similar to: Asterisk "answering machine" replacement, "WaitForRing()", application return values

Hi all, I have a client that is having problems with phantom calls. I have not been able to see it happen myself, but they say when it happens, the display on the phone (polycom 301's) says "Device" is calling, but when they answer the phone, there is only silence and then they hang back up and it sometimes rings again. I've been trying to track this down for a couple of

I''m getting my zones confused. Help. I need to have a bunch of systems using OpenVPN to gain an IP in the virtual subnet 10.100.1.0/24, on interface tun0. I will then route whole subnets to those IPs, like 10.100.2.0/24 via 10.100.1.12, etc. I want to have a policy for: - all hosts behind tun0 - all hosts in 10.100.1.0/24 - individual subnets being routed through IPs in

Does anyone have any recommendations on implementing Answering Machine detection for call generation programs? What I would like is * to determine what picks up the other line (Answering Machine, Voicemail, or Human) to determine which action to take. For example: If * detects Answering Machine or Voicemail, hangup call & the AGI will log (ANSWERING MACHINE DETECTED) and at that point,

I have connected a PSTN line to a Digium FXO card. There is also an ordinary analogue phone attached to the same line. The Asterisk answers the line on the first ring. I would like it to wait for a few seconds so that someone can answer the PSTN line with an analogue phone. This would allow a person to directly pick up the line if they wanted to or if not, let it go to the Asterisk where it

I'm trying to fit a multinominal logistic model using package mlogit. I have 15 independent variables. The code looks like this: m<-mlogit(score~0|f1+f2+f3+f4+f5+f6+f7+f8+f9+f10+f11+f12+f13+f14+f15, data, reflevel="1") And it gives the following error message: Error in parse(text = x) : unexpected ')' in "score ~ 0 + alt:(f1 + f2 + f3 + f4 + f5 + f6 + f7 + f8 + f9

is there an updated wiki or "how to", for installing install LOTRO either via Wine or Crossover?

I'm having a major problem at one of my branch offices with "Phantom Rings" on their asterisk-based phone system. The system was originally built using 2 X100P cards and was recently upgraded to a Rhino R4FXO-EC card. The upgrade severely increased the frequency of the phantom rings. I've read everything I can find on-line about automatic testing and noise on the line and

I've set up a very small * system for a small local paper. The system works great. Here's the issue: I have one of their phone's plugged into the phone port on the x100p and if the phone ring more than 2x then asterisk kicks in and doesn't recognize it as being picked up and starts playing the menu. Can i use wait or something to let the phone ring more and not start the menu?

I am getting the following information and asterisk 1.0.7 is not answering the call. Any ideas? jerry ------------------ Sip read: INVITE sip:2828;phone-context=cdp.udp@qg.com:5060;maddr=161.49.198.102;transport=udp;user=phone;x-nt-redirect=redirect-server SIP/2.0 From: <sip:3173241052;phone-context=+1@qg.com;user=phone>;tag=c22da8c0-13c4-429efa71-657b8da-2ed To:

Hello, I'm trying to investigate and solve some postgres latency spikes that I'm seeing as a result of some behaviour in the syncer. This is with FreeBSD 8.2 (with some local modifications and backports, r231160 in particular). The system has an LSI 9261-8i RAID controller (backed by mfi(4)) and the database and WALs are on separate volumes, a RAID 6 and a RAID 1 respectively. It has

Howdy I have a question regarding support for picking up when the telco sends a MWI message. My client's setup is a small office with three incoming lines on a TDM400P with iaxy's and a Grandstream as extensions. I am using CVS Head from yesterday. (I was resolving a different issue.) Since they only have two voice lines, with the third as a fax, I am using voicemail from the telco.

I am using the following macro to dial a ZAP channel. When I dial in, * answers and I go to voicemail. I never hear any ringing, though. It doesn't work with the Ringing command before or after the Dial command. [macro-zapdial] ; ; call a ZAP extension for ${ARG2} seconds, and then voice mail ; ${ARG1} - Extension ; ${ARG2} - Time to ring exten => s,1,Dial(ZAP/${ARG1},${ARG2}) exten

Hey Guys! I have perl script for allpage which is working fine with asterisk 1.8.2.3 version but same script same dialplan wouldn't working on asterisk-1.8.3.2 is there anything changes ? If i run this script from command like it works but not from asterisk dialplan. This script nothing but just connecting AMI interface and using Variable: SIPADDHEADER=Alert-Info: Ring Answer variable to

Hello. I have been beating my head over this problem for about 6 hours now. I have a SIP peer, who I register to (successfully), who should be directing all incoming calls at my [default] stanza in my extensions.conf: [ Context 'default' created by 'pbx_config' ] 's' => 1. Wait(1) [pbx_config] 2.

Godd evening! What happens if someone calls and asterisk doesn't "Answer()" itself, but another analog phone does? Can I somehow catch this situation in my dialplan. I have an ISDN line, but with it I got a box with an adapter for good old analog phones. This doesn't seems to be directly connected to the ISDN line asterisk sees. But somehow, it must know, that the call

Hi, I'm trying to write a simple perl script that will run the following: Action: Originate Channel: local/xxx@callback/r/n Exten: 1234 Context: callback Priority: 1 Extensions.conf exten => 500,1,agi,callback.pl callback perl script: use Net::Telnet (); $mgrUSERNAME='fred'; $mgrSECRET='bloggs'; $server_ip='127.0.0.1'; $tn->print("Action:

Dear all i have a TE205P connected to an Asterisk 1.2.18. Yes i know, the version is old but since now the system was stable and i don't have the necessity of an upgrade. The system provide an IVR service that: 1) receive the call 2) verify the queue length 3) hangup if queue length is > 1 4) put the call in the queue othervise Then, there is an AGI php script that 1) verify the queue

I am using Asterisk and an X101P card as a glorified answering machine. We have a residential PSTN line with about six phones connected to it. Like an answering machine, I want Asterisk answer the line *only* when an incoming call is not answered after four rings. This mostly works. My extensions.conf is at the end of this message. The problem is that Asterisk will sometimes answer the line when

I've been seeing some random Proliant DL380 G4 64bit crashes. Each time, on the console are messages relating tojbd2/cciss and something about a waitfor 120 seconds. Is anybody else seeing anything like this? Oddly, I can't seem to find this in the logs. I guess it can't write when this happens. -- John Hinton 877-777-1407 ext 502 http://www.ew3d.com Comprehensive Online

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