similar to: dial option 'g' not working

I have Meetme conferencing compiled for Debian as per http://powerontech.com/freepbx-on-debian.htm . I drop a callfile in the outgoing dir, and it intitiates a call to a local extension as a channel, but the call seems to block on the Meetme() command. That extension completes the outgoing Dial(SIP) command to my phone, announcing that leg is the only member of the conference, and just waits. If I

I'm trying to use Asterisk (v1.2.11) make a callback that dials both legs of the call into a Meetme() room together, but I keep getting "conf-invalid" messages. I created a callfile (/var/spool/asterisk/outgoing/out.call) that specifies a Local channel (extension) which contains a Dial() command to the "dialer", and an extension which contains a Dial() command to the

Hi: When should we use "unnumbered priorities"(n) in extensions.What is the different between these 2 forms of extensions.conf? and ,Are both true? extensions.conf: form1: [Conferencerooms] exten => 333,1,Answer exten => 333,n,meetme(8000|cim) exten => 333,n,playback(vm-goodbye) exten => 333,n,hangup form2: [Conferencerooms] exten => 333,1,Answer exten =>

Hey All, I need to forward an extension to an other depending on the current time but I could not get it done with GotoIfTime. What I'm trying to do is ring on the extension 1 if time is between 8:00AM and 2:00PM and on extension 2 if is between 2:01PM 11:00PM. exten => 111,1,GotoIfTime(8:00-14:00|*|*|1-12?333) exten => 111,2,Dial(${Person1}) exten => 111,3,Dial(Hangup) exten

Hello, In Asterisk 1.4 beta 3, the UPGRADE.txt file says: Variables: * The builtin variables ${CALLERID}, ${CALLERIDNAME}, ${CALLERIDNUM}, ${CALLERANI}, ${DNID}, ${RDNIS}, ${DATETIME}, ${TIMESTAMP}, ${ACCOUNTCODE}, and ${LANGUAGE} have all been deprecated in favor of their related dialplan functions. You are encouraged to move towards the associated dialplan function, as these

Hello, Asterisk. I am currently using Asterisk (asterisk-1.2.13) and asterisk-addons-1.2.3_1 on FreeBSD 6.1-RELEASE-p10 So, after setup asterisk for realtime extension: res_mysql.conf: [general] dbhost = 127.0.0.1 dbname = asterisk dbuser = asterisk dbpass = asterisk dbport = 3306 dbsock = /tmp/mysql.sock res_odbc.conf: [mysql] enabled => yes dsn => asterisk username => asterisk

Quoting Adam Tauno Williams <awilliam at whitemice.org>: >>>> It should work, it sounds like a mis-configuration somewhere, can you >>>> post the smb.conf, /etc/nsswitch.conf, /etc/resolv.conf and >>>> /etc/krb5.conf from the member server. >>> "wbinfo -u" lists 415 lines >>> "getent passwd" returns 93 lines

Modem connect speeds on calls through * seem to be lower than calls made through the telephone company lines or our old Rolm PBX. All data calls have 2 wire analog modems on both ends. For my set up I have channels of a Zhone channel bank tied to 2 modems. The Zhone channel bank interfaces my * server with a T400P card. modem --- Zhone Channel bank - * via T400P card - Zhone channel bank -

Hi All, I have a query about time based sequences. I know such questions have been asked a lot on forums, but I couldnt find the exact thing that I was looking for. I want to create a time-based sequence which will mimic the trading window AND would span multiple days. Something like below: "2011-01-03 09:15:00 IST" "2011-01-03 09:15:01 IST" .... .... .... "2011-01-03

Alternatively, just type debug(optim) before using it, then step through it by hitting enter repeatedly... When you're done, do undebug(optim). Andy > From: Spencer Graves > > Have you looked at the code for "optim"? If you > execute "optim", it > will list the code. You can copy that into a script file and walk > through it line by line to

Hi, I am new to Asterisk and I am having a setup problem that I am trying to resolved for the last couple days without any success. I am pretty much desperated on this issue and I don't know why. Can someone please kindly help me to troubleshoot this? I can't hear any audio from Asterisk when running Playback or VoiceMail tests. I have my Asterisk server ( running on Debian,

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On 14/04/15 20:59, Adam Tauno Williams wrote: > On Tue, 2015-04-14 at 15:20 +0100, Rowland Penny wrote: >> On 14/04/15 14:59, Adam Tauno Williams wrote: >>> On Thu, 2014-10-30 at 13:41 -0300, Horacio G. de Oro wrote: >>>> Hi! I'm trying to add a member to be used as fileserver, following the >>>> guides at: >>>> -

I'm trying to find some "live" examples on how to use the h, H and g parameters on the dial command (http://www.voip-info.org/wiki-Asterisk+cmd+dial) Any ideas? I was testing with the code below but after pressing * nothing happens (only after a long pause the "goodye" file was played) [testset] exten => 1023,1,NoCDR() exten => 1023,2,Dial(SIP/1023,30,Hg) exten

Hello, I've begun working on an AGI in PERL to provide the ability to dial a string, similar to composing text messages on cell phones. The current idea is that it would allow you to use a telephone to enter a string of text, (number 2 can be used for letters a, b, c, number 3 for d, e, f, Etc), and one could review the text thus far, backspace over characters, and "commit"

I have a case where I need the option g to continue execute after the hangup (I'm using 1.2.1) and I have the following in my extensions: exten => 309,1,System(echo > /tmp/file) exten => 309,2,Dial(Console/dsp,,g) exten => 309,3,System(rm -f /tmp/file) exten => 309,4,Hangup However, after the hangup priority 3 is not executed. Does 'g' not work with console/dsp or do

Hello! There's the "g"-option for the Dial-cmd that allows to execute the next extensions in the current context when the callee hangs up. I would need the same for a call where the caller hangs up, concretely i have to inform a agi-application of the end of a call. Does someone know a way to do this from the dialplan? thanks Christian

I'm unable to get the Dial option 'g' to work with callback agents. The plan is to use it so that I can redirect a customer to a menu so they can rate the call they just had with the agent. However, when the agent hangs up the call does not continue in the dialplan. I login with the agent. Call joins the queue. The agent and call get connected. The agent hangs up and the call

I'm using the Dial application in the extensions file with the G option to execute an AGI script after the Dial (I need to track the call status) as follows: exten => _X.,3, Dial({DIAL_STRING},,G(_X.^4)) exten => _X.,4, Answer() exten => _X.,5,AGI,agiScript.php The problem is that testing between two internal phones (with two ATA) I loose the audio when I include the G option in the

Hi all, I use the G option in my dials for redirect both parties in the conference. There is a way for auto-include in a conference other parties that first two without using AGI? I try with: [from-internal] exten => 9999,1,Dial(IAX2/DIP02/9999||G(fromiax^9999^1) [fromiax] exten => 9999,1,MeetMe(9999,qdxAa) exten => 9999,2,MeetMe(9999,qdx) exten =>