[asterisk-users] DTMF detection problem with analog card
Hi all. I have a DTMF detection problem by my new analog card (ATCOM 2 FXO port). When i`m playing a voice with 'GET DATA' AGI command, sometimes asterisk do not receive DTMF from caller while the voice is playing. But if user waits to the end of playing voice, there is no problem. I`m using Asterisk 10.3.1, dahdi-2.6.1 on CentOS.6.4. Could you please help me? Here is my configs: system.conf: fxsks=1 fxsks=2 loadzone = nl defaultzone = nl chan_dahdi.conf: -- [channels] ;=== ;General options ;=== usecallerid = yes hidecallerid = no busydetect=yes busycount=3 ;=== ;FXO Modules ;=== group = 1 signalling = fxs_ks context = my-context channel = 1,2-- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DTMF detection issues
*David Matías Hernández didi you have any luck?* *I have the same problem.* -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DTMF detection issues
Title: DAVIDMATASHERNNDEZ Hi list, I'm having trouble with DTFM tones detection. Usually, some tones are being received duplicated in Asterisk, some not. As you can imagine, that's a very big problem involving IVR menu options, Meetme conferences protected with passwords, and so on. We are currently using DAHDi 2.2.0.2, module wct4xxp, which is managing a Digium TE220B card, with a hardware echo canceller attached (VPM450). Our service is provided by 2 ISDN E1 PRI circuits (Telefonica): [ 17.188727] Found a Wildcard: Wildcard TE220 (4th Gen) [ 17.188727] TE2XXP: Launching card: 0 [ 17.188727] TE2XXP: Setting up global serial parameters [ 17.687422] About to enter spanconfig! [ 17.687422] Done with spanconfig! [ 17.687422] About to enter spanconfig! [ 17.687422] Done with spanconfig! [ 17.687422] dahdi: Registered tone zone 6 (Spain) [ 17.687422] About to enter startup! [ 17.687422] TE2XXP: Span 1 configured for CCS/HDB3/CRC4 [ 17.687422] wct2xxp: Setting yellow alarm on span 1 [ 17.687422] timing source auto card 0! [ 17.687422] SPAN 1: Primary Sync Source [ 17.687422] timing source auto card 0! [ 17.699422] VPM400: Not Present [ 17.719825] firmware: requesting dahdi-fw-oct6114-064.bin [ 17.731424] VPM450: echo cancellation for 64 channels [ 23.794606] VPM450: hardware DTMF disabled. [ 23.794610] VPM450: Present and operational servicing 2 span(s) For example, when I press 0180... I get this on Asterisk (as you can see, number 1 is duplicated): [Jun 17 12:39:05] VERBOSE[1935] logger.c: [ TYPE: DTMF End (1) SUBCLASS: 0 (48) ] [DAHDI/6-1] [Jun 17 12:39:06] VERBOSE[1935] logger.c: [ TYPE: DTMF End (1) SUBCLASS: 1 (49) ] [DAHDI/6-1] [Jun 17 12:39:06] VERBOSE[1935] logger.c: [ TYPE: DTMF End (1) SUBCLASS: 1 (49) ] [DAHDI/6-1] [Jun 17 12:39:07] VERBOSE[1935] logger.c: [ TYPE: DTMF End (1) SUBCLASS: 8 (56) ] [DAHDI/6-1] [Jun 17 12:39:08] VERBOSE[1935] logger.c: [ TYPE: DTMF End (1) SUBCLASS: 0 (48) ] [DAHDI/6-1] Our telco provider has told us that signalling is sent via inband mode, so I've messing with "toneduration", "relaxdtmf" parameters on chan_dahdi.conf, but I don't get better results... As I have read in Asterisk lists the echo canceller module can be configured to detect DTMF tones via hardware, but it's disabled by default. Maybe activating it I'll get an improvement on DTMF detection? Or should I check other configuration choices? Any help would be appreciated :) Thanks in advance, David -- C/JosEchegarayn8 Edificio3,1Planta,mdulo1 ParqueempresarialAlvia 28230LasRozas Madrid,Espaa DAVIDMATASHERNNDEZ SYSTEMSENGINEER T+34902154604 Ext.1085 F+34913575433 dmat...@optenet.com www.optenet.com M+34636396203 This information is private and confidential and intended for the recipient only. If you are not the intended recipient of this message you are hereby informed that you shall notify the sender immediately and delete the message. Dissemination, distribution or copying of this message is strictly prohibited. OPTENET, S.A. is entitled to complete the corresponding legal actions against people that access unlawfully to the content of the messages that have been sent from any of the companies within the structure of OPTENET, S.A. This communication is for information purposes only and should not be regarded as an official statement from OPTENET, S.A. Email transmission cannot be guaranteed to be secure. Therefore, we do not represent that this information is complete or accurate and it should not be relied upon as such. All information is subject to change without notice. This email has been scanned by our Antivirus system. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DTMF detection issues
Hi list, I'm having trouble with DTFM tones detection. Usually, some tones are being received duplicated in Asterisk, some not. As you can imagine, that's a very big problem involving IVR menu options, Meetme conferences protected with passwords, and so on. We are currently using DAHDi 2.2.0.2, module wct4xxp, which is managing a Digium TE220B card, with a hardware echo canceller attached (VPM450). Our service is provided by 2 ISDN E1 PRI circuits (Telefonica): [ 17.188727] Found a Wildcard: Wildcard TE220 (4th Gen) [ 17.188727] TE2XXP: Launching card: 0 [ 17.188727] TE2XXP: Setting up global serial parameters [ 17.687422] About to enter spanconfig! [ 17.687422] Done with spanconfig! [ 17.687422] About to enter spanconfig! [ 17.687422] Done with spanconfig! [ 17.687422] dahdi: Registered tone zone 6 (Spain) [ 17.687422] About to enter startup! [ 17.687422] TE2XXP: Span 1 configured for CCS/HDB3/CRC4 [ 17.687422] wct2xxp: Setting yellow alarm on span 1 [ 17.687422] timing source auto card 0! [ 17.687422] SPAN 1: Primary Sync Source [ 17.687422] timing source auto card 0! [ 17.699422] VPM400: Not Present [ 17.719825] firmware: requesting dahdi-fw-oct6114-064.bin [ 17.731424] VPM450: echo cancellation for 64 channels [ 23.794606] VPM450: hardware DTMF disabled. [ 23.794610] VPM450: Present and operational servicing 2 span(s) For example, when I press 0180... I get this on Asterisk (as you can see, number 1 is duplicated): [Jun 17 12:39:05] VERBOSE[1935] logger.c: [ TYPE: DTMF End (1) SUBCLASS: 0 (48) ] [DAHDI/6-1] [Jun 17 12:39:06] VERBOSE[1935] logger.c: [ TYPE: DTMF End (1) SUBCLASS: 1 (49) ] [DAHDI/6-1] [Jun 17 12:39:06] VERBOSE[1935] logger.c: [ TYPE: DTMF End (1) SUBCLASS: 1 (49) ] [DAHDI/6-1] [Jun 17 12:39:07] VERBOSE[1935] logger.c: [ TYPE: DTMF End (1) SUBCLASS: 8 (56) ] [DAHDI/6-1] [Jun 17 12:39:08] VERBOSE[1935] logger.c: [ TYPE: DTMF End (1) SUBCLASS: 0 (48) ] [DAHDI/6-1] Our telco provider has told us that signalling is sent via inband mode, so I've messing with toneduration, relaxdtmf parameters on chan_dahdi.conf, but I don't get better results... As I have read in Asterisk lists the echo canceller module can be configured to detect DTMF tones via hardware, but it's disabled by default. Maybe activating it I'll get an improvement on DTMF detection? Or should I check other configuration choices? Any help would be appreciated :) Thanks in advance, David -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DTMF detection on dahdi with b4xxp (again, some more details)
Hi, Am Dienstag, den 05.01.2010, 15:38 +0100 schrieb Christian Theune: Hi, I tried again getting DTMF detection on my ISDN devices with dahdi going again. I used the channel debug to see whether asterisk sees the frames and detects them as DTMF. Interestingly here's what works: 1. GSM phone - chan_dahdi g1 - asterisk - can_sip - SIP phone Both the GSM phone and the SIP phone can issue DTMF that will be detected as features (transfer) 2. GSM phone - chan_dahdi g1 - asterisk - chan_dahdi g4 - ISDN phone The GSM phone can issue DTMF that will be detected. The ISDN phone won't. (That's my issue.) I don't see any messages of asterisk recognizing the DTMF frames when pressing the keys. I do hear the DMTF sound on both phones. 3. ISDN phone - chan_dahdi g4 - asterisk - chan_dahdi g1 - GSM phone The ISDN phone can issue DTMF that will be recognized and so does the GSM phone. So. When the ISDN phone is receiving a call on g4 its DTMF sounds won't be recognized. OTOH when the GSM phone on g1 is being called it's sounds are recognized. I *think* there are two possibilities to transfer DTMF on ISDN: - as audio on B-Channel - as Key-Press events (Info-Elements) on D-Channel DTMF on GSM can not be signalled as audio (because of codec with high compression). I guess in case GSM = asterisk via chan_dahdi g1 in Your example, the DTMF is signalled as Info-Elements on D-Channel. I guess in the cases where Your DTMF is not working, audio path is used. In this case DTMF detection is done by DSP-Software. Look for the relaxdtmf statement (in case of zaptel this worked for me in a simmilar scenario). HTH, Karsten ___ -- 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] DTMF detection on dahdi with b4xxp (again, some more details)
