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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 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, 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, 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, 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 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 --

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 Rosario, I have a problem about CDR: inbound calls are not correctly logged in CDR, it says they are always answered even if they are not. It is very strange since outbound calls and internal calls don't suffer this problem. I'll tell you more: I made Asterisk print the DIALSTATUS variable and it is ok, says BUSY when my internal hardphone SIP is busy. Or maybe it is allright and

Hi, I'm using a HFC card on my asterisk box. I tried to make a conference but it doesn't work. I read on internet to use ztdummy but my server has no uhci (only ohci but it doesn't work) so I cannot use it. I tried zaprtc but after loading the module (it appears when typing lsmod) nothing has changed. Should I buy a x100p to get the right timing? Or there is another solution? TIA

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

... Thanks, anyway for telling as at least, it reflects your needs. > > > You want NT PtMP and i second that, > not being limited on the asterisk > side is a must in the > telephony ecosystem, since the legacy PABX aren't alwsys easy to > reconfigure. > > _______________________________________________ > -- Bandwidth and Colocation Provided by

such file or directory" on pure-IP platform in which I installed asterisk-libpri-dahdi trilogy. Maybe, it's me while following README instructions, maybe README instructions could be improved or maybe it's wrongly labeled messages ? That's why I told myself : I'm waiting too much from doc ? is a pure-IP platform too specific ? what is the official policy ? README starts with

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. >

That's the main reason I opened this thread as it surprised me a bit ... > > > Any 2-wire analog leg will be a source of echo. Many, many, many calls > do not have a 2-wire leg. Even in handset audio circuit ? I was thinking that any handset is a potential echo source due to this audio circuit ... Do you agree ? > Think cell/mobile or endpoints with PRI or T-1. > >

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

ast_waitfordigit that accepts milliseconds as input. Douglas Garstang wrote: > Admittedly I have not used the ExternalIVR app. Is it any good? > > I'm not sure I agree that Asterisk is GOOD for building IVR's. Sure, > it can do it, but boy it is UGLY. There's also the fact that you can't > call Backgound() in a macro, which forces you to use Read() which >

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, 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

might be your best bet to get the information you want. I'd look at voip-info.org for information. _____ From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Olivier Sent: Wednesday, September 16, 2009 9:20 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] How to list ongoing calls