Displaying 20 results from an estimated 2000 matches similar to: "Creating an IAX/SIP-to-ISDN PRI gateway"
2009 Aug 13
2
PRI Gateway - Worth it?
Hi all,
I was looking to build a SIP-to-PRI gateway using Asterisk (as in my other post), but there is also an alternative of using a Cisco router with something like an NM-HDV module with a T1 VIC module and DSP channel banks.
The question is, would it be more reliable to offload all dahdi/zaptel/libpri type stuff to a dedicated gateway device (Asterisk or Cisco) and have the Asterisk PBX only
2003 Jul 04
3
switch => priority in the dialplan.. (probably an issue for Mark)
Hi,
It seems that the "switch" parameter has a priority in the dialplan that is higher than the wildcard extensions.. This I am finding to be a problem..
My setup..
UA1--[AST1]--{IAX}--[AST2]--UA2
| |
PSTN1 PSTN2
I use switch on AST1 to connect to AST2... As you can see I have PSTN connections on both and also the IAX connection is not permanent..
I
2009 Apr 09
2
DTMF
[image: Post]<http://accessanywebsite.com/search.php?u=Oi8vd3d3LmVmbG8ubmV0L1ZJQ0lESUFMZm9ydW0vdmlld3RvcGljLnBocD9wPTI4NjU%3D&b=2#28652>Posted:
Thu Apr 09, 2009 8:34 pm Post subject: DTMF and IVR ... Sorry but
URGENT[image:
Reply with quote]<http://accessanywebsite.com/search.php?u=Oi8vd3d3LmVmbG8ubmV0L1ZJQ0lESUFMZm9ydW0vcG9zdGluZy5waHA%2FbW9kZT1xdW90ZSZwPTI4NjUy&b=2>
2024 Mar 04
1
[External] Re: capture "->"
Maybe someone has already suggested this, but if your functions accepted
strings you could use sub or gsub to replace the -> with a symbol that
parsed at the same precedence as <-,
say <<-. Then parse it and deal with it. When it is time to display the
parsed and perhaps manipulated formulae to the user, deparse it and do the
reverse replacement.
> encode <-
2004 Feb 02
6
Transfer
-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1
Hi,
As I've been unable to get app_transfer to work, could someone explain how it
is supposed to work?
Currently I have two Asterisk boxes. A call comes in via zaptel to ast1. ast1
dials ast2 using iax2 and gets instructed to transfer the call to a different
extension. iax2 debug shows that a transfer cmd is sent to ast1, but nothing
happens
2024 Mar 04
1
[External] Re: capture "->"
Dear Barry,
In general, I believe users are already accustomed with the classical
arrows "->" and "<-" which are used as such in quoted expressions.
But I agree that "-.>" is a very neat trick, thanks a lot. A small dot,
what a difference.
All the best,
Dmitri
On Mon, Mar 4, 2024 at 11:40?AM Barry Rowlingson <
b.rowlingson at lancaster.ac.uk> wrote:
2005 Feb 25
2
407 Proxy Authentication Required
Hi everybody:
I configured my Asterisk to register to my VoIP provider, and I can make
outgoing calls, but I can't receive any calls with it.
I used Ethereal to sniff the activity of it, and I found something that
might be causing the problem:
When my provider's gateway does the "Request: INVITE
mynumber@my-voip-provider.tld ..." my Asterisk asks for "Status: 407 Proxy
2009 Mar 26
1
IAX problem through intermediate asterisk box
I'm having a problem with IAX running through an intermediate asterisk
box. Perhaps a small diagram will explain the situation better:
*A ------- [cloud (public internet)] ------- *B --------[cloud
(private network)]----------- *C
Asterisk server's A, B, and C, are all connected together with IAX
All asterisk servers are 1.6.0.6
Server A and B are geographically close, but connected over
2003 Jun 13
2
Asterisk asterisk => statement
As I understand it (and my understanding is obviously incorrect) the
switch => statement sells the Asterisk box to resolve (aka lookup)
extensions by querying the remote Asterisk server defined in the switch
=> statement. The switch => statement is used to centralize dialplans.