Hi, I tried again getting DTMF detection on my ISDN devices with dahdi going again. I used the channel debug to see whether asterisk sees the frames and detects them as DTMF. Interestingly here's what works: 1. GSM phone - chan_dahdi g1 - asterisk - can_sip - SIP phone Both the GSM phone and the SIP phone can issue DTMF that will be detected as features (transfer) 2. GSM phone - chan_dahdi g1 - asterisk - chan_dahdi g4 - ISDN phone The GSM phone can issue DTMF that will be detected. The ISDN phone won't. (That's my issue.) I don't see any messages of asterisk recognizing the DTMF frames when pressing the keys. I do hear the DMTF sound on both phones. 3. ISDN phone - chan_dahdi g4 - asterisk - chan_dahdi g1 - GSM phone The ISDN phone can issue DTMF that will be recognized and so does the GSM phone. So. When the ISDN phone is receiving a call on g4 its DTMF sounds won't be recognized. OTOH when the GSM phone on g1 is being called it's sounds are recognized. Sounds like a configuration issue to me. Does anybody have an idea what to look out for? Thanks in advance, Christian -- Christian Theune · c...@gocept.com gocept gmbh co. kg · forsterstraße 29 · 06112 halle (saale) · germany http://gocept.com · tel +49 345 1229889 0 · fax +49 345 1229889 1 Zope and Plone consulting and development ___ -- 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] DTMF detection problem on DISA
Hi everybody, I am having DTMF detection problem on DISA with my callback system. For many users, it keeps playing the dialtone even after they have input their number. I have trunk setup to both g729 and ulaw. What could be the reason for this problem. Some users have to dial a few times before the system can recognize their dialed number. -- Zeeshan A Zakaria ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DTMF detection problem on DISA
On Wednesday 17 September 2008 11:25:36 Zeeshan Zakaria wrote: I am having DTMF detection problem on DISA with my callback system. For many users, it keeps playing the dialtone even after they have input their number. I have trunk setup to both g729 and ulaw. What could be the reason for this problem. Some users have to dial a few times before the system can recognize their dialed number. Check your dtmfmode setting in sip.conf, if you're using SIP lines. Trial and error, see what works with your service. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DTMF detection problem on DISA
I had it setup on rfc2833. Now I've set it up to auto. Will see how it will work. But I was thinking is it possible that DTMF tones get distorted on their way from my server to the provider's server, which cause this problem? On Wed, Sep 17, 2008 at 12:43 PM, Tilghman Lesher [EMAIL PROTECTED] wrote: On Wednesday 17 September 2008 11:25:36 Zeeshan Zakaria wrote: I am having DTMF detection problem on DISA with my callback system. For many users, it keeps playing the dialtone even after they have input their number. I have trunk setup to both g729 and ulaw. What could be the reason for this problem. Some users have to dial a few times before the system can recognize their dialed number. Check your dtmfmode setting in sip.conf, if you're using SIP lines. Trial and error, see what works with your service. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Zeeshan A Zakaria ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DTMF detection problem on DISA
On Wednesday 17 September 2008 12:02:44 Zeeshan Zakaria wrote: I had it setup on rfc2833. Now I've set it up to auto. Will see how it will work. But I was thinking is it possible that DTMF tones get distorted on their way from my server to the provider's server, which cause this problem? Anything is possible. However, the most reliable form of DTMF for SIP has always been SIP INFO. Unfortunately, not every provider supports it. -- Tilghman ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] dtmf detection not working on sip trunks using asterisk-1.4.15
Hi all, I am using an asterisk-1.4.13 connected to our carrier via SIP trunk. I use rfc2833 as dtmf detection method. After upgrading to asterisk-1.4.15 our system would not detect dtmf from a caller from PSTN anymore. When investigating the SIP traffic at call initiation I realized that in the SDP message asterisk is no longer offering the telephone-event/8000 capability. So the carrier does not send the rfc2833 messages anymore. Does anyone know about this or has seen an open bug case for it (I haven't found any myself)? Thanks for help and feedback. Kind Regards, Andreas Brodmann ___ --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] dtmf detection
Hi, Below is my case. phoneA (PSTN) phoneB (SIP) phoneC (PSTN) phoneA -- asterisk -- phoneB phoneA (music on hold), phoneB --attended call transfer-- phoneC phoneA --connect-- phoneC after phone B hangup phoneA type some keys in keypad but phoneC always has wrong dtmf detection. In my case, I would like to know any factor that will cause the wrong dtmf detection. ___ --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] DTMF detection -- Zaptel
So, I am not sure whether its a zaptel issue. It have TE212P card which has
echo based hardware cancellor.
--
Deepak
Deepak Naidu [EMAIL PROTECTED] wrote: Hi,
I have Asterisk-1.2.18 install with FreePBX more than 75 extnsion,
daily I come accross an issue try resolving them its either user learning
curve or my ignorance.
But, I dont know what to say regarding this issue.
I have my Dial Plan for internal users to have a 3 Digit Extensions.
So instance my Ext is 239 someone dials the main #, its gets the greeting
message to dial 3 digit ext. So when dialing its from my motorola Razor using
T-mobil I try to purposefully hold 2 button For more than a second then dial 3
9 which means dialing my extension 239. But I get an message saying Invalid
option, but in this case should ring my extension.
So I did the same thing running asterisk in debug mode. So there is see that
when dialing 239(that time when I hold 2 button for more than a second) its
sends 2 twice ie 22 then when I press 3 its 233, so I get Invalid option, bcos
there is no extension with 223.
I had to do this bcos I got feedback from many users saying that when
reaching their extension they get these invalid options, so using my phone was
the only way to replicate it. Further contacting Digium support they asked me
to enable. the relaxdtmf=yes option in zapata.conf. I did still the same
issue.
What is this a bug to live with or issue which has a solution.
==
Asterisk DEBUG Message
==
-- Playing 'custom/Greet1' (language 'en')
-- Invalid extension '223' in context 'ivr-2' on Zap/1-1
== CDR updated on Zap/1-1
-- Executing Playback(Zap/1-1, invalid) in new stack
-- Playing 'invalid' (language 'en')
-- Executing Goto(Zap/1-1, loop|1) in new stack
-- Goto (ivr-2,loop,1)
zapata.conf
; If you are having trouble with DTMF detection, you can relax the DTMF
; detection parameters. Relaxing them may make the DTMF detector more likely
; to have talkoff where DTMF is detected when it shouldn't be.
;
relaxdtmf=yes
---
Deepak
==
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[asterisk-users] DTMF detection problem on wctdm24xxp
hi all, i have problem with dtmf detection on wctdm24xxp with full fxo and vpm module. after pushing dtmf tones on my phone for several times the card just detects one or two digits randomly.so now i can't use any voice menu on my box with this card. i have tried the following scenarios: - the card with / without vpm module has the same dtmf detection problem. - relaxdtmf=yes/no didn't solve the problem - toneduration=300 / 350 / 400 didn't help also. - vpmdtmfsupport=1 / 0 didn't solve again. what else could be the possible cause for this problem? please help! - paradise dove ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DTMF detection problems on PRI channels?
I am using Asterisk 1.2 with a TE410P connected to E1 PRI trunks. The application relies on a DTMF digit string sent by the phone after the call has connected. This DTMF is detected by Asterisk under the control of WAIT FOR DIGIT commands send from an AGI processor over a FastAGI connection. Usually the DTMF is detected without error, but on a significant minority of calls, Asterisk is missing digits. In order to diagnose this, I modified chan_zap to save the received Alaw audio direct to a file, BEFORE the dsp is called for DTMF detection. I needed to do this because the detection routines do not pass the DTMF audio on, so using the standard recording or monitoring commands from the dialplan does not actually capture the tones as received from the wire. This capturing is turned on and off by an AGI command, so that my AGI program can turn it on before waiting for the DTMF string and off again afterwards. Examining this captured audio in an audio editor such as Goldwave does not provide any clue why the digits might have been missed. On most occasions the digits are clear, long enough and well spaced. Yet Asterisk still misses them. The system does not seem to have been heavily loaded at the time either. Can anyone offer any clues as to why this might be the case, and what I could do to solve it? Hacking the code doesn't bother me, although I know very little about DSP. Last I knew, the TE411P board could do on-board DTMF detection, but that the newer TE412P could not. Is that still the case? Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] DTMF detection problems on PRI channels?
