similar to: Hearing "transfer" during call

Dear The Expert, I am very new with this, I have installed AsteriskNow, X-Lite as my SoftPhone, I am using SPA-3102. I have 3 extensions, me at 250, 998 is my Linksys SPA-3102 and 999 for PSTN Line (see below) My problem is, I am unable to call 998, I thought this is registration problem, (because the Linksys screen info said Registration Failed) Could any body please help? Many thanks in

Hello, everyone. Today I've installed an asterisk svn trunk (r61667). The problem I'm having is no matter what context I set in the config file for that peer, "default" is always being used. The output of "sip show peers" shows the context correctly, but when I try to make a call, using that peer, I can only dial the numbers set in the "default" context.

Hi all. I am trying following scenerio for call park & pickup. voice is flowing established between B & C, after call-pickup ( instead of A & B ). can anyone please clarify why it is happening like this, ( or ) do i need some more configuration for park&pickup ? A B

Hi, I am attempting to setup an asterisk server, current specs: CentOS release 5 (Final) Asterisk 1.4.11 Asterisk-gui checked out from SVN last week I started with a fairly basic setup involving one VOIP provider who provided one dial in number, and a couple of handsets. Config files are below. It was pretty much totally built by Asterisk-gui, except for the fact I had to add

Greetings all, I'm trying to connect to an AT&T teleconference, but the call is never marked as ANSWERED by asterisk and therefore won't bridge and continue. The only work-around I've come up with so far is to dial like this: Exten => 744,1,Dial(Zap/g1,,p) The "private" mode keeps the line open without trying to do a bridge, but requires the

Update on this: I found that by changing insecure = very to insecure = invite, adding the second trunk no longer stopped calls working. I've read the documentation on this switch and still don't see how it applies/is meant to get used. Anyway, with this change in place, the following may help: asterisk*CLI> sip show registry Host Username

Hi, I am trying to setup a simple home voip service w/ * I have compiled and installed the svn source as a first step I am trying to configure SIP for inside my network. I have a handful of softphones and a few hardphones that I want to all be able to call each other I have configured users.conf with a single softphone(kphone) and have tried calling itself (ext 6000) and the demo from the

Hi, I would like to use the new atxfer (native assisted transfer, see mantis item #3241) , but I've partially been able to make it work. I can receive a call and then having the caller hear MOH while talking with another extension (the one I want to transfer to), but then I can't make the caller and the trasferred talk hanging up or pressing any key combination I'm aware of. My

Hello everyone, I have previously asked this question on the asterisk-video list, but I got directed here. I have a setup consisting of asterisknow beta4 (not sure if that is crucial) and a few clients all running X-Lite 3.0 (not eyebeam) on the local network. My computer has a USB-Camera installed, and now I would like to do some video calling with it, at least, so that the other user can

Hi, I'm having trouble capturing calls using the chan_dahdi with dynamic span. Here my settings: chan_dahdi.conf [trunkgroups] [channels] context=default switchtype=national facilityenable=yes rxwink=300 ; Atlas seems to use long (250ms) winks ; where the ring cadence is changed *after* the callerid spill. usecallerid=yes hidecallerid=no

Hi, I recently moved all of my conf files over to a new Asterisk 1.2.4 server and every works except the features enabled in features.conf. Was there a syntax chnage in 1.2.4? Or is there something else... Here is my features.conf: ******************** [general] parkext => 880 ; What ext. to dial to park parkpos => 881-890 ; What extensions to park calls on context

Hi, I've just built my first asterisk server. Current information: OS Version: Linux asterisk.visinet.com.au 2.6.18-8.1.8.el5 #1 SMP Tue Jul 10 06:50:22 EDT 2007 i686 i686 i386 GNU/Linux Asterisk Build: Asterisk 1.4.11 Asterisk GUI-version Revision: 1479 $ Server Date & TimeZone: Thu Sep 6 02:37:11 EST 2007 I've used the Asterisk GUI for setup with two IP

Hi - I'm trying to implement the applicationmap stuff in features.conf, and I can't seem to get it to work. I'm testing it out on 1.2.2 with Polycom IP500s and Snom190s. My features.conf looks like this: [general] parkext => 700 parkpos => 701-720 context => parkedcalls parkingtime => 240 transferdigittimeout => 2 ;courtesytone = beep

Hi, I try to set up attended transfer in my Asterisk Box . My features.conf look like this: [general] parkext => 100 parkpos => 1-5 context => parkedcalls parkingtime => 100 transferdigittimeout => 3l courtesytone = beep xfersound = beep xferfailsound = invalid featuredigittimeout = 500 ;adsipark = yes pickupexten = *8 [featuremap] atxfer => *2 blindxfer => # disconnect

I have put canreinvite=no for all my internal SIP-clients in sip.conf because I want Asterisk to be in the middle of the RTP-stream so he can provide MusiconHold and so... Now, what the Asterisk CLI tells me when I make a call from my one internal SIP-phone to another internal SIP-phone is : Verbosity is at least 25 == Spawn extension (intern, 51, 1) exited non-zero on

In features.conf, I have uncommented the transfer features under feature map, but I still cannot transfer using a POTS phone on an IAXy adapter. I think I am missing something here.... Any help is appreciated. Here is features.conf: ; ; Sample Parking configuration ; [general] parkext => 700 ; What extension to dial to park parkpos => 701-720 ;

I need attended transfers, but I do not have time to talk to another extension and see if they accept the transfer, my features.conf is: [general] parkext => 700 ; What ext. to dial to park parkpos => 701-720 ; What extensions to park calls on context => parkedcalls ; Which context parked calls are in parkingtime => 220 ; Number of

i am configured asterisk-gui the "Queue Extension Configuration" but configure and register into queue.conf : [66666] fullname = Call Center strategy = ringall timeout = 5 wrapuptime = 5 autofill = yes autopause = no maxlen = 0 joinempty = no leavewhenempty = no reportholdtime = yes musicclass = default member => Agent/60010 member => Agent/60011 member => Agent/60014 but not

I recently upgraded from asterisk 1.2.13 to 1.4.2 and am looking at using the users.conf file to setup my users, before i was using real time SIP which worked fine. However when i create a user in users.conf i am unable to register the user form a softphone, however that same softphone can still register a different the users i currently have setup form the sip.conf from real time. i've

Connected to Asterisk 1.4.11 currently running on asterisk (pid =3D = 31999) -- Remote UNIX connection Verbosity is at least 8 -- Executing [00425298582 at numberplan-custom-1:1] Macro("SIP/8001-b7d0bb20", "trunkdial|SIP/trunk_3/0425298582") in new stack -- Executing [s at macro-trunkdial:1] Dial("SIP/8001-b7d0bb20", "SIP/trunk_3/0425298582")