similar to: This is what I want to do...

I have asterisk running more or less ok but I would like to turn off call waiting and be selective about the incoming sip connections. This is running asterisk 1.2.8 with a fxs and fxo card and a configured voip (sip) line. Currently I'm using freePBX 2.1.1 to configure asterisk. Problem 1) if someone is on the phone already and another call comes in for an already engaged extension I

Basically, I am looking to set up an extension which will be used as a "help-line". I want it to function kind of like the bat phone from the old Batman series, where Commissioner Gordon would pick up the extension in his office and it would ring the phone back at Wayne's mansion. Is there a way to duplicate this functionality with Asterisk? I just need asterisk to auto-dial an

Has anyone on here used kiax before? I am asking because I have it installed on several computers and have been able to get it to connect and register to my Asterisk box. I can even call between them and my SIP softphones. The problem I am having is this: when I use kiax to call someone else, they get some kind of background music playing while I am talking to them. I have called from kiax to

Hello everyone. I had heard about this open-source PBX once a while back. I wasn't too interested in it at the time but I kept the info filed away for possible future use. A couple of days ago, I was walking around Barnes and Nobles and I found this book, called Asterisk: The Future of Telephony. I paged through it a little and I was really excited by what I read. Then I remembered the

Hey everyone, I recently bought an Act-Tel G11112DS telephony gateway (the web interface says it's model # is G1111S though.) Has anyone else on this list used one of these? It has one FXO and one FXS port. I have an account for it set up in sip.conf on my Asterisk box and it apparently logs in correctly because I can dial the extension I set up in extensions.conf and the FXS port rings

I have a TDM-400P with one FXO module. On an incoming call, I have set Asterisk to dial my phone (exten => s,1,Dial(IAX2/carey)), which is basically the only thing in my dialplan. When the call is answered by the PSTN phone first, or when the ringing call is hung up, Asterisk keeps ringing for 5+ seconds, which causes trouble (the answering of already answered calls). I noticed in the

I have been looking around some and I can't seem to find anything which will answer my question. If I have two Asterisk boxes in different locations which are linked to each other over the internet, can I configure the boxes to use each other's lines as local? In other words, let's say Site A has Phone1 for a 1FB line going into it on an FXO port. Site B has Phone2 for a 1FB line

Hello! We are using the Digium 405PP card, and getting the following messages: Jun 16 16:16:17 NOTICE[81926]: chan_zap.c:6832 pri_dchannel: PRI got event: 6 on Primary D-channel of span 1 Jun 16 16:16:17 NOTICE[81926]: chan_zap.c:6832 pri_dchannel: PRI got event: 8 on Primary D-channel of span 1 My config file is below. We are trying to set up D-Channel on channel 24, 1-23 in trunk group 1,

Hi All, I have just migrated from Asterisk 1.0.0 to Asterisk 1.0.5 and I have an X100P installed. The old asterisk was working, but now the new version isn't picking up any calls! However, I did notice that after installation, I performed modprobe zaptel and modprobe wcfxo and they worked fine, but when I executed ztcfg, I get the following errors: ioctl(ZT_LOADZONE) failed: Invalid

>Easy to do on the Linksys PAP2, if that helps. The functionality > probably depends on the make and model of the phone... maybe if you gave > those details as well? > > James > Well, there's the rub. I don't have any of the hardware yet. I am looking at the various options before buying anything. I know that sounds like an odd way to do things: research it before

Hi, I am trying to configure * 1.0.7 box with a Digium Wildcard TE110P T1/E1 and libunicall latest code. All libs compiled successfully and the E1 have a green light! I am able to receive a call (or at least) testcall shows some information when an incoming call is received so the drivers and basic configuration is working. My * box has 2 cards, one TDM400B (2 fxs and 2 fxo) and one TE110P

Steve, Here is the config, I pulled from my server, that works with D'Link Phones: Main Menu -------------------------------------------------------------------------------- SIP.CONF [general] port = 5060 ; Port to bind to (SIP is 5060) ;bindaddr = 0.0.0.0 ; Address to bind to (all addresses on machine) bindaddr = 67.109.153.236 disallow=all ;allow=ilbc allow=gsm allow=ulaw

#I would like to use the below data to make a scatter plot matrix with #code similar to that below the data #conceptually this is the right approach I think #thanks in advance melt.gg <- structure(list(stream = c("Bonham Lower", "Bonham Lower", "Bonham Lower", "Bonham Lower", "Bonham Lower", "Bonham Lower", "Bonham Lower",

I have a bit of a weird problem that I'm having great trouble debugging. I have a TDM400P PCI card with two FXO and two FXS modules. Both FXO modules are connected to BT lines here in the UK. Both BT lines have V23 Caller-ID, which works fine with Asterisk. Both asterisk and zaptel are fresh from CVS. Both FXO modules (channels 3 and 4) are in "group 1" for outgoing calls. My

Hello Wine Users, I know that my question has already been posted once. (http://forum.winehq.org/viewtopic.php?t=2130&highlight=hamachi)But that thread did not really help me. Maybe I did something wrong [Rolling Eyes] This is the situation: I want to play "The Lord of the Rings: The Battle for Middle Earth - The Rise of the Witchking" with wine in a hamachi VPN. I am online in the

Hey all, I know that there are several posts on this subject but there seems to be no actual answer, however there are alot of things to try. Reason I've created a new post is that the others go so far then no 1 answers :P. Ok so I've got hamachi working fine, pinging and all with a linux native for obvious reasons. What I'd like to get working is Command and conquer 3 kanes wrath and

I was attempting to install Hamachi 1.0.3 via Wine (1.1.17) and got the following error message: wine: Call from 0x7b82fcc0 to unimplemented function setupapi.dll.SetupDiDeleteDeviceInfo, aborting wine: Unimplemented function setupapi.dll.SetupDiDeleteDeviceInfo called at address 0x7b82fcc0 (thread 002a), starting debugger... wine: Call from 0x7b82fcc0 to unimplemented function

Hi, I wanted to know if anyone had tried to use hamachi with Asterisk. I am trying to connect remote extensions (XLite softphone on winxp with hamachi) to my asterisk pbx. I had installed hamachi on my asterisk pbx (Asterisk @ Home 2.2). The boxes are able to see each other through hamachi. But somehow I couldn't establish a call with a remote extension. But I am able to talk to an

(Im' answering cc the list, so the knowledge keeps there, and maybe some more qualified answers become). Am Mittwoch, den 15.10.2008, 18:00 -0700 schrieb Francisco del rosario: > Hey Rodolfo... Need some help from you ... > I need to know what hardware do I need to make SIP calls if I set-up > asterisk > So the situation is that I have a PC and configure the software of my PC to

Hi again, I am trying to get my DTMF to use RFC 2833 rather then inband, so that I can utilize lower bandwidth codecs through my FXO. After much tinkering I was able to get my gateway (wellgate 3701A) configured to a point where I have some success, but no real joy. I have configured the RTP Payload type (or RFC2833 Payload type) to 101. I don't have a clue what this means, but I took