similar to: Dial Plan Agents (1 of 2) agent-dialplan.conf

This version contains a fix to mbr.asm, plus a version of Murali's menu system, but ported to gcc. Although gcc produces much bigger 16-bit code (since it's really 32-bit code with prefixes) I figured it would be much easier for people to deal with since it requires the configuration to be compiled in. That being said, it's a *very* powerful menu system, partially *because*

Hi, We are starting a new project, and are trying to decide the best way to proceed. We want to setup a LAMP configuration using Centos, something we have been doing in the past with great success. The question is load balancing. We antisipate the potential for the system to receive 500,000 requests/ day with in the next year. We want to plan for that extra load now as we start the

Hai, I did see that you are using Administrator, and thats the problem. Administrator is mapped to root ( most of the time ), if you assigned Administrator UID = 0 then you have a problem, because only root = uid 0. Never ever give Administrator a UID/GID, create a new one assign that one a UID/GID. So try again with a normal user, that does have a UID/GID. If that does not work, please

Thanks for the prompt reply! > I did see that you are using Administrator, and thats the problem. > Administrator is mapped to root ( most of the time ), > if you assigned Administrator UID = 0 then you have a problem, because only root = uid 0. > > Never ever give Administrator a UID/GID I am using tdb backend. It mapped administrator account to 12000:10000. > So try again

Ronald, Grandstream products have a one year warrantee. If you don't have any luck with Pulver, contact us and we can probably get your phones exchanged. Please don't assume that your experience with Grandstream is typical. We sell a lot of these phones and the overwhelming majority of the purchasers are very happy with their units. The quality has improved tremendously over the last

Hi All, I'm upgrading some PBX's from 1.2 to 1.4 and having a bit of trouble with converting the Realtime application to the REALTIME function. I have the method down and understand simplistically what is going on, at least enough to get my old 1.2 apps to run in 1.4 functions. I do not understand why change from the app to the func? What the benefits? To me, the app seemed so

We often find ourselves reading through all sorts of contests on the Internet that never seem to echo our own personal skill set or interests. Perhaps you've even fantasized about a type of contest with the types of prizes and goodies that YOU'D actually enjoy. Maybe you've wished there were something along the lines of a asterisk phone system diagram contest? With prizes ranging from

There is a problem that bothered me for a long time: Since one of the 1.6.0.x patch releases up until 1.6.2.5 a "dialplan reload" works only once with a bigger dialplan. If I issue "dialplan reload" again, it won't do anything. After doing so the cli won't show responses to any commands anymore. So if I have to do another change to the dialplan, I have to stop/start

Hi, I'm using Asterisk 1.2.1 on Sarge. it seems like if I call a phone and nobody answers, asterisk does not jump to the next priority...it freezes. Take a look at this: exten => 777,1,NoOp(before) exten => 777,2,Dial(SIP/7|60|g) exten => 777,3,NoOp(after) priority 3 is never executed but this worked with Asterisk 1.0.7!!! TIA Giorgio Incantalupo

Recently I setup samba 2.23 on a Red Hat 7.3 machine to be the Primary Domain controller, my win2k machine logs in just fine with username and password I setup sees the shares it should, however the problem I have is if I try to copy something to one of the shares it copies hte file 3 times, one is the origianl one is blank, one is coded and is the same file. And I can't seem to figure this

Hi guys, I need to realize a sort of automatic call monitoring dialplan. This is exactly what I need: - A user originate a call - When the call is bridged (or just before) I need to invite automatically a third party to the conversation that should hear the audio channel but not speak (it's a monitoring application for a callcenter) The person in charge of monitoring cannot use

I have a very strange problem. I have two Asterisk servers running 1.4.2. On the first one I have the following options: Connected to Asterisk 1.4.2 currently running on pbxoficina (pid = 7057) Verbosity is at least 3 pbxoficina*CLI> help dialplan dialplan add extension Add new extension into context dialplan add ignorepat Add new ignore pattern dialplan add include Include

Hi all, there is an application called changrab with quite interesting capabilities: http://www.freeswitch.org/asterisk_stuff/app_changrab.c I think it is available in the 1.4 version by default!? This application can connect to channels which are already UP. The only possibility AFAIK to connect channels after they are UP are the well known conferencing applications meetme and conference. If

Hello, I'd like to implement something similar to an attended transfer, but with a little more control (I'd like to be able to use MixMonitor and StopMixMonitor to control the call recording, set the account code, etc. I'm on Asterisk 1.4.26. All of the ways I have seen to do this are complicated plans using MeetMe and applicationmap features, and playing with those over the

Hi List, How to put multiple call on hold by dialplan in asterisk? -- ----- Thanks and regards Virendra Bhati +91-9172341457 Asterisk Engineer -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110517/0396c10f/attachment.htm>

Hello, I would like to known how to use DUNDi with a Lua dialplan ? In extensions.conf, we should do like these: |[lookupdundi] switch => DUNDi/priv [internal] include => dundiextens include => lookupdundi exten => _XXXX,2,NoOp(calling ${EXTEN}) exten => _XXXX,n,Dial(SIP/${EXTEN}) exten => _XXXX,n,Hangup()| priority 1 is either defined in dundiextens (local registered

Hi all, Does asterisk support the following scenario? I need when a customer who own an endpoint registered on asterisk open the line, the asterisk will run a specific AGI script inside the endpoint context? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090421/4dce9c34/attachment.htm

Hello, everyone Here part of my dialplan context [globals] CMD_NOOP=0 CMD_DOSTUFF1=1 CMD_DOSTUFF2=2 CMD_DOSTUFF3=2 [blah-context] same => n,Set(COMMAND=${CMD_NOOP}) same => n,UserEvent(blah-event,CHANNEL:${CHANNEL(name)} same => n(COMMAND_SWITCH),GoToIf($["${COMMAND}"="${CMD_DOSTUFF1}"]?LBL_DO_STUFF1) same =>

Hello Asterisk community, I have written a document that covers an Asterisk implementation I am building. I want to place it on the lists so USERS can view and make comments on, the ideas contained within. I think it is an important issue to develop a standardised Dialplan for applications, not just for Asterisk, but for pbx systems in general. As they become cheaper and more common place,

Hello listers, Still plodding along in the 1.4 tree, though I've started to dabble in 1.6 land. Today's adventure involves a 2600 line dialplan. My friend Google only points me to an antique java script and a bunch of GUI dialplan creators. What is out there that will point out dialplan errors/problems besides just watching the CLI output and trying to figure out with