Thank you very much. I will try this! It seems to be what I'm looking for.
I'm in most cases working with 1.2 asterisks, so I'm not up to date on
newer features.
My current project however needed a newer version. I tried some googleing,
but I did not find these variables.
Glad to
Guys,
I am using
Asterisk 1.8.20.0 built by mockbuild @ buildvm-24.phx2.fedoraproject.org on
a x86_64 running Linux on 2013-01-18 19:52:25 UTC
How can I set variable in one context and then Redirect a channel to
another context and use variable there? The code below doesn't work, so
I've got
Hi all
We have an Asterisk server thats been running for a few years now without
problems. We have IPTables running, as well as fail2ban and have followed
all the security recommendations we have found.
Every few weeks we get an attack that lasts about a minute or two, resulting
in our
On Thursday 20 Feb 2014, Brynjolfur Thorvardsson wrote:
Every few weeks we get an attack that lasts about a minute or two,
resulting in our AGI script being overloaded.
What happens is that somebody seems to be trying to connect from our server
in my cdrs log I can see that they use a four
On 20/02/14 11:27, Brynjolfur Thorvardsson wrote:
Hi all
We have an Asterisk server that's been running for a few years now
without problems. We have IPTables running, as well as fail2ban and
have followed all the security recommendations we have found.
Every few weeks we get an attack
On 20/02/14 10:24, Igor Dvorzhak wrote:
Guys,
I am using
Asterisk 1.8.20.0 built by mockbuild @
buildvm-24.phx2.fedoraproject.org
http://buildvm-24.phx2.fedoraproject.org on a x86_64 running Linux
on 2013-01-18 19:52:25 UTC
How can I set variable in one context and then Redirect a channel
On 14-02-20 06:24 AM, Igor Dvorzhak wrote:
Guys,
I am using
Asterisk 1.8.20.0 built by mockbuild @ buildvm-24.phx2.fedoraproject.org
http://buildvm-24.phx2.fedoraproject.org on a x86_64 running Linux on
2013-01-18 19:52:25 UTC
How can I set variable in one context and then Redirect a
I haven't been able to find the answer online, and am not currently
able to conduct an experiment to find the answer...
I understand that in a SIP call where G729 has been negotiated as the
preferred codec, a G.729 licence is not consumed until there is a need
to perform transcoding, e.g. play a
On Thu, Feb 20, 2014 at 10:40 AM, Tony Mountifield t...@softins.co.uk wrote:
I haven't been able to find the answer online, and am not currently
able to conduct an experiment to find the answer...
I understand that in a SIP call where G729 has been negotiated as the
preferred codec, a G.729
In my experience when you run out of g729 licenses additional calls will fail.
Simple as that. Make sure you run out of licenses.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tony Mountifield
Sent:
In article CALLKq0RpimD05jz=osbgjydx-41uebohxmft_skwfjt51ko...@mail.gmail.com,
Paul Belanger paul.belan...@polybeacon.com wrote:
On Thu, Feb 20, 2014 at 10:40 AM, Tony Mountifield t...@softins.co.uk wrote:
I haven't been able to find the answer online, and am not currently
able to conduct an
On 20/02/14 17:16, Paul Belanger wrote:
On Thu, Feb 20, 2014 at 10:40 AM, Tony Mountifieldt...@softins.co.uk wrote:
I haven't been able to find the answer online, and am not currently
able to conduct an experiment to find the answer...
I understand that in a SIP call where G729 has been
I have a setup with asterisk-11.7.0 and kamailio-4.1.1. I am following the setup guide at http://kb.asipto.com/asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb . I want to run asterisk and kamailio on the same server, with SIP realtime configuration
(MySQL database) so that kamailio
On Thu, Feb 20, 2014 at 3:45 AM, Torbjörn Abrahamsson
torbjorn.abrahams...@gmail.com wrote:
I tested SIPFROMDOMAIN, and it worked. Important thing to note is that I
needed to have at least one underscore at the beginning of the variable, as
your example did, it needs to be inherited at least
On Tue, Feb 18, 2014 at 10:53 PM, Gholamreza Sabery gr.sab...@gmail.com wrote:
Hello, a few days ago I sent a question:
http://lists.digium.com/pipermail/asterisk-users/2014-February/282241.html
but no one answered me! I just want to know is it possible or not?
Hi! As many others mentioned,
I tested SIPFROMDOMAIN, and it worked. Important thing to note is that I
needed to have at least one underscore at the beginning of the variable, as
your example did, it needs to be inherited at least one level. I don't
really see
way this should be needed, shouldn't Dial be able see
On Wed, Feb 19, 2014 at 2:55 AM, A J Stiles
asterisk_l...@earthshod.co.uk wrote:
On Wednesday 19 Feb 2014, Gholamreza Sabery wrote:
Hello, a few days ago I sent a question:
http://lists.digium.com/pipermail/asterisk-users/2014-February/282241.html
but no one answered me! I just want to know
Am 20.02.2014 19:48, schrieb Alex Villacís Lasso:
My concern is that asterisk is left listening for SIP through all
interfaces and with no SIP passwords. I want to secure the setup against
directed traffic to the asterisk UDP port (5080), that bypasses the
kamailio process. I tried setting
On Wed, Feb 19, 2014 at 11:53 AM, Markus unive...@truemetal.org wrote:
Hi list,
I have a fresh install of Asterisk 12.0.0 and I'm going to use it only as a
client. I'm trying to SIP REGISTER with a remote SIP provider.
The situation is that Asterisk is running in a VMware VM with a RFC IP
El 20/02/14 15:07, Markus escribió:
Am 20.02.2014 19:48, schrieb Alex Villacís Lasso:
My concern is that asterisk is left listening for SIP through all
interfaces and with no SIP passwords. I want to secure the setup against
directed traffic to the asterisk UDP port (5080), that bypasses the
Am 20.02.2014 22:20, schrieb Rusty Newton:
To force RFC3581 support for outbound REGISTER messages, you can set
nat=force_rport in the general section of your sip.conf. (This also
forces RFC3581 compliance for inbound messages, for any peers that
inherit this general option)
[...]
Thank you
To be fair NAT is rewriting your SIP packet source port. This happens all day,
on almost every NAT device out there.Stop thinking it is purely a port
rewriting issue, something else is going on.
Have you set localnet and externip in sip.conf. Maybe the NAT device has a
short UDP
Am 21.02.2014 01:33, schrieb Eric Wieling:
To be fair NAT is rewriting your SIP packet source port. This happens all day,
on almost every NAT device out there.Stop thinking it is purely a port
rewriting issue, something else is going on.
In the meantime, the provider has reconfigured
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