similar to: Dtmf issues solved

Hi I have a problem with dtmf recognition an analog lines connected to Sangoma A200. The digits (in most cases the first one) are doubled and so my IVR is useless. I tried to adjust the rxgain, toneduration and relxing the dtmf but nothing worked. I also noticed one thing it only happens during the background application: exten => s,1,Background(soundfile) exten => 111,1,Dial(SIP/111)

Hi List, I am using Asterisk 1.6.2.18. One strange problem come into my knowledge after using this version of asterisk. Without pressing any digits or key from my mobile, I am getting DTMF into my asterisk server. For getting DTMF I have use one opensourse application which gets events from asterisk server and store into database. And after that I made my own script to gets these DTMF keys and

What would be the best way to solve this problem? Anyone else that have got the same problem with Android’s native SIP client, especially on Samsung phones? I do not know if the bug is in Android native SIP, or Samsung’s build of the SIP client, or if the bug is even with the OpenVPN client, or where the bug actually is. The ACK might even be sent for real, but have the incorrect source IP so

Hi, I'm looking for some advice on how to solve DTMF issues. I have 2 boxes, one which is the connection to the PSTN (PSTN) through PRIs and SIP trunks, and a second (PBX) which has UAs registered to it. We have a customer that has an existing pbx that we trunk analog lines to using a GXW-4008. The GXW is set to dtmfmode inband. This seems to provide the best outbound DTMF. The issue I'm

http://bugs.digium.com/bug_view_page.php?bug_id=0002928 This should correct dtmf with asterisk and things like MAX TNT's and MANY MANY more devices out there. bkw

Hi, I have a system with the following configuration: - Asterisk 1.6.0.3-rc1 - DAHDI Linux 2.1.0.4 - DAHDI Tools 2.1.0.2 - wanpipe-3.3.15 - Sangoma A108E card with ISDN PRI interfaces Calls arrive over the PRI interface and users need to supply DTMF digits before asterisk decides what to do with the call. The DTMF digits are read with the Read application. With low call volumes everything works

Hi, I just deployed new Aastra 9480i phones and when I attempt enter digits on other systems, like host pin in a GoToMeeting, the servers on the other end do not get my entries. I am assuming this is a DTMF issue but do not see anything in this phones config other than turning on the display of the digits. I have the DTMF method set to "SIP INFO". I am using AsteriskNow w/FreePBX.

Hello Does someones know a good low-level way to debug the DTMF that is arriving (or not arriving) at Asterisk? We've already used the DTMF logger level. The picture is the following: We are developing an application that calls to a customer to get authorization information through DTMF. We've used at development environment SIP/IAX channels and everything was great and the work was

Today, sessions 320-321 of the VoIP Users Conference will take place at the usual time, 12 Noon Eastern [ http://vuc.me/next for local times ] We'll be talking to Sangoma's Frederic Dickey about NetBorder 4.0. You can download or watch his accompanying slide presentation here: http://vuc.li/Sangoma-2011 Next, Pascal Dor?, mVoIP chief over at Media5 Corporation will talk about the

Hi everyone, My problem was that any time a user dialled into my IVR application (AGI on asterisk), it would close the channel if they were entered too quickly. I looked through the code for any possible hints and inserted debugging code - to no avail. It just struck me there that, since no one else was having this problem, that it could be a configuration issue. Lo and behold, I had set

Hello, When using Asterisk version 0.7.2, FreeBSD port with Mediatrix 1124 gateway, there is problem with DTMF "out-of-band". See debug below: Mediatrix forces (*) to use Payload Type as 96: [...] a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:4 G723/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:96 telephone-event/8000 a=fmtp:96 0-15 [...] Then we've got this nice debug from (*): May

For some reason 1.4.4.x was not reading chan_dahdi.conf. When I symlinked it to zapata.conf it worked. -----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Eric Wieling Sent: Wednesday, May 30, 2012 2:35 PM To: asterisk-users at lists.digium.com Subject: [asterisk-users] Sangoma Card Issue Has anyone

Hi all, i have been using asterisk for a few years but i am about to do my first t1 setup. After terrible quality issues between two business locations, we have decided to purchase a point to point t1 from the local phone co. The internet is too crappy, too much lag, queing and jitter. Most calls were dropped. I was about to order two cisco routers with csu cards and remembered our wonderful

On Sat, Mar 14, 2020 at 2:02 PM John Roman <john at dev1ce.com> wrote: > greetings asterisk users :) > ive just deployed version 17 and migrated as best I can to pjsip. I can > receive calls, and get to my mailbox prompt, however placing calls seems > impossible with the following error on dial: > > Connected to Asterisk GIT-master-0cde95ec89 currently running on dunkel

On Jul 2, 2015, at 5:33 PM, Chris Murphy <lists at colorremedies.com> wrote: > > On Thu, Jul 2, 2015 at 4:12 PM, Warren Young <wyml at etr-usa.com> wrote: >> On Jul 2, 2015, at 11:32 AM, Chris Murphy <lists at colorremedies.com> wrote: >>> >>> It's effectively impossible. Very few people know how to do this. >> >> Relatively few

I have tried beta2, beta3 and now back to 1.2.12.1 and I have correct DTMF tones 25% of the time. I have to call several times to enter an extension. I have a router and a packet shaper and some other stuff. Anyone have any other ideas why this might happen. I do not have any Zap channels but I am running CentOS4. I also do not have any cards installed. Thanks

I'm fascinated by the Pure algebraic/functional language[1]. The Pure interpreter uses the LLVM JIT compiler as its backend. I would like to compile[2] Pure so that it runs on Android / ARM. Pure has a dependency on the LLVM JIT. So I need to compile LLVM source for Pure to run. Is it possible to compile LLVM source[3] for Android (ARM) devices? There really seems to be no information about

Hi Guys, Looking for a powerful box that is compact, can take two hard drives for Raid-1 (no SSD, too expensive), have at least two Gig ports or two 10/100mbps ports. Fit two PCIe or one PCIe card plus it's daughter card which needs as much room as a PCIe and doesn't need the actual slot. That is for Sangoma A200 + Daughter card. I would be really glad if I can get some sort of a KVM

On Fri, Jan 30, 2009 at 9:23 AM, Lanny Marcus <lmmailinglists at gmail.com> wrote: > If I use "system-config-display" or "system-config-display --reconfig" > although I get a message that I need to log out of GNOME, to have the > new file written, for new Display Resolution, logging out or > restarting the box does not result in the new settings for the

My Sangoma Experience in Asterisk: Followup 2005-05-06 original review can be found at: http://astguiclient.sourceforge.net/Sangoma_experience.txt It's now been a month since I finished my last round of tests with the Sangoma A104u board in Asterisk. I have had a lot of conversations with the guys at Sangoma about how my system runs so that they could set up their test systems to try to