Displaying 20 results from an estimated 400 matches similar to: "ERROR[8192]"
2004 Jun 16
2
embedded Asterisk
Hi All,
I have a thin cliente here that i want to run asterisk:
- National Semicondudor Geode GX1 266MHz Geode 266MHz single chip
- NS Cx5530a Southbridge National Semiconductors SC2200
- NS PC97317 in chipset
- 32MB Compact Flash
- 64MB Ram
- 10/100Mbps, Autosense 10/100Mbps, Autosense Realtek 8139C National DP83815 / DP83816
Some tip?
I have a ide>flash
2003 Aug 04
14
Mysql CDR
hello all,
I am using the msql cdr module to store cdr in db, I realised that it does't capture the start and end time af a particular call record.
Therefore I dive into the source code to add the start and end time into the query (add something like cdr->start, cdr->end), but end up getting segfault.
the original version of cdr_mysql.so works fine but I need the start time and end
2003 Oct 31
2
MOH problem
Hi all!
Every time i receive a sip call MOH begin to play and i can?t talk to the caller.
My setup is the default.
Someone knows what is the problem?
thanks
Miklos
iPFONE Telefonia IP
Rua Caio Graco 735 S?o Paulo SP
iPBX +55 11 3801-3702
FWD 64662
ICH 31451543
www.ipfone.com.br
info@ipfone.com.br
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2003 Sep 23
2
error message playing .mp3
> -----Original Message-----
> From: listas iPfone [mailto:listas@ipfone.com.br]
>
> Somebody knows why asterisk gives me that error wile playing .mp3
files?
>
> The files play well but the message aperas any way:
> WARNING[131089]: File format_mp3.c, Line 120 (mp3_read): Short read (0
of
> 4
> bytes) (No such file or directory)!
Listas,
You might try down-sampling
2005 May 26
1
VIDEO ON 1.0.7 stable
--- listas iPfone <listas@ipfone.com.br> wrote:
> Hi all
>
> I need to know if the video support for h.263 is
> active in version stable
> 1.0.7 to use with eyeBeam in asterisk
it works for me...
[2222]
type=friend
secret=xxxx
auth=md5
callerid="myCallerId" <2222>
canreinvite=no
host=dynamic
disallow=all
context=default
allow=alaw
allow=ulaw
allow=speex
2008 Dec 22
1
Web-driven SIP call thru Asterisk IPBX
Hi,
I think that the web-driven SIP Phone (free) doddle (beta version) can be useful with your Asterisk applications.
You can pre-fill it with your sip settings (Asterisk host name or IP /?realm / sip user), you just need to setup the HTML link as that: (Attached is the HTML page example)
?
/**************************/
simple HTML code example:
/*************************/
<html>
<head>
2003 Dec 08
2
snom X MOH
Hi all!
I updated my snom200 to 2.02t and now MOH from * don?t works anymore... only the MOH from snom server and if i clear the MOH server field in the phone i have no MOH at all..( with the transfer button, moh plays using a extension).
Someone with that problem?
I downgrade to 2.01s but nothing changes.
Miklos
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2004 Jan 23
6
rc.local dont works
Hi All
I have a problem with initialization of asterisk using my rc.local file. when i call asterisk from the prompt it works well but don?t in the initialization...
I have in my file that comands:
touch /var/lock/subsys/local
modprobe zaptel
modprobe wcfxo
safe_asterisk
I read in somewere that it can be an interrup problem and i use the cat proc/interrupt to see what is happening
Somebody
2003 Nov 27
13
Asterisk behind NAT << How to do it.
Thanks to ww and his patch on bug #104, I have successfully implemented
Asterisk behind NAT without using STUN or anything crazy. It's quite
straight forward.
Until this gets tested enough and put into CVS, you will have to patch
your chan_sip.c file to do this. I'm sure within the next few days this
will get put merged into CVS if no one finds any problems.
I tried this on chan_sip.c
2003 Oct 08
7
chan_capi and latest Debian package
After
apt-get update && apt-get upgrade -y
wget http://www.junghanns.net/asterisk/downloads/chan_capi.0.2.5c.tar.gz
tar xfvz chan_capi.02.5c.tar.gz
cd chan_capi-0.2.5c
make && make install
shutdown -r now
asterisk seg faults upon calling in via ISDN.
