Displaying 20 results from an estimated 400 matches similar to: "Asterisk is not designed for University with large user base?"
2009 Jan 22
1
oslec + dahdi
Hi list, I install dahdi-linux successfully with the module of oslec
for the echo, but when I specify it in the system.conf the echo
canceller oslec it shows me errors:
DAHDI_ATTACH_ECHOCAN failed on channel 4: Invalid argument (22)
I see that the echo cancellers is supported: mg2, kb1, sec2, and sec
because oslec is not supported?, but he has support to compile it with
dahdi_linux!
best
2009 Apr 08
1
Call Pickup Works w/Linksys ATA, not with Cisco 7940G
<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN">
<html>
<head>
<meta content="text/html;charset=ISO-8859-1" http-equiv="Content-Type">
</head>
<body bgcolor="#ffffff" text="#000000">
I have an Asterisk 1.4.18 with a mix of cordless phones connected using
Linksys SPA2102 ATAs and Cisco 7940G
2008 Jun 12
2
Reg. setting Domain name on Cento 5 pc
Hi all,
I am running centos 5.1 and I wish to change the domain name and
dnsdomainname of my PC. currently the settings are--
$ hostname
sipx.com
$ hostname --fqdn
sipx.com
$ domainname
(none)
$ dnsdomainname
com
I have searched in the net for tips but everywhere only the hostname change
is provided. I need to change/set the domain name and the dnsdomain name on
my pc to sipx.com and this
2007 May 02
1
SIP Proxy
Hi all,
I want to deploy a SIP Proxy but I just don't know which one to choose.
Researching in the Internet I found the following ones:
* SIP Express Router
<http://www.voip-info.org/wiki/view/SIP+Express+Router>: SER is
used by many SIP providers standalone or in conjunction with Asterisk
* Vovida.org <http://www.voip-info.org/wiki/view/Vovida.org>
* sipX
2012 Jan 03
4
Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE.
Hi,
Please help me understand the following applications and what are its
advantages if we compare between each of them.
Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE.
Regards,
Kaushal
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2008 Feb 05
6
External MWI question for Asterisk
Hey there. I've been working on a project to integrate Asterisk with
Exchange Unified Messaging via sipX using large parts borrowed from:
http://blog.lithiumblue.com/2007/04/accessing-exchange-2007-unified_29.html
... and everything works surprisingly well. The one problem I have is MWI,
or a lack thereof. Exchange 2007 doesn't support MWI of any kind (!), so
I've been looking into
2004 Sep 14
1
Comparisons between * and sipXpbx (PingTel's open source product)
Has anyone compared * to sipXpbx? From a cursory look, this open source
version of PingTel's PBX has many features that make it more suitable
as a replacement for a traditional PBX, including the ability for users
to tell if a phone/trunk is in use. What I am trying to figure out is
what I'd give up using sipX instead of * (and vice versa).
/carmi
2010 Feb 26
3
: PSTN calls
Hi All,
I have installed astriesk 6 and am able to make calls using sip x-lite.
Its working as I expected.
Now I want to make call from sipx-lite to PSTN using asterisk.
can any please suggest me which Hardware card that I can buy? and use ( pl give me all ths list of cards.... which are good.).
2) what is that I need to do after buying the card to make it talk to the real world PSTN network?
2007 Apr 15
2
Custom CentOS5 DVD
Hello,
Does anyone have an up-to-date page describing, step by step, how to make a
customized CentOS5 DVD?
I noticed that CentOS5 already comes with ~240MB of updates. So for starters,
I'd like to create a new DVD with all the current updates. (And I have other
custom scripts I need to install on top of that).
I've googled around and tried various suggestions on the net:
2007 Aug 29
0
re:Cisco cfgfmt.exe tool
Hi
I am trying to your wine with sipXecs. We are testing Cisco ATA-186. When we attempt to update the ATA software via sipX we send a text file to the sipX server but it needs to be convert from a text file into a binary via the cfgfmt.exe tool
I installed sipX and wine on the same box and used wine config tool to point or grab the tool. Wine puts it or a copy of it in the system32 folder.
