similar to: HUD Lite server on Debian

Hi all, since Asterisk 1.4 seems to have too many differences from previous versions, wouldn't be nice to have a new mailing list? Giorgio Incantalupo

pbx18*CLI> module load codec_g729-ast14-gcc4-glibc-pentium3.so Unable to load module codec_g729-ast14-gcc4-glibc-pentium3.so Command 'module load codec_g729-ast14-gcc4-glibc-pentium3.so ' failed. [Dec 22 15:52:45] WARNING[4491]: loader.c:757 inspect_module: Module 'codec_g729-ast14-gcc4-glibc-pentium3.so' does not provide a license key. [Dec 22 15:52:45] WARNING[4491]:

Hi, we have an asterisk 1.2.1 box and 2 polycom SIP phones. We'd like to show the called name on the calling polycom display instead of his /her extensions as I do with the caller name on the called polycom. Is it possible? If yes, how? TIA Giorgio Incantalupo

This is a function of the Phone itself. Asterisk has nothing to do with it as it does not know anything about the call until after the SIP device 'sends' it. To my knowledge it is not posible. I don't even think a SIP standard is available for this. This 'feature' along with changing CallerID Display after a call has been answered is something missing from the RFC. >

Hi, I'd like to build a personal PBX connecting 4 or 5 analogic phones with a ADSL line and I'd like to know what is the right card I need I visited digium site and I think TDM400 could be the right choice but I cannot understand how it works...I think it has 4 slots where 4 modules (FXS or FXO) can be inserted. How many cards do I need to connect my ADSL line to 5 phones? I think I

Hi, I am trying to use zaptel module on an Ubuntu 5.10 distro (2.6.x kernel) using gcc 4.0.2. Compilation does not give me errors so after a 'make install' I try to load zaptel module with insmod but the following error arise: *insmod: error inserting '/lib/modules/2.6.12/misc/zaptel.ko': -1 Invalid module format* Is there anybody who can help me?? TIA Giorgio --

Hi, Is it possible for a Manager/Supervisor to intercept and listen in on live calls for training and evaluation purposes? Thanks -- ______________________________________________ http://www.linuxmail.org/ Now with e-mail forwarding for only US$5.95/yr Powered by Outblaze

Afternoon, I was hoping someone could point me in the right direction. I have an asterisk PBX deployed in China using a TDM400P based card. The incoming calls are being picked up correctly, but are not being hung up. I suspect that this might be an issue with the signaling that has been selected. If anyone here has deployed asterisk in china using an analog card, it would be a great help

Hi, does anybody know what *zttranscode *module* *is for*?* Thanks!! Giorgio -- _________________________________________________ Giorgio Incantalupo, mailto:gincantalupo@fgasoftware.com FG&A srl - http://www.fgasoftware.com - Voice@Work - The Agile PBX http://www.voiceatwork.eu Tel: 02997663.14, Fax: 0291390172

Hi, every second I get on the console: Remote UNIX connection Remote UNIX disconnected which gives no problem but makes console unusable. Is there anybody who has encountered the same problem? How did you solve it? TIA Giorgio Incantalupo

Hi all, I need to connect my Asterisk 1.4.26 with a Sangoma PRI card (configures with signalling=pri_net)) to a Cisco 2800 PBX. After connecting the cables everything seems fine (ifconfig w2g1 is ok, wanpipemonitor gives no errros, the span is up and active, green light on the card) but when I make a test with my iax phone, there's no way to dial the PBX and I get this WARNING: [Jul 2

Hi, is it possible to force FOP to reload its configuration files (op_buttons.cfg and op_style.cfg) while it is working? I tried to click on the refresh icon but nothing happens. TIA Giorgio Incantalupo

Hi, I'm trying to make native music on hold work on my Asterisk 1.2.9.1 server with a Sangoma PRI card. If I use a IAX phone connected to the PBX, I hear the music, but if I make a call from outside I hear nothing even if Asterisk console says music has started... it seems something related to zapata.conf but I cannot understand what's wrong. I also put musiconhold=native for every

Hi, I'm installing an Asterisk box with a TDM2400P in Brazil. I can make analog phones work while lines are not working. Since I do not know anything about brazilian lines, is there anybody who can tell me what is wrong/missing in my conf files (below)? TIA Giorgio _zaptel.conf:_ fxoks=9-16 fxsks=17-24 defaultzone=br loadzone=br* * _zapata.conf:_ context = inbound_zap echocancel = 128

Hi, I'm using Asterisk 1.2.1 on Sarge. it seems like if I call a phone and nobody answers, asterisk does not jump to the next priority...it freezes. Take a look at this: exten => 777,1,NoOp(before) exten => 777,2,Dial(SIP/7|60|g) exten => 777,3,NoOp(after) priority 3 is never executed but this worked with Asterisk 1.0.7!!! TIA Giorgio Incantalupo

Hi, is it possible to remove the "no timeout" combo box in flash operator panel? How can I reduce the flash area? I set small buttons and half of the area is white and I want to resize it. TIA Giorgio Incantalupo

Hi, I'm using Asterisk 1.2.1 on a Debian Sarge with a TDM400P and a monoBRI using chan-mISDN from beronet site. It seems to work all right except for autodial calls, monoBRI ISDN channel behaves differently waiting for the caller to answer and then continue. Asterisk console says: analog: -- Attempting call on Zap/2/3391818250 for 104@inbound_originate:1 (Retry 1) > Channel

Hi, I get a lot of these messages on my Asterisk CLI: "Failed to authenticate user 1000<sip:1000 at MY_OWN_IP_ADDRESS>;tag=03f82bb9" as if my PBX machine is trying to authenticate to itself. It seems someone is attacking my asterisk PBX. Is there a way to fix this problem? Thank you. Giorgio Incantalupo

Hi, I'm using asterisk 1.2.1 on a debian sarge distro. I've followed notes in http://www.voip-info.org/wiki/view/Polycom+auto-answer+config and http://www.voip-info.org/wiki/index.php?page=OptiPoint+600+SIP+-+Distictive+ring+using+ALERT_INFO but I still cannot change ring style via asterisk using exten => 666,1,SipAddHeader(ALERT_INFO="ring3") in extensions.conf . Is it

Hi all, When I do a test call into the box (which is running latest version of Trixbox) it all goes fine. If i decide to hangup the cellphone (during the ivr playing options) the system does not recognize the hangup and the system continues through and ends up at the timeout option. What settings do I need to change to fix this. Is it the rxgain? If so is there something i can use to figure