Displaying 20 results from an estimated 100000 matches similar to: "Sieve filter doesn't respect mailbox separator"
2007 Nov 16
4
Samba Upgrade on Centos 3
After today's samba update, Centos 3 boxes can not use samba to
communicate with each other, although Windows and the Centos 3 boxes
see each other correctly as do RHEL5 and the Centos3 boxes. The du
command works well, but ls, cp, cat ,etc produces the error:
PANIC: push_ascii - dest_len == -1
in the server log and
smb_trans2_request: result=-5, setting invalid
in the client.
A similar
2004 Aug 06
4
Icecast in Macromedia Flash
> on linux - tcpdump
> on win32 - ethereal
Ok. I've made the network dumps. You can find 'em at
http://www.sardegnaoggi.it/downloads/tcpdump.zip
Please read the readme file for more info.
Feel free to write me a line if you need anything.
BTW
For as much as I can undestand (very few indeed) icecast 2.0 server
was sending data even on the unsuccessful test.
--- >8 ----
List
2018 Mar 14
3
Sieve not working.
<!DOCTYPE html PUBLIC "-//W3C//DTD XHTML 1.0 Transitional//EN"
"http://www.w3.org/TR/xhtml1/DTD/xhtml1-transitional.dtd">
<html xmlns="http://www.w3.org/1999/xhtml" xml:lang="en" lang="en"><head>
<title></title>
<meta http-equiv="content-type" content="text/html;charset=utf-8"/>
2005 Mar 29
3
help w/ basics
Hello, I am new to Asterisk and new to this list. I got Asterisk setup and
running using Asterisk@home, and purchased a PolyCom SoundPoint IP500 phone
to test out.
I cannot get the phone to talk to the Asterisk box. On bootup of the phone,
it tells me that it cannot contact boot server. Why is that? It gets an IP
fine, and I have also tried manually setting the IP of the phone and the
Asterisk
2012 Feb 07
10
Ruby Developer position
Please let me know your interest in following.
Location: Columbia, SC
Duration: 12 months+
Rate: $65/hr 1099/c2c
Required Skills:
RUBY, RAILS, GIT, MYSQL, CUCUMBER, RSPEC, JQUERY, EXCELLENT ORAL AND WRITTEN
COMMUNICATION SKILLS, TEST-DRIVEN DEVELOPMENT, LINUX, OS X, JSON, COMMAND
LINE, SQL, SSH, HAML, SCSS
Thanks
Sandeep
Sandeep Jain
Software People Inc.
www.softwarepeople.us
2004 Jun 11
2
Asterisk PRI calls to SER problem
Hi all,
I need help. I have a Linux box with SER as a proxy server with ip phones
attached on it , and another linux box with Asterisk and T410 card connect
to an E1 line .Whenever there is a call from PSTN it is passed to Asterisk
and then to SER box and then to the phone .every time an invalid number
dialed from PSTN to SIP phones connected to SER asterisk says
that the call is progressing
2018 Jan 23
2
LAYOUT=fs
<!DOCTYPE html PUBLIC "-//W3C//DTD XHTML 1.0 Transitional//EN"
"http://www.w3.org/TR/xhtml1/DTD/xhtml1-transitional.dtd">
<html xmlns="http://www.w3.org/1999/xhtml" xml:lang="en" lang="en"><head>
<title></title>
<meta http-equiv="content-type" content="text/html;charset=utf-8"/>
2018 Jan 21
2
LAYOUT=fs
<!DOCTYPE html PUBLIC "-//W3C//DTD XHTML 1.0 Transitional//EN"
"http://www.w3.org/TR/xhtml1/DTD/xhtml1-transitional.dtd">
<html xmlns="http://www.w3.org/1999/xhtml" xml:lang="en" lang="en"><head>
<title></title>
<meta http-equiv="content-type" content="text/html;charset=utf-8"/>
2005 Aug 05
1
Problem running Astrerisk after defined card in /etc/asterisk/zapatal.conf
I have configured /etc/asterisk/zapata.conf, but now Asterisk refuses to start:
Aug 5 10:47:29 ERROR[1076842624]: chan_zap.c:5976 mkintf: Unable to get parameters
Aug 5 10:47:29 ERROR[1076842624]: chan_zap.c:9478 setup_zap: Unable to register channel '1-15'
Aug 5 10:47:29 WARNING[1076842624]: loader.c:328 ast_load_resource: chan_zap.so: load_module failed, returning -1
== Unregistered
2007 Dec 10
2
asterisk linkedin group
asterisk linkedin group
I have created an asterisk linkedin group for anyone interested.
http://www.linkedin.com/e/gis/45252/66270A773F53
Thank You,
Steven BerkHolz
- MCSA - MCSE -
Manager of Information Systems
HIROTEC AMERICA
Board member of
Connectech Greater Detroit
www.connectech.org
________________________________
Please visit us on the web at www.hirotecamerica.com
HIROTEC AMERICA Ph.
