similar to: SIP phone status

Prepend *67 if your carrier allows it Thanks, Steve Totaro http://www.asteriskhelpdesk.com > -----Original Message----- > From: Andre Courchesne - Consultant [mailto:courchea@net-forces.com] > Sent: Thursday, April 13, 2006 12:02 PM > To: asterisk-users@lists.digium.com > Subject: [Asterisk-Users] Display "Confideltial" or "unknown" on called > iddisplay

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 has good/bad experience with SIP providers in upstate NY? Any recommendations of such provider who works great with Asterisk? 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

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

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

Hello - I have just purchased an Aastra 9133i SIP phone for testing with Asterisk. Its a little flakey but overall is a far superior phone to the others in the $179 range. I have an issue regarding the message waiting indicator. The phone does not seem to respond to the "NOTIFY" command from Asterisk. Searching archives seems to indicate that this was previously an issue on the 480i

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 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

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

I upgraded my office server from 1.4.22 to 1.6.0.5 on the weekend and after some testing there seems to be a compatibility problem when using Aastra phones. With 1.6.0.5 all incoming calls to all Aastra phones were dropped after a minute or so. I installed 1.6.1-rc1 and now the newer Aastra phones (5xi) work properly. The problem remains with the older phones (9112i, 9133i and 480i). If I dial

Ok, So I just installed some aastra 9133i phones. They work great. One small problem. When I do an 'all page' anyone who is on the phone gets their call placed on hold for the duration of the page! THIS IS NOT GOOD! How do I make the page only go to people who are not on the phone, and ignore people who are on the phone?

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

Actually, I have tested this here with an Aastra 9133i and an Asterisk@Home server, and the 9133i will re-subscribe on its own after an Asterisk reboot, if you wait long enough. It took on the order of an hour to do so. Of course, a phone reboot will get it done faster, if necessary, but it _will_ eventually re-subscribe on its own. In another thread, I've seen a response that the GXP2000

Hello. Just rx'd the sip - aastra 9133i. Haven't done sip before. My initial attempt at setup has failed. "No Service" Anyone want to contact me off-list or on irc ? Regards...Martin

Fellow list members, I have run into an issue where I encounter a delay at the beginning of a phone conversation when I make outgoing calls through Asterisk with an Aastra 9133i hardphone. This is most noticable when I call a voicemail system with the Aasta and then with a land line or other VoIP phone. The first word or two of the voicemail message is generally cut off. According to

I am having a strange problem with an external SIP phone. It can register and receive calls but it cannot initiate any calls. A softphone on the same network works without problems. As far as I can notice the difference is that the hard phone is not sending the proper contact info. In the fullcontact field I can see its private IP address "sip:1008 at

Hi, 2013/4/19 Olivier <oza_4h07 at yahoo.fr> > Hello, > I've just realized that several phones display both caller name and number > while ringing but when on call, caller name is not displayed anymore. > Could you recommend a sip phone that still displays caller name during > phone call ? > Regards > I've been testing Aastra 6757i SIP phone and it appears

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