similar to: Asterisk 1.6 WaitForSilence Problem

I am originating a call to a Local channel using an Originate Action: Action: Originate Channel: Local/dial at outdial Context: outdial Exten: answer Priority: 1 Timeout: 45000 ActionID: some_id In my dialplan, I have this: [outdial] exten => dial,1,Dial(${DIAL_STRING}, ${DIAL_TIMEOUT}) exten => dial,n,NoOp(Dial Status = ${DIALSTATUS}) exten =>

Using Asterisk 1.6.1.14 and dahdi 2.2.0.2+2.2.0. We're placing outbound calls over an analog line. Some of these calls are going to cell phones that play music rather than providing a standard ring. As a result, the Dial command sometimes returns a DIALSTATUS of CHANUNAVAIL and sometimes it returns BUSY. The problem is that this is happening on calls that are being answered. Has anyone else

First off, I also posted this on the digium forums so if anyone here also reads those, sorry for the cross-post. When I place an outbound call using SIP to my cell phone, asterisk immediately starts processing the dialplan without waiting for the call to be answered. We could handle this on DAHDI using callprogress, but I don't know of a similar setting for SIP. Here is the contents of

Is there a way to limit the number of simultaneous outbound SIP calls made by Asterisk? We've tried using the 'Asterisk sip call-limit' parameter but that doesn't seem to be working and one of our engineers says that parameter has been depreciated. Thanks, Deric.Page at nisc.coop -------------- next part -------------- An HTML attachment was scrubbed... URL:

The card you installed has FXO or FXS modules in it ????? are you getting your lines directly from the telco co??? Doug D On Mon 23/08/10 8:37 AM , Cassius Smith cassius at cassius.org sent: * -----Original Message----- * From: Todd Reese * Reply-to: Asterisk Users Mailing List - Non-Commercial Discussion * To: asterisk-users at lists.digium.com [3] * Subject: [asterisk-users] Dahdi

Hi, I know it sounds weird, and this is part of the reason I have not reported that sooner. As I upgraded from 1.6.2.x to 1.8.x several months ago I am experiencing this problem. If a call is initiated from a DAHDI extension after no DAHDI extensions were used for some time, arbitrary DTMF digits are skipped and the call fails. If the call is redialed it goes through. Normally just one (1)

Hello group, I am installing Asterisk on a new pc running Ubuntu 12.04.3 Server. ?I have dahdi-complete-2.7.0.1+2.7.0.1, and have tried 2.6.1 and 2.8.0-rc1 and 2.8.0-rc2. ?I am following along in the ASTERISK The Definitive Guide 4th Edition and on page 116 when I issue the command $ sudo /etc/init.d/dahdi start I get the following error: Loading DAHDI hardware modules: ? ?wctdm24xxp: done

I'be been having problems with the WaitForSilence application in my dialplan. The behavior I've noticed, and hope to demonstrated with the fallowing examples, is that WaitForSilence (WFS for short) does not work until Background is called. Note that the lenghts of the sounds are approximately 5 seconds for bring, and 10 for presses. [Testing] exten => s,1,Answer() exten =>

I have a call server that runs on a few custom AGI scripts initiating calls and then managing the calls. I'm getting stuck on the detecting silence functions. I wanted to use the silence detecting as a quick method of substituting Answering Machine Detection. However, whenever WaitForSilence is supposed to be detecting silence, it always just ends the interval whether or not there is

I have a call server that runs on a few custom AGI scripts initiating calls and then managing the calls. I'm getting stuck on the detecting silence functions. I wanted to use the silence detecting as a quick method of substituting Answering Machine Detection. However, whenever WaitForSilence is supposed to be detecting silence, it always just ends the interval whether or not there is

Looking for some help from the UK please. I backed up all my Asterisk configuration before re-installing the server from 32 -> 64 bit. Unfortunately I did not transfer the backup to another machine!!!!! I now have a TDM400P that is not picking up the line. Can you see what I have done wrong when I have rebuilt the config please: dahdi_scan ---------- [1] active=yes alarms=OK

Hi all. I have a customer who does some automatic messaging.? Back when we were running Asterisk 1.4.x, they used waitforsilence after amd, to wait for an answering machine greeting to finish before leaving their message.? We've upgraded to 1.6, and made no other changes and things don't work. What we're seeing it that a call will get hung up, waiting for silence, well after any sane

I am unable to dial out over a Wildcard TDM400P. This was working previously, so must have messed up the config somehow. I'm running Asterisk 1.6.0.15, with FreePBX 2.5.2.2. When I dial, I see: -- Executing [s at macro-dialout-trunk:19] Dial("DAHDI/1-1", "DAHDI/g0/9239220,300,") in new stack == Everyone is busy/congested at this time (1:0/1/0) It looks like the

Hi All This may be a silly question but... I have this dahdi_genconf generated file: ; Autogenerated by /usr/sbin/dahdi_genconf on Fri Jul 4 22:05:29 2014 ; If you edit this file and execute /usr/sbin/dahdi_genconf again, ; your manual changes will be LOST. ; Dahdi Channels Configurations (chan_dahdi.conf) ; ; This is not intended to be a complete chan_dahdi.conf. Rather, it is intended ; to

I'm trying get this working. I've looked through the list, and can't see how to get AMD to print out more. I have it call and say Hello like I normally would. I've tried to say more and less doesn't seem to matter. After I hangup it does recognize hangup. Here's logging during an attempt where I make outbound call and answer, but then hangup after 1-2 seconds: Jan 24

Is it possible to place multiple phone numbers in a single outbound .call file? If I try doing this, only the last phone number in the file is called. However, if I use 1 file per phone number, then Asterisk attempts to process all generated CALL files at once, incrementing the retry count for each that cannot be called because the designated channel is busy. For example if I have a list of 10

Asterisk - 1.8.4.1 Dahdi-linux - 2.4.1.2 Dahdi-tools - 2.4.1 Kernel: 2.6.37.6 Kernel BKL: enabled I am upgrading Asterisk on this box. It has an OpenVox A400P PCI analog card with 1 FXO and 1FXS module. This server has been running just fine for two years with Asterisk 1.6.1.0 I've just upgraded all the OS and installed Asterisk 1.8.4.1. On trying to configure the A400P I get: # modprobe

I'm having a problem receiving incoming faxes and I'm hoping someone here can help me out. My system is a PBX in a Flash with one dahdi card for my incoming analog lines and another dahdi card for my analog devices (fax and modem). My dahdi-channels.conf file looks like: ; Autogenerated by /usr/sbin/dahdi_genconf on Tue Jun 23 14:56:24 2009 ; If you edit this file and execute

Hi all, As i tested with Asterisk+dahdi+libpri with openvox BRI with NT mode, the ISDN phone does not work. There is the setting and error. ***********************Enviroment***************************** Asterisk-1.6.1.18 dahdi-linux-2.3.0.1 dahdi-tool-2.30. libpri-1.4.11.2 CentOS-5.5 OpenVox B400P ****************************************************************** In my case , I set prot 1 and

I am having problems getting a Nicherons TDM400P wildcard clone to dial out. Everything appears to be configured correctly, but although I see call progress, it never seems to actually pick up the phone. (The following is a test of 911 emergency, where I substitute 811 [repair service] as the actual number dialed.) *CLI> -- Executing [911 at from-internal:1]