TDM800P Board 1" (MASTER)
;;; line="1 WCTDM/0/0"
;Cellphone
signalling=fxs_ks
callerid=asreceived
context=incoming_calls
callerid=
group=2
busydetect=yes
usecallerid=yes
hidecallerid=no
callwaiting=no
threewaycalling=yes
transfer=yes
echocancel=yes
echotraining=yes
pulsedial=no
callprogress=yes
busycount=5
toneduration=500
subscribecontext=GXP_BLF
overlapdial=no
channel => 1
;;; line="2 WCTDM/0/1"
;Landline
signalling=fxs_ks
callerid=asreceived
context=incoming_calls
callerid=
group=1,2
busydetect=yes
usecallerid=yes
hidecallerid=no
callwaiting=no
threewaycalling=yes
transfer=yes
echocancel=yes
echotraining=yes
pulsedial=yes
callprogress=yes
busycount=5
toneduration=300
subscribecontext=GXP_BLF
channel => 2
*/etc/zaptel.conf*
# Autogenerated by /usr/sbin/zapconf on Wed Jan 16 12:23:09 2008 -- do
not hand edit
# Zaptel Configuration File
#
# This file is parsed by the Zaptel Configurator, ztcfg
#
# Span 1: WCTDM/0 "Wildcard TDM800P Board 1" (MASTER)
fxsks=1
fxsks=2
fxsks=3
fxsks=4
# channel 5, WCTDM/0/4, no module.
# channel 6, WCTDM/0/5, no module.
# channel 7, WCTDM/0/6, no module.
# channel 8, WCTDM/0/7, no module.
# Global data
loadzone = za
defaultzone = za*
*
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tzafrir.cohen at xorco...
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| Posted: Wed Jan 30, 2008 5:37 am Post subject: [asterisk-users] Problem with DTMF dialing |
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On Wed, Jan 30, 2008 at 09:21:31AM +0200, Ian wrote:
| Quote: | Hi all
I have a small problem here. I asked this question on another asterisk
mailing list, but nobody seemed to be able to help me there.
We are running
* Asterisk 1.4.17
* Libpri 1.4.3
* Zaptel 1.4.8
on a 1.6 dual core, 2GB ram and a digium TDM800P wildcard, hardware echo
cancelation and a quad FXO card.
We have 4 analog lines, one of which is a Cellphone line for least cost
routing.
The problem I am having is dialing out using DTMF signalling. At the
moment I am making do with Pulse dialing through the 3 analog lines. I
can recieve calls on the Cellphone line without any problems, but cant
dial out through it, as a cellphone cant do pulse dialing. I have run
"ztmonitor 1 -f gains", where 1 is the zap channel where the cellphone
is located, while dialing the number 072 031 1294. I then went to
audacity, on my own pc, and converted the raw file into mp3 format,
|
mp3 is a compressed format, and hence may lose some quality. Generally
you should stick with wav. ztmonitor should spit the appropriate sox
command to do the conversion. Maybe it would look slightly different in
the original format.
Is that the whole tone? It is too short to be a valid DTMF.
--
Tzafrir Cohen
icq#16849755 jabber:tzafrir.cohen at xorcom.com
+972-50-7952406 mailto:tzafrir.cohen at xorcom.com
http://www.xorcom.com iax:guest at local.xorcom.com/tzafrir |
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gandresin at gmail.com
Guest
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| Posted: Tue Feb 12, 2008 11:50 am Post subject: [asterisk-users] Problem with DTMF dialing |
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I am having similar problems running the same versions of Asterisk,
libpri & zaptel.
The Asterisk bug (http://bugs.digium.com/view.php?id=11855) was
supossed to be related to FXO only, but I am having issues with a PRI
line and Digium's TE120P.
Do you guys think it can be the same issue?
--
Andres Jimenez
GPG : http://www.andresin.com/gpg/gandresin at gmail.com.asc |
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joakimsen at gmail.com
Guest
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| Posted: Tue Feb 12, 2008 5:03 pm Post subject: [asterisk-users] Problem with DTMF dialing |
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On Feb 12, 2008 10:40 AM, Ian <asterisk at iancoetzee.za.net> wrote:
| Quote: |
Hi all,
its been quite a busy few day with pc's packing up etc, I recompile my
whole asterisk today using zaptel 1.4.7.1 and now the problem is
miraculously fixed, I will be sending this report to Digium bugs as well.
Just a quick heads up for the order in which I had to recompile in order
for this to work
Recompile Zaptel
Restart Asterisk, asterisk doesn't pick up the zap channels
Recompile Libpri
Retart Asterisk, still no zap channels
Doing the thing I was hoping to skip, Recompile Asterisk
Everything in working order Did I miss something for me to have to only
recompile zaptel, or is that the way of doing things?
Thank you all for your support
Please scroll down to see the answers to my own stupid questions 
|
Asterisk depends on Zaptel (well chan_zap and the respective codecs
do) so always make sure to install first LibPRI, then Zaptel then
Asterisk
FWIW in the wav recording you sent there is alot of static. I am
playing back with amaroK 1.4.7 of openSuSE.
On Feb 12, 2008 11:50 AM, Andres Jimenez <gandresin at gmail.com> wrote:
| Quote: | I am having similar problems running the same versions of Asterisk,
libpri & zaptel.
The Asterisk bug (http://bugs.digium.com/view.php?id=11855) was
supossed to be related to FXO only, but I am having issues with a PRI
line and Digium's TE120P.
Do you guys think it can be the same issue?
|
Maybe it is related but with PRI Asterisk does not generate any tone
it sends a signal regarding your keypress. If you are using SIP phones
make sure the dtmfmode in use is RFC2833. |
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