Displaying 20 results from an estimated 3000 matches similar to: "beronet DTMF detection problem with some Telecom Italy lines"
2007 May 18
0
mISDN: long delay when making outbound calls
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
2006 Jun 29
1
beronet BNS40 led blinking: not working or not connected?
Hi,
I've just installed a beronet BNS40 on Asterisk 1.2.9.1. Everything
seems ok, asterisk gives no error (nothing inside logs) but the 4 led on
the back of the card (which is NOT connected to an ISDN line) are red
and flashing....what does it mean? Is it not properly working or it
means the card is not connected to any ISDN line? The card handbook says
the card has red led but not their
2006 May 04
1
TDM400P and monoBRI auto-dial call difference: caller phone does not ring
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
2005 Sep 02
1
Italy FastWeb problem: ISDN line crashes every time cisco router turns off
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
--
2007 Jul 12
0
No subject
...
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
2007 Jul 12
0
No subject
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
2007 Feb 22
0
cannot get whole DNID with ISDN line
Hi,
I have an Asterisk 1.2.9.1 with chan_misdn 0.3.1-rc23 and a beronet
octoBRI on a Debian box I have to set up instead of an old legacy PBX.
My problem is I get only the base DNID and not the extensions (the last
two digits) in Asterisk but the old PBX got all the DNID number so I
think it is the card.
Is there anybody experiencing a problem like this? How can I solve it?
Any ideas?
TIA
2005 Oct 17
1
module loading error with Ubuntu: insmod: error inserting '/lib/modules/2.6.12/misc/zaptel.ko': -1 Invalid module format
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
--
2007 May 10
1
module zttranscode: what is it?
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
2007 Mar 16
4
proposal: a new mailing list for asterisk 1.4, why not?
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
2006 Mar 15
5
how to show called name on calling polycomdisplay
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.
>
2005 Aug 26
0
SV: Maximum retries error.
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]
2006 Mar 15
3
how to show called name on calling polycom display
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
2004 Dec 21
2
SOHO PBX using asterisk
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
2007 Apr 26
1
asterisk slows down when unplugging internet cable with VoIP lines
Hi,
I have an Asterisk 1.2.9.1 on a Debian Sarge distro connected to a VoIP
provider via internet.
I noticed Asterisk gets slow and behaves in strange manner if I unplug
my internet cable from the PBX: for example I get incoming calls after
seconds or I get no audio during calls.
I thought it was something connected to DNS resolution so I put VoIP
provider addresses inside /etc/hosts but
2006 Nov 27
0
BRIcard not sending DTMF
Hi,
I'm getting crazy about a DTMF problem. I have an Asterisk 1.2.9.1 box
with 2.6.18 kernel using a beronet BRI card who refuses to send DTMF.
I noticed there are a lot of parameters for tuning DTMF detection but
not for sending..is there a good guy out there who knows what parameter
to enable to send DTMF?
TIA
Giorgio Incantalupo
2005 Aug 26
2
WARNING[27309]: chan_sip.c:8875 reload_config: Section '10' lacks type
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
2010 Dec 22
5
* 1.8: cannot load g729 free codec (on 1.4 it worked!)
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]:
2010 Jul 02
1
asterisk and cisco 2800
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
2013 Oct 01
1
Failed to authenticate user 1000<sip:1000@MY_OWN_IP_ADDRESS>; tag=03f82bb9
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