similar to: Conference and HFC card conflict: no solution??

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 --

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 --

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, 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'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, 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

... 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'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

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

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. >

I have a need to have the two sip phones register with the same extension (at least I think I have the need :) A client wants an incoming call to ring at the receptionists desk and also at their desk. If the receptionist is in it will be answered there and put on hold followed by a "Joe, you have a call on line 1". Is there a way to do this w/ asterisk? I've played with two

Good day all Is there a fix for these errors yet for the junghanns cards Jul 26 17:15:05 pbx1 kernel: qozap: dropped audio card 1 cardid 0 bytes 6 z1 1 z2 107 Jul 26 17:15:18 pbx1 kernel: qozap: dropped audio card 1 cardid 0 bytes 6 z1 10 z2 116 Jul 26 17:15:18 pbx1 kernel: qozap: dropped audio card 1 cardid 0 bytes 5 z1 118 z2 97 Jul 26 17:16:33 pbx1 kernel: qozap: dropped audio card 1 cardid

I often get a Maximum retries error while making outgoing calls. Why does this happend? Most of the time a reload solves the problem, but not all the time? What to do? Aug 26 09:52:46 WARNING[6613]: chan_sip.c:694 retrans_pkt: Maximum retries exceeded on call 32166-1B75-151E-E6DA-E37AEEAA2882@10.100.4.252 for seqno 1 (Non-critical Response) Regards, Arne Morten.

I cannot reach voip-info - is it just me or is the site not available ? Julian

Guys. What do you think about Dell hardware and Asterisk? Whos using it, comments, any special specs recommended or models?

hi, i am trying to do autoattendant but failing. as in the manual i inserted the background(welcome-mainmenu) file so that after the sound the caller can dial the extension he wants to call. i figured that the background sound wasn't coming in the asterisk. how do we do this without first loading the welcome message? for example after certain rings the caller can dial the extension no to