Re: [asterisk-users] Problem with DTMF dialing
On Fri, Feb 15, 2008 at 9:05 AM, Andres Jimenez [EMAIL PROTECTED] wrote: On Wed, Feb 13, 2008 at 10:48 AM, Andres Jimenez [EMAIL PROTECTED] wrote: On Tue, Feb 12, 2008 at 10:03 PM, Andrew Joakimsen [EMAIL PROTECTED] 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 Please 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/[EMAIL PROTECTED] ___ -- 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
Re: [asterisk-users] Problem with DTMF dialing
On Wed, Feb 13, 2008 at 10:48 AM, Andres Jimenez [EMAIL PROTECTED] wrote: On Tue, Feb 12, 2008 at 10:03 PM, Andrew Joakimsen [EMAIL PROTECTED] 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/[EMAIL PROTECTED] -- Andres Jimenez GPG : http://www.andresin.com/gpg/[EMAIL PROTECTED] ___ -- 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
Re: [asterisk-users] Problem with DTMF dialing
On Tue, Feb 12, 2008 at 10:03 PM, Andrew Joakimsen [EMAIL PROTECTED] 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 -- Andres Jimenez GPG : http://www.andresin.com/gpg/[EMAIL PROTECTED] ___ -- 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
Re: [asterisk-users] Problem with DTMF dialing
On Feb 12, 2008 10:40 AM, Ian [EMAIL PROTECTED] wrote: 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 [EMAIL PROTECTED] wrote: 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. ___ -- 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
Re: [asterisk-users] Problem with DTMF dialing
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/[EMAIL PROTECTED] ___ -- 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
Re: [asterisk-users] Problem with DTMF dialing
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 1. Recompile Zaptel 2. Restart Asterisk, asterisk doesn't pick up the zap channels 3. Recompile Libpri 4. Retart Asterisk, still no zap channels 5. Doing the thing I was hoping to skip, Recompile Asterisk 6. 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 :-) Regards Ian Ian said the following on 04-Feb-08 09:38 AM: 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: 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. Btw I found the right command, I just had to do a bit of READING the usage when doing ztmonitor 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. The puase is still there though, but atleast it dials now. If you want I can upload the raw data to my server 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
Re: [asterisk-users] Problem with DTMF dialing
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 ___ -- 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
Re: [asterisk-users] Problem with DTMF dialing
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 : [EMAIL PROTECTED] Skype : vddb_igcoetzee ___ -- 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
Re: [asterisk-users] Problem with DTMF dialing
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 -- Anthony - http://messinet.com - http://messinet.com/~amessina/gallery 8F89 5E72 8DF0 BCF0 10BE 9967 92DC 35DC B001 4A4E signature.asc Description: This is a digitally signed message part. ___ -- 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
Re: [asterisk-users] Problem with DTMF dialing
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 : [EMAIL PROTECTED] Skype : vddb_igcoetzee ___ -- 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
Re: [asterisk-users] Problem with DTMF dialing
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:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- 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
[asterisk-users] Problem with DTMF dialing
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 : [EMAIL PROTECTED] 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* * ___ -- 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
