Re: [asterisk-users] DTMF rfc2833 missed when transfering to another server

well looks likes we solved it the rtpkeepalive was set to 5 seconds on the trunk and every time asterisk sends a rtpkeepalive a cn packet is sent the same time a cn packet is sent asterisk loses the dtmf it was sent On Wed, Dec 16, 2020 at 7:43 PM Israel Gottlieb wrote: > Hi all > i have a

[asterisk-users] DTMF rfc2833 missed when transfering to another server

Hi all i have a asterisk server 16.11.1 (server A) that gets a call (leg A) and then calls a second server (leg B) server B is a freeswitch server the servers are configured all thru with rfc2833 for dtmf the caller enters a number a long 15 digit number like a credit card number or even a phone

Re: [asterisk-users] DTMF not working on incoming calls

Have you done a wireshark capture and then seen if the DTMF is coming in from your provider? What does the SDP show? On Thu, Dec 5, 2019 at 12:17 AM Carlos Chavez wrote: > What is the best way to debug DTMF on a PJSIP trunk? I have cycled > through all available options

[asterisk-users] DTMF not working on incoming calls

    What is  the best way to debug DTMF on a PJSIP trunk?  I have cycled through all available options ('rfc4733','inband','info','auto','auto_info') but my IVR does not recognize any options from the remote end. I have also tried changing codecs from g729 to alaw or ulaw with the same result. 

Re: [asterisk-users] DTMF tones in MixMonitor recording

I agree! I have my SBC and asterisk servers all configured with rfc2833, so it should be ok! No need for auto mode! Thanks again! Cheers Patrick On Tue, 1 May 2018, 20:07 Joshua Colp, wrote: > On Tue, May 1, 2018, at 6:52 AM, Patrick Wakano wrote: > > Thanks very much for the

Re: [asterisk-users] DTMF tones in MixMonitor recording

On Tue, May 1, 2018, at 6:52 AM, Patrick Wakano wrote: > Thanks very much for the reply Joshua! > So I guess that setting dtmfmode=auto would be the safest choice in order > to strip out the DTMFs from the recording, right? > Cheers! It should work. Personally I prefer explicit configuring

Re: [asterisk-users] DTMF tones in MixMonitor recording

Thanks very much for the reply Joshua! So I guess that setting dtmfmode=auto would be the safest choice in order to strip out the DTMFs from the recording, right? Cheers! Patrick Wakano On Tue, 1 May 2018, 19:36 Joshua Colp, wrote: > On Mon, Apr 30, 2018, at 11:23 PM, Patrick

Re: [asterisk-users] DTMF tones in MixMonitor recording

On Mon, Apr 30, 2018, at 11:23 PM, Patrick Wakano wrote: > Hello list, > Hope you are all doing fine! > > I have stumbled over some piece of dialplan code in which apparently they > were trying to avoid recording the DTMF tones in the wav file. It is really > messy and I am not sure if this

[asterisk-users] DTMF tones in MixMonitor recording

Hello list, Hope you are all doing fine! I have stumbled over some piece of dialplan code in which apparently they were trying to avoid recording the DTMF tones in the wav file. It is really messy and I am not sure if this really works. So after a bit of research I found this comment (

Re: [asterisk-users] DTMF emulation with SIP INFO and direct media

Asterisk is in version 14.7.1. One end is a SIP Trunk to another Asterisk, the other end a home-made SIP phone. SIP INFO requests are coming from the other Asterisk. Both endpoints use chan_sip with "dtmfmode" set to "info". This is not strictly speaking a one-to-one setup since we're

Re: [asterisk-users] DTMF emulation with SIP INFO and direct media

Hello Jean, 1. Can you describe a bit further how both ends of the above call were both made of and configured ? DTMF receiving is Asterisk/SIP channel but which version ? Is the other end a SIP phone or a SIP trunk ? 2. Do you observe such behaviour in a one-to-one setup (one end emits, the

