[Asterisk-Users] How to make caller groups ???

Hello All !!!   I have 4 PSTN lines in the PBX server 1,2,3,4.   First line will be used by only one extension (i.e. for the boss) for incom ing and outgoing. This line is dedicated for him only.(   The remaining lines will be shared by the employees   1) Group A have access to lines 2

Re: [Asterisk-Users] Call Pickup Woes

Melcon Moraes wrote: On Mon, 2006-03-20 at 09:24 +1100, Adam Dale wrote: Hello all, I have an asterisk @ home system running 1.2.4. Call pickup seems to be a bit of a problem. I’ve looked at a lot of posts and the wiki, which states that you need to define the pickup extension in feature

Re: [Asterisk-Users] simple perl-agi - where's the error?

no, this doesn't make a difference On Mon, 20 Mar 2006 13:01:00 +0100 René Enskat [Teamware GmbH] <[EMAIL PROTECTED]> wrote: > Tried: > $DIALSTRING??? > > > > > -Ursprüngliche Nachricht- > Von: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] Im Auftrag von Lenz > Gesendet: Montag,

[Asterisk-Users] How to setup Proxy info to * box , [* box behind a squid proxy and firewall ]

Hi All I had successfully tried out asterisk on the LAN , now I want to call outside using sipdiscount or using http://exgn.net my asterisk box is behing a Firewall and the Internet usage is through a proxy server located at 192.168.20.20:8080 Now I want to configure asterisk bo

Re: [Asterisk-Users] simple perl-agi - where's the error?

of course, but this doesn't make the difference(i just simplified the input-variable to verify it's not a regexp-issue). It should at least try to use to dial the single number i've set, but it looks like the variable is empty... On Mon, 20 Mar 2006 12:55:38 +0100 Lenz <[EMAIL PROTECTED]> wrote:

Re: [Asterisk-Users] Annoying Asterisk Realtime Limitation

> Anyway, so I went back to a plain text file for > sip.conf. What a dissapointment. This is kind of backwards but you can make a script that will pull all the info from the DB and save it as sip.conf. __ Do You Yahoo!? Tired of spam? Yahoo! Mai

[Asterisk-Users] AMP and ABE

Hi, Has anyone had experience installing AMP/FreePBX on Asterisk Business Edition? The main issue we have come across is FreePBX requires a dependency "PHP-PEAR" &"PHP-GD" which is not available on RedHat RHEL3 (ES) Thanks James ___ --Bandwidth and C

AW: [Asterisk-Users] simple perl-agi - where's the error?

Tried: $DIALSTRING??? -Ursprüngliche Nachricht- Von: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Im Auftrag von Lenz Gesendet: Montag, 20. März 2006 12:56 An: Asterisk Users Mailing List - Non-Commercial Discussion Betreff: Re: [Asterisk-Users] simple perl-agi - where's the error?

Re: [Asterisk-Users] simple perl-agi - where's the error?

Try setting it to sth like SIP/200 instead of a single number. l. On Mon, 20 Mar 2006 11:56:50 +0100, Christian B <[EMAIL PROTECTED]> wrote: Hello! I'm trying to setup a perl-deadagi, but my perl skills lack. can someone tell me why the following code doesn't work: #!/usr/bin/perl use Ast

[Asterisk-Users] MixMonitor and transferred calls

Hi; I'm trying to record all inbound and outbound calls at a site, and I have a problem with inbound calls that are transferred by a receptionist using Snom's handset buttons (i.e. SIP transfer rather than using the key sequences defined in features.conf). The first leg of the call is recorded fi

Re: [Asterisk-Users] Call Pickup Woes

What about setting up DYNAMIC_FEATURES=>pickupexten inside your [globals] ? This is needed for, as the variable name says, dynamic features. And don't forget to set callgroup/pickupgroup to each one in your sip.conf Does anyone tested the new application Pickup()? []'s MM On Mon, 2006-03-20 a

RE: [Asterisk-Users] Numbered Voicemails even with delete option!

