similar to: op_div: non-numeric argument

Hi, For some reason (outbound call tracking) I've got a few different outbound call process (using a macro for queuemetrics logging, or direct call) i wanted to factorise the routing process so i came up with something like the following. All in one it's working like expected, however every "ael reload" command trigger a lot of warning like that "application call

Hi list: I'm having the next problem, it appear that the application ChanIsAvail is not working on Asterisk 1.4.5 always return me 0 in AVAILSTATUS. I add my dialplan and the output to the cli. THanks. In the example i'm dialing from extension SIP/112 My DialPlan Secction: [macro-callonlyiffree] exten => s,1,ChanIsAvail(${ARG1}|s) exten => s,n,NoOp(${AVAILCHAN}) exten

Hello I'm trying to use ChanIsAvail() to check when the landline is back to idle after a call, but for some reason, Asterisk doesn't detect that the callee has hung up after listening to MoH for a few seconds: ========== extensions.conf ;Play MoH for a few seconds, hang up, and ;check ChanIsAvail() able to detect when line idle again exten => 8888,1,Answer() exten =>

Hi all, For those of you people that use Agents (with Agentlogin, not AgentCallbackLogin) on a call center, I have this need: when the agent logs in, a channel keeps running all the time that the agent is logged in to receive the incoming calls. How do I know which agent logged in (code)? Right now, if I query the login channel, there is no information about which agent is logged on: #

Since 1.4 ALERT_INFO variable has been deprecated. I used to send this via AMI: Action: Originate Channel: Sip/1234 Application: AgentLogin Data: 1234 Variable: _ALERT_INFO=info=alert-autoanswer Callerid: AutoLogin[1234] In order to send an autologin and autoanswer call to the agent 1234 on an Aastra phone at extension 1234. (just for example). Now in * 1.4 with ALERT_INFO deprecated I

pstn incoming on a TDM400P, sometimes i* won't answer, going into a loop like this: -- Starting simple switch on 'DAHDI/4-1' [Jan 16 10:38:55] NOTICE[5808]: chan_dahdi.c:7130 ss_thread: Got event 18 (Ring Begin)... [Jan 16 10:38:57] NOTICE[5808]: chan_dahdi.c:7130 ss_thread: Got event 2 (Ring/Answered)... [Jan 16 10:38:57] NOTICE[5808]: chan_dahdi.c:7299 ss_thread: MWI: Channel 4

Hi all, the .Last.value sometimes contains a strange $visible FALSE value. This poses problems when using R with ESS (and ess-developer-mode) from within org-mode. >From http://permalink.gmane.org/gmane.emacs.ess.general/7299: --8<---------------cut here---------------start------------->8--- Here is how to reproduce. Put df <- data.frame(a=1:3, b=1:3) in test.R and then:

I have been testing * with some Cisco 7912G's, in hope to trash our Nortel system. One feature our Nortel system has that I will need to fiqure out on the * system is paging. Is it possible to page a group of phones (all phones) with announcements? We are a k-12 school and we use our current phone system to make announcements on the phones monitor speaker. Any direction I can be pointed in

Hello all. This is my very first message to the list. I have a TDM400P card, It has 2 FXO channels which are connected to extensions of my PBX (Ericsson BP250), so I can dial from any SIP softphone directly to physical (analog and digital) extensions on my company. My PBX is configured so when I dial 8 on any extension, it will redirect to the first free FXO channel on my TDM400P card.

Does anyone know of a way to make an originate request coming over the management interface (e.g. AstTapi click-to-dial) include the relevant Alert-Info SIP headers to enable the originating phone to auto-answer? I've tried setting up a custom context (see below), but the dial plan is only entered AFTER the originating call is answered, so the SIP header is added to the terminating call leg,

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Is there a way to differentiate between a SIP address which hasn't registered (but is within sip.conf) and one that's not there at all (i.e. not in sip.conf) using a straight dialplan. I'd like to differentiate actions depending the state of a SIP device and whether it's in my config or not (if that makes sense, basic automap of dial-in lines to sip phones, but if they've

But where do can you get this later firmware from? I'm still on 1.0.0.78 on my 480i. Regards Lee -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Peter Passchier Sent: 05 August 2005 00:04 To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] AASTRA 480i Firmware 1.2.0.162 SIP ALERT_INFOproblems

It is not first application which i'm trying to run with wine, but first one failed to do so. If somebody can suggest what to do then please! Download available here: http://download.videohelp.com/DVDSubEdit/ Also, you need mfc42.dll and register it (or winetricks vcrun6). The problems after: Code: Beast:/# wine /home/user/.wine/drive_c/Program\ Files/DVDSubEdit/DVDSubEdit.exe wine:

Hi everyone! I'm quite a newbie at this Asterisk stuff so please bear with me. We've recently decided to start training in Asterisk via AsteriskNow! Asterisk version is 1.4.18.1 through AsteriskNow! 1.02 The box we have is paired with a Digium TE110P and we've managed to get it to the point where incoming calls via a single DID (from NTT Japan) can be received and answered

Hi there, bristuff comes with these two applications - and too little info to understand what they are for. Anyone has a clue and is willing to share it? Thanks, Philipp -= Info about application 'Autoanswer' =- [Synopsis]: Autoanswer a call [Description]: Autoanswer(exten):Used to autoanswer a call for an extension. -= Info about application 'AutoanswerLogin' =-

Hello everybody, i have two gsm line (extra channels) and i'd like to schedule the outgoing calls with a round-robin strategy. If all the gsm lines are busy, the call must be sent to the pri lines with a linear strategy. here is the dialplan: exten => gsm,ChanIsAvail(EXTRA/r2&DAHDI/g1) same => n,GotoIf($["${AVAILORIGCHAN}" = ""]?unavail,1) same =>

A friend of mine is running Samba (samba-2.0.3-8) on a couple RedHat Linux 6.0 machines. He is sharing files bilaterally at work with a WinNT machine and at home with a Win9x machine. In both places, he is getting a large number of SMB error messages. I'm not sure whether these are from the smbfs (smbmount) or from the Windoze machines accessing the Linux drives, though I suspect the

Is there a linux sip or iax phone that will autoanswer and connect to the console or soundcard? I found linphonec but it does not autoanswer from what I can tell. Jerry -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050916/8bc76bbb/attachment.htm

My colleague left our company, then I have to manage all our phones lines and asterisk: please, apologize me because I'm 'absolute beginner' about voip/asterisk!! Well... all seems work fine; we have some queues and some agents; the "music on hold" works fine when the agent press the hold button on the phone (thomson); the agents have the 'autoanwser' flag