similar to: Problems calling between two local SIP extensions

Hi, I have following one-line macro extension: ------------------------ [macro-oneline] ; ; Standard extension macro (with call forwarding): ; ${ARG1} - Device(s) to ring ; #exten => s,1,AGI(misterhouse.agi,"CallerID") exten => s,1,NoOp exten => s,2,DBget(temp=CFIM/${MACRO_EXTEN}) ; Get CFIM key, if not existing, goto 103 exten => s,3,Dial(Local/${temp}@default/n) ;

Same here, I just changed the b to u. Unavailable message is more generic, but it beats it saying busy when its not. -Tim -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Henry Devito Sent: Tuesday, October 05, 2004 8:31 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE:

Hi, I have attached configuration settings and cannot get ring signal when calling internal extensions. I'm probably doing something wrong so would kindly ask for a tip how to do it properly : exten => 11,1,Macro(oneline,SIP/11) Calling 11 (this is the same with BT or iax softphones) doesn't give me a ring - what is missing ? Thanks, Rob. [macro-oneline] ; ; Standard extension

I think you will find the functionality you are looking for is in * already. Here is an excerpt from the sample extensions.conf file that is included with the source: exten => s,1,Dial(${ARG2},20) ; Ring the interface, 20 seconds maximum exten => s,2,Goto(s-${DIALSTATUS},1) ; Jump based on status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER) exten

I'm struggling to get my dialplan to work with a simple analog fax machine. I have TDM400B zaptel card with an FXO and FXS port. I have the FXO port connected to the POTS machine and the FAX machine connected to the FXS port. The FAX machine itself works fine, I can FAX outgoing messages fine. I can also dial the FAX extension from the internal context, the FAX machine answers and I

Hi, I'd like to organize my Asterisk to properly handle incoming SIP/IAX/H323 callerids so they can be called back if needed. I have three incoming contexts for sip, iax and h323 calls. To each incoming call I'd like to prepend certain number that will be catched with pattern matching on output calls. For instance for iax I have: [from-iax] exten => s,1,NoOp(IAX call from outside

At work I'm used to tools like eTrust Access Control (aka SEOS). eTrust takes away the ability to manage the eTrust config from root and puts it in the hands of "security admin". So there's a good separation of duties; security admin control the security ruleset, but are limited by the OS permissions (so even if they granted themselves permission to modify /etc/shadow, the

Dear All, I'n trying to make a simple call forwarding, however I have small problem when evaluating an expresion. Here is my extensions.conf ... ; Unconditional Call Forward exten => _#21*X.,1,Set(DB(CFIM/${CALLERID(num)})=${EXTEN:4}) exten => _#21*X.,2,Hangup() exten => #21#,1,Set(ignored=${DB_DELETE(CFIM/${CALLERID(num)})}) exten => #21#,2,Hangup() ... exten =>

On 01/23/2015 06:01 PM, Stephen Harris wrote: > At work I'm used to tools like eTrust Access Control (aka SEOS). eTrust > takes away the ability to manage the eTrust config from root and puts it > in the hands of "security admin". So there's a good separation of duties; > security admin control the security ruleset, but are limited by the OS > permissions (so

I set up a call forwarding script in extensions.conf which uses the AstDB but I'm wondering if I reboot the server, will the entry in AstDB still reside? What the script does is when a call comes in, it check to see if there is a null value or a call forward number. If null, it will call the local office connections. If there is a number, it calls that. Now I just need to know if I reboot

We have an Asterisk server acting as a hosted PBX system for many clients, and we're going through an upgrade to Asterisk 1.6 by moving our most important (and complicated) clients one at a time. But we're having a problem with one customer that I really can't explain. I can place calls directly to one phone at the customer's location (they also have an IVR that asks for an

How do I get a context to continue to execute commands after the caller hangs up after a Dial command? I'm using the "e" option to the Dial application. I though the "e" option would allow the context to continue. This doesn't want to work for me. I'm using asterisk-1.6.beta5 I never get to "3" below. I get a message saying the "2" ended

hello all, another desperate request for help debugging my dialplan... from a certain extension i do the following: DBput(CFIM/${CALLERIDNUM}=${CALLERIDNUM}) a NoOp to the console says DBput: family=CFIM, key=2122022001, value=2122022001 and database show says /CFIM/2122022001 : 2122022001 so far, so good. but in a macro, when i try to get the data, exten

Hello I upgraded to CVS head yesterday (due to the lack of zaptel drivers working with 2.6.10) And noticed that now DBGet and DBPut have been deprecated in favour of the new Set/DB one. In the UPGRADING.txt in Asterisk it says: * The applications DBGet and DBPut have been deprecated in favor of functions. Here is a table of their replacements: DBGet(foo=family/key)

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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}

Hello, I'm trying to do call forwarding based on this: http://www.voip-info.org/wiki/view/Asterisk+call+forwarding In the extensions.conf file I have several context defined (local, longdistance, mobile, international and so on). Each user can be associated with different context (so can make only i.e. local calls). How to set calls forwarding only to numbers that are available in

Hello, I want to enable call forwarding for asterisk 1.6.0.6 I couldnt seen any config or option on gui or extensions.conf about it. I found some dialing plans to enable it on web as follows: [apps] ; Unconditional Call Forward exten => _*21*X.,1,DBput(CFIM/${CALLERIDNUM}=${EXTEN:4}) exten => _*21*X.,2,Hangup exten => #21#,1,DBdel(CFIM/${CALLERIDNUM}) exten => #21#,2,Hangup ;

Hello list, We run Asterisk CVS-HEAD-06/02/04-11:25:18 built by root@Gate01 on a i686 running Linux. All works fine except Audio is lost 10minutes into the call. This happens for every call PSTN-SIP, SIP-PSTN, SIP-SIP Example of one call setup using Snom200 and Grandstream 486: -- Executing Macro("SIP/xxxx1251-d638", "CFW|xxxx1251|SIP/xxxx1253") in new stack -- Executing

Hi All, I'm using asterisk 1.2.15 and call forwarding doesnt work for me. from my extensions.conf: ; Unconditional Call Forward exten => _*21*X.,1,NoCDR exten => _*21*X.,2,Set(DB(CFIM/${CALLERID(NUM)})=${EXTEN:4}) exten => _*21*X.,3,Playback(vm-saved) exten => _*21*X.,4,Hangup exten => #21#,1,NoCDR exten => #21#,2,DBdel(CFIM/${CALLERID(NUM)}) exten =>