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
calleridgroup=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
calleridgroup=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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On Wed, Jan 30, 2008 at 09:21:31AM +0200, Ian wrote:> 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.> 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.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
Sorry for taking so long to reply, This email got lost in translation, again. Ian Ian said the following on 30-Jan-08 03:57 PM> Thaks for the speedy reply > > Tzafrir Cohen said the following on 30-Jan-08 12:37 PM: >> On Wed, Jan 30, 2008 at 09:21:31AM +0200, Ian wrote: >> >>> 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. >> > Ok I tried this everywhich way I could but everytime I came up short > of an answer. Meaning I am unable to find the right sox command to get > this converted to wav on the same computer, so once again I got it to > my pc, and then using my favourite friend, audacity I imported it as a > raw format at 8000Hz, and exported it as a wav file this time, > available for download from http://www.iancoetzee.za.net/gain.wav. it > has the same effect, the numbers I dialed and the feedback I got is > two different things. >> >>> 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. >>> >> >> Is that the whole tone? It is too short to be a valid DTMF. >> > Yes that was the dial bit, this time I included the whole recording > from beginning to end. if you count the tones you get to 10, which is > the correct amount for South Africa. Another thing that got me worried > is the fact that the last digit has a fair ammount of pause (about the > same length of another tone) before it is sent. > > If you want I can upload the raw data to my server as well. > > 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 >-------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080201/08b37f09/attachment-0001.htm
On Thursday 31 January 2008 11:52:09 pm Ian wrote:> Sorry for taking so long to reply, > > This email got lost in translation, again. > > Ian > > Ian said the following on 30-Jan-08 03:57 PM > > > Thaks for the speedy reply > > > > Tzafrir Cohen said the following on 30-Jan-08 12:37 PM: > >> On Wed, Jan 30, 2008 at 09:21:31AM +0200, Ian wrote: > >>> 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 1.4.8did you use zaptel-1.4.7 prior to this? did it work then? if so, it may be related to http://bugs.digium.com/view.php?id=11855 -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/pgp-signature Size: 189 bytes Desc: This is a digitally signed message part. Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20080201/93bdbba5/attachment.pgp
Hi Thanks for the response Anthony Messina said the following on 01-Feb-08 03:36 PM:> On Thursday 31 January 2008 11:52:09 pm Ian wrote: > >> Sorry for taking so long to reply, >> >> This email got lost in translation, again. >> >> Ian >> >> Ian said the following on 30-Jan-08 03:57 PM >> >> >>> Thaks for the speedy reply >>> >>> Tzafrir Cohen said the following on 30-Jan-08 12:37 PM: >>> >>>> On Wed, Jan 30, 2008 at 09:21:31AM +0200, Ian wrote: >>>> >>>>> 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 >>>>> > > did you use zaptel-1.4.7 prior to this? did it work then? if so, it may be > related to http://bugs.digium.com/view.php?id=11855 >No, this is a clean install, I will download 1.4.7 tonight and recompile. Any specific things I should watch for, like would I need to recompile Asterisk when I compile Zaptel, etc. Thanks for the link, I am leaving a comment there as well. Regards Ian> > ------------------------------------------------------------------------ > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > ------------------------------------------------------------------------ > > No virus found in this incoming message. > Checked by AVG Free Edition. > Version: 7.5.516 / Virus Database: 269.19.18/1255 - Release Date: 2/1/2008 9:59 AM >-------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080204/1d7f3908/attachment.htm
Thanks for the speedy reply Tzafrir Cohen said the following on 30-Jan-08 12:37 PM:> On Wed, Jan 30, 2008 at 09:21:31AM +0200, Ian wrote: > >> 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. >Ok I tried this everywhich way I could but everytime I came up short of an answer. Meaning I am unable to find the right sox command to get this converted to wav on the same computer, so once again I got it to my pc, and then using my favourite friend, audacity I imported it as a raw format at 8000Hz, and exported it as a wav file this time, available for download from http://www.iancoetzee.za.net/gain.wav. it has the same effect, the numbers I dialed and the feedback I got is two different things.> >> 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. >> > > Is that the whole tone? It is too short to be a valid DTMF. >Yes that was the dial bit, this time I included the whole recording from beginning to end. if you count the tones you get to 10, which is the correct amount for South Africa. Another thing that got me worried is the fact that the last digit has a fair ammount of pause (about the same length of another tone) before it is sent. If you want I can upload the raw data to my server as well. 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080204/b9599e5e/attachment-0001.htm
On Wed, Feb 13, 2008 at 10:48 AM, Andres Jimenez <gandresin at gmail.com> wrote:> On Tue, Feb 12, 2008 at 10:03 PM, Andrew Joakimsen <joakimsen at gmail.com> wrote: > > > > > 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. > > I have just double check and my phones use DTMF in RFC2833 mode. > > I wil try to downgrade my zaptel later today >CONFIRMED. The problem disappears after downgrading zaptel from 1.4.8 to 1.4.7> > -- > Andres Jimenez > > GPG : http://www.andresin.com/gpg/gandresin at gmail.com.asc >-- Andres Jimenez GPG : http://www.andresin.com/gpg/gandresin at gmail.com.asc
On Fri, Feb 15, 2008 at 9:05 AM, Andres Jimenez <gandresin at gmail.com> wrote:> On Wed, Feb 13, 2008 at 10:48 AM, Andres Jimenez <gandresin at gmail.com> wrote: > > On Tue, Feb 12, 2008 at 10:03 PM, Andrew Joakimsen <joakimsen at gmail.com> wrote: > > > > > > > > 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. > > > > I have just double check and my phones use DTMF in RFC2833 mode. > > > > I wil try to downgrade my zaptel later today > > > > CONFIRMED. The problem disappears after downgrading zaptel from 1.4.8 to 1.4.7Please forgive me because I was wrong. After downgrading zaptel DMTF works much better, but for some reason numbers "1" & "2" are not send through DTMF. Every other key (including "*" & "#") work like a charm. DTMF works nicely in the LAN side (i. e. voicemail login) , but if I try to reach our voicemail from the outside I see any key pressed except "1" & "2". Telephone is Grandstream GXP-2000, but I think I should blame my * . I know we are having this problem when dialing through Zap channels (Digium TE120P card) Any hint? Cheers, -- Andres Jimenez GPG : http://www.andresin.com/gpg/gandresin at gmail.com.asc