similar to: Wait for an extension and dial. Why does this not work?

Hi all! I need a simple plan for the following: *answer call *wait for 4 digit extension *send call to 4-digit extension entered. I tried the following, but that doesn't work... exten => 998,1,Answer() exten => 998,2,Background(agent-newlocation) exten => 998,n,WaitExten(20) exten => 998,n,Dial(SIP/${EXTEN}@${SERADDRESS},60,tr) WaitExten obviously does not fill EXTEN with

Hi All, After searching and didnt found it, im just sending my situation here, maybe someone knows where i should look. Im using Asterisk 1.6.1.10 Internally the user with a sip phone dials a number for instance 0623456789 It goes fine to the specific dial rule: which is: exten => _0[6].,2,Dial(SIP/0${EXTEN:1}#@xs4all-out,60,tTwWkK) This works fine without a charm, but the situation is that

I'm working on an IVR that gives the users the option (number 5 in the main menu) to dial by extension: exten => 5,1,Set(TIMEOUT(digit)=5) ; Dial Extension exten => 5,2,Set(TIMEOUT(response)=10) exten => 5,3,Background(LCL/prompt-60) exten => 5,4,WaitExten(15) When going option 5 you can dial some extensions such as 2802, it goes to the extension (all extens start with 28 on the

Hi, What i want to do - is to give ability for answered call to hear regular dial tone and be able to enter phone number - that i would dial later. I tried to look at WaitExten and PlayTones, but they seem to not work together - WaitExten doesn't interrupt going on PlayTones. Is there any way how i could do that - so that it looks really natural? It would be silly to create long-long-long

I would appreciate hearing from anyone who has figured this one out. Here's the scenario: I have a context wherein I give the called party the option to dial the digit 9. If he does so, he is transferred a la this extension entry: exten => 9,1,Playback(pls-hold-while-try) exten => 9,n,Noop(Attempting to bridge to ${agentext}) exten =>

Hi everyone! I'd like to create the following: a user picks up the phone (gets a dial tone), dials '0' for an 'outside' line, gets a second (different?) dialtone, and is able to enter an external phone number. How do I implement this in extensions.conf...? Regards, Evert

Hi, I have the problem that WaitExten is not responding to key presses. Here are the sections from my extensions.conf: [globals] incoming_call=0 menu=0 announce=0 [internal] exten => 777,1,Goto(hotline,${EXTEN},1) [hotline] exten => _X.,1,Set(CALLERID(name)=Hotline) exten => _X.,n,Set(original_extension=${EXTEN}) exten => _X.,n,GotoIf($[${announce}=1]?4:10) exten =>

Hi, I have a problem with people that are calling from analog lines. We have a block of numbers 12345 - 0 to -99. Most calls are transmitting the whole number including the extension. There's no problem with that. But people calling from analog lines are connected to our asterisk box as soon as they finish dialing 12345. They don't get a chance to dial an extension. Just inserting a

When I do a 'asterisk -vvvvvc' I get following, but asterisk does NOT stay up: == Parsing '/etc/asterisk/asterisk.conf': Found == Parsing '/etc/asterisk/extconfig.conf': Found

Hi everyone! The following: Any calls coming in on extension 12121212 should get a message telling them to dial the extension of the person they are trying to reach, and then press #. The call should then go to the entered extension. This is as far as I got... *********************************************************** exten => 12121212,1,Wait,1 exten => 12121212,2,Answer exten

Is anyone interested in some part of this filesystem to figure out how it failed? Or can I erase and start again? Kind regards, -Evert Vorster- On Sun, Aug 8, 2010 at 8:18 AM, Evert Vorster <evorster@gmail.com> wrote: > Hi there. > > I have a btrfs on a raw device. (/dev/sda insted of in a partition, like > /dev/sda1 ) > The device in question is a USB hard drive with a 1TB

For those of you who saw my gargantuan post the other day I'd just like to say thanks for listening and sorry for the lengthy post! It turns out that my key issue was with the WaitExten app. I saw this on the wiki which really helped out: http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial (See example 2: Dial macro) I replaced WaitExten with Read, did a little shuffling of

[Sorry if this arrives more than once. I have sent this twice and it never arrived, despite other messages getting to the list O.K.] ----------- Hello, I would like an incoming caller to be able to choose from the menu options in my extension.conf below. Once They have dialled the appropriate digit, * should call two extensions simultaneously: one SIP phone on this * server, and one over a

SIP user => Asterisk 1.6 server => SIP Trunk => external destination: works AMI script => Asterisk 1.6 server => SIP Trunk => external destination: Failed to authenticate on INVITE to '"asterisk" <sip:asterisk@(ipaddr)>;tag=alphanumeric' I?ve tried doing things like ?include => contextwithtrunk" in various places, googling, re-reading relevant

Hi list, I am running TDM11b + Asterisk-1.4.9 + Zaptel-1.4.4 + Libpri-1.4.1 on CentOS 5. I am having trouble to make my simple IVR extension work, here is relevant config: zapata.conf ---- context=incoming signalling=fxs_ks channel => 4 context=internal signalling=fxo_ks channel => 1 ----- extensions.conf: ---- [office] exten => s,1,Dial(Zap/1,30) [home] exten =>

Hi all I'm being really stupid today. i simply want asterisk to answer a incomming call, then wait for digits dialed. and then dial that extenstions but i keep on getting: WARNING[3314]: pbx.c:2017 ast_pbx_run: Invalid extension '5', but no rule 'i' in context 'zap-in' my config: exten => 0,1,answer() exten => 0,2,digittimeout,5 exten =>

I've been trying to setup AddQueueMember() as a replacement for the deprecated AgentCallbackLogin(), but I get _tree_ Queue()'s. Massaged extensions.conf (can provide the original if need be): ----- s n i p ----- [default] include => agent-loginout include => local ; ---------- [agent-loginout] exten => _100.,n,Macro(queue-addremove,I${EXTEN:3},dispatch,10)

Hello, I am running Asterisk 11 on CentOS 6.4 with about 150 local SIP clients on a LAN. The SIP clients are a mixture of Yealink phones (e.g SIP-T32G, SIP-T42G, etc). I have configured the system as follows: sip.conf: [169] secret=111111 dtmfmode=rfc2833 directmedia=no directrtpsetup=yes canreinvite=no context=main host=dynamic type=friend port=5060 call-limit=5 nat=force_rport,comedia

I have some problem to configure the call from asterisk to ser. [globals] SERADDRESS=xxx.xxx.xxx.xxx:5060 exten => 77,1,Dial(SIP/phonenumbertocall@${SERADDRESS},20,r) Error in Sip Debug ------------------------------------------- NOTICE[25541]: chan_sip.c:6848 handle_response: Failed to authenticate on INVITE to '"Alexg"

I am using Polycom 501s with asterisk 1.2.4. When transfering to call parking wih "#1" -> 700 the user is able to hear asterisk tell him what extension the call was parked on. However, when I press "transfer" -> blind -> 700 . The user is not able to hear what extension the call was parked on. It seems like the polycom is hanging up before asterisk is able to