similar to: Problem with DTMF detection in ast_app_getdata (*1.2)

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Followed this; #cd /usr/src #export CVSROOT=:pserver:anoncvs@cvs.digium.com:/usr/cvsroot #cvs login (password is anoncvs) #cvs checkout zaptel libpri asterisk #cd zaptel ; zaptel equipment #make clean; make install #cd ../libpri ; isdn #make clean; make install #cd ../asterisk #make clean ..but

I have a couple of quick questions that I do not see answered on the wiki or the Asterisk archives. I dropped an note to the Dev list and got an "awating moderator" notice. I realized I had left out some important info, so I cancelled the message and resubmitted it, but have not gotten another "awaiting moderator" message. I suspect it was too similar to the original that I

Hi all, I have browsed through a couple of posts that deal with the failure of applications that originally worked on asterisk 1.2 but fail on asterisk 1.4, but can't seem to understand what I need to change in my installation. I also went through the CHANGES.txt file in my asterisk source directory, and still was none the wiser. I have a a MySQL integrated Prepaid application which worked

Hi all, I am running Fedora Core 5 on an Intel Server board with 2GB memory and kernel 2.6.18-1.2798.fc6 . I am running asterisk-1.2.10 from bristuff-0.3.0-PRE-1s. I have a mysql-version of app_prepaid which builds properly but when I run asterisk -gc I get the following: [root@myserver /]# asterisk -gc Asterisk 1.2.10-BRIstuffed-0.3.0-PRE-1s, Copyright (C) 1999 - 2006 Digium, Inc. and others.

Hi all, I have been trying to get my asterisk (v1.2.10) to lookup voicemail config data from my mysql database as opposed to voicemail.conf + sip.conf for my users. Users register with SER and get passed through to asterisk when they dial out. I followed the instructions as per http://www.voip-info.org/wiki/view/Asterisk+voicemail+database so basically I have 1) Build asterisk-addons-1.2.5 and

Hi all, I downloaded a copy of bristuff 0.4.0-RC11 on my Ubuntu 10.04.2 LTS server machine with 2.6.32-24-generic-pae kernel. The prereq.sh script executes without complaints (BTW on my system, libncurses-dev evaluates to libncurses5-dev and libz-dev evaluates to zlib1g-dev). With the asterisk 1.4.41 package that is installed, a make menuselect operation indicates that all dependencies are met

Hi, How do I disabled asterisk to use database and storage voicemail in directory. Im getting the below error [Dec 3 19:08:53] WARNING[4934]: app_voicemail.c:3430 inboxcount: Failed to obtain database object for 'asterisk'! [Dec 3 19:08:55] WARNING[5129]: app_voicemail.c:2353 last_message_index: Failed to obtain database object for 'asterisk'! [Dec 3 19:09:04]

I recently moved my Asterisk configuration to a new server and re-built Asterisk from CVS. Now, I'm experiencing the following issue with SIP: Executing Dial("Zap/1-1", "SIP/100|20") in new stack NOTICE[-1232077904]: File app_dial.c, Line 518 (dial_exec): Unable to create channel of type 'SIP' == Everyone is busy at this time Has anyone seen this issue before?

On 4/20/2023 7:16 AM, Rhys Hanrahan wrote: > > Hi All, > > I’m having issues figuring out how to set no DTMF timeout on the AGI > GET DATA command as “0” still has a multi-second timeout if no input > is given. I am trying to find the source code of the AGI command to > figure it out, but I can’t for the life of me find the underlying > source code. The closest I’ve

Hello all, I have been searching thru all docs that I can find on wiki and such but can not get an answer. I am trying to collect a date from user input in the form of digits dialed from the phone to use in an agi script to do a database look up. I have tried to use "Get Data filename, timeout, maxdigits " in the agi script. In * console I get message saying playing filename but it

Since www.bkw.org seems not to exist anymore (getting response from some hosting provider), does anyone happend to have a copy of app_valetparking.c from www.bkw.org - the one that should work with * stable 1.0.X ? If so please contact me. One that can be downloaded from www.loligo.com dosn't compile with 1.0.X, and SuperValletParking (www.asterlink.com/svp/) seems to be for * HEAD

Nope ! This is the one that tries to include PRE 1.0.X header file <parking.h>. It cannot compile on * 1.0.X (I have tried also to include <features.h> instead of <parking.h> (as far as I know features.h is successor to parking.h), but still without results). Thanks anyway. Nenad > > Try this > >> Since www.bkw.org seems not to exist anymore (getting

Hello, I need to be able to send a DTMF to an existing channel remotely. So I made a php script to do such with the Manager command PlayDTMF. I need it for example to start a transfer. isb177*CLI> features show Builtin Feature Default Current --------------- ------- ------- Pickup *8 *8 Blind Transfer # #8 Attended Transfer

Hello, I need to be able to send a DTMF to an existing channel remotely. So I made a php script to do such with the Manager command PlayDTMF. I need it for example to start a transfer. isb177*CLI> features show Builtin Feature Default Current --------------- ------- ------- Pickup *8 *8 Blind Transfer # #8 Attended Transfer

Hello, how are you? I need to be able to send a DTMF to an existing channel remotely. So I made a php script to do such with the Manager command PlayDTMF. I need it for example to start a transfer. isb177*CLI> features show Builtin Feature Default Current --------------- ------- ------- Pickup *8 *8 Blind Transfer # #8 Attended

Hello, I need to be able to send a DTMF to an existing channel remotely. So I made a php script to do such with the Manager command PlayDTMF. I need it for example to start a transfer. isb177*CLI> features show Builtin Feature Default Current --------------- ------- ------- Pickup *8 *8 Blind Transfer # #8 Attended Transfer

Hello ! My asterisk log is full of messages like this: [2011-03-06 19:01:15] DEBUG[20556] res_rtp_asterisk.c: Ignoring RTP 2833 Event: 0000009f. Not a DTMF Digit. [2011-03-06 19:01:20] DEBUG[20556] res_rtp_asterisk.c: Ignoring RTP 2833 Event: 0000009f. Not a DTMF Digit. [2011-03-06 19:01:25] DEBUG[20556] res_rtp_asterisk.c: Ignoring RTP 2833 Event: 0000009f. Not a DTMF Digit. [2011-03-06

Are there any known problems converting dtmf from oob over iax2 to inband over rtp/ulaw? Obviously it works when converting to inband over pri/ulaw et al, but how about rtp? I've got packet traces that confirm that 2833 packets are properly generated when I have 2833 configured for the rtp link, but the other side seems to be ignoring those packets. So I tried inband on that link; nothing

I have a telco providing DTMF inband, they say they can't provide it any other way. This is creating headaches for me. What is the common method for SIP DTMF? Kpml, or 2833 or inband? My handsets don't support inband so I'm tying up some expensive resources to convert the inband DTMF to out-of-band DTMF... Can you recommend a vendor in US that provides SIP with DTMF in RFC