Sounds like the DTMF tones are too far from spec, or noisy. Is the DTMF being transcoded somewhere along the way? If you have time to killtry to separate the two frequencies in your software (I don't know goldwave) - are both present and clean and same amplitude and on freq? Remove the two frequencies and what's left? If there's a lot of noise, then the other party is doing a bad job encoding the DTMF. Otherwise we can start to chase your machine causes Michelle Dupuis Technical Support Specialist Generation Software - Linux and Asterisk solutions and support. Visit us at www.generationd.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Tony Mountifield Sent: Friday, March 02, 2007 10:56 AM To: asterisk-users@lists.digium.com Subject: [asterisk-users] DTMF detection problems on PRI channels? I am using Asterisk 1.2 with a TE410P connected to E1 PRI trunks. The application relies on a DTMF digit string sent by the phone after the call has connected. This DTMF is detected by Asterisk under the control of WAIT FOR DIGIT commands send from an AGI processor over a FastAGI connection. Usually the DTMF is detected without error, but on a significant minority of calls, Asterisk is missing digits. In order to diagnose this, I modified chan_zap to save the received Alaw audio direct to a file, BEFORE the dsp is called for DTMF detection. I needed to do this because the detection routines do not pass the DTMF audio on, so using the standard recording or monitoring commands from the dialplan does not actually capture the tones as received from the wire. This capturing is turned on and off by an AGI command, so that my AGI program can turn it on before waiting for the DTMF string and off again afterwards. Examining this captured audio in an audio editor such as Goldwave does not provide any clue why the digits might have been missed. On most occasions the digits are clear, long enough and well spaced. Yet Asterisk still misses them. The system does not seem to have been heavily loaded at the time either. Can anyone offer any clues as to why this might be the case, and what I could do to solve it? Hacking the code doesn't bother me, although I know very little about DSP. Last I knew, the TE411P board could do on-board DTMF detection, but that the newer TE412P could not. Is that still the case? Cheers Tony -- Tony Mountifield Work: [EMAIL PROTECTED] - http://www.softins.co.uk Play: [EMAIL PROTECTED] - http://tony.mountifield.org ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DTMF detection during Call
Hi I have calls comming from a SIP-ATA-Box via Asterisk to PSTN Phones by outbound SIP. Now i want to detect DTMF-Tone Code coming from the called party to trigger a signal. Can this be done with asterisk? I read that the codec with DTMF detection are ulaw and alaw. But i couldn't find a command to detect dtmf's within a normal call. thanks and mani greetings Christian ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DTMF detection during Call
[EMAIL PROTECTED] wrote: Hi I have calls comming from a SIP-ATA-Box via Asterisk to PSTN Phones by outbound SIP. Now i want to detect DTMF-Tone Code coming from the called party to trigger a signal. Can this be done with asterisk? I read that the codec with DTMF detection are ulaw and alaw. But i couldn't find a command to detect dtmf's within a normal call. pbx-1*CLI show application dial pbx-1*CLI -= Info about application 'Dial' =- [Synopsis] Place a call and connect to the current channel [Description] Dial(Technology/resource[Tech2/resource2...][|timeout][|options][|URL]): This applicaiton will place calls to one or more specified channels. As soon as one of the requested channels answers, the originating channel will be answered, if it has not already been answered. These two channels will then be active in a bridged call. All other channels that were requested will then be hung up. Unless there is a timeout specified, the Dial application will wait indefinitely until one of the called channels answers, the user hangs up, or if all of the called channels are busy or unavailable. Dialplan executing will continue if no requested channels can be called, or if the timeout expires. This application sets the following channel variables upon completion: DIALEDTIME - This is the time from dialing a channel until when it is disconnected. ANSWEREDTIME - This is the amount of time for actual call. DIALSTATUS - This is the status of the call: CHANUNAVAIL | CONGESTION | NOANSWER | BUSY | ANSWER | CANCEL DONTCALL | TORTURE For the Privacy and Screening Modes, the DIALSTATUS variable will be set to DONTCALL if the called party chooses to send the calling party to the 'Go Away' script. The DIALSTATUS variable will be set to TORTURE if the called party wants to send the caller to the 'torture' script. This application will report normal termination if the originating channel hangs up, or if the call is bridged and either of the parties in the bridge ends the call. The optional URL will be sent to the called party if the channel supports it. If the OUTBOUND_GROUP variable is set, all peer channels created by this application will be put into that group (as in Set(GROUP()=...). Options: A(x) - Play an announcement to the called party, using 'x' as the file. C- Reset the CDR for this call. d- Allow the calling user to dial a 1 digit extension while waiting for a call to be answered. Exit to that extension if it exists in the current context, or the context defined in the EXITCONTEXT variable, if it exists. D([called][:calling]) - Send the specified DTMF strings *after* the called party has answered, but before the call gets bridged. The 'called' DTMF string is sent to the called party, and the 'calling' DTMF string is sent to the calling party. Both parameters can be used alone. f- Force the callerid of the *calling* channel to be set as the extension associated with the channel using a dialplan 'hint'. For example, some PSTNs do not allow CallerID to be set to anything other than the number assigned to the caller. g- Proceed with dialplan execution at the current extension if the destination channel hangs up. G(context^exten^pri) - If the call is answered, transfer both parties to the specified priority. Optionally, an extension, or extension and context may be specified. Otherwise, the current extension is used. h- Allow the called party to hang up by sending the '*' DTMF digit. H- Allow the calling party to hang up by hitting the '*' DTMF digit. j- Jump to priority n+101 if all of the requested channels were busy. L(x[:y][:z]) - Limit the call to 'x' ms. Play a warning when 'y' ms are left. Repeat the warning every 'z' ms. The following special variables can be used with this option: * LIMIT_PLAYAUDIO_CALLER yes|no (default yes) Play sounds to the caller. * LIMIT_PLAYAUDIO_CALLEE yes|no Play sounds to the callee. * LIMIT_TIMEOUT_FILE File to play when time is up. * LIMIT_CONNECT_FILE File to play when call begins. * LIMIT_WARNING_FILE File to play as warning if 'y' is defined. The default is to say the time remaining. m([class]) - Provide hold music to the calling party until a requested channel answers. A specific MusicOnHold class can be specified. M(x[^arg]) - Execute the Macro for the *called* channel before connecting to the calling channel. Arguments can be specified to the Macro using '^' as a delimeter.
[asterisk-users] DTMF detection problem in PABX trunk
Hi for all, i 've installed asterisk with isdn trunk with alcatel pabx. When alcatel users dial for external numbers, a channel on asterisk is allocated for dial. When we call to an number that is an IVR the digits isn't recognized by IVR. In sip.conf i putted dtmfmode as rfc... and info, inband is only for 64k codecs, and still don't work. How can i resolve this issue ?? Thanks. -- Frederico Madeira[EMAIL PROTECTED]www.madeira.eng.br ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DTMF detection problem in PABX trunk
Check your dtmfmode I use dtmfmode=rfc2833 Check with your provider Best regards, Al Bochter Bochter Services (Voip PBX) Toll Free: 866-638-1254 EXT: 250 (Voip PBX) Free World DialUp: 780217 EXT: 250 (Voip) Cellular: 712-432-5401 http://www.BochterServices.com/?t=Email BUY and sell Coins, Silver and Gold http://www.bochterservices.com/?j=goldt=email For new and used security items http://www.bochterservices.com/?j=storet=email_security GOLD PLATING SERVICES http://www.bochterservices.com/?j=platingt=email Frederico Madeira wrote: Hi for all, i 've installed asterisk with isdn trunk with alcatel pabx. When alcatel users dial for external numbers, a channel on asterisk is allocated for dial. When we call to an number that is an IVR the digits isn't recognized by IVR. In sip.conf i putted dtmfmode as rfc... and info, inband is only for 64k codecs, and still don't work. How can i resolve this issue ?? Thanks. -- Frederico Madeira [EMAIL PROTECTED] www.madeira.eng.br ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Inbound (clean). Database: 0643-6, 10/26/2006 - 10/27/2006 7:52:49 AM ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DTMF detection problem in PABX trunk
Hi Frederico, I had digits detection problems with my ISDN beronet cards too, do not know if u are using those cards but in case try to add s parameter to Dial command: dial(mISDN/1/123/s) It worked for me. :) Giorgio Incantalupo Frederico Madeira wrote: Hi for all, i 've installed asterisk with isdn trunk with alcatel pabx. When alcatel users dial for external numbers, a channel on asterisk is allocated for dial. When we call to an number that is an IVR the digits isn't recognized by IVR. In sip.conf i putted dtmfmode as rfc... and info, inband is only for 64k codecs, and still don't work. How can i resolve this issue ?? Thanks. -- Frederico Madeira [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] www.madeira.eng.br http://www.madeira.eng.br ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DTMF detection problem in PABX trunk
I'm already try this configuration, but don't have sucess.-- Frederico Madeira[EMAIL PROTECTED]www.madeira.eng.br 2006/10/27, Al Bochter [EMAIL PROTECTED]: Check your dtmfmode I use dtmfmode=rfc2833 Check with your provider Best regards,Al BochterBochter Services(Voip PBX) Toll Free: 866-638-1254 EXT: 250(Voip PBX) Free World DialUp: 780217 EXT: 250(Voip) Cellular: 712-432-5401 http://www.BochterServices.com/?t=Email BUY and sell Coins, Silver and Gold http://www.bochterservices.com/?j=goldt=email For new and used security items http://www.bochterservices.com/?j=storet=email_security GOLD PLATING SERVICES http://www.bochterservices.com/?j=platingt=email Frederico Madeira wrote: Hi for all, i 've installed asterisk with isdn trunk with alcatel pabx. When alcatel users dial for external numbers, a channel on asterisk is allocated for dial. When we call to an number that is an IVR the digits isn't recognized by IVR. In sip.conf i putted dtmfmode as rfc... and info, inband is only for 64k codecs, and still don't work. How can i resolve this issue ?? Thanks. -- Frederico Madeira [EMAIL PROTECTED] www.madeira.eng.br ___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Inbound (clean). Database: 0643-6, 10/26/2006 - 10/27/2006 7:52:49 AM ___--Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DTMF Detection Problems with certain phones incoming zap channels