2007 Dec 10
1
T.38 fax solution, opinions?
Hi,
I'm putting together a fax solution for my company that utilizes T.38. I
wanted to throw out my plan and get some feedback if anyone is doing
something similar or sees a blatant problem with it.
We're currently rolling out SPA-942 phones for the standard desk phone with
vanilla Asterisk 1.4.15 (just upgraded it today) on the back end. Most calls
for satellite offices are handled by
2005 May 13
4
Polycom configuration
How do you configure your Polycom phones? Is it enough to configure one line
appearance? Or is there a way to configure a roll over?
Chris Mason
US Number: (646)722-0001 US Fax (815)301-9759
Skype: netconcepts
2007 Jan 15
1
I have to register asterisk/sip with a sipproxy that does not support authentication?
I have to register asterisk/sip with a sipproxy that does not support authentication, I do not know how to tell Asterisk not to send authentication request?
# sip.conf
[general]
insecure=very
permit=207.148.115.10/255.255.255.0
[myproxy]
type=friend
host=217.118.115.10
context=from-sip
# Logging:
<--- Reliably Transmitting (NAT) to 207.148.115.10:5060 --->
SIP/2.0 407 Proxy
2007 Mar 20
2
Updating the DVD ISO howto?
I seem to remember this being addressed before, but I can't find the
howto anywhere.
I've got a friend heading to another country with VERY limited
bandwidth. He'd like me to update the 4.4 DVD to include all of the
updated RPMS from updated.
Where can I find the scripts to update the meta data on the RPMS and
create a new bootable DVD?
Thanks!
Ben
2007 Oct 25
2
T.38 Faxing and Asterisk
I understand that Asterisk 1.4 should support T.38 pass-through, but I
need Asterisk (or something on the Asterisk box) to act as a T.38
endpoint. Judging from the unclaimed $12,000USD bounty, it doesn't
appear that Asterisk itself can do this.
http://www.voip-info.org/wiki-Asterisk+T.38+Bounty
Does anyone have any experience with this, or are able to point to an
example of this working?
2005 Apr 15
2
sipXphone
Maybe I just woke up too early today. I have SJPhone and X-Lite working
perfectly but I cannot for the life of me get sipXphone working properly
with Asterisk. Its probably something stupid on my part, but does anyone
have a quick setup sheet for it?
-Kerry
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2009 Jun 01
1
piping to littler
Hi,
This code used to work well until a few months ago (I haven't used it
since), but now it's giving this:
---<--------------------cut here---------------start------------------->---
$ cat <<EOF | r -
locs <- read.csv(list.files(pattern="ds_.+_.+\\.csv"),
colClasses=c(rep("character", 2), "numeric", "character",
2008 Aug 27
2
specifying compiler path in configure
I'm having trouble building some packages while running Debian Lenny
(testing) and suspect that the issues are related to the default use
of gcc-4.3. With Lenny, build-essentials depends on 4.3, so I'd like
to leave it installed but have also installed 4.2.1. How do I tell
./configure the path to 4.2.1 ? I"m sure it's an option, but I don't
see it documented in the R-admin
2023 Feb 11
1
scan(..., skip=1e11): infinite loop; cannot interrupt
On Fri, 10 Feb 2023 23:38:55 -0600
Spencer Graves <spencer.graves at prodsyse.com> wrote:
> I have a 4.54 GB file that I'm trying to read in chunks using
> "scan(..., skip=__)". It works as expected for small values of
> "skip" but goes into an infinite loop for "skip=1e11" and similar
> large values of skip: I cannot even interrupt it; I
2005 May 26
0
Q : registering sipXphone
Hello all,
I have problems trying to register sipXphone to asterisk. It always print
the message :
May 26 13:04:33 NOTICE[2781]: chan_sip.c:7691 handle_request: Registration
from
'sip:2503@172.25.50.52' failed for '172.25.49.219'
Anyone has an idea to help? Asterik seems to fail in register_verify() but i
don't know why...
My sip.conf is the following :
;
; SIP