similar to: IVR Survey

Hi I'm using asterisk 1.8 on CentOS 5 I'm initiating call recordings with MixMonitor and trying to pause them with the features.conf. Whenever I try to pause the recording the call dies. Is PauseMonitor incompatible with MixMonitor? Here are some key log excerpts features reload == Parsing '/etc/asterisk/features.conf': == Found == Registered Feature

I have got call recording working on our 1.4.30 asterisk box together with a recording pause ability and being able to play different audio to each party at the start and end of the pause. This all works perfectly but one wish is to have the audio files have a beep or something in them so when you listen later you can tell where the audio was paused. So I changed things around so that instead

Hi All, I'm at the end of my tether here and would really appreciate some help. I'm trying to implement DTMF based pause/resume of call recording. I'm using Asterisk 1.4.22.1. Here's the scenario: The caller (SIP or ISDN, doesn't matter) dials into the asterisk which executes the following code: exten => _X.,1,Monitor(wav,${CALLDIR}${UNIQUEID},mb)

Greetings all, I'm trying to connect to an AT&T teleconference, but the call is never marked as ANSWERED by asterisk and therefore won't bridge and continue. The only work-around I've come up with so far is to dial like this: Exten => 744,1,Dial(Zap/g1,,p) The "private" mode keeps the line open without trying to do a bridge, but requires the

After upgrading from asterisk-10.5.0 to asterisk-11.1.2, I am getting a Segmentation fault. [root at localhost asterisk-11.1.2]# asterisk -vvvvvvc Asterisk 11.1.2, Copyright (C) 1999 - 2012 Digium, Inc. and others. Created by Mark Spencer <markster at digium.com> Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details. This is free software, with components

I'm just trying to make a real simple Survey via php. Just want it to play the Question Files, wait for a response, save the response into the correct variable and then email it all. I have no issue playing the audio or emailing. But I can't get it to wait for digits or to properly capture those digits into the variables. I know the code is technically right since the emails have this

I have the below script that doesn't seem to be working. I don't know if I have something in the script wrong that I am just missing. Or if I don't have the php.ini set correctly for emailing This is the CLI output -- Executing [4099XXXXXX at port3_real:1] Goto("DAHDI/50-1", "newhire,s,1") in new stack -- Goto (newhire,s,1) -- Executing [s at

Alright I have a SIP phone located off premises with a very annoying issue. Well I say a sip phone it is actually two phones hooked to a Cisco Spa 2102 Link: http://www.cisco.com/en/US/products/ps10026/index.html Each phone being a different line/extension. Alright either line can ALWAYS make outbound calls no issue. The problem is on the Inbound side. I'm completely stumped as

There are times when I need to call a client from my cell and I want a recording of the call. I was trying to put into an * a way of doing that. Below is what I'm using in my extensions.conf exten=> X,1,Read(num,"/var/lib/asterisk/sounds/mtas/10digit",10,,,5) exten=> X,2,SayDigits(${num}) exten=> X,3,Background(/var/lib/asterisk/sounds/mtas/verify) exten=>

Telco, location, ect?) At X times of day? =20 Ect, ect. =20 It sounds like bleed over, which can be causes by some many things the best place to start is to find a pattern if there is one. =20 James Shigley Monroe Telephone Answering Service =20 From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of David @ULC Sent: Tuesday, May

Alright I've been having an issue when trying to dial out locally when coming from SIP. This used to work no problem, now it doesn't. Now the local PRI to Bell Is working fine I have calls coming in and out of it constantly right now. BUT if I try and make a local call from SIP (from X-Lite or one of our Linksys SPA2102s) It fails every time with errors like these == Using SIP RTP

Hello, I have a recording started in the dialplan with the MixMonitor application. I want to be able to stop it during a call and maybe restart it. I tried using the value defined in [featuremap] but it starts another MixMonitor application even if there already one instead of stopping it. Any idea on how I can stop the MixMonitor application while it is running? [featuremap] automixmon =>

Other than having stripping out IPs this is what I am receiving for my voip calls. Now I normally use ${CALLERID(rdnis) for RDNIS, this works fine calls that come in on my PRI. BUT at least from this VOIP source the To field which is my RDNIS information for these calls, doesn't actually fill into ${CALLERID(rdnis). But as you can see I'm getting the information. My question is, Does

Ok I have a queue that is working perfectly. The problem is when one of the agents is outside the building on an external phone line (say a cell phone). My telco hangs up on the call . I think the telco is hanging up on these calls because there is no CID attached. (I know my telco wont connect calls without ANI, so that is what it is my assumption) So first I need to prove my assumption

First off this is not my work for extensions.conf it is modified from http://leifmadsen.wordpress.com/2009/07/15/migrating-from-agentcallbackl ogin-to-standard-dialplan-methods-part-1/ So credit to Leif Madsen <http://www.leifmadsen.com> But as to my question [AgentLogin] ;A replaced version of AgentCallbackLogin() using a GoSub() ; exten =>

Trying to configure IAX for use I think I have everything set right. But my IAX phone wont connect. When I run wireshark I'm seeing this Note if above screenshot from wireshark does not show here is a link for it: http://img402.imageshack.us/i/tempe.jpg/ I've tried a variety of setups in my IAX.conf (they all end up with the same issue, tried just bindaddr=0.0.0.0 with

Hi, I'm currently working on a setup between Asterisk and VoiceGenie (which is a IVR system). The way my setup is done, is that I have a PRI line coming in my Asterisk server, and then VoiceGenie is connected to Asterisk via SIP, like any other softphone basically. I'm able to receive calls in Asterisk and then link them with VoiceGenie. But one of my issues is that when I get an

Ok I have a queue that is working perfectly. The problem is when one of the agents is outside the building on an external phone line (say a cell phone). My telco hangs up on the call . I think the telco is hanging up on these calls because there is no CID attached. (I know my telco wont connect calls without ANI, so that is what it is my assumption) So first I need to prove my assumption

AGI is executable. =20 Then run 'agi debug' from the asterisk cli, place a call and see what was send and receive from your agi =20 From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of James A. Shigley Sent: April-23-09 12:26 PM To: asterisk-users at lists.digium.com Subject: [asterisk-users] AGI PHP script =20 I have the

Thank you for the rapid response. I will try and investigate getting answers to some of your points but I'm a little new to Solaris so I'll need some time. Glancing at the configure output, it looks like it built against v0.1.7 of libusb (yes i think that is derived from the one you mention), checking for libusb version via pkg-config... 0.1.7 found checking for libusb cflags... checking