Displaying 20 results from an estimated 4000 matches similar to: "Excessive Asterisk delay to answer on ZAP inboundcall"
2006 Apr 26
4
Excessive Asterisk delay to answer on ZAP inbound call
Hi,
I have an asterisk 1.2.1 on a Debian Sarge distro with *three* TDM400P
(12 fxo ports). I noticed Asterisk is slow to answer inbound calls so I
connected an analog phone in parallel to make a test:
__________Asterisk fxo
---- line -----|
-----------------Analog phone
The analog phone rings immediately when calling, while asterisk shows
the message
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 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
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
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
2006 Oct 18
2
random one way audio and noise betweenSIP phoneson same LAN
Giorgio,
I'll answer in reverse order:
I've not had reports of "noise" from my users. However, when I went down to
get the s/w version from the phone that seems to be acting up the most, the
user reported that earlier they were actually on a call that was ok then
spontaneously dropped the audio. Per my instructions (based on another
similar report I read on Digium's site),
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
--
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
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
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 Oct 18
2
random one way audio and noise between SIP phoneson same LAN
I'm having the same "random" problem.
I have "canreinvite=no" on all extensions. I have "qualify => yes" on all
non-NAT extensions. I do have several NAT extensions, but I've not had
reports of problems from those. 95% of my extensions (all polycom 501/601)
are on a brand-new network comprised of 2 48-port Cisco 3560 1GB switches.
In all cases, the
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
2006 Oct 16
4
Remote UNIX connection, Remote UNIX disconnected displayed every second
Hi,
every second I get on the console:
Remote UNIX connection
Remote UNIX disconnected
which gives no problem but makes console unusable.
Is there anybody who has encountered the same problem? How did you solve it?
TIA
Giorgio Incantalupo
2006 Jan 27
2
WARNING: chan_sip.c:3470 process_sdp: Unknown SDP media type in offer: image 5004 udptl t38
Hi,
I'm using asterisk 1.2.1.
Is there anybody out there who knows what this warning means?
*WARNING: chan_sip.c:3470 process_sdp: Unknown SDP media type in offer:
image 5004 udptl t38*
Google does not help at all.
TIA
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.
>
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
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
--
2006 Mar 29
3
FOP flash panel: how to reload config files when running
Hi,
is it possible to force FOP to reload its configuration files
(op_buttons.cfg and op_style.cfg) while it is working? I tried to click
on the refresh icon but nothing happens.
TIA
Giorgio Incantalupo
2006 May 25
2
connecting asterisk to hylafax via t38modem: is it possible?
Hi,
I'm trying to use Hylafax without a modem. Is it possible to use
t38modem to make Hylafax send and receive fax via Asterisk?
If yes, how? I'm searching on internet but still haven't found anything
useful.
TIA
GIorgio Incantalupo
2006 Dec 06
3
Asterisk freezes when DNS not working: a BUG??
Hi,
I'm using Asterisk 1.2.9.1. I have big problem with SIP VoIP providers
registrations: Asterisk freezes when it cannot (re-)register with VoIP
provider (registration timeout). The problem is related to DNS names
resolution: if DNS server is very slow to respond Asterisk stops every
activity (no zap or restart commands on CLI). The bad news is VoIP
providers usually do not give their IP