Hello,
I'm having a problem with the autoattendant. It won't recognize the
DTMF signals from certain people that call in. I have relaxed DTMF,
upgraded Asterisk from 1.2 to 1.2.12 to 1.2.12.1 as well as the zaptel
drivers. I have stopped X from running then only thing I didn't do
that was on Digium's support website was to reconpile vpmdtmfsupport
to 0 in wctdm24xxp.c or wct4xxp.c
Extansions.conf
[incoming]
exten=s,1,Answer
exten=s,2,Wait,1
exten=s,3,Set(TIMEOUT(digit)=3)
exten=s,4,Set(TIMEOUT(response)=10)
exten=s,5,Background(welcome)
;exten=s,2,WaitExten
exten=i,1,Playback(invalid)
exten=i,n,Goto(incoming,s,1)
exten=6,1,Set(TIMEOUT(digit)=6)
exten=6,2,Set(TIMEOUT(response)=10)
exten=6,3,Background(dir-intro)
;exten=6,2,WaitExten
exten=0,1,Dial(SIP/12|20)
exten=0,2,VoiceMail(u0)
exten=t,1,Goto(incoming,s,1)
exten=h,1,Hangup()
exten=627,1,Background(johnis)
exten=627,2,Goto(mastro,10,1)
exten=372,1,Background(fradeis)
exten=372,2,Goto(frade,11,1)
exten=386,1,Background(eileenis)
exten=386,2,Goto(eileen,12,1)
exten=10,1,Goto(mastro,10,1)
exten=11,1,Goto(frade,11,1)
exten=12,1,Goto(eileen,12,1)
exten=13,1,Goto(conference,13,2)
exten=8500,1,VoiceMailMain
exten=17,1,Dial(SIP/s12|20)
[internal]
exten=10,1,Goto(mastro,10,1)
exten=11,1,Goto(frade,11,1)
exten=12,1,Goto(eileen,12,1)
exten=13,1,Goto(conference,13,1)
exten=110,1,VoiceMail(u10)
exten=111,1,VoiceMail(u11)
exten=112,1,VoiceMail(u12)
exten=_NXX,1,Dial,Zap/g1/w${EXTEN}w
exten=_NXX,103,Congestion
exten=_NXX,104,Hangup()
exten=_1NXXNXX,1,Dial,Zap/G1/w${EXTEN}w
exten=_1NXXNXX,103,Congestion
exten=_1NXXNXX,104,Hangup()
exten=_011.,1,Dial,Zap/G1/w${EXTEN}w
exten=_011.,103,Congestion
exten=_011.,104,Hangup()
exten=8500,1,VoiceMailMain
[mastro]
exten=10,1,Dial(SIP/10|20)
exten=10,2,Set(TIMEOUT(digit)=3)
exten=10,3,Set(TIMEOUT(response)=10)
exten=10,4,Background(john)
;exten=10,3,WaitExten
exten=10,5,VoiceMail(u10)
exten=10,101,VoiceMail(b10)
exten=1,1,VoiceMail(u10)
exten=2,1,Goto(eileen,12,1)
exten=3,1,Dial(Zap/G1/w19175450294w|10)
exten=i,1,Playback(invalid)
exten=i,n,Goto(mastro,10,2)
exten=t,1,Congestion(5)
exten=h,1,Hangup()
exten=8500,1,VoiceMailMain
[frade]
exten=11,1,Dial(SIP/11|20)
exten=11,2,Dial(SIP/s11|20)
exten=11,3,Set(TIMEOUT(digit)=3)
exten=11,4,Set(TIMEOUT(response)=10)
exten=11,5,Background(manny)
;exten=11,4,WaitExten
exten=11,6,VoiceMail(u11)
exten=11,101,VoiceMail(b11)
exten=1,1,VoiceMail(u11)
exten=2,1,Dial(Zap/G1/w15165325627w|10)
exten=i,1,Playback(invalid)
exten=i,n,Goto(frade,11,3)
exten=t,1,Congestion(5)
exten=h,1,Hangup()
exten=8500,1,VoiceMailMain
[eileen]
exten=12,1,Dial(SIP/12|20)
exten=12,2,VoiceMail(u12)
exten=12,101,VoiceMail(b12)
exten=t,1,Congestion(5)
exten=h,1,Hangup()
exten=8500,1,VoiceMailMain
[conference]
exten=13,1,Dial(SIP/13|20)
exten=t,1,Congestion(5)
exten=h,1,Hangup()
exten=8500,1,VoiceMailMain
Zapata.conf
[channels]
signalling=fxs_ks
context=incoming
language=en
rxgain=25
txgain=0
usecallerid=no
;callprogress=yes
;hanguponpolarityswitch=yes
;callerid=asreceived
;cidsignalling=bell
;cidstart=ring
echocancel=128
echotraining=800
echocancelwhenbridged=yes
transfer=yes
immediate=yes
;relaxdtmf=yes
;busydetect=yes
;busycount=5
group=1
channel=1-4
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Re: [asterisk-users] DTMF Detection Problems with certain phones incoming zap channels
On Tue, Sep 19, 2006 at 05:29:17PM -0400, asterisk asterisk wrote: I'm having a problem with the autoattendant. It won't recognize the DTMF signals from certain people that call in. I have relaxed DTMF, Are you sure? Zapata.conf [channels] signalling=fxs_ks [ ... ] ;relaxdtmf=yes I'm no expert, but it looks like you haven't. Unless you did earlier, and turned it back off. Or I'm an idiot. :-) Cheers -- jra -- Jay R. Ashworth[EMAIL PROTECTED] Designer Baylink RFC 2100 Ashworth AssociatesThe Things I Think'87 e24 St Petersburg FL USA http://baylink.pitas.com +1 727 647 1274 That's women for you; you divorce them, and 10 years later, they stop having sex with you. -- Jennifer Crusie; _Fast_Women_ ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DTMF Detection Problems with certain phones incoming zap channels
I did turn it on and off as it does not seem to make a difference. On 9/19/06, Jay R. Ashworth [EMAIL PROTECTED] wrote:On Tue, Sep 19, 2006 at 05:29:17PM -0400, asterisk asterisk wrote: I'm having a problem with the autoattendant. It won't recognize the DTMF signals from certainpeople that call in. I have relaxed DTMF,Are you sure? Zapata.conf [channels] signalling=fxs_ks[ ... ] ;relaxdtmf=yesI'm no expert, but it looks like you haven't.Unless you did earlier, and turned it back off.Or I'm an idiot.:-)Cheers-- jra--Jay R. Ashworth[EMAIL PROTECTED]DesignerBaylink RFC 2100 Ashworth AssociatesThe Things I Think'87 e24St Petersburg FL USAhttp://baylink.pitas.com +1 727 647 1274That's women for you; you divorce them, and 10 years later, they stop having sex with you.-- Jennifer Crusie; _Fast_Women--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DTMF detection and Sangoma cards
On 7/12/06, El Flynn [EMAIL PROTECTED] wrote: Are you only having this problem for call parking? Any issues when the caller is navigating an IVR? We're not running an IVR on this particular system. Here's the strange thing: the DTMF is not coming from the inbound caller but rather, the agents who are using Polycom SIP phones and trying to park the calls. I'm not sure how Asterisk's DTMF detection works but in this instance, despite what I said earlier, I'm starting to think that the Sangoma card is not involved. It gets even stranger: when I call into the system using a Polycom phone (over POTS, on a different PBX in a different state), the agent can park my call. When I call in with my cell phone, the agent cannot park the call. Strange! Any ideas? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] DTMF detection and Sangoma cards
I have not read through this entire thread, but I used to experience an issue on Polycom phones where if you were on a call, interacting with an IVR menu, if a call came in on the second line, you could not interact with the IVR via DTMF, until the call coming in on the second line went away. Cory Andrews Executive Vice President ++ VoIPSupply.com PBXSelect.com ++ 454 Sonwil Drive Buffalo, NY 14225 voice - 800.398.VoIP X3402 fax - 716.630.1548 e - [EMAIL PROTECTED] m - 716.907.4059 aim - B2Cory -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Christopher Snell Sent: Thursday, July 13, 2006 1:31 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] DTMF detection and Sangoma cards On 7/12/06, El Flynn [EMAIL PROTECTED] wrote: Are you only having this problem for call parking? Any issues when the caller is navigating an IVR? We're not running an IVR on this particular system. Here's the strange thing: the DTMF is not coming from the inbound caller but rather, the agents who are using Polycom SIP phones and trying to park the calls. I'm not sure how Asterisk's DTMF detection works but in this instance, despite what I said earlier, I'm starting to think that the Sangoma card is not involved. It gets even stranger: when I call into the system using a Polycom phone (over POTS, on a different PBX in a different state), the agent can park my call. When I call in with my cell phone, the agent cannot park the call. Strange! Any ideas? ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] DTMF detection and Sangoma cards
Hi, I posted earlier about Call parking breaks suddenly. I believe that I have narrowed this down to a problem with DTMF detection and the Sangoma A101 card that we use. Earlier, DTMF detection was not working at all. Then, I set 'relaxdtmf=yes' in zapata.conf and it works...sort of. When I call into the PBX from a digital desk phone, Asterisk is able to detect the agent's DTMF and parks the call as requested. However, if I dial in from my cell phone, the agent's DTMF is not detected and the caller (me) hears the DTMF on the lines. Does anybody have any ideas? thanks, Chris ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DTMF detection and Sangoma cards
Christopher Snell wrote: Hi, I posted earlier about Call parking breaks suddenly. I believe that I have narrowed this down to a problem with DTMF detection and the Sangoma A101 card that we use. Earlier, DTMF detection was not working at all. Then, I set 'relaxdtmf=yes' in zapata.conf and it works...sort of. When I call into the PBX from a digital desk phone, Asterisk is able to detect the agent's DTMF and parks the call as requested. However, if I dial in from my cell phone, the agent's DTMF is not detected and the caller (me) hears the DTMF on the lines. Does anybody have any ideas? We've been running a Sangoma A104 at a client site for the past 12 months without any DTMF issues whatsoever, neither from the inbound nor the outbound side. That unit's connected to a multitude of analog and IP phones, as well as a large legacy PABX behind the * box. Some numbers on the PRIs are provisioned to hit a lengthy IVR menu tree, so I know the DTMF works. Bear in mind it's running much older versions of Asterisk, Zaptel and libpri. Are you only having this problem for call parking? Any issues when the caller is navigating an IVR? Flynn ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DTMF detection and Sangoma cards
I can attest to this. We've got one of these Sangoma cards at a customer site and have no DTMF detection issues whatsoever.AlexOn 7/12/06, El Flynn [EMAIL PROTECTED] wrote: Christopher Snell wrote: Hi, I posted earlier about Call parking breaks suddenly.I believe that I have narrowed this down to a problem with DTMF detection and the Sangoma A101 card that we use. Earlier, DTMF detection was not working at all.Then, I set 'relaxdtmf=yes' in zapata.conf and it works...sort of.When I call into the PBX from a digital desk phone, Asterisk is able to detect the agent's DTMF and parks the call as requested. However, if I dial in from my cell phone, the agent's DTMF is not detected and the caller (me) hears the DTMF on the lines. Does anybody have any ideas? We've been running a Sangoma A104 at a client site for the past 12 monthswithout any DTMF issues whatsoever, neither from the inbound nor the outboundside. That unit's connected to a multitude of analog and IP phones, as well as a large legacy PABX behind the * box. Some numbers on the PRIs are provisioned tohit a lengthy IVR menu tree, so I know the DTMF works.Bear in mind it's running much older versions of Asterisk, Zaptel and libpri. Are you only having this problem for call parking? Any issues when the caller isnavigating an IVR?Flynn___--Bandwidth and Colocation provided by Easynews.com --asterisk-users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Alex Robar[EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] DTMF Detection: Where it happens actually?
