similar to: send text to a device

... Activating "sip debug" shows the register packets but nothing in return. ... I think that this is a network related issue, but you have to solve it by using a Asterisk config file. Unfortunately I think that the faster way to solve your problem is trying to understand if sip messages are correctly sent to tnet. I strongly suggest to use http://www.wireshark.org/ previoulsly named

tnet.itand SIP register messages are not replied. I suggested to check if your Asterisk box is really sending SIP messages, you can use a net sniffer. Did you alerady used different sip client with the same sip account of your Asterisk box? Did you use zoiper from the same box? Marino p.s. Are you Italian? On Mon, Jul 14, 2008 at 5:27 PM, Giorgio Incantalupo < gincantalupo at

Hi, I've upgraded my Asterisk to 1.0.7version patched with bristuff 0.2.0-RC8c. I'm using the same extensions.conf but it seems now include instruction doesn't want to work, here follows an extract: [inbound_menu] include => ins_exts exten => _X.,1,Answer exten => _X.,2,Wait(1) exten => _X.,3,Background(msg) exten => _X.,4,Background(3-sec-pause) exten =>

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 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

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

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, 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

Hi, is there anybody who knows what this warning means?? WARNING[27309]: chan_sip.c:8875 reload_config: Section '10' lacks type TIA Giorgio -- ____________________________________________________________________ GIORGIO INCANTALUPO Tel. +39 02 9350 4780 (104) FG&A Software 20017 Rho - Via Puccini, 8 E-Mail : gincantalupo@fgasoftware.com Internet: http://www.fgasoftware.com

Hi, I have an Asterisk 1.2.9.1 box connected to an ISDN line via a beronet card (with ports in PTP mode). I noticed a long delay when making outbound calls, more precisely between (taken from Asterisk CLI) "Called 1/XXXXXXXXX/s" and "mISDN/1-u43 is proceeding passing it to SIP/8-5486" I searched on misdn.org but found nothing. I'd like to understand if this delay is

There is no static interval. But i found out that it was my IP-Phone Service Provider that was having serviceproblems today. -----Opprinnelig melding----- Fra: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] P? vegne av Giorgio Incantalupo Sendt: 26. august 2005 11:33 Til: Asterisk Users Mailing List - Non-Commercial Discussion Emne: Re: [Asterisk-Users]

Hi, I live in Italy and I have an Asterisk 1.0.7 box with a HFC monoBRI card connected to my cisco router which is connected to FastWeb provider: does anybody knows why every time my cisco router turns off, my telephone connection to Fastweb drops (while internet connectior is ok)? Restarting Asterisk is worth nothing. TIA Giorgio --

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, 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 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, I'm getting crazy about ztdummy. I have to replicate a PBX where ztdummy is working fine but for some reason I cannot. The two machines have the same kernel, motherboard, the same gcc version and the same zaptel 1.4.8. On the second machine zaptel compiles without errors and ztdummy.ko is generated but when I modprobe it I get the following error in messages: rtc: lost some interrupts