similar to: Asterisk 1.6 Displaying in Debug that it is playing a ulaw file using BackGround() but no audio is heard from the phone

Thought a different succinct subject line must drum up an answer or two... Also, this has been tested from two different carriers: We're getting an average of 2/10 call success rate. ---------- Forwarded message ---------- From: Joe Wood <schmoe at gmail.com> Date: Thu, Aug 26, 2010 at 6:58 PM Subject: Asterisk 1.6 Displaying in Debug that it is playing a ulaw file using BackGround()

On 8/27/10, asterisk-users-request at lists.digium.com <asterisk-users-request at lists.digium.com> wrote: > Send asterisk-users mailing list submissions to > asterisk-users at lists.digium.com > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.digium.com/mailman/listinfo/asterisk-users > or, via email, send a message with subject or body

Hello, I have a problem of DTMF duplication. I receive call from my provider with SIP protocol. These calls pass through an interactive voice menu, using the application Waitexten to enter a client code. The menu works fine, but sometimes I have DTMF duplication that prevent proper code entry. All DTMF come twice. my sip.conf ----------- [general] context=default allowguest=no

Hi, I have a Digium TE410p T1 card and I've noticed that under asterisk 1.4.17/18 I have problems detecting DTMF in IVRs. I think I've narrowed the problem down to some sort of interference between the greeting that is playing and the DTMF tones. DTMF detection seems to work very reliably when I am in Read() or WaitExten(), but is absolutely unusable while in Background(). I hope someone

Gang, I'm having this error pop up when I do a ForkCDR, and I'm not sure how to get around it. Here are a few log lines: Nov 8 10:37:08 VERBOSE[28079] logger.c: -- Executing ForkCDR("Zap/49-1", "") in new stack Nov 8 10:37:08 WARNING[28079] app_forkcdr.c: Channel does not have a CDR The scenario occurs like this: I use a .call file to generate a call on

I'm a bit confused about how WaitExten works. I assumed that when it returns 0, the next priority in the extension would be executed, but that doesn't seem to be the case. When I get to WaitExten and enter extension 8, it plays the message, then Waits another 10 seconds and times out. [local] exten => s,1,Wait,1 ; Wait a second, just for fun exten =>

I am trying to setup an interactive menu where a caller hits the main menu and can then dial an extension. As far as I can tell the "Waitexten" app is failing to read 3 digits and just reading the first and then announcing that it is invalid since all extensions are 3 digits. How do I make Waitexten wait for 3 digits? I have setup the extension "100" for users to reach the

All: Some of the people who dial into to our system will press the pound key when entering an extension for the directory key. When waitexten gets that, I get an error messages as, for example 123# doesn't match any extension. I was going to use ${EXTEN} to just use the first three numbers, but I'm not sure how to use this with WaitExten. so I have exten =>

Having some issues with getting sound from a call. I have 4 systems. 3 main systems which handle calls for our 3 locations. The 4th system is the central voice mail system. When an inbound call gets passed to someones voice mail its done with an IAX2 connection. The same happens after hours when we have our night mode set. If you dial the main number after hours you are passed straight to the

I would appreciate hearing from anyone who has figured this one out. Here's the scenario: I have a context wherein I give the called party the option to dial the digit 9. If he does so, he is transferred a la this extension entry: exten => 9,1,Playback(pls-hold-while-try) exten => 9,n,Noop(Attempting to bridge to ${agentext}) exten =>

Here is my Extensions.conf file (Default Context). When an individual calling in dials the extension, the response time seems very slow. It doesn't immediately go to the next step, but hangs out for a few seconds (silence)... Suggestions? Thanks in advance... /pj [default] exten => _XX.,1,Wait,2 ; Wait a second, just for fun exten => _XX.,n,Answer

Hi, Firstly, what I'm trying to do is: * Get asterisk to pick up a SIP call via a DID * Prompt the user * When the user puts in a number, go to IAX.conf and route it according to what I've specified there, i.e Least Cost Routing, etc. I've set-up something similar to what I've found online, but it doesn't work! Asterisk doesn't pick up the call at all..... :( The files

I've got the following set up: Someone calls into my PBX on a single number (via SIP trunk from my carrier), and the get a voice menu of extensions. On one of the extensions, it rings a bunch of internal SIP hardphones, plus ringing my cellphone via a hairpin through the cairrier's SIP/PSTN gateway. The issue is that my cellphone shows my PBX's number, not the original calling

Hi list, I enabled the transfer function in my dialplan, but I may configure it wrong because sometime when I call a SIP extension number from one FXS port, the SIP side will hear word "transfer", I hear nothing, after that, the call conversation is fine.I'v had this problem for a long time, could not get clue where I configure it wrong. here is my related config part: sip.conf:

I'm in the process of setting up a 'phone tree', and are running into some problems. My goal is for users to dial a phone number, the asterisk system picks it up, plays the greeting, and users can type whatever they want into the system. What actually happens is users dial the phone number, asterisk picks up and additionally goes off-hook on another line, plays the greeting and

I cant seem to get outgoing calls to be placed properly .. No matter what I try I get an error from the PSTN company saying that the "call can not be completed as dialed" or "you need to dial a one..." The asterisk debugging seems to show the correct number being dialed out of the zap interface... the "9" is being stripped and there is a "1" where it is

Hello all, I'm planning to create a simple Auto-Attendant (IVR Menu) for my home PBX yet all callers from incoming (trunk) calls must only press the extension numbers from the [analog-ext] else will play the "pbx-invalid". How do you do that using the GotoIf() (or probably using the other applications) but will check if the numbers entered belongs to a specific context? Also, how

I just brought up an asterisk server. On dialing "2" from grandstream hardphone, I get the beginning of the welcome message, but each segment is cutoff. Specifically "Asterisk is an open source full"-1s silence-"if you'd like to learn more technical information about Asterisk"-11s silience-"goodbye" Any help or pointers on how to gather more debug

I have a Asterisk PBX System with Redhat Linux Fedora 4, Webmin version 1.400 and I am simply trying to configure into the "Extensions.conf" script an entry that will add to the "Auto-Attendant" a line that will allow a "Caller" to enter a "0" (Zero) will then ring the extension(s) of the "Operator" to speak directly with the "OPERATOR"

<br>My dialplan is as follow<br><br>exten =&gt; 1002,1,Answer<br>exten =&gt; 1002,n,Wait(2)<br>exten =&gt; 1002,n,Background(thank-you-for-calling)<br>exten =&gt; 1002,n,Background(vm-enter-num-to-call)<br>exten =&gt; 1002,n,WaitExten()<br> exten =&gt; 1002,n,Hangup<br>exten =&gt;