similar to: [Zaptel] "numberplan-local" context from nowhere?

Hello The following, dead simple Bash script ran as AGI doesn't reply to Asterisk: ============= extensions.conf [from_fxo] exten => s,1,Wait(2) exten => s,n,Set(CID=${CALLERID(num)}) exten => s,n,AGI(/var/tmp/basic.agi) exten => s,n,Hangup() ============= /var/tmp/basic.agi #!/bin/bash #Ripped from #http://lists.digium.com/pipermail/asterisk-users/2003-July/008554.html while

Hi list, I enabled the transfer function in my dialplan, but I may configure it wrong because sometime when I call a SIP extension number from one FXS port, the SIP side will hear word "transfer", I hear nothing, after that, the call conversation is fine.I'v had this problem for a long time, could not get clue where I configure it wrong. here is my related config part: sip.conf:

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! i have an error using the meetme aplication, and just dont work.. my meetme.conf is: [rooms] conf = 700 i calling from a sip phone, the extension number is 600. there is the error: Executing [700@numberplan-custom-1:1] MeetMe("SIP/600-09111e58", "700|MI") in new stack WARNING[20055]: channel.c:3024 ast_request: No channel type registered for 'zap' WARNING[20055]:

Hello No matter what I try, Asterisk still fails dialing back through a callfile built through an AGI script. The whole thing works fine when the original call that triggers Asterisk is from an internal extension (Xlite), but it fails when it's from my cellphone ringing through the FXO/Zaptel port and I want to wait a few seconds and call back through the FXO/Zaptel. Could it that even

Hello Incoming calls from the FXO trigger an AGI script which simply NOOP data sent by Asterisk through stdin. The first two NOOP work fine, but after this, Asterisk isn't happy: ============= extensions.conf ... [from_fxo] exten => s,1,Wait(2) exten => s,n,Set(CID=${CALLERID(num)}) exten => s,n,AGI(/var/tmp/test.lua) exten => s,n,Wait(5) exten => s,n,Hangup =============

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

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

Hello I need Asterisk (1.4.39.2) to simulate a flash hook (ie. hitting the "R" key on European handsets) so I can put a call on hold, dial a second number, and set up a conference call. By default, linux/include/dahdi/kernel.h sets the flashtime to 750ms, which appears to be too long for European telcos, as they seem to expect a line cut of about 100ms. After editing the

Hi everyone, having a issue with asterisk and my new Voip providers service. Iv set up many asterisk systems before but never seen this and have tried to fix this with no luck.. I have used this exact same sort of setup for 5 other providers and never had this issue, If i replace the trunk login details with my works voip account and set it to IAX then it works perfect, Just not the new

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'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 everyone! I'm quite a newbie at this Asterisk stuff so please bear with me. We've recently decided to start training in Asterisk via AsteriskNow! Asterisk version is 1.4.18.1 through AsteriskNow! 1.02 The box we have is paired with a Digium TE110P and we've managed to get it to the point where incoming calls via a single DID (from NTT Japan) can be received and answered

How to I "recode" a factor into a binary data frame according to the factor levels: ### example:start set.seed(20) l <- sample(rep.int(c("locA", "locB", "locC", "locD"), 100), 10, replace=T) # [1] "locD" "locD" "locD" "locD" "locB" "locA" "locA" "locA"

Xen Host Server OS: SLES10 Kernel: 2.6.16.21-0.8-xen x86_64 Hardware: Sun 4200 Memory: 12Gb Guest Domains (2 total) RHEL4-U3 2.6.16-xen x86_64 2Gb Swap 4Gb RAM 2 Nics bridged to seperate physical Nics (Public and Private) Systems come up normal and we are able to mount filer space to the guest domains and do work. However, at some point, the guest

Hello everyone, Proceeding with an effort to optimize pointer-arg-to-struct-with-data-pointer arrangement, I'm taking a shot at writing a specific AA pass. I've got a scaffold up modelling after BasicAA and bumped into a few issues. Here's a snippet of IR I'm dealing with. Slice is the wrapper structure and "data" pointer is element 0 and we're looking at two of

I have an issue with one of my installations running Asterisk 1.4.20 that I need some help with. Not sure if this is new or not but Zaptel or Libpri is not releasing channels properly. I have had issues with calls that stay up for eight hours, long distance on the telco side, so it is more than just a nuisance. Does anyone know if there was a related bug in the 1.4 branch of Asterisk, Zaptel,

I have multiple Asterisk instances set up in different locations and would like to modify the callerID of inbound calls to identify which instance the call is coming from.? I knew how to do that with the old sip format, but can't seem to figure it out with PJSip. For example: Currently Location A, extension 10 calls Location B, extension 20.? CallerID on Extension 20 displays

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")