similar to: Display "Confideltial" or "unknown" on called iddisplay

Hi, When making a call from an Asterisk box over a PRI connection, I am able to set the Caller ID phone number to what ever I want. This works find. How to I make the called party callerid display "Confidential" or "unknown" as we sometimes see ? Andre

Maybe hidecallerid=yes in Zapata.conf Thanks, Steve Totaro http://www.asteriskhelpdesk.com > -----Original Message----- > From: Rich Adamson [mailto:radamson@routers.com] > Sent: Thursday, April 13, 2006 12:14 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] Display "Confideltial" or "unknown" on > calledid

Hi, For thanks to everyone that answered the "dial from pph". On an other subject, how would I go about playing a wav file while talking to someone over a Zap channel ? Let me explain. I am on line with someone. I want him to hear a WAV (or mp3) sound file. I punch a key on my phone keyboard and he hears the sound file and after we can continu talking. Any hints

Hi all, Here is the situation. Linux workstation access a web page on a web server (not the one running Asterisk). From that web page, we need to initiate a dial-out on the Asterisk server and once the call is connected, it must ring on the agent's hard phone. Any pointers about how to initiale an Asterisk call from a remove web server? Thanks, Andre Courchesne

Hi, Anyone knows if there is a way to play a list of sound file in a round robin mode (at specific interval) while someone in waiting in moh in a queue? Ok, you enter a queue and wait listening to moh, every X minutes a sound file is played from a list of sound files to be played. If that possible and if so how? Thanks for any pointers. Andre

We have 1 server that after a few hours operating has multiple process of asterisk running. Here is the pstree output: # pstree init-+-atftpd |-auditd---{auditd} |-bash---safe_opserver---op_server.pl |-crond |-cwASTcall.pl |-dbus-daemon |-events/0 |-hald-+-hald-addon-acpi | `-2*[hald-addon-stor] |-httpd---3*[httpd] |-khelper |-klogd

Hi, Any reason why I can not get the php exec() function to execute a shell command inside an agi script? Thanks. Andre

There seems to be something about SL for queues since when the show queues CLI command is used, it give something like "SL:0.0% within 0s": pbx*CLI> show queues 1 has 3 calls (max unlimited) in 'rrmemory' strategy (243s holdtime), C:174, A:9, SL:0.0% within 0s Members: SIP/1242 (dynamic) has taken no calls yet SIP/1251 (dynamic) has taken 4 calls

Hi all, First, this is not my first PRI/T1 Asterisk deployement. Did several with Bell, Telus, AllStream, Rogers but this is my first with Videotron. Just spoke with the person taking the order and on top of the standard settings (switch, coding,...) she asked me about data rate (56k or 64k). Since I have never been asked this question before and can find anything relevant in the

Hi, Anyone has good/bad experience with SIP providers in upstate NY? Any recommendations of such provider who works great with Asterisk? Thanks, Andre Courchesne

Hi, Is there a way to create trunk groups while asterisk is running. For exemple let's say that zapata.conf defines g0 as channels 1-23 I would like (while asterisk is running) define g1 as 1-10 and g1 as 10-23 Any hints appreciated. Andre Courchesne

Hi, Anyone can point me to a way to get the SIP phones status information (off-hook, on-hook,...). Either through Asterisk or directly from the phone (standard API?). I'm working with the Aastra 9133i. Thanks for any pointers. -- ---- Andre Courchesne

Hi, Anyone has hints to share about dialing result detection. By that I mean the ability to detect what answered: - Human - Answering machine - Fax - Disconnected number. Any hints or pointers appreciated. ---- Andre Courchesne

Hi, It's my first attempt to run asterisk 1.4 (have been on 1.2 for a while) and I have a problem where playback and background are played very very fast. When I say fast is you get a few sounds that's it... Running kernel 2.6.20.4 and latest released asterisk packages (asterisk, libpri, zaptel). Any hints? Andre COurchesne

Hi, Running Asterisk 1.2.12.1 when dialing a known disconnected number the calls just rings and rings. We never get the "The number you are trying to reach...". If we dial the same number from an Asterisk 1.0.11 server again over PRI, we get the message on the 1st ring. Here is the PRI debug of such a call that just rings and rings. Any ideas? PRI debug sur CPL: -- Executing

Ok, Here is what I got working: A call comes in from a Zap line. 5 SIP extension ring if nobody picks up, the call is transfered to a cell phone number. That works. I not want to add a playback of a file ("Please waite while you are being transfered") before transfering the call to the cell phone. How can I do this? Andre

Hi, I have a problem where some PRI channels get stuck in a "Call" mode. If I do a zap show channel XX, it shows as "PRI Flags: Call". However there is no calls on that channel. Trying to force a hangup does not work: [root@neil1 Dialer]# asterisk -r -x "soft hangup zap/27-1" -- Remote UNIX connection zap/27-1 is not a known channel Any ideas?

Hi, Ok, here is the setup. Asterisk conected to a PRI line (23 lines). 3 tool-free phone numbers are routed to this PRI line. Customer wants to have a way to have shown on the receptionist phone that the call comes from which of the 3 tool-free lines. Possibly display on the phone that the call comed from tool-free number 1, 2 ou 3 or even better a name or text id associated with this

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

Hi all! Every time i receive a sip call MOH begin to play and i can?t talk to the caller. My setup is the default. Someone knows what is the problem? thanks Miklos iPFONE Telefonia IP Rua Caio Graco 735 S?o Paulo SP iPBX +55 11 3801-3702 FWD 64662 ICH 31451543 www.ipfone.com.br info@ipfone.com.br -------------- next part -------------- An HTML attachment was scrubbed... URL: