similar to: Call Forward - Need Help

Hello How do I fix this.... [IPComms-in] exten => ${IPCCIDN01},1,Noop(${DATETIME} ${CALLERID}) exten => ${IPCCIDN01},2,SetCallerID(${CALLERID}) exten => ${IPCCIDN02},1,Noop(${DATETIME} ${CALLERID}) exten => ${IPCCIDN02},2,SetCallerID(${CALLERID}) exten => ${IPCCIDN03},1,Noop(${DATETIME} ${CALLERID}) exten => ${IPCCIDN03},2,SetCallerID(${CALLERID}) exten =>

I'm trying to connect an asterisk server via oh323 to a Lucent iMerge. I patched the code due so that Lucent can handle asterisk's ver4 h323 http://www.voip-info.org/wiki-Asterisk+Lucent+iMerge+Configuration I can now successfully dial in to our company over multiple lines. The issue is when I dial out The first outgoing call connects to an outside user A The second call drops the first

I am trying to SetCallerID to a variable I have defined. This obviously is wrong. It actually sets the caller ID to $NEWCALLERID. I have search through the examples on wiki but wasn't able to find something similar to see what I was doing wrong. Could someone tell me the correct way to SetCallerID to a defined variable? exten => 2125551212,5,SetCallerID({$NEWCALLERID}) exten =>

Greetings all.. We have an inbound/outbound PRI installed and terminated on a T400P ? Digium Quad T1 card. We?re seeing an odd problem when sending $CALLERIDNUM when calls from the PRI are forwarded back out to the PSTN over the PRI. The $CALLERIDNUM is not being sent out along with the call. It?s sending the phone number of the PRI itself, rather than the $CALLERIDNUM information. Yes, we can

My number is not submitted. I updated my asterisk but this error still occurs coz of the "" in the SetCallerID tag thats why it will be a empty SetCallerID is submitted. Is there a fix to correct this error? -- Executing SetCIDNum("SIP/31-752a", "4989427xxxx") in new stack -- Executing SetCIDName("SIP/31-752a", "4989427xxxx") in new stack

When somebody call me on my pstn # cable connected to my fxo card it does not work when I check my computer the following error shows Connected to Asterisk CVS-v1-0-12/05/04-19:46:25 currently running on asterisk1 (pid = 2160) Verbosity is atleast 3 -- Remote UNIX connection -- Starting simple switch on 'Zap/1-1' == Starting Zap/1-1 at incoming,s,1 failed so falling

I have no idea what this really should be called, so for lack of a better name, I called it "reverse hold". Hopefully someone else can make use of it, or even make it better, as its the first thing of its kind I've made for asterisk. Like most people, I'm very busy, so when I call other companies, sitting on hold really sucks. If you have speaker phone, its not so bad, but then

Hey guys. Thanks for the help on the Pattern matching, I got that working pretty nicely. the next problem I have is that I'm using an include file, but its not really working... In my extensions.conf: [incoming] exten => _NXXNXXXXXX,1,SetCallerID("Unknown Called Number") #include "numbers.conf" exten => _NXXNXXXXXX,3,Macro(Number,1000,${EXTEN}) [macro-Brand]

> -----Original Message----- > From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users- > admin@lists.digium.com] On Behalf Of Robert Withrow > Sent: Tuesday, June 15, 2004 12:32 PM > To: Asterisk-users > Subject: RE: [Asterisk-Users] making * more like a normal pbx (ciscoata- > 186) > > On Mon, 2004-06-14 at 19:34, Reid A. Forrest wrote: > > I've

Hi forum, I have been fighting days and days configuring FWD and asterisk with NO success I have the following scenario. My sister in Spain with FWD dialup client My question is if she can dial my FWD dialup number, which is registered in Asterisk and the call being forwarded to ring my IP Phone. Spain LAN FWD

I am making some changes to the dial plan at the request of the company president and have run into some problems. I have a couple of layers of menu's and I am not sure how to handle them. Here is how it should work (sorry for the crappy diagram) main menu --------Dial 1 for support | Dial 2 for special | Dial 3 sales

Hi, Does anyone know how to dial toll-free (800) numbers through FWD or Siphone? Using the configuration below, I can dial out to SIPphone.com users by simply dialing their number (1747XXXXXXX) and can dial out to FWD users by dialing 1383<FWD#> However, when I dial 18005551212 through SIPphone, or through FWD (depending upon which line is selected in "; 800 Toll Free Numbers"

Hei! I have a little problem with the subject. I use Asterisk CVS-HEAD-09/06/04-12:42:56 as a production *, but I do tests with a newer version Asterisk CVS-HEAD-11/18/04-10:01:32. Ok the problem is: in extension.conf I use macro for redirection, found on wiki pages: [macro-stdexten] ; ; Standard extension macro (with call forwarding): ; ${ARG1} - Extension(we could have used ${MACRO_EXTEN}

Is it possible to build grousp of channels? I have a series of extensions which are receiving incoming calls to various virtual organizations. Something like this... [foo-incoming] exten => 2122222222, 1, Goto(ACorp|1000|1) exten => 2123333333, 1, Goto(BCorp|1000|1) . . . exten => _NXXNXXXXXX, 1, Goto(GenericCorp|1000|1) [GenericCorp] exten => 1000, 1, SetCallerID("Generic

Folks, Eric is spot-on--both phones are happy with the modified ${CALLERIDNAME} value since I removed the quotes in the call to SetCIDName(). I've replace my calls to SetCallerID() since I think using SetCIDName() is cleaner. Cheers, Rob > -----Original Message----- > Date: Sat, 29 Jan 2005 16:44:16 -0600 > From: Eric Wieling aka ManxPower <eric@fnords.org> >

hi, i've a wirded problem, i try to dial out, using this dialplan [default] exten => _*7.,1,Macro(anrufextern-sip,${EXTEN:2}) [macro-anrufextern-sip] exten => s,1,SetCallerID(SIP-ID) exten => s,n,Dial(SIP/${ARG1}sip-out) exten => s,n,Hangup() when i use my analog telephone, everything is okay: - Starting simple switch on 'Zap/3-1' -- Executing

This is with an x100p (the motorola chipset) Two problems. Looks like CALLERIDNAME is being used uninitialized. On my other phones the callerid is fine and my buttset shows that the callerid passes the checksum. This is the relevant portion of extensions.conf exten => s,1,Answer exten => s,2,SetCallerID(H ${CALLERIDNAME} <${CALLERIDNUM}>) exten => s,2,Dial(${MGCP_ALL}) Here is

Quick question...Im authenticate all exten except this one(2006). If I call from ext 2006 I still have to authenticate. If I call form any other ext I have to authenticate. Any suggestions? Thanks extex => s,1,GotoIf($[${EXTEN} = "2006"]?3) exten => s,2,Authenticate(731) exten => s,3,GotoIf($[foo${ECID${CALLERIDNUM}} = foo]?4) exten =>

Hello, Does anyone know how to set the outgoing CallerID properly when using Snom200/SIP/CAPI/BRI? Following doesn?t work: exten => _0.,1,SetCallerID,526910 exten => _0.,2,Dial,CAPI/526980:${EXTEN:1} Asterisk writes: *CLI> -- Executing SetCallerID("SIP/226-ada0", "526910") in new stack -- Executing Dial("SIP/226-ada0",

Here is the setup: SIP/3044 -> SetCallerID(5551212) -> Call out PRI The CDR shows a src of 5551212. That is a lie! The src of that call was not 5551212, the source was 3044! The "translated source" of that call was 5551212. How can I get "real" source of this call and not some faky nonsense? The "reason" behind using the SetCallerID is because if I