similar to: Blind transfer on Queue-CDR

I'm using FreePBX 2.5.2.2 with Asterisk 1.6.1.4. If I make a call and then decide to blind transfer them using ## my side of the call is not hung up. Instead it sends me to voicemail. If somebody calls me and then I blind transfer them with ## I am hung up on as expected. I called from 8678 to 28688. I then transferred the call to 8532. Asterisk acts like it wants to hang up, but then

Hi, I'm trying to edit an AMP-derived dialplan: the macro "dial" uses the AGI script "dialparties.agi" to find the extension to call. I'd like to drop this script: does anyone can explain me what is its main job? Thanks -- Domenico Viggiani

Hi, I use: Trixbox-2.2.4 FreePBX-2.3.1.0 Asterisk-1.2.17 BRIstuffed-0.3.0-PRE-1y-e Zaptel-1.2.19 ..with two ISDN cards, often but occasionally the dial out failed but is possible to receive external call. My zapata.conf conf is: [trunkgroups] [channels] language=it context=from-pstn signalling=bri_cpe_ptmp rxwink=300 pridialplan=unknown prilocaldialplan=local switchtype=euroisdn

I have been shaking down a dialplan for SIP fax to efax. The basic senario is an ATA on the same subnet as the Asterisk 1.2 box (avoid RTP packet lose and thus fax crash), calling a 'fax extension' and envoking rxfax then email. I leverage off of context: from-internal-additional-custom, so as not to have it overwritten by FreePBX. In extension_custom.conf I have:

Hi, I can't get Asterisk to save CDRs for calls transferred via SIP blind transfer. My extensions.conf: [globals] __TRANSFER_CONTEXT = transfer [common] exten => 123,1,Playback(demo-congrats) exten => 123,n,Hangup() exten => _0X.,1,Dial(SIP/${EXTEN}@PSTN-GW,60) exten => _0X.,n,Hangup() exten => i,1,Hangup() exten => h,1,Hangup() exten => t,1,Hangup() [transfer] exten

Hello, Sorry for a bit of a newbie post but we all had to start somewhere right .. I'm wondering if someone can briefly explain the difference between blind and attended transfers and why they would generate two very different CDR entries.? From my own research, it seems that transfers are both ultimately a SIP REFER and thus seeing two different CDR entries just confuses me further.

I am using Asterisk 11.17.1 with my program that uses AMI Originate calls to generate a bunch of calls for a callcenter. The PBX configuration is handled by FreePBX 2.11. I want to understand the dialplan behavior in order to figure out why the CDR(recordingfile) is blank on the CDR records despite the dialplan setting it. My program generates the calls by setting Channel=Local/NUMBERTODIAL at

Hello, I have a TDM400 and currently have 2 of the ZAP Trunks configured on it. Zap/1-1 and Zap/2-1. I am Running Asterisk@home Version 2.4 with AMP version 1.10.010 In my Outbound Routing I have the Trunk Sequence set up so that 0 is Zap/1-1 and 1 is ZAP/2-1 What I see is that when Trunk Sequence 0 is full, it does not open Trunk Sequence 1. I have found that this is true even if I

Hi i have installed AAH 2.6 and configured some extensions the calls are working fine. but if i dont answer a call then it says " the person at extension " and hangs up . it doesnt spell out the extesion number nor it goes to voice mail box. *************************** Asterisk CLI log **************************** dialparties.agi: Extension 200 is available...skipping checks --

Hi folks You are welcome to try our (Xorcom)'s latest debs (for Xorcom Rapid, or Debian Sarge in general) "Unstable": Asterisk and AMPortal: The repository is available at: deb http://rapid.dotsrc.org/ unstable/ #deb-src http://rapid.dotsrc.org/ unstable/ The commands you are looking for are: apt-get update apt-get dist-upgrade apt-get install amportal

change asterisk.conf: mkdir /var/run/asterisk chown it to your asterisk user. change astrundir => /var/run to astrundir => /var/run/asterisk My guess would be that you are running asterisk as a non-root user and that this user can not write to /var/run . if so, the ctl and PID files are not created. -- -- Steven http://www.glimasoutheast.org "Tom Castleman"

Hello, I have setup a test server with asterisk/AMP and have several 7960's connected to it. The asterisk server has a public ip and all the 7960's are behind nat'd routers. When I try to call from extension to extension I get directed straight to voicemail. I do not have any cards installed and instead direct everything to an Ondo server. I have been told it's not an AMP

Ok, I've been trying to figure out why my A@H won't answer the lines when I can call out and the panel shows the call coming in - well something bizarre has happened. I set up inbound routing to ring my extension if a call comes in - and my extension rings but when I pick it up I get a dial tone. The whole time after I answer I hear the phone I originated the call on just ring and ring

Any body have an idea why my first attempt to use "observe_field" isn''t working? <%= start_form_tag(:name => ''my_form'', :controller => ''boo'' , :action => ''ya'') %> <%= radio_button_tag(''foo'', ''true'', checked = true) %> <%= radio_button_tag(''foo'',

Previously, by mistake I added request in wish list for slide like MSN. Actully it was Blind like msn. However it was my mistake so I tried to created script by my slef - and able to do it. But I am new to this, so my script is not in good condition. Effect.Scale2 = Class.create(); Object.extend(Object.extend(Effect.Scale2.prototype, Effect.Base.prototype), { initialize: function(element,

Greetings, I am trying to add a row to a table using the blind down effect. I have this working for lists (aka <li>) but not tables. With tables it seems to display the newly inserted row then blind. I basically want to start with the row hidden and then blind down. For example this works with lists: <ol style="" id="list1"> <li

I wrote a simple toggle function similar to Element.toggle that will toggle a section between being up and down using the Blind effect. I''m relatively new to the code, so I''m not sure if it''s implemented 100% correctly or if it can be done better. I am using it though and it seems to be working fine. Anyway, here''s the code and if you think it''s

HI callee side blind transfer is failed in 1.8 but caller side blind transfer is succes,Transfer doing by refer method,please help me on this Nikhil

Hi, I have a SIP phone which does not natively support SIP transfers (REFER etc...). So far all that is possible is to enable blind transfers using the t and T arguments in Dial from the # DTMF key. The phone has an R button on it and this can be setup to either send an RFC2833 hook flash message (value 16) or a SIP INFO message which you can edit the contents of (since there seems to be

Hi, I setup an asterisk system (version 1.8.0-rc5). While using a SIP only environment I discovered a problem using blind transfer. The phones are SNOM or Aastra and are using the SIP REFER Method. The following is working: User A calls user B, B accepts the call, user A than transfers to user C The following is NOT working: User A calls user B, B accepts the call, user B than transfers to user