similar to: overlapping extensions

I have couple of questions about the following. Currently I have 2 phone lines going into my house, and I would like to have both of those coming into asterisk. I also want to have a backup asterisk, so here are the main questions (I am knew to this so I apologize if I ask something stupid): - Is there a dual FXO card available from digium or do I need 2 x single FXO (if this is the case then

This is mildly off topic since it has more to do with TCP/IP networking than with CentOS per se. However, at the risk of ruffling a few feathers I would like some advice on how to resolve the following routing problem: Given: dual homed host running CentOS-5.2 with eth0 = 2xx.yyy.zzz.23 eth1 = 192.168.219.1 and a Cisco gateway at 2xx.1xx.y7y.1 with the following networks: interface

Folks: I've got several systems attached to a 2003 domain where we use kerberos to authenticate. When I upgraded a system to the latest greatest samba things stopped working. Just to find where it happened in the different versions of samba I downloaded, built, & ran 3.0.23d to 3.0.25c using the same smb.conf file. Turns out the 2.0.23d and 3.0.24 works but from 3.0.25 on it fails.

Hi, I am lost trying to figure out how to enable overlap dialing for calls coming in and coing out on a pri span. DISA looked promising at first, but does not seem to support overlap dialing. Just picking up a call by and trying to dial out does not seem the way to do it either. I tried: [dialincontext] exten => 12341234,1,Goto(dialoutcontext,s,1) [dialoutcontext] exten => s,1,Wait,1

Hello, We are having a problem with linker lookup on Windows. When the driver tries to locate a program by its name, e.g. a linker, it scans the paths provided by the toolchain using ScanDirForExecutable and then, if fails, uses llvm::sys::findProgramByName, which on Windows searches the CWD and system path. ScanDirForExecutable disregards file extensions, so search in the toolchain-provided

I have a SIP phone (Cisco 7960) registered to 2 * pbx, is there anyway to control which * pbx will be used for making calls? I know by default the cisco will use and I want to change that. thanks micko

Hi, How do I display/log the results of variables from extenstions.conf? I've several macros, where I'd like to use ${CONTEXT} to help in GotoIf's, but I'm not convinced what value CONTEXT is being set to. I tried using NoOp(${CONTEXT}) and then set debug on for messages, but all I see is: Dec 7 16:09:25 VERBOSE[815] logger.c: -- Executing

I need to write a script that will contact a remote system using TELNET with VT52/1xx/2xx/320 and perform some operations. I am familiar with TELNET but it appears that it won't work for my purposes. TIA, Frank M. Ramaekers Jr. Systems Programmer MCP, MCP+I, MCSE & RHCE American Income Life Insurance Co. Phone: (254)761-6649 1200 Wooded Acres Dr. Fax:

I am grouping my extensions by building like so: 1XX is Building 1 2XX is Building 2 7XX is Office [Office] extensions has the following includes 7xx Include => Local Include => International Include => Building1 Include => Building2 [Building1] has 1xx Include => Office Include => Building2 Include => Local I don't want building1 to access international, but

I have 2 hosts that Asterisk is in between of...and for both I have canreinvite=no - but asterisk still sends re-invite to get out of the media path. ? Proxy 1 --> Asterisk--> Proxy 2 ? I want asterisk to anshor media.. ? In extenstions.conf I have an entry to send calls for say 5551000 to proxy 2 and if? I suffix that entry with ||t , asterisk does anchor media.... ? However, it is

Hi all I'm being really stupid today. i simply want asterisk to answer a incomming call, then wait for digits dialed. and then dial that extenstions but i keep on getting: WARNING[3314]: pbx.c:2017 ast_pbx_run: Invalid extension '5', but no rule 'i' in context 'zap-in' my config: exten => 0,1,answer() exten => 0,2,digittimeout,5 exten =>

I've got a PAP2 that I've got working with asterisk. At the moment, its configured so that when a phone is picked up on it, it connects to Asterisk. My hope is that I can let Asteirsk handle the entire dialplan, including dial tone generation. What would my context in extenstions.conf look like for this sort of dialing. More accurately, how can I get Asterisk to generate the dial tone on

Hello, I've been writing a program in C that will be called by R. I seem to have stumbled upon an odd error that seems to suggest there is a limit on the number of times "Realloc" (the R version as defined in the manual "R-extenstions" not the C version "realloc") when I try to use the following program: #include <R.h> #include <Rinternals.h> SEXP

Hello - I am trying to decide if I have stumbled across a bug in PJSIP or I am just missing something. My Asterisk has two interfaces, an "internal" eth0 and an "external" eth1. In pjsip.conf, I define the following transports: [trusted] type=transport protocol=udp bind=10.xx.yy.zz:5060 [untrusted] type=transport protocol=udp bind=12.4.aa.bb:5060 My internal endpoints use

I am grouping my extensions by building like so: 1XX is Building 1 2XX is Building 2 7XX is Office [Office] extensions has the following includes 7xx Include => Local Include => International Include => Building1 Include => Building2 [Building1] has 1xx Include => Office Include => Building2 Include => Local I done't want building1 to access international, but does

Hi all. I just installed bunch of IP450's and everything went well and my customer is happy.... except that they are unable to transfer calls to other extenstions. They can dial them directly just fine. However, when the user is in a call and presses the transfer soft key, they get dial tone, and start typing the extension, say 1008. But by the time they get 100 typed in, the phone tries

I'm trying to setup my first Asterisk setup on a CentOS 5 installation on VMWare Workstation 6. Got two Linksys SPA941s working fine. But X-Lite softphones can't answer phone calls, and when one of them calls on of the Linksys phones they "connect" but neither party can hear hear the other. I noticed that the Linksys phones are connected via Native bridging while the

Hi all, I want to make a SIP trunk between a Cisco 2811 router and a Asterisk. Both 2811 and Asterisk are working fine (2811 has 1XX and asterisk 2XX). Now I want to configure a trunk so that 2811 users can call * users. I've been reading a lot but I'm still confused. Hope you can help me, -- Diego Quintana a.k.a. RouterMaN Ingeniero de las Telecomunicaciones Linux Registered User

I love it... I buy a half a million dollars worth of Trashiba's finest ... I download Asterisk for free... I now refer to it as legacy 18 months and 300 extensions later! Anyway, I am trying to integrate my dial plans acrossed platforms. PSTN>>>CTX670>>>Asterisk The dialplan I would like to setup, 1xx,2xx,3xx,7xxx to the CTX 670 4xx,6xxx to a remote ctx100 (this is setup

Hi there! I''ve using HTB for a while and now I an faced with a ''problem''. How hierarchical is HTB? Let''s say I have this 3 layer HTB setup: root class 1: (rate=100, ceil=100) 1: children classes 1:10 (30,100) and 1:20 (70,100) 1:10 children classes 1:100 (10,100) and 1:101 (20,100) 1:20 children classes 1:200 (30,100) and 1:201 (70,100) I managed to have