[asterisk-users] DTMF emulation with SIP INFO and direct media

Hello, I think there is an issue when DTMF are handled with SIP INFO and direct media is enabled. When I receive a SIP INFO, the logs tell me that a "DTMF begin" is generated, but no related "DTMF end" is generated, unless the call is ended. Here is an excerpt of the logs : *--- SIP INFO

[asterisk-users] DTMF not detecting from Mobile phones

Hi, I have a Asterisk running in Amazon AWS with an IVR configured. The problem I'm having is I'm not getting DTMF from mobile phones. The Landlines works without any issues. I have configure DTMF to rfc 2833. I checked with dtmf debug but I'm not receiving dtmf from mobile devices. please let me

Re: [asterisk-users] DTMF issues between Asterisk and Callmanager with Zoiper

Carlos Chavez wrote: I had an old Asterisk installation die recently and we decided to upgrade to Asterisk 13 to replace the old server. Everything seems to be working with PJSIP but there is one issue. Asterisk talks to a callmanager via a SIP trunk and send calls to PSTN (another country).

[asterisk-users] DTMF issues between Asterisk and Callmanager with Zoiper

I had an old Asterisk installation die recently and we decided to upgrade to Asterisk 13 to replace the old server. Everything seems to be working with PJSIP but there is one issue. Asterisk talks to a callmanager via a SIP trunk and send calls to PSTN (another country). Most of the

[asterisk-users] DTMF talkoff beep (still)

Hi all, I am still receiving reports from some users that calls they make or receive contain loud deafening beeps that can last a couple of seconds. I understand this is DTMF talkoff and is being triggered because the phone interprets speech as a key press (say if someone is pressing 1 at an

Re: [asterisk-users] DTMF talkoff beep (still)

On 9/10/2015, at 5:16 AM, Jamie Rees wrote: > > > I understand this is DTMF talkoff > > > My question is how do people running SIP phone systems mitigate against this? My personal answer to this question has been to completely avoid the use of any ATAs at all. Since

Re: [asterisk-users] DTMF issue

- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jamie Rees Sent: 08 July 2015 10:27 To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] DTMF issue Indeed, thanks. I'll let you know how it goes

Re: [asterisk-users] DTMF issue

' Subject: Re: [asterisk-users] DTMF issue You probably have to reload asrerisk after making the change. Thomas M. Peters | Systems Administrator | tpet...@mcts.org Desk: 414.343.1720 | Helpdesk: x3400 or helpd...@mcts.org Jamie Rees jr...@gmlnt.com 7/7/2015 3:53 PM Ah I see, in theory it's

Re: [asterisk-users] DTMF issue

, Jamie -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tom Peters Sent: 07 July 2015 19:14 To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] DTMF issue It's called DTMF Talk-off. We have it too

Re: [asterisk-users] DTMF issue

It's called DTMF Talk-off. We have it too. Seems worse when talking to mobile phones but it happens at random on many external calls. If this happens to you, especially on voice peaks (when the outside party said a particularly loud syllable) then you probably have DTMF talk-off. I think it's

Re: [asterisk-users] DTMF issue

[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tom Peters Sent: 07 July 2015 19:14 To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] DTMF issue It's called DTMF Talk-off. We have it too. Seems worse when talking to mobile phones but it happens at random on many external calls

Re: [asterisk-users] DTMF issue

Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] DTMF issue In my humble opinion, adjusting this setting will (for you) do nothing, since you don't use the dahdi channels for transport. See this discussion, which I found after I posted my first response: http

Re: [asterisk-users] DTMF issue

: [asterisk-users] DTMF issue In my humble opinion, adjusting this setting will (for you) do nothing, since you don't use the dahdi channels for transport. See this discussion, which I found after I posted my first response: http://www.voip-info.org/wiki/view/Asterisk+DTMF Particularly this sentence

Re: [asterisk-users] DTMF issue

From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jamie Rees Sent: Monday, July 06, 2015 5:54 PM To: asterisk-users@lists.digium.com Subject: [asterisk-users] DTMF issue Hello folks, We have an issue with several Cisco SPA512G phones

