similar to: Problem configuring voice conference

hello i want to kick participant in a meeting by pressing the digit on sip phone.when i entry the meeting ,no matter how i press the button,the dtmf does not work. here is my dialplan and my agi script,and sip.conf [from-internal] exten =>121,1,MeetMeCount(900,CONFCOUNT) exten =>121,2,GotoIF($[${CONFCOUNT}<10]?3:100) exten =>121,3,Authenticate(123456) exten

--- segue il messaggio inoltrato (the forwarded message follows) --- -------------- next part -------------- An embedded message was scrubbed... From: "QAMAR MUHAMMAD UZAIR" <d029307 at polito.it> Subject: help needed Date: Thu, 24 May 2012 18:36:30 +0200 Size: 1289 URL: <https://stat.ethz.ch/pipermail/r-help/attachments/20120524/cad7d73b/attachment.mht>

Hi, I am trying to register SIP clients which are behind NAT on different network. In order to achieve this goal I think I need STUN Server . I downloaded STUN Server from http://internap.dl.sourceforge.net/sourceforge/stun/stund_0.96_Aug13.tgz But I don't know how to install/configure it. And please advice me that STUN server is good idea for this scenario? Thanks in advance Wazb

Dear R users, I have weekly data in the following manner [,1] [,2] [,3] [,4] 6 2 2 2 5 4 4 3 6 35 2 13 7 32 3 5 4 4 423 3 3 6 4 5 5 6 5 3 I drew curve of each column against days (1,2,3,4,5,6,7). Now I want to draw an average curve (a representative curve) of the whole data. Is there a way in R to perform such operation? please reply ASAP as i have an assignment to submit on monday. Thanks

Hi, I get error when my DID hit to asterisk box which I am using for FAX to Email Service. Sometimes Fax goes through but mostly I get communication error on Fax Machine and on Asterisk I get Comfort noise support incomplete in Asterisk (RFC 3389) error. I am using SIP with G711. My Did provider cannot turn off VAD and Echo from his side, so is there any option or setting I can do at my side to

dear all, it will just take you a minute to tell me the difference between qnorm and qqnorm. are they same or is there any difference between them?? regards

Hi, I have read a lot about the thread of faxing support in Asterisk as well as spanDSP. However, either I don't fully understand other people's applications or may be what I'm trying to do is different from what others are trying to do. I have a very simple setup. I have an asterisk server with a TE110P connected to the PSTN via T1 PRI (Asterisk A). I have another asterisk

Hello, I have installed FreeRadius server on Trixbox Server. My problem is mysql is not letting FreeRadius to login either locally or remotely. I also insert proper entries in HOST and USERS tables. But it does not work I always get ERROR 1045 (28000); Access Denied for user 'root'@'localhost' Thanks Wazb

asterisk-users-request@lists.digium.com wrote: >Send Asterisk-Users mailing list submissions to > asterisk-users@lists.digium.com > >To subscribe or unsubscribe via the World Wide Web, visit > http://lists.digium.com/mailman/listinfo/asterisk-users >or, via email, send a message with subject or body 'help' to > asterisk-users-request@lists.digium.com > >You can

Hello, I have come across an unusual behavior where instruction domination rule is violated "Instruction does not dominate all its uses." It concerns with StackColoring pass present at llvm/lib/CodeGen/StackColoring.cpp. I am reaching out to the LLVM community to help me understand the cause of this issue and the working of the pass. The IR produced at the end of the pass seems to be

hello people, i m new to rails and i m thinking of buying Ruby on Rails 3 Essential Training with Kevin Skoglund tutorial http://www.lynda.com/Ruby-on-Rails-3-tutorials/essential-training/55960-2.html but problem is the tutorial is created on 2010 oct => so its 2 yrs old so my question it is fine to Buy these Tutorial in the changing rails world? Please Help me -- You received this

Hi, I have Asterisk 12.7.1 installed through Asterisk@Home CD. and explicitly I have installed UnixODBC and FREETDS in order to access MS SQL 2000 Database which in on Windows 2003 Server on remote location. I tested connectivity through isql and tsql, both utilities are working fine. I need to access MS SQL 2000 Database through PHP. When I tired to check the connectivity through a Test PHP

I am grouping my extensions by building like so: 1XX is Building 1 2XX is Building 2 7XX is Office [Office] extensions has the following includes 7xx Include => Local Include => International Include => Building1 Include => Building2 [Building1] has 1xx Include => Office Include => Building2 Include => Local I done't want building1 to access international, but does

So, I can't figure out how to get the parkandannounce application to work the way I want it to... I have cisco 7960 IP phones using SIP, and this is what I have in my extensions.conf: exten => 700,1,ParkAndAnnounce(pbx-transfer:PARKED|90|SIP/${EXTEN:1}|internal,${EXTEN:1},1) exten => 700,2,Hangup and in my parking.conf: [general] parkext => 700 ; What ext.

I have been working on integrating some FXS ports into my dial plan delivered via a channel bank and testing with an analog handset. The receptionist is on Extension 700. All other SIP phones are 7XX. >From a SIP phone I can dial 700 and all other extensions. >From the analog handset I can dial any other extension but not the 700 number. Weird? Yep. The CLI does not show any dialing when I

We've had a regular Park function in the past but recently I found the ParkAndAnnounce() application and I love the idea behind it. Here's a snip from the wiki (http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+ParkAndAnnounce) so that we're all talking the same language: || ParkAndAnnounce(announce:template|timeout|dial|return_context) || || Park a call into the

Good Day list, Anyone know if there is a way to have the AgentCallBackLogin function play a voice file after the agent picks up the phone? If this is not an available feature, any ideas on the difficulty in making this feature? Example: Extensions.conf exten?=>?700,1,AgentCallbackLogin(${CALLERIDNUM}|?AnnounceCAllQue-TechSu pport?); ....... exten => s,6,Queue(queue1)

hello, i am french student and i want configure a Asterisk server. when I want launch the server with the command safe_asterisk -vvvvvcf the server answer : Asterisk ended with exit status 1 Asterisk died with code 1 what is the signification of it please ? thank you lucas _________________________________________________________________ MSN Hotmail :

Hello, Asterisk 1.4.26.2, FreeBSD 8.0-RELEASE We have a very simple setup, using SIP softphones and a simple diaplan as follows in the examples below. When I dial the 700 extension it asks me for the extension and password, and it always says "login incorrect". The mail system send the email ok and Ekiga shows that I have vaoicemail, so the only thing that is failing is the actual

hi there! I have a couple phones connected to a sipura ata and if I go into *- IVR, I press options on the regular phones and it all works fine and dandy. then I connect an xten softphone, a new extension in my dialplan, I dial the ivr, * asks me to dial something to go through it, I press keys on xten, but nothing happens, * just times out through as if I did not press anything! is there some