Thanks Bret for the input. Your solution seems a lot neater=) I had problems with "globbing" I think it is called. I kept getting files name being created called "msg*.txt" which caused me problems later. I think your way removes this. The reason I was doing this was for testing purposes. I was

[Asterisk-Users] Re: Local Channel

In article <[EMAIL PROTECTED]>, Darren Wiebe <[EMAIL PROTECTED]> wrote: > I'm using the Local channel in an app of mine and I'm finding that > the app is being cut out of the call path. You used to be able to > avoid this using the \n command but that doesn't seem to work any > more. This is

[Asterisk-Users] Help: Using asterisk and mysql for a university project

Hello all,   I want to use mysql for to save the users of my asterisk PBX. I use the realtime solution with mysql but when I made the ‘sip show peers’ command doesn’t appear my users. My configurations are:   res_mysql.conf  [general] dbhost = 127.0.0.1 dbname = asterisk dbuser = a

[Asterisk-Users] simple perl-agi - where's the error?

Hello! I'm trying to setup a perl-deadagi, but my perl skills lack. can someone tell me why the following code doesn't work: #!/usr/bin/perl use Asterisk::AGI; $AGI = new Asterisk::AGI; $dialstring = $AGI->get_variable("DIALSTRING"); $res = $AGI->exec("DIAL $dialstring"); the asterisk output

[Asterisk-Users] ISDN Protocol Unknom Error with Junghanns OctoBRI

Hello, I recently bought a Junghanns Octobri Card. I have some problems with this card to make outbound calls but I can receive calls. I have 3 lines to PSTN and 3 lines to my existing PBX FRANCE TELECOM <-- OctoBRI --> Asterisk 1.2.4-BRIstuffed-0.3.0-PRE-1h <-- OctoBRI --> PABX e-Generis

Re: [Asterisk-Users] Numbered Voicemails even with delete option!

On Mon, 2006-03-20 at 09:32 +, David Waugh wrote: > NOTE: This is my first shell script so I'm sure it can be improved! > noted, in that spirit see notes below ... > *** > [EMAIL PROTECTED] INBOX]# more /etc/asterisk/voicemail-clean > >

[Asterisk-Users] Problems loading res_odbc.so and cdr_odbc.so

Hi. I am having troubles loading the res_ and cdr_odbc modules, they fail because they cannot find libodbc.so.1 I have unixODBC properly installed and the needed DNS setup correctly. Any ideas why I am having this troubles? Where is asterisk looking for the libodbc.so.1 file? And were can I c

[Asterisk-Users] Grabbing the billsec and duration after a hangup.

Hello, I am wondering if someone has got any ideas that can help solve this problem. I have a dial plan that you call into, and depending on certain conditions it calls out on a number grabbed from a database. Something like this : exten => s,n,Do something exten => s,n,Do something

[Asterisk-Users] Numbered Voicemails even with delete option!

Hello, Thought people might be interested in this. I want my voicemails emailed to a person and not stored on my asterisk server. However, I want them to have a sequential number. I found that if I set the option delete=1 in my voicemail.conf file for the mailbox, then the numbering would keep be

RE: [Asterisk-Users] Countries supporting SMS on PSTN (ISDN)

Unfortunately in Italy doesn't work: Italy and Spain uses Protocol Type2 and app_SMS doesn't support it (to my knowledge). http://www.rtx.dk/Files/Filer/tekniske%20artikler/SMStransmissionwithinthePS TN.pdf Mimmus > > -Original Message- > > From: [EMAIL PROTECTED] > [mailto:asterisk-user

Re: [Asterisk-Users] g729 and latency measures

On Mon, 2006-03-20 at 11:38 +0530, ram wrote: > Hi > > what is mtr ? > > where can i find that http://www.google.com/linux?hl=en&lr=&q=mtr&btnG=Search Pete ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list

[Asterisk-Users] Using IAX

Hello, I am a newbie, so I apologize for this maybe simple question. I want to connect two Asterisk machines with IAX. >From one machine I want to call to the other Asterisk,but sometimes I want to place the call on one context and sometimes in another one. I how can I do this?? When dialing on

[Asterisk-Users] asterisk and DDI

Hi,   Somebody has some infos for asterisk and swyx connected via DDI? Somebody has a example config for ddi wiith asterisk?   regards rene ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or updat

Re: [Asterisk-Users] Re: Attended Transfer - transfer timeout, how to change?

Tomislav Parčina wrote: > In article <[EMAIL PROTECTED]>, [EMAIL PROTECTED] says... >> you are using the attended transfer feature.. >> ist it already possible to hang up before the other person lifts the handset >> without loosing the caller when you are doing an attendet transfer? >> >> (person

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