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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 -- ____________________________________________________________________ GIORGIO INCANTALUPO Tel....

...20:51:41 +0200 (CEST) X-Virus-Scanned: amavisd-new at gruppocrimi.it Received: from mail.gruppocrimi.it ([127.0.0.1]) by localhost (mail.gruppocrimi.it [127.0.0.1]) (amavisd-new, port 10024) with LMTP id NC3YcizeDFPO; Mon, 6 Jul 2015 20:51:40 +0200 (CEST) Received: from mr003msb.fastweb.it (mr003msb.fastweb.it [85.18.95.87]) by mail.gruppocrimi.it (Postfix) with ESMTP id 4380DA1A7C; Mon, 6 Jul 2015 20:51:40 +0200 (CEST) Received: from ceres.assyoma.it (93.63.55.57) by mr003msb.fastweb.it (8.5.140.03) id 55501C9F0432631D; Mon, 6 Jul 2015 20:51:40 +0200...

...ing: 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 http://debian.fastweb.it unstable/main Packages *** 2.2.2-6 0 -10 http://debian.fastweb.it testing/main Packages 100 /var/lib/dpkg/status 2.0.4-4 0 990 http://debian.fastweb.it etch/main Packages effem at tux:~$ lsusb [cut] Bus 001 Device 002: ID 051d:0002 American Power Conversion Bac...

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'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...

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

...b3,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 any call on the PSTN PRI Channel. Actually I don't know 1) ccs or cas ? my provider (Fastweb Italy) does not answer..... 2) ami or hdb3 ? same answer (no answer) 3) crc or not ? my provider says it does not matter moreover : what is supposed to show the zttool utility ? If I push "Loop" nothing happens. If I run ztcfg -v i see SPAN 1: CCS/HDB3 Build-out: 0 db (CSU)/0-133 feet...

...ly linked, BuildID[sha1]=ef07f9566623c84b92b85a37d173c1b02dfed727, stripped so it seems libpng16.so.16.21.0 is used (vrs 16, which "does not have" `PNG12_0'). I installed the following packages from the debian mirror listed here http://ftp.it.debian.org/ that is deb http://debian.fastweb.it/debian/ testing main contrib non-free l at m740n:~$ dpkg -l | grep "r-base" ii r-base-core 3.2.4-revised-1 amd64 GNU R core of statistical computation and graphics system ii r-base-dev 3.2.4-revised-1 all GNU R in...

hi, I've been running a pure sip asterisk box for a while now with no problems, and i've recently added an isdn2e line from bt in the uk. everything is hooked up and i've got it ringing my sip extensions, but the logs don't quite look perfect and i can't see any description of what i should consider to be normal behaviour. would someone be able to look this over and tell me

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