[asterisk-users] DTMF issue

Hello folks, We have an issue with several Cisco SPA512G phones connected to an Asterisk platform where several users hear loud, random beeps during calls to external recipients. The noises are akin to button press tones, are very loud and a significant annoyance. I've tried changing the DTMF

Re: [asterisk-users] DTMF issue

On 7/6/15 5:53 PM, Jamie Rees wrote: Hello folks, We have an issue with several Cisco SPA512G phones connected to an Asterisk platform where several users hear loud, random beeps during calls to external recipients. The noises are akin to button press tones, are very loud and a significant

Re: [asterisk-users] DTMF behavior in asterisk 12 with PJSIP

Thank you Mathew, We tested the feature flag workaround and it worked. We opened a ticket - Asterisk-24459. If you need any information please get back to us and we will do our best. Thanks again, Yaron. On Mon, Oct 27, 2014 at 3:48 PM, Matthew Jordan mjor...@digium.com wrote: On Mon, Oct

Re: [asterisk-users] DTMF behavior in asterisk 12 with PJSIP

Hello Mathew, Thank you for the reply. I will open an issue and send debug information. Can you explain more about the workaround? A reference to the documentation would be fine. Thanks again, Yaron. On Sun, Oct 26, 2014 at 10:46 PM, Matthew Jordan mjor...@digium.com wrote: On Sun, Oct

Re: [asterisk-users] DTMF behavior in asterisk 12 with PJSIP

On Mon, Oct 27, 2014 at 1:20 AM, Yaron Nachum nachum.ya...@gmail.com wrote: Hello Mathew, Thank you for the reply. I will open an issue and send debug information. Can you explain more about the workaround? A reference to the documentation would be fine. Sure - really, what you are

[asterisk-users] DTMF behavior in asterisk 12 with PJSIP

Hello all, We have recently upgraded some of our services to Asterisk 12 with PJSIP. We have 2 issues related to DTMF: 1. in the regular SIP channel we had DTMF auto mode, which adapted the DTMF settings according to the incoming INVITE - RFC2833 or inband. The is no such settings in PJSIP. Do you

Re: [asterisk-users] DTMF behavior in asterisk 12 with PJSIP

On Sun, Oct 26, 2014 at 3:22 AM, Yaron Nachum nachum.ya...@gmail.com wrote: Hello all, We have recently upgraded some of our services to Asterisk 12 with PJSIP. We have 2 issues related to DTMF: 1. in the regular SIP channel we had DTMF auto mode, which adapted the DTMF settings according to

[asterisk-users] DTMF transmitting letter A

Dear list, maybe not really an Asterisk question, but... all my users dial in via PSTN (via SIP DIDs) and enter a target number via DTMF through my Asterisk 1.4. Out of about 150,000 calls per month I see on average about 1 call per month where an arbitrary caller enters the letter 'A' via

Re: [asterisk-users] DTMF transmitting letter A

[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Markus Sent: Tuesday, June 17, 2014 11:30 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] DTMF transmitting letter A Dear list, maybe not really an Asterisk question, but... all my users dial in via

[asterisk-users] DTMF relay in meetme is not reliable

Hello List, I am facing some issue while passing DTMF (RFC2833 set globally in sip.conf) in meetme (asterisk 1.8). The issue I have observed that - if two users tries to pass DTMF simultaneously at the same time from their phones only one DTMF is detected in asterisk and broadcasted to other

[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-users] DTMF over IAX trunk ignoring last digit

Hello. Scenario: 9 servers connectec to each other over IAX trunks. Users used to call to remote extensions and remote conferences (meetme) via IAX. Problem: all extensions from one server (just one) when try to attend remote conferences had problems with PIN validation. If they use their local

