similar to: How can I log call forwards?

I'm simply trying to forward calls to users who have the call forwarding feature enabled (FWD Status and FWD Ph Number kept in the astDB). The problem is that I want users to be able to forward calls to numbers that they would normally be allowed to dial within their own context. (I don't want a local call only person forwarding to a long dist number, for example.) I'm able to

I can't seem to retrieve a parked call. Here is what I do: Call from phoneA to phoneB Answer phoneB Press Flash/Callwait on phoneB Press 700 to park the call A voice says that the call is parked at 701 When I try to dial 701, I don't get connected to the parked call Below is the asterisk output when I tried to park

I just thought i'd post a small sample that uses the new chan_local to show one way of doing variable callfwding This sample extension.conf uses's the ast DB to store a users current extension, in a db family of CallFWD and the unique Key is based on the current channel the user is assigned. In the globals var section each key is hardcoded EXT1, EXT2 this is used in the [incoming] context

Hi, I am fairly new at working with Asterisk. I am having a call quality issue that I really need to get ironed out before we go to rollout the system in a week. Any help would be greatly appreciated!!! Even if it is just pointing me in the right direction. My current setup: I have Asterisk Setup on a Dell Server. It has 2 T100P cards. One will be for out T1 PRI from the Phone Company (We

>On Sat, 29 Jan 2005 12:53:11 -0600 > -----Original Message----- >From: <asterisk@draughon.org> >Subject: [Asterisk-Users] RE: Q: Can I over-ride the value of > ${CALLERIDNAME} ? >To: <asterisk-users@lists.digium.com> >Message-ID: <001a01c50633$d9e10a30$6701a8c0@calhoun> >Content-Type: text/plain; charset="us-ascii" > >Folks, > > Many

Hi all, We keep getting these and all the calls between these two asterisk boxes get dropped. what is going on here, I have been trying to solve this problem on my own but maybe I don't have the trunk setup right. also I have posed the output of my full log of the machine with the zap interface, the other is using ztdummy. IAX.conf on machine 1: [general] port=5036 ;iaxcompat=yes

<div><FONT size=2>Hello,</FONT></div> <div><FONT size=2>two questions: </FONT></div> <div><FONT size=2></FONT>&nbsp;</div> <div><STRONG><FONT size=2>1: How can I open/enable network connection to B?</FONT></STRONG></div> <div><FONT

Greetings folks; I guess I must be missing something, because for the life of me I can't seem to make this work. I have remote clients connecting to Asterisk using IAX2, these clients have changing IPs so we're using the useful register tool. The client can call out successfully, that's not an issue at all. Calling coming from the server to the client, however, do not appear to go

I have a issue with two Zap clone cards where they used to work. I am using Asterisk@Home 2.5 which includes Asterisk 1.24 and Zaptel drivers 1.2.4. The system is a new Intel Celerion machine. I used to have the same cards running in a Intel PIII system. in this system, they worked. In this older system, I was able to call into the machine and call out from it. Now that I have upgraded

I am using IAX to send a call to my cell phone. I want to be able to hit # and transfer it back into the office. I have added tTr to the dial command and hitting # prompts me for the transfer, but after I start dialing 103, it stops at 1 and tries to transfer it within nufone instead of my dialplan. This is the debug output: -- Called me@NuFone/1515480XXXX -- Call accepted by

Hi, A most peculiar and vexing problem for you all. I hope I have been verbose enough without being a firehose ;) The set up: I have a channel bank, using the r1t1 rhino driver with a rhino T1 card (the channel bank itself is a very legacy piece of equipment)- this supplies FXS for all the house phones. Also, a Wildcard TDM400P, using the wctdm module with 1 FXO module, this supplies FXO to the

Hi, I'm using the macro below in extensions.conf for most of my outbound calls. One issue with my current configuration is that when I make an outbound call it doesn't properly detect that my PSTN line (Zap/1) is busy with another call and then overflow to my outbound IAX connections. I think the root cause is that DIALSTATUS gets reported as BUSY. The debug output is below. My desired

Hello, I've got very annoying behaviour from our asterisk PBX. We have 12 channels T1 e&m wink start for TDM and using iax softphones internally (iaxcomm, but tried firefly-thirdparty and discarded for bad sound quality). Slackware 9.1 w/ kernel 2.4.26+ digium TE110P card. In some cases when call is placed from softphone to TDM, system does not detect call answered on Zap channel and

I'm having some trouble with placing some iax calls over an openvpn: Setup A is a 1.8GHz Celeron, T100P attached to a Zhone Zplex. Setup B is a 266MHz P2, T100P attached to a Zhone Zplex. Setup C is a 700MHz P3, T100P attached to an Adtran TA 750. Setup D is a 233MHz Pentium, with an X100P. Setups A and B are on the same physical network. IAX calls routed between them work fine. Setup D is

Don't know enough to properly term the problem I'm seeing... sorry if subject appears vague. And I have other questions too, but "Newbie Help Wanted" isn't exactly more specific... ;-) My setup, intended for testing and all, "*" version 1.6.0.6, dahdi with an X100p clone. Regular phone line provides PSTN access with one port (and my DSL). Calls come in and are

Hi, I am setting up a small call center using *. I have ZAP setup for incoming calls and IAX setup for agents. Agents login using AgentCallbackLogin. When customers call, it's getting picked up and when queue is trying to call back the agents, I am getting error. I am using CVS HEAD, and updated just now. The error is: -- Executing Answer("Zap/1-1", "") in new

Hello - I recently offloaded some of the SIP traffic on to a seperate Asterisk box and interconnected our main Asterisk system with the new system via IAX. The SIP clients are running 7960's. When a call is put on hold, often times when the call is pulled off hold, there seems to be no RTP in at least one direction. There seems to only be voice in one direction. Basically the call comes

Is it just my box, or is there something flaky in the implementation of "show manager commands"? Note: I'm using putty. About half way through this, I toggled my KVM over to the desktop and logged in to try and recreate it. The output was the same as the last two entries in this dump. bebop*CLI> show manager commands bebop*CLPing Ping bebop*CLLogoff Logoff Manager

Hello, I'm trying to set up call recording for a queue. Right now the recording appears to work correctly, but when I call and chatter for a minute or so, at the end of the call I end up with a very small file (less than 100 bytes), which contains about .06 seconds of silence. If I talk for another minute, this file will get up to 200 bytes or so. In my queue configuration, I have: [testq]

My boss is complaining that the delay between speaking and hearing in a meetme conference is noticeable and doesn't want to roll out our system until I can eliminate the delay. Personally, I don't think the delay is significant, but I don't sign his check. The system consist of 3 1u's, each with a single quad t1 card. Each card has 2 t1's running NFAS. The "t1