similar to: Why does it take several seconds to interpret DTMF-input ?

Hello, I execute the following php script when a call ends and the h-context is executed : /exten => h,n,System(/usr/bin/php /var/log/asterisk/loggingAST/loggingAST.php /var/log/asterisk/loggingAST/${CHANNEL:4}.csv)/ The script loggingAST.php writes some information in a MySQL database on a remote webserver. I have noticed that when the webserver is unreachable, this channel

Hello, when picking up an incoming call from one ip phone on another ip phone, the call terminates after about 5 to 10 seconds. When reading out the hangup cause variable in the h-extention of the dialplan, the hangup cause seems to be 111. In the dialplan output, you can see that SIP-peer sipacc3 picks up the incoming channel SipAgenT01-00001454, and the call is answered. After 7 seconds,

Hello using Asterisk 1.8.32. I notice that there is a spontaneous reboot of the Asterisk system from time to time. When I look in the logs (verbose file) I noticed that every time this occurs it's at a moment that there is a MySQL action, be it a lookup or an insert/update/delete. I must say I do have some MySQL queries that occur in my dialplan when a call comes in, to look up

Hello thank you for your answer. If I read your (and others) reaction correctly I can conclude that this is an Asterisk problem and not a problem of MySQL or dialplan logic ? You should know that the MySQL database is heavily questioned : mysql> show status like '%onn%'; +--------------------------+--------+ | Variable_name            | Value  |

I am trying to send a DTMF digit automatically every 15 seconds to keep a call connected to an alarm panel. I tried using the dial command L and recording a dtmf tone for the beep, but obviously that didn't work. Does anyone have a suggestion for merging the L option and the sendDTMF or the D option? Any other suggestions would be appreciated! Thanks! Paul Gentilini

An application I am running provides a dial tone to my users, read 9 digits, checks whether or not the called party number should be allowed and then dials out using overlap dialing on a pri channel. I.e. exten => _XXXXXXXXX,1,AGI(pm-check-destination.agi) exten => _XXXXXXXXX,2,Dial,Zap/g1/BYEXTENSION|60|CH The AGI-Skript takes about 0.3 to 0.5 seconds (it does a number of rather complex

Hello! I would like to call a number (e.g.35), and when i press a secret code (12345), it should jump to my voicebox menu. On this page http://www.voip-info.org/wiki-Asterisk+cmd+background i found something about "Silently Wait for DTMF Input". In my case it wouldn`t be silence. It woudl just play the away message. Now how can i include such a secret code to my background funktion? I

I'm new to this list, and I apologize if this is an already answered question, but my Google-fu was not strong enough to find the answer if it was. I'm having a problem with DTMF on incoming IAX calls. For the first few seconds of the call (between maybe 1 and 15, it varies from call to call) everything works fine. After that I continue get DTMF_E messages from the remote IAX server

Hi! I'm using asterisk 1.4.17 with twinkle and a custom phone based on iaxclient 2.0.2 and I'm struggling a bit with DTMF and features.conf. While the twinkle client is able to initiate an attended transfer using *2 (as configured in features.conf), the iax client is not. I can see the DTMF messages showing up on the asterisk console, but asterisk does not invoke the features

I have this silly problem : When I place a call, being either to an extention or to an outside line, DTMF signals are ignored by Asterisk. This is serious because I can't even transfer calls (#) or park them (#70). When I receive a call there's no such problem. When I recover a call from parking (71) all goes OK too, and so goes call capturing with *8... I already tested dtmfmode=inband,

On Jul 26, 2014, at 2:05 PM, Jonathan Rudenberg <jonathan@titanous.com> wrote: > I’m running into an issue with libvirt-lxc networking. I have an init program that configures the eth0 interface with an IP and gateway when the container starts. I noticed that programs running in the container encountered “no route to host” errors and looked into it further. What I found is that ARP

I’m running into an issue with libvirt-lxc networking. I have an init program that configures the eth0 interface with an IP and gateway when the container starts. I noticed that programs running in the container encountered “no route to host” errors and looked into it further. What I found is that ARP packets are not making it onto the gateway during the first few seconds of the container’s life.

I realize I cannot use inband audio for phones (voicemail and internal ivr, password for external trunks and other thing not working) So I put everywhere rfc2833. Doing this, anyway, make any EXTERNAL IVR NOT working. I see a lot of posts about this, but no solution, becouse using inband audio (which works for outside...) breaks inside IVR Is it possible to define to use inband audio ONLY on

Hi, Trying to properly broadcast / relay DTMF digits to other confbridge users, but does not appear to work. Goal is to have a conference user be able to receive the DTMF, so it has the effect of being 'broadcasted.' I have the following set up in 'confbridge.conf': dtmf_passthrough=yes From logger.conf, I can see the DTMF tones via setting "console => dtmf". When

Thanks a lot Danny. I have done the part of playing a file by creating a context in my dialplan. Now I am puzzled as i wish to store the DTMF inputs done by the users who is listening to the playback. I found there are ways, but some specific way by which it is not stored in file but conveyed directly to the asterisk server. When the call landed up on the softphone, i pressed keys the

Hi, I have a 1.8.22 Asterisk (Box A) connected to a 1.4.32 Asterisk box (Box B) through SIP. The 1.4.32 box is then connected to the PSTN through PRIs. I've noticed there are occasions where I am seeing duplicated DTMF. I've verified from the SIP trace from the phone that there is only a single '3' being pressed. It appears as though the DTMF end (without a begin) that is detected

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

Hi, in asterisk 1.6.2.16 I get spurious DTMF recognition over SIP from an Audiocodes. I think the DTMF recognition is the Audiocdes' fault, the Audiocodes log seems to say so as well, but I want to make sure, and fixing the Audiocodes is not an option in this particular case - don't ask. Can someone explain to me what the following means *exactly* [Feb 10 21:15:40] DTMF[2538] channel.c:

https://bugzilla.mindrot.org/show_bug.cgi?id=1950 Bug #: 1950 Summary: sshd tries to bind over and over to ::1 for several seconds Classification: Unclassified Product: Portable OpenSSH Version: 5.9p1 Platform: All OS/Version: All Status: NEW Severity: normal Priority: P2

I have a couple of Linksys PAP2T-NA & Grandstream HT-502 extensions that are receiving random DTMF tones on their side, but that are not heard by the outside party. I have been using Asterisk 1.6.6 through 1.6.10 and have always had this issue. I am only using SIP on the Asterisk server and all extensions and trunks are set to rfc2833; outside of this issue DTMF operation works fine.