[asterisk-users] DTMF

Anyone see this before? I have a main Asterisk box 11.4 connected to Windstream via SIP trunks in my colo. So as a did comes in they are routed to appropriate customers, in this case another asterisk 11.4 box. All is working well with the exception of DTMF. Losing the last digits so say

[asterisk-users] DTMF recognized after call establishment

Hi, I am receiving DTMF without any reason after call establishment. The log as follows, and I suspect something related to directmedia, [May 17 00:33:35] VERBOSE[4238] app_dial.c: -- SIP/MyTrunk-000a4b49 is making progress passing it to SIP/MAN-000a4b48 [May 17 00:33:35] VERBOSE[4238]

Re: [asterisk-users] DTMF recognized after call establishment

i had this in past there was an ATA configured to send 9 at the end of dialing in my case. On Tue, May 28, 2013 at 8:21 AM, Gopalakrishnan N gopalakrishnan...@gmail.com wrote: Hi, I am receiving DTMF without any reason after call establishment. The log as follows, and I suspect something

Re: [asterisk-users] DTMF recognized after call establishment

So any resolution for this? I suspect it could be related to RE INVITE On Tue, May 28, 2013 at 2:09 PM, Asghar Mohammad asghar...@gmail.comwrote: i had this in past there was an ATA configured to send 9 at the end of dialing in my case. On Tue, May 28, 2013 at 8:21 AM, Gopalakrishnan N

Re: [asterisk-users] DTMF recognized after call establishment

work around was block dtmf. set wrong type of dtmf in incoming trunk. On Tue, May 28, 2013 at 11:15 AM, Gopalakrishnan N gopalakrishnan...@gmail.com wrote: So any resolution for this? I suspect it could be related to RE INVITE On Tue, May 28, 2013 at 2:09 PM, Asghar Mohammad

Re: [asterisk-users] DTMF recognized after call establishment

Let me try with dtmfmode as auto... On 28 May 2013 19:32, Asghar Mohammad asghar...@gmail.com wrote: work around was block dtmf. set wrong type of dtmf in incoming trunk. On Tue, May 28, 2013 at 11:15 AM, Gopalakrishnan N gopalakrishnan...@gmail.com wrote: So any resolution for this? I

Re: [asterisk-users] DTMF Blips at end of Record() - 1.8.18

On Wed, Feb 20, 2013 at 10:49 AM, James Lamanna jlama...@gmail.com wrote: Hi, I've noticed on asterisk 1.8.18 I'm hearing the blip of '#' DTMF to end the recording on the recording itself. Is there an easy way to truncate the last 200ms of the recording or so to eliminate this? The DTMF is

[asterisk-users] DTMF Blips at end of Record() - 1.8.18

Hi, I've noticed on asterisk 1.8.18 I'm hearing the blip of '#' DTMF to end the recording on the recording itself. Is there an easy way to truncate the last 200ms of the recording or so to eliminate this? The DTMF is coming in through rfc2833 and not inband. Thanks. -- James --

[asterisk-users] DTMF are not shown when dialed by PSTN phone

  Dear all, having verbose level 5, and enabling dtmf logging in /etc/asterisk/logger.conf console = notice,warning,error,debug,dtmf I receive dtmf detected, in a SIP-PSTN call, as follows [code][Nov 28 16:12:19] DTMF[2532]: channel.c:2351 __ast_read: DTMF begin '1' received on

Re: [asterisk-users] DTMF are not shown when dialed by PSTN phone

mohammad aliasgari wrote: Dear all, Hola, having verbose level 5, and enabling dtmf logging in /etc/asterisk/logger.conf console = notice,warning,error,debug,dtmf I receive dtmf detected, in a SIP-PSTN call, as follows snipped Why don't I receive DTMF that are dialed by a PSTN phone?

