similar to: Agents and agent counts

Hello, I am new on asterisk and do some tests on freepbx. I have 2 SIP provider: Provider1: In-/Out- Flatrate, only 1 Number Provider2: Incoming Flatrate, Outgoing Cost depend on destination, 3 numbers On Asterisk site i have 3 phones (branch ??, don't know how its called in asterisk) Is it possible to do something like: Phone 1: Incoming Call: Number1/Provider1 Outgoing Call:

Kevin Larsen - Systems Analyst - Pioneer Balloon - Ph: 316-688-8208 asterisk-users-bounces at lists.digium.com wrote on 12/14/2017 09:36:06 AM: > From: "basti" <mailinglist at unix-solution.de> > To: asterisk-users at lists.digium.com > Date: 12/14/2017 09:36 AM > Subject: Re: [asterisk-users] Rewrite Outgoing Number > Sent by: asterisk-users-bounces at

I have a 'day' and a 'night' mode. In the day mode, I play a 'background' message which is interruptable by the pushing of a DTMF key - ie - all is normal. In night mode - I decided to get smarter... I play two backgrounds with a 'sayunixtime' in between and now DTMF does nothing - the menu times out to my 'lets get the operator then'... If I change the

Can any suggest why the mount failed? The IP address of number1 is 192.168.0.1. Here's the script: Thank you. Script started on Mon Aug 16 12:49:14 2004 [root@number4 root]# smbclient -L number1 added interface ip=192.168.0.4 bcast=192.168.0.255 nmask=255.255.255.0 Password: Domain=[HOME] OS=[Windows 5.1] Server=[Windows 2000 LAN Manager] Sharename Type Comment

Hi all, I received a report of a client which stated that two of its agents are ?logging in? to the queues when they actually aren?t there working. They appeared to be logged on all night. They thought they weren?t logging off correctly, but they checked one of them and he was following the procedure. Any ideas of what can be happening? Is there a way to prevent logins to queues afterhours?

I cannot seem to make the directory work properly ( as I see it ). When I call directory with this setup and press the 3 letters, it locates the person and plays their name. When I hit '1' to call it hangs up without any fanfare or message on the console ( verbose 999 ) It doesn't play goodbye or anything, just a drop. What am I missing? TIA, Bill console... -- Accepting

I need to create a legend for a simple scatter plot in the following format. This is Blah1 number1 number2 This is Blah2 number3 number4 . . . This is Blah6 number11 number12 I looked up these help pages and found the following solution. lStr<-c(Blah1, Blah2,....Blah6, number 1, number2, ...number12) legend(x="topright",lStr,ncol=3) So this creates the tabular format I am

Greetings folks, new to this, trying to get the syntax correct for a day of week routing. exten => 345,1,Answer() exten => 345,n,GotoIfTime(10:00-17:00|tue&thu&sat|*|*?open,345,1) exten => 345,n,GotoIfTime(10:00-19:00|wed&fri|*|*?open,345,1) exten => 345,n,Playback(afterhours) exten => 345,n,Hangup() I'll get an error stating incorrect day of week

I have Asterisk 1.2.13 installed as a Debian package and I edit only sip.conf and extensions.conf in this way: sip.conf: [general] realm=work.com.ar ; Realm for digest authentication bindport=5060 ; UDP Port to bind to (SIP standard port is 5060) bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all) srvlookup=yes

Hi, I have two X100P cards in the same system. I can use both of them to initiate and/or receive PSTN calls. I want now to define separate context for each of them, in oder to route inbound calls to different extensions. This is what I have now in zapata.conf file: [channels] language=en context=inbound-analog signalling=fxs_ks usecallerid=yes callwaiting=yes echocancel=yes

Hi people, I want to make a meetme between 2 numbers. First I enter the number1 into the meetme. It is waiting for the other number, but the other number never entered, so, how can I finish the meetme from the dialplan?. Is it posible by using MeetmeAdmin and kick all the users? Thanks, Anahi Ludue?a _________________________________________________________________ Descubre

Hi, I am having trouble passing variables via the call files, here is my call file via the php: fputs($oSocket, "Action: login\r\n"); fputs($oSocket, "Events: off\r\n"); fputs($oSocket, "Username: $strUser\r\n"); fputs($oSocket, "Secret: $strSecret\r\n\r\n"); fputs($oSocket, "Action: originate\r\n"); fputs($oSocket,

Hello, I am calculating 'Betweenness' of a large network using R. Currently, I have the node-node information (City1-City2) in an excel file, present in two columns where column A has City1 and column B has City2 that city1 is connected to. These are the steps that I go through to calculate betweenness of my network. a) Convert the City1-City2 (text) into Number1-Number2 in the excel

Hi guys, I have an asterisk box with sip 20 internal extensions and 100 lines registered on a external voip provider. For most part of time, it work fine, but in few moments it act ignoring sip packets becouse my ip phones can't register in asterisk and asterisk can't register his 100 lines in external voip provider. I have log's only for external registration error: [Apr 16

Hey everyone, I have a more of an opinion question then a technical question. The asterisk server I am setting up is going to host 3 different businesses. Each business is in the same building, and on the same network. My question is regarding calls coming in and going out. We are a small ISP and have a lot of numbers that are forwarded to our phone system. The other companies have about 3

Hi, I'm Marianna I'm trying to apply the command "repeat" to my matrix but the repeat process doesn't work as I would. In particular I would like to apply the function robustm () _that I have created_ to my two matrices, if the difference between the two matrices is less than 0.001, R give me back the last matrix. The code thus created allows me to repeat the process only

Hi Folks, I want to use key() to position the legend on top of the page in the following example. Prefer not to use locator()> par(mfrow = c(2,2))> plot(1:4, title="Plot Number1")> plot(rnorm(10), title="Plot Number2")> plot(1:10, title="Plot Number3")> key(x,y, text=list(c("Alpha","Beta")), text=list(c(1,2)))> > > x,y

Hi guys, I have a customer with asterisk registering 100 lines from my Voip Provider. In some times during a day we receive this messages: [Feb 21 17:09:34] DEBUG[26223]: sched.c:204 sched_settime: Request to schedule in the past?!?! [Feb 21 17:09:34] DEBUG[26223]: sched.c:204 sched_settime: Request to schedule in the past?!?! [Feb 21 17:09:34] DEBUG[26223]: sched.c:204 sched_settime: Request

I have the following asterisk configuration (sip.conf) : [General] externip=82.79.81.3 localnet=192.168.10.0 localmask=255.255.255.0 [Phone1] type=friend host=dynamic nat=yes qualify=yes context=sip callerid="Phone1" <1> disallow=all allow=gsm [Phone2] type=friend host=dynamic qualify=yes context=sip callerid="Phone2" <2> disallow=all allow=gsm [Phone3]

I think your last approach should work (the m3u file), except you specified port 8002 instead of 8000... which would cause the behavior you're getting, winamp opens but does not play the song. Fred -----Original Message----- From: icecast-bounces@xiph.org [mailto:icecast-bounces@xiph.org] On Behalf Of Wolfgang Schwurack Sent: Friday, June 03, 2005 10:18 AM To: icecast Subject: [Icecast]