similar to: Can't get simple config working!

So I have a problem. A customer of mine wants a PBX, owns an office building. I want to sell him on asterisk. He has 4 tenants. I am using my asterisk box to simulate it. My asterisk box has a TDM400P card, not a PRI card. Don't know if it makes any difference. Anyway, I want to route incoming phone calls to different contexts based on the phone number being called. Here is my

Hi Lee, You need to specify the VM context that you are using.. so using your examples.. extensions.conf entry.. exten => 1000,1,Dial(SIP/1000,20) exten => 1000,2,Voicemail2(u1000) exten => 1000,102,Voicemail2(b1000) exten => 1000,103,Hangup should be.. exten => 1000,1,Dial(SIP/1000,20) exten => 1000,2,Voicemail2(u1000@sip) exten => 1000,102,Voicemail2(b1000@sip) exten

I want to record all phone calls made inbound and outbound. I'm new so having a hard time getting this started. Here is what I have so far but isn't working. Can someone help me out? Thanks, [macro-record-on] exten => s,1,SetVar(CALLFILENAME=${DATETIME}-${ARG2}-${ARG1}) exten => s,2,Monitor(wav,${CALLFILENAME}) [sip] include => macro-record-on include => iaxtel exten

Here is what I have now. Where should the line " exten => _.,1,macro(record-on,${EXTEN},${CALLERIDNUM})" go should it be under [sip]? Right now if I call sip to sip monitoring starts and the calls connect but I only get 44 byte files. If I call and iaxtel number monitoring starts but call never gets placed and again 44byte files with nothing in them. Thanks for the help. [iaxtel]

Hello all. Now I aleady installed the Asterisk. I could make communication between 2 XLite client through Asterisk. I tryed to test the voicemail function as follow. 1, I make a call to 1001 from 1002 2, Start ringing 3, Wait untill time out for ringing If no problem, 1001 go to voicemail and unavailable message will be played. But 1001 receive a 403 forbidden massage and connection go

ive got asterisk running with chan_sccp and three cisco phones (2 7910's and 1 7960). lots of bugs. when i press the speed dial button on either 7910, asterisk dies. also, if i dial from the 7910 to 7910, everything works fine. i can dial from or to the 7960 once, and then one of the 10's and the 60 die and try to reregister. if i take the 7960 out of the mix and remove its

Hi, Every time I dial into my asterisk box i hear nothing but asterisk jittering. The following is an example of what I get on the asterisk CLI Thanks *CLI> DEBUG[81926]: File chan_sip.c, Line 3826 (check_user): Setting NAT on RTP to 0 DEBUG[81926]: File chan_sip.c, Line 4807 (handle_request): Check for res DEBUG[81926]: File chan_sip.c, Line 952 (find_user): Call from user

I'd like to know how everyone else is going about getting correct CDR information for calls. Right now I notice that if a call come in and gets parked the CDR info doesn't how the correct info on who picked that call up, also when someone transfer a call there isn't a new record being made so the duration of the call shows up for who received the call and transferred it. I started

On Aug 26, 2013, at 2:39 PM, Stephen Crane <sjcrane at uci.edu> wrote: > We have been working on adding randomness into code generation > to create a diverse population of binaries. This diversity prevents > code-reuse attacks such as return-oriented-programming (ROP) by > denying the attacker information about the exact code layout. Putting on my security hat (as opposed to

I am not having any luck placing out going calls I dial the number 08 82420173 ( our outside line ) But all I get is engaged signal and log this. Oct 14 08:40:14 DEBUG[16401]: File pbx_wilcalu.c, Line 65 (autodial): Entered Wil-Calu fd=20 Oct 14 08:40:14 DEBUG[8201]: File chan_sip.c, Line 657 (create_addr): Setting NAT on RTP to 0 Oct 14 08:40:14 DEBUG[8201]: File chan_sip.c, Line 548

Hi, I would like to prepare postlogin a script that allow imap connection to roundcube for all but restrict imap access for selected users. My question is that: Is possible in condition IF use IP addresses as range or with mask (because I've more than one web servers) ? My script: #!/bin/sh if [ "$IP" = "172.11.0.28" ] ; then printf "* [ALERT] Access allowed from

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I have a T100P card connected to an Adtran and then a T1. I have added the following configurations to Asterisk...but, when I dial 9 and then a local phone number, it bounces between the dial tone and silence and the *error* light on the Adtran blinks. zaptel.conf span=1,0,0,esf,b8zs fxsks=1-8 loadzone=us defaultzone=us zapata.conf [channels] context=from-sip signalling=fxs_ks

I can't do threeway from my Grandstream phone. Looking through the server config files, I figured out why - zapata.conf has Threeway turned off for the channels I use. I do my work on someone else's Asterisk box and don't want to modify zapata.conf for several reasons, the biggest being that the guy who owns the box has a couple clients using it and I am deathly afraid of breaking

I came across an interesting test sample the other day while trying to compress some of my music and trying out different encoders. You can grab the sample in question from here: http://www.animus-facticius.org drone_clip.zip is a 9 second clip of the track where the artifact is most apparent, drone_short.pac is a 1 minute clip of track. The sample is from the song "Drone" from

Thx, prips works as I expected, gr8 tool, not available in Gentoo repository but after compilation Dovecot doing what I wanted. Regards, Jack 2017-11-09 21:19 GMT+01:00 Gedalya <gedalya at gedalya.net>: > A bit clunky but perhaps you could find another command. > > https://packages.debian.org/stretch/netmask > > $ IP=172.11.0.28 > $ if [ "$(netmask -n $IP/24)"

Hello, I'm trying to set up a conference room. When I dial it's extension, I get an audible error saying "Not a valid conference room, please try again" followed by a disconnect. I've got debug sip peer 1001 (my X-Lite client) and I see this in the logs: (I'm pretty sure it has something to do with ztdummy, but I dunno... I have the port installed, but I

Greetings, In reviewing the changelogs it?s unclear in which release FLAC began supporting a sample rate of 192kHz. The reason for my question is that there are many forums and university studies that state that FLAC does not support a sample rate of 192kHz however the current documentation (assumed 1.2.1b) under FORMAT under FRAME_HEADER does note that it is supported. If it was not

Hello, everyone. Right now, we are trying launch our own PBX system based on Asterisk(Fedora) with Cisco 2611. Cisco has 2 port FXO card installed on it. For testing, I have 2611 hooked into phone line with number of xxx-xxx-xxxx fine. (I'll call it F). Using softphone, I can dial in extension 1001 on asterisk, which should talk to cisco. After initial connection to Asterisk, I have try to

Hello, I have Asterisk running on a RH9 box; Everything seems to be working as it should, except for Festival. Every time that Festival is called from Asterisk, Asterisk silently shuts down. Festival doesn't give any error indication and Asterisk just plain dies without a peep. Festival was installed per the Wiki, using source and patched with festival-1.4.3-diff; it works fine at the