Re: [asterisk-users] DTMF are not shown when dialed by PSTN phone

digits, they are displayed and logged.what's missing?  Regards --- On Wed, 11/28/12, Joshua Colp jc...@digium.com wrote: From: Joshua Colp jc...@digium.com Subject: Re: [asterisk-users] DTMF are not shown when dialed by PSTN phone To: Asterisk Users Mailing List - Non-Commercial Discussion

[asterisk-users] DTMF Payload Settings

Hello, The service provider wants me to setup dtmfmode to rfc2833 and dtmf payload to 101. I can configure SIP trunk as dtmfmode=rfc2833 but how to configure payload? -- Necati DEMİR -- _ -- Bandwidth and

Re: [asterisk-users] DTMF Payload Settings

Necati Demir wrote: Hello, Hola, The service provider wants me to setup dtmfmode to rfc2833 and dtmf payload to 101. I can configure SIP trunk as dtmfmode=rfc2833 but how to configure payload? Asterisk already uses payload 101 for RFC2833 so you should be fine with dtmfmode=rfc2833

Re: [asterisk-users] DTMF inband with telephone-event in SDP

Jakob Hirsch wrote: Hello everyone! Hola, We use Asterisk for various services like voicemail. Our SIP clients usually use rtp events (rfc2833) for DTMF, which works just fine and independent from the codec (g711 vs. g726 etc.). Now we noticed there are some SIP clients that announce

[asterisk-users] DTMF inband with telephone-event in SDP

Hello everyone! We use Asterisk for various services like voicemail. Our SIP clients usually use rtp events (rfc2833) for DTMF, which works just fine and independent from the codec (g711 vs. g726 etc.). Now we noticed there are some SIP clients that announce telephone-event in their SDP, but

Re: [asterisk-users] DTMF digits are coming through twice

Any ideas? On Thu, Oct 11, 2012 at 2:32 PM, Vik Killa vipki...@gmail.com wrote: Call was to 7167436110 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory

Re: [asterisk-users] DTMF digits are coming through twice

The trace is attached 3 emails back. -- _ -- 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

Re: [asterisk-users] DTMF digits are coming through twice

Why am I feeling like I'm the only one here who is not able to see any pastebin link or attachments in this thread ! On Fri, Oct 12, 2012 at 6:18 PM, Vik Killa vipki...@gmail.com wrote: The trace is attached 3 emails back. --

Re: [asterisk-users] DTMF digits are coming through twice

Sorry the attachment was too big. here is link: http://www.2shared.com/file/Ola640Pn/doubledigit.html On Fri, Oct 12, 2012 at 9:24 AM, SamyGo govoi...@gmail.com wrote: Why am I feeling like I'm the only one here who is not able to see any pastebin link or attachments in this thread ! --

Re: [asterisk-users] DTMF digits are coming through twice

Only callers calling from Earthlink internet connection On Wed, Oct 10, 2012 at 5:18 PM, Don Kelly d...@donkelly.biz wrote: Is this happening for all callers, or just iPhone callers? --Don -- _ -- Bandwidth and Colocation

Re: [asterisk-users] DTMF digits are coming through twice

Can you share your pcap trace ! On Thu, Oct 11, 2012 at 5:16 PM, Vik Killa vipki...@gmail.com wrote: Only callers calling from Earthlink internet connection On Wed, Oct 10, 2012 at 5:18 PM, Don Kelly d...@donkelly.biz wrote: Is this happening for all callers, or just iPhone callers?

[asterisk-users] DTMF digits are coming through twice

I've been running an Asterisk server (1.6.2.17.2) for over a year without any major issues. All of a sudden people are unable to login to their voicemail because Asterisk is seeing DTMF twice for each digit the caller pushes. We've noticed the problem only consistently happens to callers from

Re: [asterisk-users] DTMF digits are coming through twice

Hi, Not exactly a solution, but I'm sure you must've taken pcap traces of a few such sample calls. See in their RTPs that you are receiving repeatedly same RTPs which will tell you that any DTMF packet is coming in twice by the source or not ! just one such simple pcap will help you identify at

