similar to: DTMF Detection: Where it happens actually?

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

Às 14:46 de 27/07/2019, Ilia Mirkin escreveu: > On Sat, Jul 27, 2019 at 7:37 AM Ralph Corderoy <ralph at inputplus.co.uk> wrote: >> The video plays, CPU load is less (my aim), but there's ‘tearing’ of the >> picture as if small rectangles that are updates are appearing in the >> wrong location, off by a little. If I step through the frames with >> mpv's

Dear Forum Members, I just purchased two VoiSmart GSM cards. Tried to install one of them on my Fedora Core 5 system, The compilation was not smooth, as expected, but after a small fix, it went through. Then I put two SIM cards in the card's slots. Then I loaded the modules. Then I started the Asterisk. After all I configured the vgsm.conf file according to my settings, that is just changed

Dear Asterisk Comunity, I'm thinking about developing a callback application based on the following scenario: 1. Customer Calls the outgoing number which is a PSTN line connected to my Zap channel 2. Asterisk captures the Caller ID and calls back the customer. 3. As soon as the customer picks up the phone, asterisk plays a promt to enter the Destination number. 4. Asterisk Connects the

In short, I have an application which executes another application, a so called client loader. I would simply like to know which parameter the client loader passes to the other executable. It seems to me that this should not be hard to do, but I have not been able to figure out how. Please enlighten me in case you know how this may be achieved. In short, I'd like to execute the application

Hi All I'm trying to set up a machine with a simple samba configuration. I've written the config file with no problems, however any time I attempt to connect to the shares it doesn't work. Firstly, the smb.conf: [global] netbios name = TEMPSRV workgroup = LIPSON map to guest = Bad User [ittechs] comment = Should not see me... path = /var/smb/ittechs read only = no force user =

Just got started and stuck at the same time. Working with the C-Api, i have come to the point where i want to emit a binary file for my module. I queried the X86-Target and called LLVMCreateTargetMachine this way: > LTarget := LLVMGetTargetFromName('x86'); > LMachine := LLVMCreateTargetMachine(LTarget, 'i386-win32-msvc', nil, > nil, LLVMCodeGenLevelDefault,

Prior to "win32" i used "windows" 2017-09-19 19:09 GMT+02:00 Alexander Benikowski <sebal007 at googlemail.com>: > Just got started and stuck at the same time. Working with the C-Api, i > have come to the point where i want to emit a binary file for my module. I > queried the X86-Target and called LLVMCreateTargetMachine this way: > >> LTarget :=

I am still trying to figure out why samba wont let me create in subdirectories I've tripple checked everything and a few things leapt out at me. *) I can modify a file inside a directory that I cannot create a file in. I did not know it was possible under linux to do that. *) If I set all perms on 777 I can create. But neither 775 or 755 will allow it. *) When I go into a mount

Hi all! I've problem with playing ogg files. I'm triing to use tripple buffer method: +-------------+ +-| buffer 1 |-+ | +-------------+ | | | | +-------------+-+ | | buffer 2 | | +-------------+-+ | | | +-------------+-+

Hi All, I'm using 1.8 Asterisk and i havet set DTMF mode=rfc2833 in SIP global default settings. but when user sending DTMf event with SIP info method my asterisk accepting that DTMF. If default or global setting is rfc2833 then how come asterisk accepting SIP info dtmf event? what to check please guide Amit-- -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi. I am having problems with asterisk detection dtmf properly in asterisk 11. I am up to rev 390229. Now, when coming in off a did we have with Velocity, the dids work fine, but from extensions often it misses digits -- I can type *4 and it will miss the 4. Often, if I type quite slowly things will work properly. All dtmf modes are set to rfc2833. Strangely enough, I did not notice this with

*David Mat?as Hern?ndez didi you have any luck?* *I have the same problem.* -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20120815/6478461e/attachment.htm>

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.

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

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:

Hello I have a setup where i have 2 asterisk servers connected over the public internet with plenty of bandwidth, NAT on one side only. If i use IAX between the two *'s dtmf is flawless. If I use SIP, DTMF detection is around 30% or less. I have an exten to dial into and check DTMF: exten => NPANXXxxxx,1,Answer(); (actual number blanked for privacy) exten =>

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

We're still having problems with DTMF detection on our X100P cards. Incoming callers that hold down the "1" button for too long are being connected to extension 11. One would think fat fingers were uncommon, but it happens to alot of people. I suspected this was related to our having to increase the txgain, but I tried turning it down with no effect. I also tried disabling the

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