similar to: Blind Transfer Won't Hangup

Hello List, I am having a strange issue with a trixbox system we installed for a client, and I would appreciate any help on this one. The issue is that occasionally when they go to answer an inbound call from the Sangoma A200 - there is no one there, and they are presented with dial tone. The calling party is hung up. A bit of background: The client actually has two systems install (one at

I am trying to record calls on demand both inbound and outbound calls. I can record outbound calls just fine but not inbound calls or calls from an internally between extensions. I am using the latest asterisk 1.8.x certified version. On an outbound call I see: == Using SIP RTP CoS mark 5 -- Called SIP/ BVTrunk /7190000000 -- SIP/BVTrunk-00000163 is making progress passing it to

[also posted on Trixbox trunk forum] I am also working with Sangoma directly to debug this, but so far no real luck. TrixBox 2.6.1, A102d card with V33 firmware (latest) and WANPIPE 3.2.6 (3.2.7 is out, but nothing has changed that would affect this problem). The system gets about 200 calls inbound on the trunk, which is not very heavily used, and of those calls one or two a day is randomly

Whenever I try to drag calls to the Parking Lot or On Hold, FOP would drop my calls. I have searched online and have found similar problem, such as the link below. I have tried their solution but still the FOP is not working correctly. I even installed the HUDLite server and is getting the same results. www.freepbx.org/forum/freepbx/users/flas...ot-transfering-calls Here is the log when I tried

I am having a problem with the Call Queue Feature. If I let a user into the Queue prior to an agent being available for them to take the call, they experience the following: 1) they hear that they are the first in line 2) when the agent finally logs in (the caller on hold in the queue is sent to the users phone) 3) the AGENT is still in the login phase hearing that they are "successfully

Hello all, I have been posting some questions about this problems that I cannot yet solve, but I think I have a better diagostic, so maybe someone can give me a clue why it is happenning. I have Asterisk + AMP configured as a PBX with a Customer Center Queue with 4 agents that login/logout dinamically. If there are no agents, queue timesout and gets derived to another queue that somebody

Hello all, I've installed AAH 1.1 on my VIA C3 powered mini PC. I've made the necessary changes to the * makefile, so the compilation went well. The first thing I did was configuring two extensions from AMP, namely 200 and 201. Then I installed X-lite on two PC's and configured them with one of the extensions: System settings - SIP proxy - Default: Username: 200 Authorisation user:

There is an environment Setup uses Asterisk 1.2 and doesn't want to upgrade. There is an issue while a call goes to any queue we create, the call is being disconnected after 20 seconds and it is hangup. The following is the configuration: - vi /etc/asterisk/queues_additional.conf [8] wrapuptime=0 timeout=30 strategy=ringall servicelevel=5 retry=4 reportholdtime=No queue-youarenext=

I have a Asterisk server (PBX in a Flash) with Aastra 57i phones. When there is an incoming call the phone will display two buttons "answer" and "ignore". If you press "ignore" the call is dropped instead of sent to voice mail. The following is the log: -- Called 111 -- SIP/111-00001c14 is ringing -- Got SIP response 486 "Busy Here" back from

Here is the output. These are Panasonic KX-TG2564's. Does something need to be set for the phones? I can call out fine, but all of the extensions seem to be busy. Starting simple switch on 'Zap/5-1' -- Executing Macro("Zap/5-1", "exten-vm|200@default|200") in new stack -- Executing SetVar("Zap/5-1", "FROMCONTEXT=exten-vm") in new stack

Hey all! I'm trying to figure out how to have a queue accept an inbound caller's key press to action on. At first I'm just trying to implement a "Press 1 to leave a voice mail" announced and at any time in the queue, the user can press 1 and go to the queue's voicemail. Later I'd like to have it accept "Press 1 if this is an x issue, press 2 if this a y

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

Hi, Hi All, I've just delved into the world of asterisk and I'm having a few dtmf issues. Internally, amongst sip phones, dtmf is fine. Externally, if you ring from a GSM mobile, DTMF is fine, however if you ring from a standard phone, DTMF fails to register. I am attempting to use a quad port HFC-4S Beronet Card. I've been searching the web most of the last week and

Arch = x86_64 OS = CentOS-6.4 (freepbx) Asterisk = 11.4.0 FreePBX = 2.11.0.2 We use Snom870 handsets with firmware v.8.7.3.19. I am trying to develop a custom dial plan to invoke a distinctive ring-tone when an external call is transferred internally. Based on an earlier solution I discovered I am attempting this: [from-internal] include => set-alert-if-local [from-internal-original]

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

Hi all, as in last mail, i've installed Asterisk from CVS and AMP to manage it. I've made 4 extensions: moloch*CLI> sip show peers Name/username Host Dyn Nat ACL Mask Port Status 204/204 (Unspecified) D 255.255.255.255 0 UNKNOWN 203/203 192.167.125.9 D 255.255.255.255 5062 OK (3 ms) 202/202

Hi all, i've installed Asterisk with AMP. I've created 4 extensions (for 4 SIP phones) and this is what my "sip show users" return: moloch*CLI> sip show users Username Secret Accountcode Def.Context ACL NAT 204 moira from-internal No No 203 michele from-internal No

Hi Everyone, I have an asterisk box in my office. It does not display the correct Incomming Caller id. For incomming we are using ISDN Bri line which is terminated in a Digium 4 port bri card (B410P). Like if a number say 02 12345678 calls to our line asterisk displays it 12 12345678. Similarlay if a mobile number say 0416 123456 dials us , asterisk displays 1416 123456. I am not sure where the

Hi ALL Have installed asterisk@home 1.0 On FWD DID's, appears that 2 calls are generated to the inbound extention. I have confirmed this on a number of friends boxes also. Does anyone have a fix for this ? I set the DID simply to a custom context and it did the same... Anyone have a way to fix this ? Here is the output...... -- Accepting AUTHENTICATED call from 65.39.205.121, requested

Hi, guys, For my server, if i use my handphone to call in the PSTN line by TDM400p card, the server could not receive the caller id correctly. anyone knows the problem? I am currently using asterisk 1.2.14 with freepbx 2.2.1. The CLI is as below, "Caller ID name is 'zap1' number is '4521'" , this 4521 is one of my FXS zap extension created. dialparties.agi: Starting New