Hello, Could anyone help me to figure out the following questions, please: 1. Whenever there is an incoming DTMF signal on the Zap channel, where does the processing actually take place: In Asterisk?; or in Zaptel Drivers? 2. I'm having a problem of double (or sometimes tripple) detection of a single DTMF received whenever I'm calling in through a Mobile Phone. I guess I need to make some ammendments in the DTMF detection algorithm. Now the question is where can I find the code responsible for DTMF processing: in channel driver?; or in the Asterisk source? Your comments are appreciated. Kind Regards, Hohenzolern ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] DTMF Detection Problems on VGSM channel
Hello Asterisk Community, I'm using Voismart's GSM PCI cards to connect Asterisk to GSM cellular network. The problem I face is DTMF detection; that is, whenever I call to one of the channels (SIMs) on GSM card through my Mobile phone, and dial DTMF digits while in the call, the Asterisk receives almost all the digits in multiple samples. e.g. I dial 123456789 and Asterisk receives kind of 11122455678999. I tried to change the DTMF settings on my Mobile phone (There is one on my phone); Changed DTMF type to Long, short... did not help. I guess those GSM cards have problems in DTMF detecting algorithm (I think DTMF detection burden lies on GSM cards)... Has anyone faced this issue, or I'm the one to struggle? Thanks for your comments. Regards, Hohenzolern ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
AW: AW: [Asterisk-Users] DTMF detection when outgoing call tomobilephones
Ok.. Here we are again - with more input ;-) Probably I have sorted out the source of the problem. I think it is OVERLAPDIAL. For example: OVERLAPDIAL is set to YES - The DTMF recognition at mobiles on outgoing calls is not available OVERLAPDIAL is set to NO - Every DTMF tone is deteced VERY WELL on outgoing calls! But: If I set overlapdial to no I will have problems with incoming calls not using block dial. They are transfered to wrong extensions. Why depends OVERLAPDIAL to outgoing calls and DTMF detection - I can not see it. So I have two options - like always in life - 1st = DTMF detection well but sometimes no inbound calls 2nd = perfect inbound calls but not outgoing DTMF detection -Ursprüngliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Im Auftrag von Kevin P. Fleming Gesendet: Freitag, 5. Mai 2006 09:52 An: Asterisk Users Mailing List - Non-Commercial Discussion Betreff: Re: AW: [Asterisk-Users] DTMF detection when outgoing call tomobilephones Marc Scheuffler wrote: Yes I can hear the DTMF keys. I ve tried 2 different phones and 3 different mobile network providers. Nothing. There was a bug in various versions of Asterisk when outbound calls were placed using spool files and then could not detect DTMF from the called party. Without more details, including the version of Asterisk you are running, it will be difficult to suggest anything to you other than upgrading to the latest release. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
AW: AW: [Asterisk-Users] DTMF detection when outgoing call tomobilephones
I have made a step forward regarding my problem. It seems to be a routing problem of the telco provider. The provider is using different routings depending on the called number. We have successfully tested a different routing than the standard one. So thank you for now for the support! Cheers. Marc -Ursprüngliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Im Auftrag von Kevin P. Fleming Gesendet: Freitag, 5. Mai 2006 09:52 An: Asterisk Users Mailing List - Non-Commercial Discussion Betreff: Re: AW: [Asterisk-Users] DTMF detection when outgoing call tomobilephones Marc Scheuffler wrote: Yes I can hear the DTMF keys. I ve tried 2 different phones and 3 different mobile network providers. Nothing. There was a bug in various versions of Asterisk when outbound calls were placed using spool files and then could not detect DTMF from the called party. Without more details, including the version of Asterisk you are running, it will be difficult to suggest anything to you other than upgrading to the latest release. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: AW: [Asterisk-Users] DTMF detection when outgoing call to mobilephones
Marc Scheuffler wrote: Yes I can hear the DTMF keys. I ve tried 2 different phones and 3 different mobile network providers. Nothing. There was a bug in various versions of Asterisk when outbound calls were placed using spool files and then could not detect DTMF from the called party. Without more details, including the version of Asterisk you are running, it will be difficult to suggest anything to you other than upgrading to the latest release. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: AW: [Asterisk-Users] DTMF detection when outgoing call tomobilephones
There was a bug in various versions of Asterisk when outbound calls were placed using spool files and then could not detect DTMF from the called party. Without more details, including the version of Asterisk you are running, it will be difficult to suggest anything to you other than upgrading to the latest release. Works perfectly on mine. Using Digium Digital boards (ZAP - UK Mobile). Asterisk version 1.2.1 with a few patches. Mark ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
AW: AW: [Asterisk-Users] DTMF detection when outgoing call tomobilephones
Actually I am using Asterisk 1.2.7.1, zaptel 1.2.5, libpri 1.2.2 I ve tried many values for rx/txgain togeher with echocancel and relaxdtmf. The detection is not working with call file, manager originate and not with the dial command to the mobile. I have no ideas left. I got it sometimes to work if I use a specific channel (i.e. Dial(ZAP/14/...) But with the same vaules on a second call there is the DTMF problem again. Sometimes the DTMF tones are choppy at the other end of the line. There is also a Warning with outbound calls only: -- Called G1/016080... -- Moving call from channel 2 to channel 31 May 5 12:45:33 WARNING[1403]: chan_zap.c:7744 pri_fixup_principle: Can't fix up channel from 2 to 31 because 31 is already in use May 5 12:45:33 WARNING[1403]: chan_zap.c:9045 pri_dchannel: Unable to move channel 31! -- Zap/31-1 is ringing -- Zap/31-1 answered Zap/2-1 But everything is ok and working on not moblie connections My actual settings: Zaptel.conf # # Zaptel Configuration File # span=1,1,0,ccs,hdb3,crc4 span=2,0,0,ccs,hdb3,crc4 # bchan=1-15 dchan=16 bchan=17-31 bchan=32-46 dchan=47 bchan=48-62 # loadzone = de #loadzone = us-old #loadzone=gr #loadzone=it #loadzone=fr #loadzone=de #loadzone=uk #loadzone=fi #loadzone=jp #loadzone=sp #loadzone=no #loadzone=hu #loadzone=lt #loadzone=pl defaultzone=nl Zapata.conf ; ; Zapata telephony interface ; ; Configuration file ; [trunkgroups] [channels] ; context=default switchtype=euroisdn pridialplan=unknown prilocaldialplan=unknown ; ; sample 1 for Germany ;internationalprefix = 00 ;nationalprefix = 0 ;localprefix = ;privateprefix = ;unknownprefix = ;relaxdtmf=yes overlapdial=yes ;priindication = outofband ;rxwink=300 ; Atlas seems to use long (250ms) winks ;toneduration=100 ;usedistinctiveringdetection=yes ;busydetect=no ;immediate=no usecallerid=yes hidecallerid=no callwaiting=yes usecallingpres=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes canpark=yes cancallforward=yes callreturn=yes ;echocancel=yes echocancelwhenbridged=yes ;rxgain=2.0 ;txgain=0.0 ; group=1 callgroup=1 pickupgroup=1 immediate=no ; ;callprogress=yes ;progzone=us ; ;jitterbuffers=4 ; group=1 ;switching=euroisdn signalling=pri_cpe context=from-pmx rxgain=2.0 txgain=0.0 relaxdtmf=yes echocancel=yes callerid=asreceived channel=1-15, 17-31 group=2 ;switching=euroisdn signalling=pri_net context=to-pmx callerid=asreceived channel=32-46, 48-62 -Ursprüngliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Im Auftrag von Kevin P. Fleming Gesendet: Freitag, 5. Mai 2006 09:52 An: Asterisk Users Mailing List - Non-Commercial Discussion Betreff: Re: AW: [Asterisk-Users] DTMF detection when outgoing call tomobilephones Marc Scheuffler wrote: Yes I can hear the DTMF keys. I ve tried 2 different phones and 3 different mobile network providers. Nothing. There was a bug in various versions of Asterisk when outbound calls were placed using spool files and then could not detect DTMF from the called party. Without more details, including the version of Asterisk you are running, it will be difficult to suggest anything to you other than upgrading to the latest release. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: AW: [Asterisk-Users] DTMF detection when outgoing call tomobilephones
