Displaying 20 results from an estimated 5000 matches similar to: "IAX termination services"
2006 Jan 03
1
bridging two active calls
Can someone give me some direction on how to bridge two calls on an asterisk
server. I'm originating two calls using asterisk java manager and after
some processing in each of the individual dialplans, I want to connect the
calls together.
Jason Wolfe
jasonwolfe@bellsouth.net
c (770) 561-6956
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2009 Aug 25
0
Samba + ldap issues
Hey All
I am having a problems with using openldap and samba. We have been having issues with samba passwords expiring and I have tried several things to resolve the issues. The ldap server was setup using the smbldap-tools. When the password expires the only thing I have been able to do is to reset the password. I have tried the smbldap-usemod -B -1 &username to disable the
2009 Aug 26
0
Problems with smb-ldap tools
Hey All
I am needing to write a script for my end users to be able to reset their passwords when they expire. When I run smbldap-passwd as a user I get the following error messages below. Execute permissions are set correctly on smbldap-passwd.
Failed to modify SMB password: Insufficient access at /usr/local/sbin/smbldap-passwd line 238, <STDIN> line 3.
Failed to modify UNIX password:
2005 Nov 16
1
Very strange permissions issue with Samba 3.0.20(a/b)
Hi Guys,
First, thanks for all the hard work! You all rock.
I am running Samba 3.0.20a on RHEL 3 u5 x86, my configuration is working
perfectly except for cvs commits for 3 users. We are using ADS, pam_winbind, and pam_require to authenticate CVS users against AD.
Our CVS directories are mod 2775, and the group ownership of all dirs is
the AD group "DEN-CVS-Users". Every valid user
2006 Jan 11
1
where to get app_cepstral.c
hello
Anyone know where I can download a copy of app_cepstral.c, or can anyone
send it to me? I saw that it was available at
http://www.automated.it/asterisk/ but the site isn't up.
Thanks,
jasonwolfe@bellsouth.net
2009 Jun 15
2
Click-to-dial CTI for Windows
Hello guys,
Is there a decent click-to-dial CTI which works well with Asterisk?
We have vanilla asterisk implementation and I have tried a few (ADA,
Outcall etc) but they have poor documentation and don't work very well.
We are looking for an application which can allow us to dial a number
from Outlook and IE/Firefox for outbound calls and get a pop-up for
inbound calls with call history
2004 Nov 01
3
Read in data from Microsoft SQL
Hi,
I am trying to read data from Microsoft SQL. I tried to use odbcConnect
(in RODBC) but failed. What should the "dsn" be? Does anyone know how to
do it? I will appreciate it.
Thank you!
Xiaorong
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2003 Dec 01
7
Call Announcement - How To ...
All,
I would like to play an announcement to the user on what external line a
call came in, right before this call get bridged to this user. How would I
go about implementing this in * ?
Regards,
Hans
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2014 Apr 15
2
Old Asterisk Release wanting to upgrade ...
Hello,
I have been running Asterisk for the past 5+ years on RedHat and I never upgraded it before.
All my Asterisk software is of the following release:
1) Asterisk 1.4.21.2
2) Libpri-1.4.4
3) Zaptel-1.4.11
I would like to move the OS to CentOS and then I thought I can at the same time ponder about upgrading Asterisk releases.
However, I am bewildered by the myriad of different releases like 1.6,
2014 May 12
1
Kernel and DAHDI
Hi,
I have noticed it for a while but I just thought about confirming this with the Asterisk community.
As the compilation of DAHDI will need to reference Kernel-devel, does it mean that after DAHDI is installed, we should not yum update kernel because it will affect the operation of DAHDI?
Thanks.
The contents of this e-mail are intended for the named addressee only. It contains information that
2006 Jun 15
10
Best $300 VoIP phone for asterisk?
Polycom 601, hands down.
- Brad
-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Warren
Sent: Thursday, June 15, 2006 10:05 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Best $300 VoIP phone for asterisk?
If you had approx $300 per phone as a budget and needed to buy
2005 Mar 10
2
Asterisk and USB ISDN controllers ...
Guys,
I am planning on building a small SIP PBX with a single ISDN line. Currently
I am looking into the specs of a very tiny barebone system that has an
option Colognechip base ISDN controller. The only thing is that the ISDN
module that comes with this barebone hooks up to the motherboard using USB.
My intention is to allow incoming and outgoing calls from SIP to ISDN. Is
this setup in any way
2005 Feb 24
2
Brainstorm: Running Asterisk as cool as poss ible - AKA solid state.
Hi Kristian,
Anywhere I can read about this Soekris/AstLinux project? ...
Regards,
Hans
-----Original Message-----
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com]On Behalf Of Kristian
Kielhofner
Sent: Thursday, February 24, 2005 6:02 AM
To: jim@vanmeggelen.ca; Asterisk Users Mailing List - Non-Commercial
Discussion
Subject: Re: [Asterisk-Users]
2006 Mar 17
1
RE: DUNDi .... Halfway and CLUSTERING
At the moment I'm out of the office, but when I return I'll be certain to do
that. Note that my solution is different from what you are working on with
regexten, though I suspect some of the challenges that I've faced and
overcome are not. I'm actually using UltraMonkey for load-balancing and
failover of the Asterisk boxes, and my dialplan is set up so that it need
not be changed
2006 Mar 22
3
Voicemail limit?
Hi,
Is there an account limit for voicemail? I have 80+ users in the
voicemail and I can only reach the 70-ieth user. If there is a limit
how can I increase it to hundred for example?
Thanks,
Ryan
2006 Nov 30
4
Trouble with regexten
Can anyone help with the use of regexten? (* 1.4.3)
I've got Asterisk creating extensions for my SIP phones using regexten
but I can't seem to figure out how to make use of them once they're
registered.
Here's my dialplan for from-sip (the SIP's default context):
asterisk*CLI> dialplan show from-sip
[ Context 'from-sip' created by 'pbx_config' ]
2006 May 18
2
SIP Header Info
I remember seeing something somewhere that described how I could get SIP header information with Asterisk. It was a command or a variable. Anyone know what it is? Thanks.
Doug.
2004 Feb 02
0
VOIP/IAX Termination
If you connect an * system to a company like VoicePulse with there
Connect service you are able to get incoming calls unlimited for 7.99/mo
and outgoing at .0295/minute. VoicePulse also offers an unlimited
business plan.
Are you able to sign up with VoicePulse unlimited plan and connect an *
system to it?
Also, with VoicePulse connect you're able to make unlimited simultaneous
calls
2004 Apr 22
0
Looking for IAX/SIP termination in Israel
I'm looking for IAX (preferred) or SIP termination in Israel, with DIDs. Can
anyone give me a price, or better yet, point me to a resource for
buying/selling minutes?
Thanks,
Mark
2004 Apr 22
1
IAX or SIP termination provider
I'm in Mexico an I'll like to know wish is the best IAX or SIP Termination
provider. Im tring to start a small Pre-paid long distance service.
Thanks
Erick