I've not used the switch => statement yet, I'm just trying to understand
the ramifications of using
2009 Aug 03
1
New Digium TE207P for sale
New never used
bought by mistake ( 5V pci slot ) and stored...
pls make offer,
thanks and regards,
Jean-louis Curty
The TE207P is a bundling of our leading TE205P product and our new
VPMOCT064 Octasic DSP-based echo cancellation module. The TE207P is
Digium's and the industry's first two-port digital card featuring
hardware-based echo cancellation. The new module enables users to
2008 Jul 29
3
directory with space
I am facing one problem with rsync. in /etc/rsyncd.conf file , one of my
entry is like bellow.
path = /data/Knowledge Centre
With this, rsync report an error "chdir failed, no such directory".
Even i tried like bellow
path = /data/Knowledge\ Centre
with this also, it gives the same error ?
How to overcome this ? what is the right syntax ?
--
Thanks & Regards,
Shashi Kanth
2018 Apr 26
4
CIFS Null Session Vulnerability Fix in Samba 3.5.10
Hello Rowland,
I do not have support contract with RedHat, and due to some application
dependency i have to be on 5.8. No choice for me to upgrade the OS. I have
choice to upgrade Samba from 3.5 to 3.6.6 through RPMs, but i am not really
sure whether it solves my core issue.
Coming back to my original query " CIFS Null Session vulnerability ", just
i would like to understand whether
2007 Apr 28
8
Poor man's High Availability solution
Hi,
I'm wondering what the best option to obtain a high availability
asterisk server is.
I currently use a TE410P (4 x E1) card.
I'm thinking of 2 different solutions:
- 2 servers configured with Heartbeat + DRBD (drbd mainly for
voicemail....) and the E1 span plugged to the 2 servers (with a TE410P
in each server).
- 2 servers configures with Heartbeat + DRBD with the E1 span hooked
2009 Jul 22
2
sip configuration masking the peers
Hi all,
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2005 Oct 06
0
Issue with trunking
Hi all.
Ive recently setup two Asterisk boxes (running Asterisk@Home to be specific), and Im trying to get a trunk going between them.
So far I have tried a combination of IAX and SIP configuring them through AMP and writing the config files manually, but I cant seem to get calls going between the two.
I have named each box asterisk1 and asterisk2.
Does anyone have some working SIP and/or IAX
2005 Sep 06
0
Weird SIP behaviour
Hi All,
I've been observing a very odd behaviour of Asterisk, when relating to SIP
connections.
Here's the scenario:
Ast1 is an Asterisk box originating calls via a predictive dialer
Ast2 is an Asterisk box connected to 3E1 circuits
Ast1 originates calls to Ast2 via SIP, in order to utilize the PSTN lines.
(There is a reason
I'm using SIP here, so please don't say:
2018 Apr 25
2
CIFS Null Session Vulnerability Fix in Samba 3.5.10
Hi Volker,
Reaching to RedHat is not the option for me right now, could you please
tell me are there any configuration parameters in smb.conf which will avoid
issuing NULL or Anonymous connections or sessions.
On Thu, Apr 26, 2018 at 12:49 AM, Volker Lendecke <Volker.Lendecke at sernet.de
> wrote:
> On Wed, Apr 25, 2018 at 11:12:07PM +0530, Shashi Kanth Boddula via samba
> wrote:
2018 Apr 26
2
CIFS Null Session Vulnerability Fix in Samba 3.5.10
Hi Volker,
I am not finding anywhere the Samba 4.X RPMs for RHEL 5.X platform. Please
share if you know any place from where i can download. I am afraid to build
from source code.
On Thu, Apr 26, 2018 at 11:01 AM, Volker Lendecke <Volker.Lendecke at sernet.de
> wrote:
> On Thu, Apr 26, 2018 at 04:21:39AM +0530, Shashi Kanth Boddula wrote:
> > Reaching to RedHat is not the option
2005 Jun 08
3
AgentCallBacklogin (logout continued...)
Anyone know if
- it is possible to limit 1 agent per extension where
the last agent to log in overrides any previous agents
or
- a Command/application to clear all agents logged in
on extension
Does this look like it would require a custom mod to
do it?
J
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2018 Apr 25
2
CIFS Null Session Vulnerability Fix in Samba 3.5.10
Hi Everyone,
I have Samba server 3.5.10 running on RHEL 5.8 platform and it has joined
to our AD domain controller. Recently my Windows guys has done some changes
to AD Security by stating " CIFS Null Session Vulnerability Fix via GPO -
Security Requirement". After this change, my windows clients are not
authenticating with domain credentials while accessing the shares, but
nothing has