Mark Ackroyd wrote: There was a bug in various versions of Asterisk when outbound calls were placed using spool files and then could not detect DTMF from the called party. Without more details, including the version of Asterisk you are running, it will be difficult to suggest anything to you other than upgrading to the latest release. Works perfectly on mine. Using Digium Digital boards (ZAP - UK Mobile). Asterisk version 1.2.1 with a few patches. Also works for me with USA cell phones on at least two different Asterisk servers with version 1.2.x -- Now accepting new clients in Birmingham, Atlanta, Huntsville, Chattanooga, and Montgomery. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] DTMF detection when outgoing call to mobile phones
Hi all, I am trying to detect DTMF keys from a mobile when asterisk make an outgoing call to the mobile. The DTMF detection on incoming calls (also FROM mobiles) is working very well. The only problem is if asterisk called the phone... Nothing is detected. I am using a digium te205p with PMX/PSTN connection. Everything that I can find in forums are problems with dtmf detection on SIP. Any suggestions? Cheers Marc ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] DTMF detection when outgoing call to mobile phones
From my recent problem on this sort of thing, I'd suggest you set the timeout to around 1500ms in the feature.conf file. This is of course if your using the DTMF digit's to activate any of the features. also make the devices both sides of the call are using the same DTMF mode. Mark Hi all, I am trying to detect DTMF keys from a mobile when asterisk make an outgoing call to the mobile. The DTMF detection on incoming calls (also FROM mobiles) is working very well. The only problem is if asterisk called the phone... Nothing is detected. I am using a digium te205p with PMX/PSTN connection. Everything that I can find in forums are problems with dtmf detection on SIP. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DTMF detection when outgoing call to mobile phones
Marc Scheuffler wrote: Hi all, I am trying to detect DTMF keys from a mobile when asterisk make an outgoing call to the mobile. The DTMF detection on incoming calls (also FROM mobiles) is working very well. The only problem is if asterisk called the phone... Nothing is detected. I am using a digium te205p with PMX/PSTN connection. Everything that I can find in forums are problems with dtmf detection on SIP. Any suggestions? Can you hear DTMF tones from the cellphone when you call it? It is possible they are not produced. I'm not sure if that is a handset or network issue, but I had this happen a few years ago when implementing an IVR that called subscribers for notification, and expected input from them. Most of the time it worked, but some phones in some locations sent nothing. If the phone made a call from the same location there was no problem. Steve ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
AW: [Asterisk-Users] DTMF detection when outgoing call to mobilephones
Yes I can hear the DTMF keys. I ve tried 2 different phones and 3 different mobile network providers. Nothing. I played with the rx/txgain values from hearing nothing to too loud... I have no more ideas. Marc -Ursprüngliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Im Auftrag von Steve Underwood Gesendet: Donnerstag, 4. Mai 2006 17:00 An: Asterisk Users Mailing List - Non-Commercial Discussion Betreff: Re: [Asterisk-Users] DTMF detection when outgoing call to mobilephones Marc Scheuffler wrote: Hi all, I am trying to detect DTMF keys from a mobile when asterisk make an outgoing call to the mobile. The DTMF detection on incoming calls (also FROM mobiles) is working very well. The only problem is if asterisk called the phone... Nothing is detected. I am using a digium te205p with PMX/PSTN connection. Everything that I can find in forums are problems with dtmf detection on SIP. Any suggestions? Can you hear DTMF tones from the cellphone when you call it? It is possible they are not produced. I'm not sure if that is a handset or network issue, but I had this happen a few years ago when implementing an IVR that called subscribers for notification, and expected input from them. Most of the time it worked, but some phones in some locations sent nothing. If the phone made a call from the same location there was no problem. Steve ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
AW: [Asterisk-Users] DTMF detection when outgoing call to mobilephones
Yapp, timeout is set to 1500ms. What kind of dtmf mode? As far as i know there are just 2. Relaxdtmf yes or no Or am I wrong? -Ursprüngliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Im Auftrag von Mark Ackroyd Gesendet: Donnerstag, 4. Mai 2006 16:52 An: 'Asterisk Users Mailing List - Non-Commercial Discussion' Betreff: RE: [Asterisk-Users] DTMF detection when outgoing call to mobilephones From my recent problem on this sort of thing, I'd suggest you set the timeout to around 1500ms in the feature.conf file. This is of course if your using the DTMF digit's to activate any of the features. also make the devices both sides of the call are using the same DTMF mode. Mark Hi all, I am trying to detect DTMF keys from a mobile when asterisk make an outgoing call to the mobile. The DTMF detection on incoming calls (also FROM mobiles) is working very well. The only problem is if asterisk called the phone... Nothing is detected. I am using a digium te205p with PMX/PSTN connection. Everything that I can find in forums are problems with dtmf detection on SIP. ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] DTMF detection in TE406P ??
Hi, Im getting a lot of false DTMF detections on my system. Following is a diagram of my system: PRI-TE406P SPAN1-TE406P SPAN3-PABX Basically anyone talking to me with a higher pitch voice (Ladies) I get beeps all over the place. If I unplug PRI from Asterisk and plug it directly to PABX I do not get any beeps during conversation. I noticed that the latest wct4xxp sources allow disabling DTMF support in VPM modules. Will it help me in this situation and if disabled, will I still be able to call IVR systems from my PABX? Thanks in advance! ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DTMF detection in TE406P ??
Boris Bakchiev wrote: PRI-TE406P SPAN1-TE406P SPAN3-PABX Please contact Digium technical support. I noticed that the latest wct4xxp sources allow disabling DTMF support in VPM modules. There are other steps that can be taken if necessary first. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DTMF detection in TE406P ??
On Wednesday 09 November 2005 21:46, Boris Bakchiev wrote: I noticed that the latest wct4xxp sources allow disabling DTMF support in VPM modules. It just disables the VPM's capability to detect DTMF; DTMF detection still works within Asterisk as it'll be done on the host CPU (i.e. the old way before the TE406). Will it help me in this situation and if disabled, will I still be able to call IVR systems from my PABX? Yes, it clears this up very well, at least in my experience. -A. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DTMF detection in TE406P ??
Kevin P. Fleming wrote: There are other steps that can be taken if necessary first. Can you please elaborate on this? It may just save a lot of calls to Digium support about the same issue. (I have noticed this sporadically). ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DTMF detection in TE406P ??
