similar to: Avaya IP Office DTMF Issue

Hi i have a agi script that gets called when a user wants to dialout externally. it gets passed in the exten number and the number dialled and looks up in a db to see if they are allowed to dial the number. the problem is if someone forwards their phone to a external number the CALLERIDNUM is the CLID of the calling party not the extension forwarded thus the call is denied. Can anyone think of a

Hi How do i disable dialling out from voicemail? -- Jon Farmer Telford, Shropshire, UK

Hi I can't for the life of me work out why this is not working. When in the campon contect if you hit a DTMF key 2 you get moved to the exten => 2 defined in the mainmenu context not the exten => 2 defined in the campon context. What is wrong? The same happens if you hit key 1. [campon] exten => _*1XXX,1,Answer exten => _*1XXX,2,SetCallerID(${CALLERIDNUM}) exten =>

Been working fine for us so far. -----Original Message----- From: Andrew Joakimsen [mailto:joakimsen@gmail.com] Sent: Wednesday, December 06, 2006 6:33 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] MWI across multiple servers How well would NFS work in this situation? On 12/6/06, Porier, Jeremy M. < jporier@ccu.edu> wrote: We are about to

This depends on your application. As you say you are able to do everything you require in dialplan at that is great. I have used AGI fairly extensively becuase the stuff I want to do can't be done in dialplan alone. For instance i have written a auto attendants that can be dynamically controlled by a non-techie user with real time and in call reconfiguration. Also i have written IVR apps that

Hi all I think this could be en old question. I would like to do a realtime billing prepaid system, mainly using asterisk. I have found few things; I can not get CDR function into agi because asterisk set them once the call is absolutely finish (at least main values for the main porpouse, billsec,duration, etc..) There is a patch that allow you to use CDR

Hi, voicemail are working ok, I receive message as attach via email. My question is : how can the user call asterisk and listen to his voicemessages ? thanks Victor

Hi all, i have an * version: Asterisk SVN-branch-1.2-r45691, I need to register a linksys 922 phone thru internet and when I make sip debug command i see this debug information: -- SIP read from x.x.x.x:1024: REGISTER sip:mysipserver.com SIP/2.0 Via: SIP/2.0/UDP x.x.x.x:1025;branch=z9hG4bK-839856dc From: "SPA922" <sip:5403@mysipserver.com>;tag=685bbad1fae3325do0 To:

I have a fairly complicated setup. Extensions (1,2 and 3). In 3 - I execute AGI in java which plays few wav files depending on external parameters. Can I have a dial plan inside my AGI? If not, how do I accomodate user who needs to reach extension 2 from my agi? I have tried stream file and get data but the two commands did not work at all.

Hi! I did manage to load phone with SIP image : SIP70.8-0-3S, made SEP-MAC.cnf.xml, but phone never read the configuration from it. On the screen it's written "Unprovisioned", and phone is not trying to register with asterisk. Please help!! MihaelaMJ -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi all, I need to connect two asterisk server in same network and i'm using sip user as my clients...... plz anyone suggest me.... Regards, Thiru -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20061227/aa4e409c/attachment.htm

Hi All Does anyone know how to use the MySQL cmd in Asterisk with LIKE and % in the query? I have: exten => s,5,Set(query=SELECT name from contacts where tel like %${number}) exten => s,6,MySQL(Connect connid hostname username password dbname) exten => s,7,MySQL(Query resultid ${connid} ${query}) But there seems to be a problem with the % sign and I don't know how to

Hi, I have a basic dial plan question; Here is the scenario. Call comes through IAX and my * authenticate, then collect the digits and dials out, simple :). Here is the dial plan; [did-in] ;for did callers exten => 866219xxxx,1,Ringing exten => 866219xxxx,2,Wait,4 exten => 866219xxxx,3,Answer exten => 866219xxxx,4,Authenticate(/etc/asterisk/authenticate.txt|a) exten =>

It would appear that using :include on a model that has_many with an :order on another model makes it so that the order is ignored. Is this supposed to happen? Strikes me as odd behavior. For example class Farmer < ActiveRecord::Base has_many :cows, :order => ''position'' end class Cow < ActiveRecord::Base acts_as_list :scope => :farmer end If I do

Hi All, Has anyone hooked up * as an extension/trunk of an Avaya system that has around 2 ISDN30e's. Trying to add 100 extensions to one of our systems, but not sure where to start reading. Thanks. -- Kind Regards, Gavin Henry.

----- Forwarded message from Telford Tendys <telford@triode.net.au> ----- Delivery-Date: Thu Jan 25 22:22:57 2001 Date: Fri, 26 Jan 2001 16:26:27 +1100 From: Telford Tendys <telford@triode.net.au> To: feedback@vorbis.com Subject: Comments on your code User-Agent: Mutt/1.2.5i Dear Vorbis, I didn't even know about ogg or vorbis until I was at a conference and someone said, ``if

Hello guys, I have these models class Farmer < ActiveRecord::Base has_many :cows end class Cow < ActiveRecord::Base set_table_name :cows belongs_to :farmer end ... graph = Farmer.find(x) # Retrieves four cows of x graph.cows[1].name = "Trottolina" graph.save while adding a new cow to collection works perfectly fine, ActiveRecord doesn''t

So far I have not found a way that I can register the Avaya phone with Asterisk. From what I have found so far is that Avaya phone needs the Avaya Media Server and Avaya Gateway. Looking at the h.323.conf (in Asterisk) and the file 46xxsettings.txt (avaya file located in tftpboot) there are no settings to make the phone initialize. I have sent an email to the Asterisk Users Mailing List to see

The company I work for has deployed an Avaya IP phone system. They have deployed the Avaya 4602 and 4620 IP telephones. They might be sending me one of these phones for use in my home office. Question: Can I make this IP telephone register and work with my Asterisk server? I don't know if it is a SIP phone? I searched thru the Avaya site, but can't find whether it's a SIP phone or

Hi! First at all, sorry for my bad english ... I m trying to connect an Avaya IP Office 412 to Asterisk using E1 I ve compiled/installed libpri -> zaptel -> asterisk correctly and now, im trying to get the link working. I think, first step is to have green light on the TE110P, isnt it? I setup zaptel.conf: span=1,1,0,css,hdb3,crc4 bchan=1-15 dchan=16 bchan=17-31 defaultzone=es So, i