2006 May 10
2
REPOST: features.conf *1 Call Recording
Hi all. I posted this earlier but never got any advice that helped. If
anyone knows how to get this going, I'd appreciate some advice.
I am attempting to setup Asterisk to allow me to press *1 while in a
call to use automon to record the call but have had absolutely no
success. Is there a trick to this?
In extensions.conf
[globals]
DYNAMIC_FEATURES=>automon
[default]
exten =>
2002 Sep 27
8
Longer synonym for R?
[This message is not always serious, but it addresses a real problem.]
A problem with R's present name is that it is not very well suited to
search engines, although Google at least seems to cope as long as you have
sufficient other terms to filter out Toys R Us. Initials remain a problem.
I suppose "R" is just too culty to do away with, but perhaps we could come
up with a longer
2009 May 26
2
Domains
Hi,
I'm trying to understand an issue I'm seeing between two Asterisk
servers. I think it has to do with Domain definitions.
Server A), has extension 5550 defined. Has a sip client 2000 defined,
and has guest-invites enabled.
Server B), Dials to server A for any 5550 dialled. Has sip client 2000
and 2001 defined.
If I register at server B as client 2001, and dial 5550 then
2010 Mar 03
6
Identify scripts connecting to the asterisk manager
Is there any easy way to identify which script or service is
connecting to the Asterisk manager? Somewhere on my system a script or
service is trying to connect with a bad user name or password. I get
the following error: connect attempt from '127.0.0.1' unable to
authenticate
I thought maybe I could do a tcpdump on port 5038 and try to fish out
the bad username or password but I
2006 Mar 23
1
spam filtering with amavis
I'm filtering that is being deliverd to postfix mail server with
amavisd-new .
I want spam with spam f level 1 - 8 to ad a tag any everything above to
be delete is this posebol?
If yes how?
Met vriendelijk groet,
Bas van Dikkenberg
GISkit bv
BFVD1-RIPE
Tel: +3130-6340430
Fax: +3130-6342433
Prive Tel: +3130-6372769
Mob:
2007 Nov 08
3
Asterisk as a SIP to XMPP Jingle voice gateway
Hello,
I'm looking for a SIP to XMPP Jingle voice gateway.
I see that Asterisk has Jabber and Jingle support, but it looks like Asterisk acts as a Jabber client.
Are there any Jabber server solutions, where Jabber users can call SIP users by using the SIP URI and vice versa?
--
Eric Chamberlain, CISSP
Chief Technical Officer
Voxilla - http://voxilla.com/
2004 Sep 21
2
SIP termination in Brazil
Is there an up and running provider of SIP termination in Brazil?
I know that there are some people building on a SIP termination solution.
But who as it up and running ?
Best regards,
Han
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2005 Aug 22
1
Hangup Faster
Hello -
My single line extension users (connected via channel banks) need to be
able to hang up faster. If they just flash the hook it doesn't
disconnect right away. Any ideas on how to resolve this?
Thanks,
Dave
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2008 Jan 12
2
Asterisk RFC2833 to SIP INFO DTMF conversion erros.
Hi,
I am using asterisk 1.4.17 which is connected to a SIP trunk supporting
rfc2833 dtmf events. Asterisk stays in the media path. In sip.conf I have
set dtmfmode=rfc2833 for the outbound sip proxy (SIP Trunk account) and for
SIP clients I have set dtmfmode=info. So when I make a call to a cell number
using the sip trunk and then press digits I can see the 2833 dtmf events
coming to asterisk
2004 Jan 02
6
hangup detection
So I made the mistake of buying a Carrier Access channel bank without
noticing the page on the wiki about the fact that they don't support
disconnect supervision (bastards!). However, apart from that, I do have
it working fine for incoming calls.
Is there some trick to get asterisk to detect the hangup tones from
SBC? I've tried busydetect and callprogress as suggested, but neither