Re: [asterisk-users] DTMF digits are coming through twice

I'm not sure I follow, the packet capture on the asterisk server shows double digits being entered. Does that mean it's the source? On Wed, Oct 10, 2012 at 11:55 AM, SamyGo govoi...@gmail.com wrote: Hi, Not exactly a solution, but I'm sure you must've taken pcap traces of a few such sample

Re: [asterisk-users] DTMF digits are coming through twice

After comparing packet captures of good and bad calls. It looks like the double digit is coming from rfc2833 and dtmf inband. It looks like the inband tone is splitting the rfc2833 in two? Is there some way to resolve this??? On Wed, Oct 10, 2012 at 12:28 PM, Vik Killa vipki...@gmail.com wrote:

Re: [asterisk-users] DTMF digits are coming through twice

-Commercial Discussion Subject: Re: [asterisk-users] DTMF digits are coming through twice I'm not sure I follow, the packet capture on the asterisk server shows double digits being entered. Does that mean it's the source? On Wed, Oct 10, 2012 at 11:55 AM, SamyGo govoi...@gmail.com wrote: Hi

Re: [asterisk-users] DTMF digits falsely detected

On 9/15/2012 6:28 PM, Matthew Jordan wrote: - Original Message - From: Vladimir Mikhelson v...@mikhelson.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, September 15, 2012 1:11:14 PM Subject: Re: [asterisk-users] DTMF

Re: [asterisk-users] DTMF digits falsely detected

PM Subject: Re: [asterisk-users] DTMF digits falsely detected On 9/14/2012 10:11 PM, Matthew Jordan wrote: - Original Message - From: Vladimir Mikhelson v...@mikhelson.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent

Re: [asterisk-users] DTMF digits falsely detected

-Commercial Discussion Subject: Re: [asterisk-users] DTMF digits falsely detected On 9/14/2012 11:04 PM, Matthew Jordan wrote: - Original Message - From: Vladimir Mikhelson v...@mikhelson.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent

Re: [asterisk-users] DTMF digits falsely detected

Hopefully the initial poster still has the configuration to produce the files for you. Are you saying the DTMF logs I attached do not provide enough evidence to support the theory of the DTMF length being the cause of this issue? -Vladimir Vladimir, What was the

Re: [asterisk-users] DTMF digits falsely detected

- Original Message - From: Vladimir Mikhelson v...@mikhelson.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, September 15, 2012 11:41:23 AM Subject: Re: [asterisk-users] DTMF digits falsely detected Please take

Re: [asterisk-users] DTMF digits falsely detected

@lists.digium.com Sent: Saturday, September 15, 2012 11:41:23 AM Subject: Re: [asterisk-users] DTMF digits falsely detected Please take a look at the case https://issues.asterisk.org/jira/browse/ASTERISK-20424?actionOrder=asc I uploaded the PCAP captured on the Soft Phone end and the RTP debug log. I

Re: [asterisk-users] DTMF digits falsely detected

[2012-09-15 22:36:44.489226] DTMF[1706] channel.c: DTMF end '4' received on SIP/alec-0009, duration 1660 Alec, Interestingly in your log DTMF durations are even greater than in my original sampling. Well, maybe my duration theory is not

Re: [asterisk-users] DTMF digits falsely detected

On 9/15/2012 5:16 PM, Alec Davis wrote: [2012-09-15 22:36:44.489226] DTMF[1706] channel.c: DTMF end '4' received on SIP/alec-0009, duration 1660 Alec, Interestingly in your log DTMF durations are even greater than in my original sampling. Well,

Re: [asterisk-users] DTMF digits falsely detected

And just to make sure. In both scenarios, normal digit press and prolonged digit press, you did not reproduce the problem we are discussing with X-Lite. Is that correct? Correct, everything with X-Lite 3.0 and asterisk 1.8.16.0 worked correctly with short, normal and long key presses

Re: [asterisk-users] DTMF digits falsely detected

- Original Message - From: Vladimir Mikhelson v...@mikhelson.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Saturday, September 15, 2012 1:11:14 PM Subject: Re: [asterisk-users] DTMF digits falsely detected snip Can you please