We disabled it in the wct4xxp.c driver as well and everything has worked great since then. I have heard from support to try raising the threshold value while leaving hardware DTMF detection on and it didn't help raising it by 100. In the big picture, doing hardware DTMF detection gives you an incredibly small performance boost so you really won't notice any load difference if you disable it. MATT--- On 11/9/05, Boris Bakchiev [EMAIL PROTECTED] wrote: Hi, I'm getting a lot of false DTMF detections on my system. Following is a diagram of my system: PRI-TE406P SPAN1-TE406P SPAN3-PABX Basically anyone talking to me with a higher pitch voice (Ladies) I get beeps all over the place. If I unplug PRI from Asterisk and plug it directly to PABX I do not get any beeps during conversation. I noticed that the latest wct4xxp sources allow disabling DTMF support in VPM modules. Will it help me in this situation and if disabled, will I still be able to call IVR systems from my PABX? Thanks in advance! ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DTMF detection
Tole spada v DTMF zgodbo... - Original Message - From: Ryan [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Sunday, October 23, 2005 6:35 AM Subject: Re: [Asterisk-Users] DTMF detection On Sat, Oct 22, 2005 at 12:53:39PM -0600, Ryan exclaimed: snip I just setup telasip and I'm having the same issue. I captured some RTP packets and realized that when I get duplicate numbers it is because an RTP packet has arrived out of order. In all my test cases it was just one packet coming 1 packet too late, but the sequence number was correct. It seems that * instead of putting the packets back in order (using the seq numbers) makes a duplicate digit. I'm not sure if this is a bug or not (I haven't read the rfc). I found this in mantis at: http://bugs.digium.com/view.php?id=4659 Unfortunately this will require upstream providers to patch asterisk before this will work (which will happen over time). ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DTMF detection
Sorry, went on wrong address Regards, Rob. - Original Message - From: Robert Rozman [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, October 29, 2005 9:22 AM Subject: Re: [Asterisk-Users] DTMF detection Tole spada v DTMF zgodbo... - Original Message - From: Ryan [EMAIL PROTECTED] To: asterisk-users@lists.digium.com Sent: Sunday, October 23, 2005 6:35 AM Subject: Re: [Asterisk-Users] DTMF detection On Sat, Oct 22, 2005 at 12:53:39PM -0600, Ryan exclaimed: snip I just setup telasip and I'm having the same issue. I captured some RTP packets and realized that when I get duplicate numbers it is because an RTP packet has arrived out of order. In all my test cases it was just one packet coming 1 packet too late, but the sequence number was correct. It seems that * instead of putting the packets back in order (using the seq numbers) makes a duplicate digit. I'm not sure if this is a bug or not (I haven't read the rfc). I found this in mantis at: http://bugs.digium.com/view.php?id=4659 Unfortunately this will require upstream providers to patch asterisk before this will work (which will happen over time). ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DTMF detection
Hello all, yes there is a lot of information about this on the wiki and in past posts on this list but have not found anything that has solved my problem. setup is: phone--PAP2-na--asterisk v1.0.9(in house on local subnet dtmf is inband)---PSTN---Telisipasterisk box at colo v1.0.9 VoIP only. I have only access to dial up so can not go VoIP out of the house. In extensions.conf on colo * i have some logic that based on callerid lets me hit a single digit to get to DISA, this work every time. the problem is that when i enter a number for DISA to dial i get duplicate digits, example i enter 6037862111 and disa tries to dial 6003778621. I have tried setting relaxdtmf=yes in sip.conf with no luck. I have read on the wiki that RFC2833 should work, but alas its a no go. I am also using ulaw which should not be distorting the dtmf through compresion, correct? Also with RFC2833 it should not matter? Everything works great otherwise. sip.conf for colo * is posted below: I just setup telasip and I'm having the same issue. I captured some RTP packets and realized that when I get duplicate numbers it is because an RTP packet has arrived out of order. In all my test cases it was just one packet coming 1 packet too late, but the sequence number was correct. It seems that * instead of putting the packets back in order (using the seq numbers) makes a duplicate digit. I'm not sure if this is a bug or not (I haven't read the rfc). Thanks, Ryan ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DTMF detection
On Sat, Oct 22, 2005 at 12:53:39PM -0600, Ryan exclaimed: snip I just setup telasip and I'm having the same issue. I captured some RTP packets and realized that when I get duplicate numbers it is because an RTP packet has arrived out of order. In all my test cases it was just one packet coming 1 packet too late, but the sequence number was correct. It seems that * instead of putting the packets back in order (using the seq numbers) makes a duplicate digit. I'm not sure if this is a bug or not (I haven't read the rfc). I found this in mantis at: http://bugs.digium.com/view.php?id=4659 Unfortunately this will require upstream providers to patch asterisk before this will work (which will happen over time). ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] DTMF detection
Hello all, yes there is a lot of information about this on the wiki and in past posts on this list but have not found anything that has solved my problem. setup is: phone--PAP2-na--asterisk v1.0.9(in house on local subnet dtmf is inband)---PSTN---Telisipasterisk box at colo v1.0.9 VoIP only. I have only access to dial up so can not go VoIP out of the house. In extensions.conf on colo * i have some logic that based on callerid lets me hit a single digit to get to DISA, this work every time. the problem is that when i enter a number for DISA to dial i get duplicate digits, example i enter 6037862111 and disa tries to dial 6003778621. I have tried setting relaxdtmf=yes in sip.conf with no luck. I have read on the wiki that RFC2833 should work, but alas its a no go. I am also using ulaw which should not be distorting the dtmf through compresion, correct? Also with RFC2833 it should not matter? Everything works great otherwise. sip.conf for colo * is posted below: [general] context=telasip port=5060 bindaddr=0.0.0.0 srvlookup=yes disallow=all; First disallow all codecs allow=ulaw register = username:[EMAIL PROTECTED] [telasip] type=peer username=* fromuser=* authname=* secret=* host=gw3.telasip.com context=default dtmfmode=RFC2833 disallow=all allow=ulaw canreinvite=no nat=no Thanks in advance for any help John Millican ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DTMF detection
I have been battling this problem for 2 months with no resolution as of yet with TelaSIP. I am told that it is a provider problem(Level 3) because all TelaSIP is doing is passing the call directly to them once the call comes through. Anyone else having this issue with TelaSIP or Level3?On 10/10/05, John Millican [EMAIL PROTECTED] wrote: Hello all,yes there is a lot of information about this on the wiki and in past posts on this list but have not found anything that has solved my problem.setup is:phone--PAP2-na--asterisk v1.0.9(in house on local subnet dtmf isinband)---PSTN---Telisipasterisk box at colo v1.0.9 VoIP only.I haveonly access to dial up so can not go VoIP out of the house.In extensions.confon colo * i have some logic that based on callerid lets mehit a single digit to get to DISA, this work every time. the problem is that when i enter a number for DISA to dial i get duplicatedigits, example i enter 6037862111 and disa tries to dial 6003778621.I havetried setting relaxdtmf=yes in sip.conf with no luck.I have read on the wiki that RFC2833 should work, but alas its a no go.I am also using ulawwhich should not be distorting the dtmf through compresion, correct? Alsowith RFC2833 it should not matter? Everything works great otherwise. sip.conffor colo * is posted below:[general]context=telasipport=5060bindaddr=0.0.0.0srvlookup=yesdisallow=all; First disallow all codecsallow=ulawregister = username:[EMAIL PROTECTED][telasip]type=peerusername=*fromuser=* authname=*secret=*host=gw3.telasip.comcontext=defaultdtmfmode=RFC2833disallow=allallow=ulawcanreinvite=nonat=noThanks in advance for any help John Millican___--Bandwidth and Colocation sponsored by Easynews.com --Asterisk-Users mailing list Asterisk-Users@lists.digium.comhttp://lists.digium.com/mailman/listinfo/asterisk-usersTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Tom VileBaldwin Technology Solutions, IncConsulting - Web Design - VoIP Telephony www.baldwintechsolutions.comPhone: 518-631-2855 x205Fax: 518-631-2856 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DTMF detection
We had this problem a few months ago but they resolved it for us. I really don't remember more than that. Darren Wiebe [EMAIL PROTECTED] Tom Vile wrote: I have been battling this problem for 2 months with no resolution as of yet with TelaSIP. I am told that it is a provider problem(Level 3) because all TelaSIP is doing is passing the call directly to them once the call comes through. Anyone else having this issue with TelaSIP or Level3? On 10/10/05, *John Millican* [EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote: Hello all, yes there is a lot of information about this on the wiki and in past posts on this list but have not found anything that has solved my problem. setup is: phone--PAP2-na--asterisk v1.0.9(in house on local subnet dtmf is inband)---PSTN---Telisipasterisk box at colo v1.0.9 VoIP only. I have only access to dial up so can not go VoIP out of the house. In extensions.conf on colo * i have some logic that based on callerid lets me hit a single digit to get to DISA, this work every time. the problem is that when i enter a number for DISA to dial i get duplicate digits, example i enter 6037862111 and disa tries to dial 6003778621. I have tried setting relaxdtmf=yes in sip.conf with no luck. I have read on the wiki that RFC2833 should work, but alas its a no go. I am also using ulaw which should not be distorting the dtmf through compresion, correct? Also with RFC2833 it should not matter? Everything works great otherwise. sip.conf for colo * is posted below: [general] context=telasip port=5060 bindaddr=0.0.0.0 http://0.0.0.0 srvlookup=yes disallow=all; First disallow all codecs allow=ulaw register = username:[EMAIL PROTECTED] mailto:username:[EMAIL PROTECTED] [telasip] type=peer username=* fromuser=* authname=* secret=* host=gw3.telasip.com http://gw3.telasip.com context=default dtmfmode=RFC2833 disallow=all allow=ulaw canreinvite=no nat=no Thanks in advance for any help John Millican ___ --Bandwidth and Colocation sponsored by Easynews.com http://Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com mailto:Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Tom Vile Baldwin Technology Solutions, Inc Consulting - Web Design - VoIP Telephony www.baldwintechsolutions.com http://www.baldwintechsolutions.com Phone: 518-631-2855 x205 Fax: 518-631-2856 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DTMF detection
Anyone else having this issue with TelaSIP or Level3? Yes, to some extend. I have had more luck with incoming calls with IAX from Telasip, but it's still not 100%. On SIP even two digit extensions would end up with double digits (12 as 112, etc). I couldn't find a resolution although Telasip has been quite cooperative and willing to change things (which I certainly appreciate and give them credit for). So I ended up getting a DID from a different provider that comes in via IAX and no problem with double digits at all... it is possible to get this to work reliably after all. ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] DTMF detection problems.
