search for: extensions_custom

...d the call is forwarded /transfered to the extension no , I use Asterisk 1.2.4, configured using AMP , on RHEL3 I did some configuration for my calls to asterisk , my aim was to ask the caller to enter the extension and the transfer the call For this purpose , I added following in "extensions_custom.conf" file [from-pstn-custom] include => ext-local exten => s,1,Answer() exten => s,n,Background(enter-ext-of-person) exten => s,n,Wait(2) exten => s,n,Goto(ext-local,s,1) the "ext-local" context is in "extensions_additional.conf" Now what happens is that...

...g the details in AMP for when caller dials 3, I have referenced it using 'custom-myapp,s,1', and if I go to 'extensions_additional.conf' I see the following line under the rest of menu item info that was created : "exten => 3,1,Goto(custom-myapp,s,1) ;" and in the extensions_custom.conf file I have [custom-myapp] exten => 3,1,SayDigits(1234) exten => 3,2,Hangup() But when you call and press option 3, it hangs up immediately. I have followed examples from the documentation, and this should be working. Any other places I can check where something is perhaps missing ?...

Hello Everyone, I am making a simple index.php file which will allow a web user to enter his $phoneNumb, $dialNumb, and callerID ($spoofNumb) and get the call bridged. Following is the index.php and the contents of extensions_custom.conf. When I submit the form nothing happens. I don't even see Manager Connected msg. Your input will be much appreciated. I am thinking I have some syntax problem. /etc/asterisk/extensions_custom.conf *[testphp] exten => _X.,1,Answer() exten => _X.,n,Dial(SIP/testTrunk/${EXTEN}) exten =...

...,System(/home/asterisk/bash_test) I don't really understand your code… I think I don't have to edit the first part of the conf file (" same = n,Dial(whatever) "), you just mean the second part of the code is executed by "n,Hangup"? Then I have to add the second part to extensions_custom.conf, context [macro-hangupcall-custom]? (I cannot edit extensions_additional.conf where're the other settings/it doesn't make sense, because FreePBX overwrites it.) Probably you mean h,1,DumpChan() instead of h,1,1,DumpChan()? last line: same → same code like in the upper line up to "...

...and good, unless you are using a plain old copper line that doesn't support DID. Anyhow, I have figured out how to make a call that comes in on a specific ZAP channel ring at a specific extension (not that it was brain surgery). I'm not certain if it would be better to use the file extensions_custom.conf instead of extensions_additional.conf, does anyone know? I have an A@H box with an unused TDM11P card in it at home in my basement. The A@H box normally handles incoming calls for my small business, but I wanted to plug my home phone line into the FXO port, and all of my phones into t...

...button". Is it possible? All the documentation I found so far, enable recording for 'all calls' to an extension. Does this code look ok? Currently Recording "on" only for 1030 when user presses *44, start recording. *55 to stop recording Extensions.conf #include extensions_custom.conf [from-internal] include => custom-startmonitor include => custom-stopmonitor [macro-dial] include => custom-startmonitor include => custom-stopmonitor Extensions_custom.conf [custom-startmonitor] exten => *44,1,GotoIf($[${EXTEN} = 1030]?2:20) exten => *44,2,Ba...

Ok so I added to my Freepbx config running Asterisk 1.2.4 in extensions_custom.conf ; intercom exten => _7XXX,1,SIPAddHeader(Alert-Info: Auto Answer) exten => _7XXX,2,Dial(SIP/${EXTEN:1},12,Tt) and configured my Polycoms via this page http://www.voip-info.org/wiki/view/Polycom+auto-answer+config for auto answer and that works fine if I dial 7 then the 3 digit...

Hi, I'm struggling to get busydetect to work. I'm using asterisk 1.2.1 and a digium TDM04B (4 port FXO) card. I've set busydetect=yes, busycount=6 and busypattern=300,200 in zapata.conf and i've modified zondata.c with a busy setting of 620+480, 300/200 which is the busysignal received from Korea Telecom. Asterisk isn't detecting the busy signal and doesn't hangup.

...;faxdetect=no ;Include genzaptelconf configs #include zapata-auto.conf group=1 ;Include AMP configs #include zapata_additional.conf extensions.conf file: ; include extension contexts generated from AMP #include extensions_additional.conf ; Customizations to this dialplan should be made in extensions_custom.conf ; See extensions_custom.conf.sample for an example #include extensions_custom.conf [from-trunk] ; just an alias since VoIP shouldn't be called PSTN include => from-pstn [from-pstn] include => from-pstn-custom ; create this context in extensions_custom.conf to include customizations...