[asterisk-users] DTMF digits falsely detected

Hi, I have a context that basically does: Wait(1) Background(message) WaitExten(10) _6XX,1,DoSomething The problem is that when I reach this context and press some digits (eg. 6566604) then I can see in the log that Asterisk reads 6655666. So it's actually reading the digits twice. How

Re: [asterisk-users] DTMF digits falsely detected

-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Vieri Sent: Saturday, 15 September 2012 8:45 a.m. To: asterisk-users@lists.digium.com Subject: [asterisk-users] DTMF digits falsely detected Hi, I

Re: [asterisk-users] DTMF digits falsely detected

On 9/14/2012 6:04 PM, Alec Davis wrote: -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Vieri Sent: Saturday, 15 September 2012 8:45 a.m. To: asterisk-users@lists.digium.com Subject: [asterisk-users

Re: [asterisk-users] DTMF digits falsely detected

On 9/14/2012 6:04 PM, Alec Davis wrote: -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Vieri Sent: Saturday, 15 September 2012 8:45 a.m. To: asterisk-users@lists.digium.com Subject: [asterisk-users

Re: [asterisk-users] DTMF digits falsely detected

- Original Message - From: Vladimir Mikhelson v...@mikhelson.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, September 14, 2012 9:24:41 PM Subject: Re: [asterisk-users] DTMF digits falsely detected On 9/14/2012 6:04

Re: [asterisk-users] DTMF digits falsely detected

On 9/14/2012 10:11 PM, Matthew Jordan wrote: - Original Message - From: Vladimir Mikhelson v...@mikhelson.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, September 14, 2012 9:24:41 PM Subject: Re: [asterisk-users] DTMF

Re: [asterisk-users] DTMF digits falsely detected

- Original Message - From: Vladimir Mikhelson v...@mikhelson.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, September 14, 2012 10:39:30 PM Subject: Re: [asterisk-users] DTMF digits falsely detected On 9/14/2012 10

Re: [asterisk-users] DTMF digits falsely detected

On 9/14/2012 11:04 PM, Matthew Jordan wrote: - Original Message - From: Vladimir Mikhelson v...@mikhelson.com To: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com Sent: Friday, September 14, 2012 10:39:30 PM Subject: Re: [asterisk-users] DTMF

Re: [asterisk-users] DTMF Issue.

Up? 2012/8/20 Luis H. Forchesatto luisforchesa...@gmail.com Thanks for your answer. The logs where posted at pastebin, here the links: - Working Phone: http://pastebin.com/q3pHcwna - Not working phone: http://pastebin.com/iiCHPMmn 2012/8/20 Rusty Newton rnew...@digium.com On 8/20/2012

[asterisk-users] DTMF Issue.

Hi I've got a little issue with DTMF/IVR on my asterisk. I got 3 types of ATA on the network who autenticate the phones: Linksys PAP2, Overtek OT-ATA200SP+ and Opticom VoIP 690. They autenticate at the VoIP server at the same network all with g729 codecs and rfc2833 for the DTMF. Making calls via

Re: [asterisk-users] DTMF Issue.

On 8/20/2012 7:19 AM, Luis H. Forchesatto wrote: Hi I've got a little issue with DTMF/IVR on my asterisk. I got 3 types of ATA on the network who autenticate the phones: Linksys PAP2, Overtek OT-ATA200SP+ and Opticom VoIP 690. They autenticate at the VoIP server at the same network all with

Re: [asterisk-users] DTMF Issue.

Thanks for your answer. The logs where posted at pastebin, here the links: - Working Phone: http://pastebin.com/q3pHcwna - Not working phone: http://pastebin.com/iiCHPMmn 2012/8/20 Rusty Newton rnew...@digium.com On 8/20/2012 7:19 AM, Luis H. Forchesatto wrote: Hi I've got a little issue

[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:

[asterisk-users] DTMF transmission problem

I am having difficulties with customer-bound DTMF being very short clipped off (and basically unusable, as systems on the customer side aren't recognizing the DTMF digits, and I can barely tell that DTMF is there when I listen on a handset). My system set up as follows: PSTN -- Metaswitch

Re: [asterisk-users] DTMF transmission problem

On Thu, Aug 02, 2012 at 12:45:28PM +, Noah Engelberth wrote: I am having difficulties with customer-bound DTMF being very short clipped off (and basically unusable, as systems on the customer side aren't recognizing the DTMF digits, and I can barely tell that DTMF is there when I listen

Re: [asterisk-users] DTMF transmission problem

-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Shaun Ruffell Sent: Thursday, August 02, 2012 11:06 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] DTMF

Re: [asterisk-users] DTMF transmission problem

-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Noah Engelberth Sent: Thursday, August 02, 2012 12:27 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] DTMF

Re: [asterisk-users] DTMF transmission problem

-Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users- boun...@lists.digium.com] On Behalf Of Noah Engelberth Sent: Thursday, August 02, 2012 1:10 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] DTMF

Re: [asterisk-users] DTMF forwarding and Page [SOLVED] [PATCH 1/1]

Nevermind, I checked the code, and A* is not using the F option in MeetMe for Page(), so it's not working by default. Attached is a patch which fixes the problem for me, if anyone needs it. Matteo Il 11/02/2012 13:53, Matteo Fortini ha scritto: Noone knows that? Where/whom could I ask?

Re: [asterisk-users] DTMF forwarding and Page

Noone knows that? Where/whom could I ask? Thanks Il 10/02/2012 12:30, Matteo Fortini ha scritto: Hi, I'd like to implement some way of controlling remote SIP clients while in a call, to execute remote commands. The call topology (think of a PA system) is this: * the caller is in a MeetMe()

[asterisk-users] DTMF forwarding and Page

Hi, I'd like to implement some way of controlling remote SIP clients while in a call, to execute remote commands. The call topology (think of a PA system) is this: * the caller is in a MeetMe() conference room * the callees are Page()d, then the dynamic conference room is connected to the

Re: [asterisk-users] DTMF Testing software to test IVR system

Easy, use Read() to capture the incoming DTMF from Server-B Server-A Server-B Initiate-Call - AnswerCall() SendDTMF(5)-- Read() Read()-SendDTMF(4) SendDTMF(3)-- Read()

Re: [asterisk-users] DTMF Testing software to test IVR system

In server B if I use SendDTMF then it means I am changing programming at server B. Actually I don't have right or permission to change programming in server B. otherwise your suggestion is best for channel base communication. On Thu, Dec 29, 2011 at 2:33 PM, Sammy Govind govoi...@gmail.com

Re: [asterisk-users] DTMF Testing software to test IVR system

o in that case you need to observer the call flow in Server-B, i.e what is the length of sound file playing. what DTMF it requires etc etc and once you detect the call flow for a successful IVR traversal then mimic the behaviour of the call from Server-A. Thats all you can do. Think of it exactly

[asterisk-users] DTMF Testing software to test IVR system

Hi list, Is there any way in asterisk by which I make a call from server and then dialplan(IVR system) gets DTMF from it. I mean to say that automatically DTMF is sended by channels as per user defined, I read there is an application sendDTMF but I don't know how we can used it? like A script

Re: [asterisk-users] DTMF Testing software to test IVR system

Create a callfile with local channel and once first call leg is answered, use wait() and senddtmf() application on second call leg. CALLFILE sample: Channel: LOCAL/1234\@test_ivr Context: senddtmf Extension: s Priority: 1 Extensions.conf sample: ;-- FIRST LEG CALL --; [test_ivr] exten =

Re: [asterisk-users] DTMF Testing software to test IVR system

Hi Satish, Thank you Satish. I did the same before your e-mail i saw. But i got another issue in such case. DTMF is passed to that channels but in case I will make the complete IVR system for calling server end. and which become so complected to do it. Is there any alternate way by which I get

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