I have an Asterisk box (Fedora Core 2, kernel 2.6.10-1.771_FC2smp) connected to an E1 (MFC/R2 signaling) through a TE110P Digium card. My IVR doesn't work because sometimes a single digit is recognized two, three times and sometimes not. I've already changed relaxdtmf to yes as well as txgain and rxgain in unicall.conf file but, unfortunately, without success. My configuration files, /etc/zaptel.conf and /etc/asterisk/unicall.conf, follow below: /etc/zaptel.conf === # MFC/R2 does not normally use CRC4 span=1,0,0,cas,hdb3 # cas=1-15:1101 cas=17-31:1101 loadzone = us #loadzone = us-old #loadzone=gr #loadzone=it #loadzone=fr #loadzone=de #loadzone=uk #loadzone=fi #loadzone=jp #loadzone=sp #loadzone=no defaultzone=us /etc/asterisk/unicall.conf == callgroup=1 pickupgroup=1 language=br usecallerid=yes hidecallerid=no immediate=yes callwaitingcallerid=yes threewaycalling=yes transfer=yes cancallforward=yes callreturn=yes echocancel=yes echocancelwhenbridged=yes relaxdtmf=yes rxgain=-1.0 txgain=-0.9 protocolclass=mfcr2 protocolvariant=br,4,4 protocolend=co callerid=asreceived context=voki group=1 channel=1-15 channel=17-31 ___ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] DTMF detection with Adit 600
-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 It seems like Asterisk are having problems detecting DTMF digits when using an Adit 600 channel bank via MGCP. I've tried to turn on RFC 2833 on both Adit and Asterisk, but no digits at all are working then. Anyone experienced simular with Adit or other channel banks? I'm also unable to use V.90 modem through my setup (Adit600 via MGCP - - Asterisk - E1). Fax worked once though.. Does the Echo Cancelling make the problems with V.90? - -- Daniel http://www.faqs.org/rfcs/rfc2833.html -BEGIN PGP SIGNATURE- Version: GnuPG v1.2.5 (MingW32) Comment: Using GnuPG with Thunderbird - http://enigmail.mozdev.org iD8DBQFCfIMg/4dZjWjLCy0RAkIDAKCKskwPa5nHURRKBADPccqNYTrSYACbB09i /TpbLaIZHRdd7K0iPLiwi2o= =HtCY -END PGP SIGNATURE- ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] DTMF detection in dial macro
Hi all,
Has anyone got the call screening sample to pickup DTMF correctly ? I
have tried with the latest HEAD release and the dial macro gets executed
all the way up until the Read command where it sits until the timeout is
triggered no matter what DTMF tones you send it. Asterisk responds with
User entered nothing.. I have tried this on a variety of extensions
with the same result although a simple DTMF test directly on the ingress
leg of the call works perfectly...
The config I am using is as follows:
exten = 123,1,Wait(0.2)
exten = 123,2,Playback(screen-record)
exten = 123,3,SetVar(SCREEN_FILE=/tmp/${CALLERIDNUM}-${EPOCH})
exten = 123,4,Record(${SCREEN_FILE}.gsm|6|25)
exten = 123,5,Dial(Zap/g1/01276459906|60|gM(screen^${SCREEN_FILE}))
exten = 123,6,Voicemail([EMAIL PROTECTED])
[macro-screen]
exten = s,1,Wait(0.2)
exten = s,2,Playback(screen-from)
exten = s,3,Playback(${ARG1})
exten = s,4,Read(ACCEPT|screen-accept|1)
exten = s,5,GotoIf($[${ACCEPT} = 1 ] ?7:6)
exten = s,6,SetVar(MACRO_RESULT=CONTINUE)
exten = s,7,System(/bin/rm ${ARG1})
I am guessing this is a bug of sorts relating to the detection code not being
applied to the egress
channel but I dont know enough about how this works yet to debug it...
Any help greatfully received !
Tristan.
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[Asterisk-Users] DTMF detection/generation
I'm hoping Asterisk can help me solve an unusual problem. I need two SIP endpoints (VoiceXML gateways) to transfer DTMF tones to each other. Both of them can detect DTMF according to rfc2833. However, one of them (host2) must generate DTMF inband. Happily, I came up with the following sip.conf to allow host1 to detect DTMF tones generated by host2. [in] type=peer host=host1 dtmfmode=rfc2833 canreinvite=no [out] type=peer host=host2 dtmfmode=inband But this is not enough, because it doesn't allow host2 to detect tones generated by host1. :-( I'm an Asterisk newbie, but thrilled that it got me this far. I'm kinda stuck now, though, and I'm hoping someone on the list can point me in the right direction. Thanks, Jim ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] DTMF Detection Problem
Hi, My set up is like this Asterisk---SipuraATA-AnalogPhone When I'm calling into asterisk from a cell phone, there's no dtmf detection problem as asterisk can detect correct extensions that I press. But when the phone is further connected to the AnalogPhone thru the ATA, the dtmf signal is really short/weak. I've tried to adjust dtmf lengths, gain, etc. on the ATA and it helps a little bit, but not much. And this seems to be a problem only if I call in from a cell phone. If I were to use a SIP phone to call in, it works much better. Is there a way to make Asterisk to regenerate the DTMF tones to improve the DTMF tones? Such as making it interpret the DTMF tones and regenerate it w/ a certain length regardless of original signal length. The reason I want to DTMF comes to AnalogPhone clearly is because I want to ultimately connect it to a FXSFXO converter and go back out to PSTN line. Thank you Ron
[Asterisk-Users] dtmf detection on modem-ISDN
hello! How to turn off DTMF detection on modem (isdn) on active channel (when the channel is open )? Problem is that dtmf is detected when someone on (ISDN) telco side speak then dtmf tones are send to from asterisk to internal line (x100p) . tnx. Tomaz ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] dtmf detection from AS5350 over SIP
Hi, Just wondering if anybody has encountered a similar problem as I have with recieving calls on Asterisk from a CISCO AS5350 (over SIP). I have dtmf relay configured on the AS, however, when someone calls in from the PSTN sometimes their digits are inputted twice, which messes up the extensions. If there is a better way to terminate calls from a AS without using SIP, that would fix this problem, then I'd be interested in that too. Have any ideas? If it would help, I could provide you with some of my config files. Brian. ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DTMF Detection and noise in TDM10B
On Tue, Jun 10, 2003 at 07:08:45PM +0200, Sergio Serrano Revuelto wrote: HI all, we get a TDM10B to probe it. I find two problems: -First, I hear a lot of noise in communication. I have tried do dd if=/dev/zero of=/dev/null but it isn't work. -Second, When I pickup phone connected to TDM10B I hear a strange dial tone and * doesn't detect DTMF. That is the same problem I had last week. The system had worked before then started with that nonsense. I swapped PCI slots for the TDM and X100P cards. That made it so the TDM card had its own IRQ and the X100P was sharing with the AC97 sound device on the motherboard, which is unused. That fixed my symptoms. These cards definately have problems with IRQ sharing, which isn't supposed to happen on a PCI bus. My laptop manages to run flawlessly with every PCI device, except graphics and IDE controller, on the same IRQ. That includes 802.11b PCMCIA card, compact flash reader, sound device, Intel EEPro, Lucent Winmodem, and some other thing. Back in the OS/2 days I was even able to share IRQs with multiple ISA based serial ports using the SIO.SYS driver. Never dropped a bit. -- Scott LambertKC5MLE Unix SysAdmin [EMAIL PROTECTED] ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DTMF detection on SIP provider ?
At 14:39 2003-03-09 -0600, Mark Spencer wrote: try the new dtmfmode parameters on the user or peer. Note they are not currently valid in the [general] section. you can set dtmfmode=inband or dtmfmode=rfc2833 Mark On Sun, 9 Mar 2003, Mikael Andersson wrote: Exactly where shoud I enter that value ? /regards Mike ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DTMF detection on SIP provider ?
Look into sip.conf.sample [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to context = default ; Default for incoming calls ;tos=lowdelay ;tos=184 ;maxexpirey=3600; Max length of incoming registration we allow ;defaultexpirey=120 ; Default length of incoming/outoing registratio ; ;register = [EMAIL PROTECTED] ; Register with a SIP provider ;register = [EMAIL PROTECTED]/1234 ; Register 2345 at sip provider as 1234 ; ;[snomsip] ;type=friend ;secret=blah ;host=dynamic ;dtmfmode=inband here is the answer ; Choices are inband, rfc2833, or info ;defaultip=192.168.0.59 - Original Message - From: Mikael Andersson [EMAIL PROTECTED] To: [EMAIL PROTECTED] Sent: Monday, March 10, 2003 12:29 AM Subject: Re: [Asterisk-Users] DTMF detection on SIP provider ? At 14:39 2003-03-09 -0600, Mark Spencer wrote: try the new dtmfmode parameters on the user or peer. Note they are not currently valid in the [general] section. you can set dtmfmode=inband or dtmfmode=rfc2833 Mark On Sun, 9 Mar 2003, Mikael Andersson wrote: Exactly where shoud I enter that value ? /regards Mike ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] DTMF detection on SIP provider ?
At 00:50 2003-03-10 +0100, Andre Bierwirth wrote: Look into sip.conf.sample [general] port = 5060 ; Port to bind to bindaddr = 0.0.0.0 ; Address to bind to context = default ; Default for incoming calls ;tos=lowdelay ;tos=184 ;maxexpirey=3600; Max length of incoming registration we allow ;defaultexpirey=120 ; Default length of incoming/outoing registratio ; ;register = [EMAIL PROTECTED] ; Register with a SIP provider ;register = [EMAIL PROTECTED]/1234 ; Register 2345 at sip provider as 1234 ; ;[snomsip] ;type=friend ;secret=blah ;host=dynamic ;dtmfmode=inband here is the answer ; Choices are inband, rfc2833, or info ;defaultip=192.168.0.59 Well.. But I need it on : the [general] part where I do the register ? or ? The clients in my case all my ATAs work fine.. But incoming calls doesnt.. /Mike ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