I'm lerning to make my custom configurations. In extensions.conf, there is #include extensions_custom.conf [from-trunk] ; just an alias since VoIP shouldn't be called PSTN include => from-pstn [from-pstn] include => from-pstn-custom ; create this context in extensions_custom.conf to include customizations include =&gt...

Hi All; Using the FreePBX, after I added the extension from the GUI, I discover that it is automatically added in the extensions_additional.conf in the context [ext-local] and [from-did-direct-ivr] How I can change these context name? I need to determine this. How? Regards Bilal

...if someone could point out a dirty trick to get the voice to play right, for internal and external callers, depending upon what number they dialed into, or what organization the user belongs to, without massive dialplan facelift. I'm using the default AMP from-internal context for inside, and extensions_custom.conf for remote IAX clients. I have partitioned extension numbers i.e. exten 1XXX belongs to company 1 exten 2XXX belongs to company 2, etc. Any hints would help, tia

hi, i have a pronlem connecting call from pstn with the following confuguration, please advice extensions.conf [from-trunk] include => from-pstn [from-pstn] include => from-pstn-custom include => ext-did include => from-pstn-timecheck exten => fax,1,Goto(ext-fax,in_fax,1) extensions_custom.conf [from-pstn-custom] exten => s,1,Answer exten => s,2,Background(demo-echodone) exten => s,3,WaitExten(60) exten => 1,1,Dial(SIP/201,10) ; as soon as i key 1 or 2, call drop when i'm calling from pstn but works on lan exten => 2,1,Dial(SIP/203,10) logs (asterisk -r -vvvv)...

...= $agi->parse_callerid(); $agi->text2wav("Hello, {$cid['name']}. Let's enter some text."); $text = $agi->text_input('UPPERCASE'); $agi->text2wav("You entered $text"); $agi->text2wav('Goodbye'); $agi->hangup(); ?> ------extensions_custom.php------------ exten => 311,1,Answer exten => 311,2 Wait(1) exten => 311,3,DigitTimeout(7) exten => 311,4,ResponseTimeout(10) exten => 311,5,AGI(input.php) ------CLI output ---------------------- -- Executing Answer("SIP/200-09b20488", "") in new stack == Spa...

Hello, I am trying to make the following... Can someone tell me if it is possible? Is someone willing to do it from an asterisk@home box? 1. I send an http request to asterisk@home box. Ex: http://asterisk@home/call.php?phone=0033102030405&code=12345 2. Application will call phone number 0033102030405 (using a sip provider) 3. Application will play a pre-recorded voice prompt 4. Application

Hi All; If I set a context other than the default context, then I do not see a generation for a configuration in the extensions_additional.conf for this context, but always the generation for the configuration is for the default context (from-internal). Normally, I have to put some Phones in a context and another Phones in a context, and give each context a privilages, but if I do this, then I

...re using an Asterisk (AAH 2.8, specifically) to sink calls. I do this by taking the ${EXTEN} and breaking it down by sections until I get to the last 4 digits (i.e., 2105551212). Once I get to the 4-digit extension, I am trying to set a flag, or semaphore, to do Busy/Idle testing. Here is my extensions_custom.conf fragment: [SATX_555_Extensions] exten => 1212,1,System(cat /tmp/{orig_num}) ; ${orig_num} is set at the beginning of [from-trunk-custom] to the full dialed digits in ${EXTEN}, before I break it down. exten => 1212,n,Busy(); if the file exists, someone else has already called this...

Hi Friends, I am facing a strange problem with DISA. I have installed and configured Trixbox. I've created a secret extension i.e., 555 and called this extension in Digital Receptionist using custom extension i.e., created in extensions_custom.conf file. When I call from my mobile phone to my PSTN number, which is connected to FXO port, my IVR is responding. After entering my DISA extension(555), its asking password. After entering correct password, its giving ringtone. Upto this, no problem. The problem is coming here only. When I ente...

Is it possible for Asterisk to set the ptime attribute on outbound calls in SDP invite? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060405/7e1e68d1/attachment.htm

I have a brief from a local hotel to build a PBX using Asterisk but they want to use their exisiting telephones and wiring from an old PBX that no longer works. Basically, I can build the system but an looking for a card that will allow for upto 20 extensions to be wired into the back of the PC. Doeas anyone know of a solution to this Sean-- ICQ: 679813 FidoNet: 2:263/950 Jabber: