similar to: Perl-AGI process

On 2007-05-15 Tony Mountifield wrote (wrt using AGI scripts to dial out): > Can't comment on this one, as I never use AGI to dial. > My AGIs just set the context, extension and priority, > and exit to the dialplan to do any dialling. (http://article.gmane.org/gmane.comp.telephony.pbx.asterisk.user/185537) I would like to do this, but I am having trouble figuring out how. I have

I am trying to install wine (1.3.24) from winehq for fedora and sketchup (8) on a Fedora-15 box. I have tried to follow the Sketchup Sage site http://sites.google.com/site/sketchupsage/problems/linux After installing Wine and Sketchup, I have ?Wine Files? under the new GNOME-3 Applications and when I open it I get what looks like a Windows window. No Sketchup in there. I have 2 Sketchup icons

Looking at the code, it would appear that the 'callstart' column of the cdrs table should really be called 'callend': $dialstr = "IAX2/$res->{path}/$phone|30|HL(" . ($maxtime * 60 * 1000) . ":60000:30000)"; $res = $AGI->exec("DIAL $dialstr"); $answeredtime =

Hi everybody, I sent an e-mail this morning regarding SIP / IAX2 with no ring-back, I now succeeded to pin-point the problem, here it is, if I dial a provider directly from extensions.conf I get ring-back, if I dial from an AGI script I don't get the ring-back but it calls anyway. I use 1.0.9. Any hint would be appreciated ! Thanks, Frederic ;Calling this one does not give me ring back

Hi, I have configured asterisk now with ENUM lookups which are working really perfect. Now I want to play a small soundfile before dial the number to inform the caller which protocl is used (SIP, IAX2 or ISDN). How can I do this? With Playback it doesn't seems to work: [iax2-sipport-out] ; with leading 3 using IAX-sipport exten => s,1,NoOp(Dialing ${DIALSTR} with iax2-sipport-out) exten

Hi all, I have been looking at the code for both the zaptel driver (wctdm.c/wcfxs.c) and the asterisk channel driver (chan_zap.c) trying to figure out how much of this that has been implemented. So far I can see that the current stable 1.0.9.1 zaptel driver don't have the SETPOLARITY ioctl that would be required to properly signal the Swedish/Dutch CLIP, but the 1.2 beta1 has this

Hi, I am having a strange problem with enum and AGI. Here is what happens: I have in my agi something like that: foreach my $resolver ("e164.arpa", "e164.info", "e164.org") { my @enums = get_enums($phone, $resolver); foreach my $enum (@enums) { $dialstring = $enum .

Hi, We are trying to use attended transfer with Asterisk 1.2.5, but when we do the transfer and dial the new number, it times out after 3 rings and then the callee is put back to the original agent. Where can I adjust the timeout which applies to the number we are transferring to? I have changed the extension for this number to timeout at 60 seconds, but that seems to make no difference. --

Hi List, I need to modify the callerID name of the call coming back when a parked call returns to the extension that parked it when it times out. Looking at app_parkandannounce.c /* Now place the call to the extention */ snprintf(buf, sizeof(buf), "%d", lot); memset(&oh, 0, sizeof(oh)); oh.parent_channel = chan; oh.vars =

Today my system coredumped (4.8-STABLE from Saturday), I believe it's somehow X11 related: X11 crashed first (signal 11). I was running it as root (I know I shouldn't). I didn't think about it and restarted X11. While it was starting, I had a look at the console, there was a bright white message: issignal. This shows up at X11 startup. Then the system coredumped. Below is more

Hello all. I am not 100% familiar with the Linux advanced routing capbalieties yet, so I thought , after reading source code, documentation and more that I might be better off asking the experts on this list. I am not using this ina production environment, but at home, as a test case. First of all, here is the network topology. Internal LAN <--> Switch <--> [NAT/FIREWALL/INTERNAL LAN

Hi, I have 2 g729 licences - what I want to do is use g729 by default but if I get more than 2 calls at a time, use gsm for the others. So, I put this on all my sip providers: disallow=all allow=g729 allow=gsm However, this just seems to use gsm for everything. If I comment out the gsm line, it then uses g729. I thought it would use the codec's in the order they are allowed - is this

Hello Everyone, I've been playing with an agi script for paging sip phones. page.agi will take all available sip extensions and assign them to the global variable PAGE_GROUP. Allowing the phones to be paged from the dialplan with the new Page cmd. Extensions to be excluded are presented as arguments to the agi. Each time a page is made this agi refreshes the global variable. This works with

Hi all. I have enabled the attended transfer feature in features.conf. I'm using it and I want to resolve some questions, I hope someone can help me :) When I transfer a call to an extension: - The extension rings during 15 seconds and the call returns to the "transferer". Is there any possibility to recover the call before the timeout of 15 seconds expires? I mean, I would like

Hi All, I have been attempting to get an AGI LCRdialout script to work. Basically what I need to have happen is when someone dials out a number the script check to see if it is local if so, go out the ZAP channel. If the ZAP channel is busy, go out the IAX channels, if IAX is all busy, go out the SIP channels. Here is a sample of what I have in my script. #!/usr/bin/perl use strict; use

Hi My hardware PBX run asterisk on vxworks,Because the vxworks not support perl. Now I want to add a callback function to my pbx. now it can store Caller and Called party numbers in queue when Called party is busy Then I malloc a new ast_channel to call.It is should use ast_get_channel_by_exten_locked() or ast_channel_alloc() , my program as follow,But it isn't work, anyone know how to

Hi all! I need your help. My server doesn't respond any more. Everything crashed. How can I find out if this is a bug or it is simply overloaded? I don't have much running, just KDE with about 10 programs on each of the 16 desktops, and a few background processes. This seems much, but I often have much more stuff running, and it is not even slow. It does respond when I ping it, but

Good Day All, I am facing a serious problem since I started to use asterisk. I don?t know if it is a bug or some one already solved this. Currently I am running 120 VoIP SIP channels on my asterisk server but each day 2, 3 calls got hanged in asterisk, and on asterisk CLI ?show channels? showing us as call UP but in real there is no call. When asterisk restarted the hanged calls removed from

Hi, Sorry, not an experienced shorewall user, this is my first basic setup. This starts to drive me crazy. I wanted to use DNAT to forward port 33890 to an internal machine (windows) port 3389. To reach my workstation when I''m not home. In my rules : DNAT:debug net loc:192.168.0.11:3389 tcp 33890 - pub.lic.ip.add #SECTION BLACKLIST #well known port scans DROP net

Hi all, We are running more than 40 active calls on our Asterisk Box. But some time we are facing problem, call is not disconnecting for a long time more than 2 and 2 hrs. in this cuase our customers charged for 1,2 hrs. even they made very small calls. i have already set rtptimeout = 60, but not disconnecting Here is my extentions. [main-ext] exten => _x.,1,AGI(main-ext.pl) exten =>