Any ideas are greatly appreciated.
rgds
pos
2008 Feb 14
4
domain name display issue in linux pc
Hi,
Thanks for your response on the kernel switching.I was away and could not reply immediately.
Right now, I am facing a differentissue. I have to set up DNS server using BIND on Centos 4.3. When Itype the hostname on Centos, it shows:
sipserver.vodcalocal.com
But the cli prompt has root at sipserver~ meaning only the sipserver part of the hostname is displayed. whyis this so? What is the
2003 Oct 06
6
Alternatives to FXS cards?
Hi everyone,
I know someone makes a product that's a POTS phone to SIP converter,
where you just plug your POTS phone in one side and the network cable in
the other. Has anyone successfully used any of these with Asterisk, and
if so how expensive were they?
I ask partly out of frustration with the FXS cards but mostly because it
would make installation MUCH easier for what we're
2004 Jan 26
0
Digium FXO Card
Hi,
I wish to know if GNUGk can work with * running as a gateway with the Digium
FXO card.
Kindly share your experiences in case there are some issues which one must
know before going in for such a setup.
Also, I've been reading about the DialTone detection capability by the
hardware in different countries. What are the issues with it?
Thanks & Regards,
Deepak
----- Original Message
2005 Oct 09
1
Problem setting SIP incoming/outgoing
Hi
I am a newbie to * and I am having a problem which appears strange as I did
not find any mention of it anywhere in my search.
Simply speaking, I have an external SIP proxy server which I am trying to
configure for incoming and outgoing calls from my asterisk installation. So
here is my configuration in sip.conf
[general]
register =>
2008 Jul 22
1
issue with high latency
Hi,
Is there a specific latency that asterisk accepts? I encountered a
problem wherein when the latency was unusually high,my xlite's (i have 2
xlite) cannot register. but when the link suddenly went stable, the
x-lite just registered. what i forgot to look at is if the registration
packet is reaching my asterisks.
------ when xlite cannot register ---------------
Pinging
2004 Jan 22
2
Polycom Reboot Script - Please wiki-size me
With my thanks to Brian West and his offering in the thread,
"Subject: Re: [Asterisk-Users] Remote reload Cisco 7960"
I offer PolyReboot.pl, a perl script for rebooting Polycom IP Phones
PolyReboot.pl takes an IP address as a single argument and reboots the
phone.
You must have a cfg file in the Polycom style, i.e., 00ab00cd00ef.cfg - all
lower case. Further,
you need to use ftp for
2005 Jan 13
3
SER vs Asterisk for SIP
Why is SER considered a better SIPserver than asterisk , why is it that SER
can handle more clients than asterisk can. And if this is just cause of say
poor SIP handling code in asterisk then is there anything being done to fix
it. Just wanted to know why SER claims to be better than asterisk as a SIP
server. ?
--
regards
Vikram (http://www.vicramresearch.com)
2009 Nov 07
1
Trouble registering Cisco 7942
I'm trying to connect a Cisco 7942 to my Asterisk box. I have a 7960
and 7912 currently connected and functioning. I'm trying to use the
recommendations from here:
http://www.voip-info.org/wiki/view/Asterisk+phone+cisco+79x1+xml+configuration+files+for+SIP
I have created a "XMLDefault.cnf.xml" and it took the latest image but
the phone states it's unprovisioned? Any
2004 Nov 29
1
Polycom Reboot Script PRI errors!!
Kevin wrote:
> There is a reboot script posted on the wiki to reboot Polycom
> telephones. When I execute this script, I get the following messages.
> I am concerned as this is causing issues with asterisk and the PRI.
> Does anyone have any ideas why this would be happening?
>
>
>
> asterisk console:
>
> -- Remote UNIX connection
> -- Remote UNIX
2003 Oct 02
3
Xten Lite Build 1079
I've just down loaded Xten Lite and it is now build 1079.
It now finds the NAT firewall type and has loads more to configure.
But it doesn't work on my poor W95 tablet PC.
--
Dave Cotton
Directeur
Linux Autrement
193 rue Marcel Cerdan
84270 Vedene
04 90 23 30 81
"Internet Sheriff Technology" revendeur en France
<http://www.linuxautrement.com>
IAX 17004902330