similar to: Console duplicate output problem

Hello, I'm running asterisk-1.0.6 on a centos3.4 box. I'm still in testing phase and so far everything is running smoothly. I'm now trying to play a soundfile or an mp3file using 'MP3Player', 'Playback' or the 'Background' commands, but don't get any sound. The logfile says: -- Executing BackGround("SIP/joa-9def", "tt-weasels") in

> I'm going blind searching - maybe you know? > > During the execution of a script I want to play fake ring to caller. > Both of these examples complain of missing option: > > $agi->exec("Ringing"); > $agi->exec("Playtones ring"); > > Notice: Undefined variable: options in > /var/lib/asterisk/agi-bin/includes/phpagi.php on line 326

Hello, In my quest to figure out the source of the random echo on our shiny new asterisk install, I have been using ztmonitor on the TDM400p channels for the good part of today. I have 2 TDM400p cards with 8 FXO modules and 6 pots lines connected to them (last 2 channels are unused but configured in zaptel). Even when the lines are onhook, the Tx values settle down to 0 but the Rx values still

Does any have or can point me to /ACR/ Anonymous Call Rejection message I can download? The one I found was not not too clear. Thanks, Bart -------------- next part -------------- A non-text attachment was scrubbed... Name: bhfisher.vcf Type: text/x-vcard Size: 253 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20090916/c3b682d5/attachment.vcf

I have a block of DID's that I ported to Vitelity about 7 days ago. The problem is if a POTS caller dials into the system, his dtmf is not heard at READ() or Background() while a prompt is played. After the prompt is finished, then dtmf is heard. I've been working with their support, but it still not resolved. SIP callers are not effected. Yesterday, I purchased a DID from

I have some old PC's I want to build as a test box - It's up and running OK now. Now I installed a TDM400P and there is nothing I can do to get the card to come up. My guess is the box is not PCI 2.2 compliant or does it need to be to see the card? Thanks, Bart Here's what I know: Processors 1 Model Pentium III (Katmai) CPU Speed 551.37 MHz Cache Size 512 KB System Bogomips

Is it possible to have a different verbose level full log than to console output? I'd like to keep console verbose at 1, but now full log is at 1 also. Bart -------------- next part -------------- A non-text attachment was scrubbed... Name: bhfisher.vcf Type: text/x-vcard Size: 253 bytes Desc: not available Url :

I've moved 1 of our facilities over from 1.2 to 1.4 two weeks back. So far, the only issue that I've encounted is. I have a scheduled CRON job that runs at 3am every Sunday, that issues a: asterisk -rx 'restart when convenient' The first Sunday that it ran, Asterisk never restarted. The CRON logs show that it issued the command successfully. This Sunday, it ran but never

I'd been putting off installing the FX3 packages because of some extensions I use that don't have upgrades. [*] However, when FX1.5 crashed on me today I decided to finally go ahead. Fired up yum and I'm told I have to install a slew of evolution28 library packages in order to install the firefox package. Why? I removed evolution from my system for a reason, and I'd just as

I've got CentOS 5 installed on my hp pavilion ze 5300 laptop and the network configuration UI says my wireless NIC has been correctly identified and assigned what appears to be the right Broadcom device driver on eth1. However, the inteface doesn't come up. Google doesn't find anything recent about using Linux with Broadcom except for someone's passing remark that Ubuntu 7 beta

Is there a way to reduce the volume of the dial tone on a zap channel? I don't want to reduce the audio volume on calls so txgain in zapata.conf will not work. I am having problems with asterisk not recognizing the first dialed digit from an analog phone about 8-15% of the time. Once the dialtone goes away, the digits are always recognized. Any other thoughts on how to solve this are also

I'm looking for a solution to determine if a PSTN call to a zaptel channel was originated from a VoIP provider or not in real time. I'd like to use the callerid(num) to reverse match to the provider. Does anyone have a clue how I could do this? TIA Bart -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi all, I would like to announce a CentOS on Laptops initiative. The aim is to allow everybody in the community (and on this mailinglist) to document their own experience with CentOS on their laptop (on the CentOS wiki). The goal of this initiative consists of 2 parts: - help and convince people with their own CentOS on laptop installation - promote CentOS on other Laptop websites and fill

Hi all, I would like to announce a CentOS on Laptops initiative. The aim is to allow everybody in the community (and on this mailinglist) to document their own experience with CentOS on their laptop (on the CentOS wiki). The goal of this initiative consists of 2 parts: - help and convince people with their own CentOS on laptop installation - promote CentOS on other Laptop websites and fill

Hi I want to submit a bug about auto-dial , but I am not sure on which project the auto-dial comes, how to know about which project , auto-dial comes Thanks Joseph ___________________________________________________________ To help you stay safe and secure online, we've developed the all new Yahoo! Security Centre.

I've searched but can't find an answer as to how many MGCP paths can a single ADIT/CMG card support? It appears it's only 24 ports, maybe 48. What I'd like to do is install 6 Telco T1's into a single (or more) Adit 600 and route inbound calls towards asterisk. Can I have more than one CMG in a single chassis? Or maybe you know of a better way to connect T1's to

> -----Original Message----- > From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users- > bounces@lists.digium.com] On Behalf Of Eric "ManxPower" Wieling > Sent: Thursday, December 14, 2006 4:30 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] Hardware TDM Switching > > asterisk@nicox.org wrote: >

Hi All! I am facing some delay in conferencing. Using DIAX for Voip calls ,no hardware used yet I am using MeetMe to achieve conferencing and am having a lot of delays. Can anyone tell me how to reduce the delay Regards, Amna Saleem -------------- next part -------------- An HTML attachment was scrubbed... URL:

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Hi List; What configuration needed to let my FXS send and receive FAX? Regards Bilal