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I would like to know how to record all calls on a queue. Anu good sugestions?

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On 12/13/06, Ed Nu?ez <enunez@netoneint.com> wrote: > > I've been trying to find where to download the Web Vmail application and > instructions on how to install it for Asterisk BE. Any ideas? Is this any different than the vmail.cgi that comes with the open version? Otherwise, you will just need to grab a compiled co...

Does anyone know how to gain access directly to the configuration files in AsteriskNow? I have dual NICs and need to change the binding in the config file. I believe they blocked ssh2 access by default. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070504/c4d288d1/attachment.htm

What is a good solution for playing music on hold on the 1.2 branch. I do not want to use mpg123 because last time I used it in a production server it caused many problems. The MPG123 process was taking about 60% of my Xeon CPU. -------------- next part -------------- An HTML attachment was scrubbed... URL:

I installed a hardware g729 codec card in my asterisk, and I'm getting the following error when calling from a g729 sip extension to a SIP trunk also set to g729. The call goes through just fine, but these error messages keep flying by until I disconnect the call. Any ideas? ERROR[11871]: channel.c:1316 queue_frame_to_spies: Translation to slin failed, dropping frame for spies Jun 5

Good morning, Pardon for this intrusion I just wanted to let everyone know about some of the specials that I have going on at HYPERLINK "http://www.astawerks.com"www.astawerks.com . From now until the end of June I will have a huge unpublished sale on all Digium products. Prices are way to low to list so I will have to be personally contacted. I also have a permanent sale on all

I am running Asterisk 1.4.5 and addons 1.4.1 in a CentOS 5 Server. My PBX has experienced several core dumps the last couple of days and I am not sure if this is what's causing it, but it always seems to happen when a particular extension on a grandstream phone uses ChanSpy SIP group. I have not been able to locate where the core dump file is being saved. I can't find it in my

With Asterisk BE I am trying to record calls coming to a queue,. I am getting the call to record, however the file name that the file saves to, is not the correct one. In my extensions.conf, I have the following entry to set the file name. exten=> 0072,1,Answer exten=> 0072,2,Ringing exten=> 0072,3,Wait(2) exten=> 0072,4,Set(AGENTFILENAME=${CALLERID(number)}-${TIMESTAMP}-${EXTEN:4})

This may be a Linux newby question, but here it goes. I was reading the instructions on downloading and installing Asterisk GUI, but I can't get this to work. svn checkout http://svn.digium.com/svn/asterisk-gui/trunk asterisk-gui What would be the equivalent command in CentOS 4? http://astrecipes.net/?n=217 -------------- next part -------------- An HTML attachment was

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Has anyone succesfully tried using ChanSpy on SIP channels with the latest Asterisk 1.4? I tried ChanSpy(SIP/5060) to monitor SIP extension 5060 and the console displays, Monitoring Sip/5060, but I don't hear anything. I am able to monitor Zap channels. -------------- next part -------------- An HTML attachment was scrubbed... URL:

I am seeing a peculiar message on my console screen on my new installation of Asterisk 1.4.5I would appreciate any comments. Really destroying SIP dialog '3f9a224b517f88c961a311324bfe24b7 at 64.211.222.23' Method: OPTIONS Really destroying SIP dialog '099a002b064129d74fc2e4cd4a88c2ef at 64.211.222.23' Method: OPTIONS Really destroying SIP dialog

I have an Avaya 4610SW IP phone which I have upgraded to SIP firmware. I have successfully registered this phone to Asterisk BE as well as Asterisk 1.4.18 Almost everything is working well. Except for two issues. One of the problems is that the phone looses registration every now and then ad I have to re register. I have found a tip for this which I am testing if it will work, which

I would like to know if anyone here knows the answer to the following question I need to implement the following conferencing feature for my agents. 1. Agent receives call from caller 2. Agent conferences a verification service 3. After finishing the verification, agent needs to drop third party (Verification service) and continue on the line with caller. My problem

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If you use both the public and private interfaces for VoIP in the Asterisk Server, make sure you don't specify one of them for the binding in sip.conf Example bindaddr=0.0.0.0 will allow SIP traffic on any of your interfaces. Ed Nu?ez IT/Telecom Engineer 4037 Metric Drive Winter Park, FL (o) 407-384-4200 x 1656 (f) 407-384-4222 (c) 732-925-0730 -----Original Message----- From:

Hello list I am having an issue with Chanspy/SIP that I?m hoping someone has come across and resolved in the past. I am sending calls that come in TDM through T1 ZAP channels and go out to a SIP trunk. If I spy on the SIP channel, I can hear the person on the SIP side of the call just fine, but the person on the ZAP channel fades in and out. If I spy on the ZAP channel, and can hear