similar to: Italy FastWeb problem: ISDN line crashes every time cisco router turns off

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

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'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 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, 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, 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

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

... 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

On 14/07/15 08:17, Steffen Kaiser wrote: > -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA1 > > On Mon, 13 Jul 2015, Gionatan Danti wrote: > >> On the other hand, private (per-user) sieve file works without >> interfering with hardlinks. In a similar manner, disabling sieve also >> permits dovecot to create multiple hardlinks for a single message. >> >>

Hello world! Please, help me :-) I have a APC RS-800 usb and it don't shutdown. I try every things but nothing: don't shutdown. Uhmmmm My actual situation: tux:/home/effem# uname -a Linux tux 2.6.26-1-686 #1 SMP Wed Aug 20 12:56:41 UTC 2008 i686 GNU/Linux tux:/home/effem# wajig policy nut nut: Installato: 2.2.2-6 Candidato: 2.2.2-6 Tabella versione: 2.2.2-7 0 -20

Hi Javier, thanks for your reply. I already checked SIS and, while interesting, is not what I want, because: 1) it can be difficult to restore a single message/attachment from a backup 2) only the attachments, and not the entire messages, are deduped. Message-based hardlinks really exists for a reason. The good news is that I found _why_ they are not working: it depends from how dovecot and

Hi, I configured an asterisk box with 1 Digium Wildcard TE110P T1/E1 Card 0 I setup the jumper in e1 position. my zaptel.conf : defaultzone=it loadzone=it #gestione PRI span=1,0,0,ccs,hdb3,crc4,yellow bchan=1-15 # set this to 1-15,17-31 for E1 dchan=16 # set this to 16 for E1 bchan=17-31 # set this to 1-15,17-31 for E1 # Asterisk starts correctly, I see th 30 channels. Anyway I cannot put

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

Hi, I'd like to use PSTN and analogic telephone with a Asterisk server which works on a LAN connected to the outside with fiber optic, on which voice stream passes too. Do I need some particular solution, some particular card to make it works or I just need a Digium or similar fxo card? Thanks, Bob __________________________________________________________________ Tiscali ADSL Senza Canone,

Calling any unreachable mobile from Fastweb or Telecom PSTN I don't get any courtesy message. The same happens when calling an inexistent number. I'm configuring two PBX's, connected to two different phone lines, both behave this way. Perhaps there's some missing zapata parameter? Regards, _fangi_ > Well, > > it's funny because here, now (Italy; Telecom Italia PSTN

Evnin... Googled around for this strange error meesage with no helpful results at all... Does somebody has any idea what this means? "Forcing Marker bit, because SSRC has changed" At the same time I only get inbound audio but other side can't hear me...sometimes I just hear my echo and nothing from other side... Asterisk version 1.2.9 and both participants with public IP